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From: Ionel C. <ion...@me...> - 2024-09-13 18:32:19
|
Quick follow up. Purely coincidence, when I migrated to the Astlinux VM version, ATT mobile decided to block my home number (where I was dialing from) I've spent many many hours trying to troubleshoot this issue and finally the vitelity folks told me is not in their end and most likely is ATT. I really appreciate your time and private messages offering to help :) This community is absolutely GREAT. On Sep 12, 2024, at 6:40 PM, Ionel Chila via Astlinux-users <ast...@li...> wrote: I recently migrated from my DAHDI system to a VM on UnRaid. Everything runs smooth with asterisk 18.X and almost perfect expect one weird oddity. I can dial out every local US number in the world except one particular number *****8840 as you can see in the logs. Is the weirdest thing. I can dial *****7000 from the same extension but it will not work for the *****8840. I tried from all my local extensions and they all behave the same. I can dial every US number except that one? Any ideas what I am doing wrong here? I attached the debug log for a the working version and not working version. I would really appreciate the help and guidance as my fluency in asterisk is not where it needs to be :) <number-WORKING-from extension-200.txt><number-NOTWORKING-from extension-200.txt> Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline <untitled attachment> |
|
From: Ionel C. <ion...@me...> - 2024-09-12 23:40:19
|
I recently migrated from my DAHDI system to a VM on UnRaid. Everything runs smooth with asterisk 18.X and almost perfect expect one weird oddity. I can dial out every local US number in the world except one particular number *****8840 as you can see in the logs. Is the weirdest thing. I can dial *****7000 from the same extension but it will not work for the *****8840. I tried from all my local extensions and they all behave the same. I can dial every US number except that one? Any ideas what I am doing wrong here? I attached the debug log for a the working version and not working version. I would really appreciate the help and guidance as my fluency in asterisk is not where it needs to be :) |
|
From: Michael K. <li...@mk...> - 2024-08-30 12:23:55
|
BTW: Asterisk is case-sensitive: Set(CALLERID(num)=0338810600) Maybe that command failed. Michael http://www.mksolutions.info > Am 30.08.2024 um 14:20 schrieb Michael Keuter <li...@mk...>: > > I don't think these are errors, these are just infos, you just could decrease the verbosity. > Or you can try to disable strictrtp in rtp.conf (yes is default). > > Do you set "directmedia=no" in your sip trunk settings? (old name was "canreinvite=no") > > Michael > > http://www.mksolutions.info > >> Am 30.08.2024 um 13:57 schrieb Ionel Chila via Astlinux-users <ast...@li...>: >> >> Thanks Michael. I am not referring to the resolve and look of my VM-HOME-PBX. I get these RTP errors and my 2 extensions never dials out. >> >> >> == Using SIP RTP CoS mark 5 >> -- Called SIP/oriunde/40338810606 >>> 0x151c2400a2d0 -- Strict RTP learning after remote address set to: 3.233.63.160:44094 >> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>> 0x151c2400a2d0 -- Strict RTP switching to RTP target address 3.233.63.160:44094 as source >>> 0x151c20052820 -- Strict RTP switching to RTP target address 192.168.0.99:16414 as source >> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 >> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 >> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 >> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>> 0x151c20052820 -- Strict RTP learning complete - Locking on source address 192.168.0.99:16414 >> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 >>> 0x151c2400a2d0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44094 >> == Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/101-00000081’ >> >> >> >> >> >>> On Aug 30, 2024, at 3:57 AM, Michael Keuter <li...@mk...> wrote: >>> >>> >>> >>>> Am 30.08.2024 um 03:50 schrieb Ionel Chila via Astlinux-users <ast...@li...>: >>>> >>>> I just recently moved to a virtual image for Astlinux and also moved away from DAHDI card that I replaced my internal land lines with a Cisco SPA box and Cisco IP phone as per below. All incoming calls work fine but outgoing I get this error / loop as per below logs. It never dials out, just loops like this. >>>> >>>> Any ideas what’s going on? Any pointers I can research this further or try something different? MANY THANKS >>>> >>>> >>>> SIP Clients —> Astlinux (asterisk 18.x) —> VoIp Provider Romania (Oriunde) >>>> —> VoIp Provider USA (Teliax) >>>> —> VoIp Provider USA (Vitelity) >>>> >>>> >>>> 101/SIP - Cisco SPA-2102 >>>> 200/SIP - Cisco IP Phone 7965 >>>> >>>> >>>> And some info from my sip.conf and extensions.conf >>>> >>>> ;--------------------------------------------------------- >>>> ; USER 101 CHILA HOME LINE 1 SPA-2102 >>>> ;--------------------------------------------------------- >>>> [101] >>>> context=voip >>>> type=friend >>>> secret=******* >>>> host=dynamic >>>> canreinvite=yes >>>> qualify=5000 >>>> qualifyfreq=60 >>>> >>>> ;--------------------------------------------------------- >>>> ; USER 200 Cisco Phone Office 7965 >>>> ;--------------------------------------------------------- >>>> [200] >>>> context=voip >>>> type=friend >>>> secret=******** >>>> host=dynamic >>>> canreinvite=no >>>> qualify=5000 >>>> qualifyfreq=60 >>>> transport=udp,tcp >>>> mailbox=200 >>>> ; >>>> >>>> in extensions.conf: >>>> >>>> ; DIAL IN ROMANIA NUMBER >>>> ; >>>> [oriunde] >>>> exten => _338810603,1,Answer() ; Answer the call >>>> exten => _338810603,2,DIAL(SIP/101&SIP/105&SIP/200&SIP/203,30) >>>> exten => _338810603,3,Voicemail(101,u) ; Go to voicemail if no answer >>>> exten => _338810603,n,Hangup() ; Hang up the call >>>> >>>> >>>> ; DIAL OUT ROMANIA NUMBERS >>>> ; >>>> [voip] >>>> exten => _XXXXXXXXXX,1,Set(CALLERid(NUmber)=0338810603) >>>> exten => _XXXXXXXXXX,n,DIAL(SIP/oriunde/4${EXTEN}, 60) >>>> >>>> >>>> VM-HOME-PBX asterisk # asterisk -rvv >>>> Asterisk 18.22.0, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others. >>>> Created by Mark Spencer <mar...@di...> >>>> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. >>>> This is free software, with components licensed under the GNU General Public >>>> License version 2 and other licenses; you are welcome to redistribute it under >>>> certain conditions. Type 'core show license' for details. >>>> ========================================================================= >>>> Connected to Asterisk 18.22.0 currently running on VM-HOME-PBX (pid = 1062) >>>> VM-HOME-PBX*CLI> >>>> VM-HOME-PBX*CLI> >>>> >>>> Extensions 101: >>>> [Aug 29 12:25:49] WARNING[1116][C-00000030]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX' >>>> == Using SIP RTP CoS mark 5 >>>>> 0x151c20052820 -- Strict RTP learning after remote address set to: 192.168.0.99:16414 >>>> -- Executing [0338810606@voip:1] Set("SIP/101-00000081", "CALLERid(NUmber)=0338810600") in new stack >>>> [Aug 29 12:25:50] ERROR[1413][C-00000030]: pbx_functions.c:700 ast_func_write: Function CALLERid not registered >>>> -- Executing [0338810606@voip:2] Dial("SIP/101-00000081", "SIP/oriunde/40338810606, 60") in new stack >>>> [Aug 29 12:25:50] ERROR[1413][C-00000030]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname >>>> [Aug 29 12:25:50] WARNING[1413][C-00000030]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX' >>>> == Using SIP RTP CoS mark 5 >>>> -- Called SIP/oriunde/40338810606 >>>>> 0x151c2400a2d0 -- Strict RTP learning after remote address set to: 3.233.63.160:44094 >>>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>>>> 0x151c2400a2d0 -- Strict RTP switching to RTP target address 3.233.63.160:44094 as source >>>>> 0x151c20052820 -- Strict RTP switching to RTP target address 192.168.0.99:16414 as source >>>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>>> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 >>>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>>> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 >>>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>>> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 >>>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>>>> 0x151c20052820 -- Strict RTP learning complete - Locking on source address 192.168.0.99:16414 >>>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>>> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 >>>>> 0x151c2400a2d0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44094 >>>> == Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/101-00000081' >>>> >>>> >>>> >>>> Extension 200: >>>> [Aug 29 12:24:33] ERROR[1384][C-0000002e]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname >>>> [Aug 29 12:24:33] WARNING[1384][C-0000002e]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX' >>>> == Using SIP RTP CoS mark 5 >>>> -- Called SIP/oriunde/40338810606 >>>>> 0x151c080264a0 -- Strict RTP learning after remote address set to: 3.233.63.160:44040 >>>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>>>> 0x151c080264a0 -- Strict RTP switching to RTP target address 3.233.63.160:44040 as source >>>>> 0x151c200547c0 -- Strict RTP switching to RTP target address 192.168.0.33:16390 as source >>>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d >>>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d >>>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>>>> 0x151c200547c0 -- Strict RTP learning complete - Locking on source address 192.168.0.33:16390 >>>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d >>>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>>>> 0x151c080264a0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44040 >>>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d >>>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d >>>> == Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/200-0000007d’ >>>> >>> >>> Hi, >>> >>> try to add the local IP address and hostname VM-HOME-PBX to the AstLinux DNS hosts table. And restart DNS/DHCP. >>> >>> Michael >>> >>> http://www.mksolutions.info > |
|
From: Michael K. <li...@mk...> - 2024-08-30 12:20:38
|
I don't think these are errors, these are just infos, you just could decrease the verbosity. Or you can try to disable strictrtp in rtp.conf (yes is default). Do you set "directmedia=no" in your sip trunk settings? (old name was "canreinvite=no") Michael http://www.mksolutions.info > Am 30.08.2024 um 13:57 schrieb Ionel Chila via Astlinux-users <ast...@li...>: > > Thanks Michael. I am not referring to the resolve and look of my VM-HOME-PBX. I get these RTP errors and my 2 extensions never dials out. > > > == Using SIP RTP CoS mark 5 > -- Called SIP/oriunde/40338810606 >> 0x151c2400a2d0 -- Strict RTP learning after remote address set to: 3.233.63.160:44094 > -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >> 0x151c2400a2d0 -- Strict RTP switching to RTP target address 3.233.63.160:44094 as source >> 0x151c20052820 -- Strict RTP switching to RTP target address 192.168.0.99:16414 as source > -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 > -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 > -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 > -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 > -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 > -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 > -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 > -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 > -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 > -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >> 0x151c20052820 -- Strict RTP learning complete - Locking on source address 192.168.0.99:16414 > -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 > -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 >> 0x151c2400a2d0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44094 > == Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/101-00000081’ > > > > > >> On Aug 30, 2024, at 3:57 AM, Michael Keuter <li...@mk...> wrote: >> >> >> >>> Am 30.08.2024 um 03:50 schrieb Ionel Chila via Astlinux-users <ast...@li...>: >>> >>> I just recently moved to a virtual image for Astlinux and also moved away from DAHDI card that I replaced my internal land lines with a Cisco SPA box and Cisco IP phone as per below. All incoming calls work fine but outgoing I get this error / loop as per below logs. It never dials out, just loops like this. >>> >>> Any ideas what’s going on? Any pointers I can research this further or try something different? MANY THANKS >>> >>> >>> SIP Clients —> Astlinux (asterisk 18.x) —> VoIp Provider Romania (Oriunde) >>> —> VoIp Provider USA (Teliax) >>> —> VoIp Provider USA (Vitelity) >>> >>> >>> 101/SIP - Cisco SPA-2102 >>> 200/SIP - Cisco IP Phone 7965 >>> >>> >>> And some info from my sip.conf and extensions.conf >>> >>> ;--------------------------------------------------------- >>> ; USER 101 CHILA HOME LINE 1 SPA-2102 >>> ;--------------------------------------------------------- >>> [101] >>> context=voip >>> type=friend >>> secret=******* >>> host=dynamic >>> canreinvite=yes >>> qualify=5000 >>> qualifyfreq=60 >>> >>> ;--------------------------------------------------------- >>> ; USER 200 Cisco Phone Office 7965 >>> ;--------------------------------------------------------- >>> [200] >>> context=voip >>> type=friend >>> secret=******** >>> host=dynamic >>> canreinvite=no >>> qualify=5000 >>> qualifyfreq=60 >>> transport=udp,tcp >>> mailbox=200 >>> ; >>> >>> in extensions.conf: >>> >>> ; DIAL IN ROMANIA NUMBER >>> ; >>> [oriunde] >>> exten => _338810603,1,Answer() ; Answer the call >>> exten => _338810603,2,DIAL(SIP/101&SIP/105&SIP/200&SIP/203,30) >>> exten => _338810603,3,Voicemail(101,u) ; Go to voicemail if no answer >>> exten => _338810603,n,Hangup() ; Hang up the call >>> >>> >>> ; DIAL OUT ROMANIA NUMBERS >>> ; >>> [voip] >>> exten => _XXXXXXXXXX,1,Set(CALLERid(NUmber)=0338810603) >>> exten => _XXXXXXXXXX,n,DIAL(SIP/oriunde/4${EXTEN}, 60) >>> >>> >>> VM-HOME-PBX asterisk # asterisk -rvv >>> Asterisk 18.22.0, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others. >>> Created by Mark Spencer <mar...@di...> >>> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. >>> This is free software, with components licensed under the GNU General Public >>> License version 2 and other licenses; you are welcome to redistribute it under >>> certain conditions. Type 'core show license' for details. >>> ========================================================================= >>> Connected to Asterisk 18.22.0 currently running on VM-HOME-PBX (pid = 1062) >>> VM-HOME-PBX*CLI> >>> VM-HOME-PBX*CLI> >>> >>> Extensions 101: >>> [Aug 29 12:25:49] WARNING[1116][C-00000030]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX' >>> == Using SIP RTP CoS mark 5 >>>> 0x151c20052820 -- Strict RTP learning after remote address set to: 192.168.0.99:16414 >>> -- Executing [0338810606@voip:1] Set("SIP/101-00000081", "CALLERid(NUmber)=0338810600") in new stack >>> [Aug 29 12:25:50] ERROR[1413][C-00000030]: pbx_functions.c:700 ast_func_write: Function CALLERid not registered >>> -- Executing [0338810606@voip:2] Dial("SIP/101-00000081", "SIP/oriunde/40338810606, 60") in new stack >>> [Aug 29 12:25:50] ERROR[1413][C-00000030]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname >>> [Aug 29 12:25:50] WARNING[1413][C-00000030]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX' >>> == Using SIP RTP CoS mark 5 >>> -- Called SIP/oriunde/40338810606 >>>> 0x151c2400a2d0 -- Strict RTP learning after remote address set to: 3.233.63.160:44094 >>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>>> 0x151c2400a2d0 -- Strict RTP switching to RTP target address 3.233.63.160:44094 as source >>>> 0x151c20052820 -- Strict RTP switching to RTP target address 192.168.0.99:16414 as source >>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 >>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 >>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 >>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>>> 0x151c20052820 -- Strict RTP learning complete - Locking on source address 192.168.0.99:16414 >>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081 >>> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081 >>>> 0x151c2400a2d0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44094 >>> == Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/101-00000081' >>> >>> >>> >>> Extension 200: >>> [Aug 29 12:24:33] ERROR[1384][C-0000002e]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname >>> [Aug 29 12:24:33] WARNING[1384][C-0000002e]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX' >>> == Using SIP RTP CoS mark 5 >>> -- Called SIP/oriunde/40338810606 >>>> 0x151c080264a0 -- Strict RTP learning after remote address set to: 3.233.63.160:44040 >>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>>> 0x151c080264a0 -- Strict RTP switching to RTP target address 3.233.63.160:44040 as source >>>> 0x151c200547c0 -- Strict RTP switching to RTP target address 192.168.0.33:16390 as source >>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d >>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d >>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>>> 0x151c200547c0 -- Strict RTP learning complete - Locking on source address 192.168.0.33:16390 >>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d >>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>>> 0x151c080264a0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44040 >>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d >>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d >>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d >>> == Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/200-0000007d’ >>> >> >> Hi, >> >> try to add the local IP address and hostname VM-HOME-PBX to the AstLinux DNS hosts table. And restart DNS/DHCP. >> >> Michael >> >> http://www.mksolutions.info |
|
From: Ionel C. <ion...@me...> - 2024-08-30 11:57:39
|
Thanks Michael. I am not referring to the resolve and look of my VM-HOME-PBX. I get these RTP errors and my 2 extensions never dials out.
== Using SIP RTP CoS mark 5
-- Called SIP/oriunde/40338810606
> 0x151c2400a2d0 -- Strict RTP learning after remote address set to: 3.233.63.160:44094
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> 0x151c2400a2d0 -- Strict RTP switching to RTP target address 3.233.63.160:44094 as source
> 0x151c20052820 -- Strict RTP switching to RTP target address 192.168.0.99:16414 as source
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> 0x151c20052820 -- Strict RTP learning complete - Locking on source address 192.168.0.99:16414
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
> 0x151c2400a2d0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44094
== Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/101-00000081’
> On Aug 30, 2024, at 3:57 AM, Michael Keuter <li...@mk...> wrote:
>
>
>
>> Am 30.08.2024 um 03:50 schrieb Ionel Chila via Astlinux-users <ast...@li...>:
>>
>> I just recently moved to a virtual image for Astlinux and also moved away from DAHDI card that I replaced my internal land lines with a Cisco SPA box and Cisco IP phone as per below. All incoming calls work fine but outgoing I get this error / loop as per below logs. It never dials out, just loops like this.
>>
>> Any ideas what’s going on? Any pointers I can research this further or try something different? MANY THANKS
>>
>>
>> SIP Clients —> Astlinux (asterisk 18.x) —> VoIp Provider Romania (Oriunde)
>> —> VoIp Provider USA (Teliax)
>> —> VoIp Provider USA (Vitelity)
>>
>>
>> 101/SIP - Cisco SPA-2102
>> 200/SIP - Cisco IP Phone 7965
>>
>>
>> And some info from my sip.conf and extensions.conf
>>
>> ;---------------------------------------------------------
>> ; USER 101 CHILA HOME LINE 1 SPA-2102
>> ;---------------------------------------------------------
>> [101]
>> context=voip
>> type=friend
>> secret=*******
>> host=dynamic
>> canreinvite=yes
>> qualify=5000
>> qualifyfreq=60
>>
>> ;---------------------------------------------------------
>> ; USER 200 Cisco Phone Office 7965
>> ;---------------------------------------------------------
>> [200]
>> context=voip
>> type=friend
>> secret=********
>> host=dynamic
>> canreinvite=no
>> qualify=5000
>> qualifyfreq=60
>> transport=udp,tcp
>> mailbox=200
>> ;
>>
>> in extensions.conf:
>>
>> ; DIAL IN ROMANIA NUMBER
>> ;
>> [oriunde]
>> exten => _338810603,1,Answer() ; Answer the call
>> exten => _338810603,2,DIAL(SIP/101&SIP/105&SIP/200&SIP/203,30)
>> exten => _338810603,3,Voicemail(101,u) ; Go to voicemail if no answer
>> exten => _338810603,n,Hangup() ; Hang up the call
>>
>>
>> ; DIAL OUT ROMANIA NUMBERS
>> ;
>> [voip]
>> exten => _XXXXXXXXXX,1,Set(CALLERid(NUmber)=0338810603)
>> exten => _XXXXXXXXXX,n,DIAL(SIP/oriunde/4${EXTEN}, 60)
>>
>>
>> VM-HOME-PBX asterisk # asterisk -rvv
>> Asterisk 18.22.0, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
>> Created by Mark Spencer <mar...@di...>
>> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
>> This is free software, with components licensed under the GNU General Public
>> License version 2 and other licenses; you are welcome to redistribute it under
>> certain conditions. Type 'core show license' for details.
>> =========================================================================
>> Connected to Asterisk 18.22.0 currently running on VM-HOME-PBX (pid = 1062)
>> VM-HOME-PBX*CLI>
>> VM-HOME-PBX*CLI>
>>
>> Extensions 101:
>> [Aug 29 12:25:49] WARNING[1116][C-00000030]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX'
>> == Using SIP RTP CoS mark 5
>>> 0x151c20052820 -- Strict RTP learning after remote address set to: 192.168.0.99:16414
>> -- Executing [0338810606@voip:1] Set("SIP/101-00000081", "CALLERid(NUmber)=0338810600") in new stack
>> [Aug 29 12:25:50] ERROR[1413][C-00000030]: pbx_functions.c:700 ast_func_write: Function CALLERid not registered
>> -- Executing [0338810606@voip:2] Dial("SIP/101-00000081", "SIP/oriunde/40338810606, 60") in new stack
>> [Aug 29 12:25:50] ERROR[1413][C-00000030]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname
>> [Aug 29 12:25:50] WARNING[1413][C-00000030]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX'
>> == Using SIP RTP CoS mark 5
>> -- Called SIP/oriunde/40338810606
>>> 0x151c2400a2d0 -- Strict RTP learning after remote address set to: 3.233.63.160:44094
>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
>>> 0x151c2400a2d0 -- Strict RTP switching to RTP target address 3.233.63.160:44094 as source
>>> 0x151c20052820 -- Strict RTP switching to RTP target address 192.168.0.99:16414 as source
>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
>> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
>> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
>> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
>>> 0x151c20052820 -- Strict RTP learning complete - Locking on source address 192.168.0.99:16414
>> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
>> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
>>> 0x151c2400a2d0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44094
>> == Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/101-00000081'
>>
>>
>>
>> Extension 200:
>> [Aug 29 12:24:33] ERROR[1384][C-0000002e]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname
>> [Aug 29 12:24:33] WARNING[1384][C-0000002e]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX'
>> == Using SIP RTP CoS mark 5
>> -- Called SIP/oriunde/40338810606
>>> 0x151c080264a0 -- Strict RTP learning after remote address set to: 3.233.63.160:44040
>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
>>> 0x151c080264a0 -- Strict RTP switching to RTP target address 3.233.63.160:44040 as source
>>> 0x151c200547c0 -- Strict RTP switching to RTP target address 192.168.0.33:16390 as source
>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
>>> 0x151c200547c0 -- Strict RTP learning complete - Locking on source address 192.168.0.33:16390
>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
>>> 0x151c080264a0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44040
>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
>> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
>> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
>> == Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/200-0000007d’
>>
>
> Hi,
>
> try to add the local IP address and hostname VM-HOME-PBX to the AstLinux DNS hosts table. And restart DNS/DHCP.
>
> Michael
>
> http://www.mksolutions.info
>
>
>
>
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
|
|
From: Michael K. <li...@mk...> - 2024-08-30 08:57:37
|
> Am 30.08.2024 um 03:50 schrieb Ionel Chila via Astlinux-users <ast...@li...>:
>
> I just recently moved to a virtual image for Astlinux and also moved away from DAHDI card that I replaced my internal land lines with a Cisco SPA box and Cisco IP phone as per below. All incoming calls work fine but outgoing I get this error / loop as per below logs. It never dials out, just loops like this.
>
> Any ideas what’s going on? Any pointers I can research this further or try something different? MANY THANKS
>
>
> SIP Clients —> Astlinux (asterisk 18.x) —> VoIp Provider Romania (Oriunde)
> —> VoIp Provider USA (Teliax)
> —> VoIp Provider USA (Vitelity)
>
>
> 101/SIP - Cisco SPA-2102
> 200/SIP - Cisco IP Phone 7965
>
>
> And some info from my sip.conf and extensions.conf
>
> ;---------------------------------------------------------
> ; USER 101 CHILA HOME LINE 1 SPA-2102
> ;---------------------------------------------------------
> [101]
> context=voip
> type=friend
> secret=*******
> host=dynamic
> canreinvite=yes
> qualify=5000
> qualifyfreq=60
>
> ;---------------------------------------------------------
> ; USER 200 Cisco Phone Office 7965
> ;---------------------------------------------------------
> [200]
> context=voip
> type=friend
> secret=********
> host=dynamic
> canreinvite=no
> qualify=5000
> qualifyfreq=60
> transport=udp,tcp
> mailbox=200
> ;
>
> in extensions.conf:
>
> ; DIAL IN ROMANIA NUMBER
> ;
> [oriunde]
> exten => _338810603,1,Answer() ; Answer the call
> exten => _338810603,2,DIAL(SIP/101&SIP/105&SIP/200&SIP/203,30)
> exten => _338810603,3,Voicemail(101,u) ; Go to voicemail if no answer
> exten => _338810603,n,Hangup() ; Hang up the call
>
>
> ; DIAL OUT ROMANIA NUMBERS
> ;
> [voip]
> exten => _XXXXXXXXXX,1,Set(CALLERid(NUmber)=0338810603)
> exten => _XXXXXXXXXX,n,DIAL(SIP/oriunde/4${EXTEN}, 60)
>
>
> VM-HOME-PBX asterisk # asterisk -rvv
> Asterisk 18.22.0, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
> Created by Mark Spencer <mar...@di...>
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
> This is free software, with components licensed under the GNU General Public
> License version 2 and other licenses; you are welcome to redistribute it under
> certain conditions. Type 'core show license' for details.
> =========================================================================
> Connected to Asterisk 18.22.0 currently running on VM-HOME-PBX (pid = 1062)
> VM-HOME-PBX*CLI>
> VM-HOME-PBX*CLI>
>
> Extensions 101:
> [Aug 29 12:25:49] WARNING[1116][C-00000030]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX'
> == Using SIP RTP CoS mark 5
> > 0x151c20052820 -- Strict RTP learning after remote address set to: 192.168.0.99:16414
> -- Executing [0338810606@voip:1] Set("SIP/101-00000081", "CALLERid(NUmber)=0338810600") in new stack
> [Aug 29 12:25:50] ERROR[1413][C-00000030]: pbx_functions.c:700 ast_func_write: Function CALLERid not registered
> -- Executing [0338810606@voip:2] Dial("SIP/101-00000081", "SIP/oriunde/40338810606, 60") in new stack
> [Aug 29 12:25:50] ERROR[1413][C-00000030]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname
> [Aug 29 12:25:50] WARNING[1413][C-00000030]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX'
> == Using SIP RTP CoS mark 5
> -- Called SIP/oriunde/40338810606
> > 0x151c2400a2d0 -- Strict RTP learning after remote address set to: 3.233.63.160:44094
> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> > 0x151c2400a2d0 -- Strict RTP switching to RTP target address 3.233.63.160:44094 as source
> > 0x151c20052820 -- Strict RTP switching to RTP target address 192.168.0.99:16414 as source
> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> > 0x151c20052820 -- Strict RTP learning complete - Locking on source address 192.168.0.99:16414
> -- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> -- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
> > 0x151c2400a2d0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44094
> == Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/101-00000081'
>
>
>
> Extension 200:
> [Aug 29 12:24:33] ERROR[1384][C-0000002e]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname
> [Aug 29 12:24:33] WARNING[1384][C-0000002e]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX'
> == Using SIP RTP CoS mark 5
> -- Called SIP/oriunde/40338810606
> > 0x151c080264a0 -- Strict RTP learning after remote address set to: 3.233.63.160:44040
> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> > 0x151c080264a0 -- Strict RTP switching to RTP target address 3.233.63.160:44040 as source
> > 0x151c200547c0 -- Strict RTP switching to RTP target address 192.168.0.33:16390 as source
> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> > 0x151c200547c0 -- Strict RTP learning complete - Locking on source address 192.168.0.33:16390
> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> > 0x151c080264a0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44040
> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
> -- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> -- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
> == Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/200-0000007d’
>
Hi,
try to add the local IP address and hostname VM-HOME-PBX to the AstLinux DNS hosts table. And restart DNS/DHCP.
Michael
http://www.mksolutions.info
|
|
From: Ionel C. <ion...@me...> - 2024-08-30 01:51:53
|
I just recently moved to a virtual image for Astlinux and also moved away from DAHDI card that I replaced my internal land lines with a Cisco SPA box and Cisco IP phone as per below. All incoming calls work fine but outgoing I get this error / loop as per below logs. It never dials out, just loops like this.
Any ideas what’s going on? Any pointers I can research this further or try something different? MANY THANKS
SIP Clients —> Astlinux (asterisk 18.x) —> VoIp Provider Romania (Oriunde)
—> VoIp Provider USA (Teliax)
—> VoIp Provider USA (Vitelity)
101/SIP - Cisco SPA-2102
200/SIP - Cisco IP Phone 7965
And some info from my sip.conf and extensions.conf
;---------------------------------------------------------
; USER 101 CHILA HOME LINE 1 SPA-2102
;---------------------------------------------------------
[101]
context=voip
type=friend
secret=*******
host=dynamic
canreinvite=yes
qualify=5000
qualifyfreq=60
;---------------------------------------------------------
; USER 200 Cisco Phone Office 7965
;---------------------------------------------------------
[200]
context=voip
type=friend
secret=********
host=dynamic
canreinvite=no
qualify=5000
qualifyfreq=60
transport=udp,tcp
mailbox=200
;
in extensions.conf:
; DIAL IN ROMANIA NUMBER
;
[oriunde]
exten => _338810603,1,Answer() ; Answer the call
exten => _338810603,2,DIAL(SIP/101&SIP/105&SIP/200&SIP/203,30)
exten => _338810603,3,Voicemail(101,u) ; Go to voicemail if no answer
exten => _338810603,n,Hangup() ; Hang up the call
; DIAL OUT ROMANIA NUMBERS
;
[voip]
exten => _XXXXXXXXXX,1,Set(CALLERid(NUmber)=0338810603)
exten => _XXXXXXXXXX,n,DIAL(SIP/oriunde/4${EXTEN}, 60)
VM-HOME-PBX asterisk # asterisk -rvv
Asterisk 18.22.0, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <mar...@di...>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 18.22.0 currently running on VM-HOME-PBX (pid = 1062)
VM-HOME-PBX*CLI>
VM-HOME-PBX*CLI>
Extensions 101:
[Aug 29 12:25:49] WARNING[1116][C-00000030]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX'
== Using SIP RTP CoS mark 5
> 0x151c20052820 -- Strict RTP learning after remote address set to: 192.168.0.99:16414
-- Executing [0338810606@voip:1] Set("SIP/101-00000081", "CALLERid(NUmber)=0338810600") in new stack
[Aug 29 12:25:50] ERROR[1413][C-00000030]: pbx_functions.c:700 ast_func_write: Function CALLERid not registered
-- Executing [0338810606@voip:2] Dial("SIP/101-00000081", "SIP/oriunde/40338810606, 60") in new stack
[Aug 29 12:25:50] ERROR[1413][C-00000030]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname
[Aug 29 12:25:50] WARNING[1413][C-00000030]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX'
== Using SIP RTP CoS mark 5
-- Called SIP/oriunde/40338810606
> 0x151c2400a2d0 -- Strict RTP learning after remote address set to: 3.233.63.160:44094
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> 0x151c2400a2d0 -- Strict RTP switching to RTP target address 3.233.63.160:44094 as source
> 0x151c20052820 -- Strict RTP switching to RTP target address 192.168.0.99:16414 as source
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> 0x151c20052820 -- Strict RTP learning complete - Locking on source address 192.168.0.99:16414
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
> 0x151c2400a2d0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44094
== Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/101-00000081'
Extension 200:
[Aug 29 12:24:33] ERROR[1384][C-0000002e]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname
[Aug 29 12:24:33] WARNING[1384][C-0000002e]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX'
== Using SIP RTP CoS mark 5
-- Called SIP/oriunde/40338810606
> 0x151c080264a0 -- Strict RTP learning after remote address set to: 3.233.63.160:44040
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> 0x151c080264a0 -- Strict RTP switching to RTP target address 3.233.63.160:44040 as source
> 0x151c200547c0 -- Strict RTP switching to RTP target address 192.168.0.33:16390 as source
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> 0x151c200547c0 -- Strict RTP learning complete - Locking on source address 192.168.0.33:16390
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> 0x151c080264a0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44040
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
== Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/200-0000007d’
|
|
From: Ionel C. <ion...@me...> - 2024-08-30 01:51:34
|
I just recently moved to a virtual image for Astlinux and also moved away from DAHDI card that I replaced my internal land lines with a Cisco SPA box and Cisco IP phone as per below. All incoming calls work fine but outgoing I get this error / loop as per below logs. It never dials out, just loops like this.
Any ideas what’s going on? Any pointers I can research this further or try something different? MANY THANKS
SIP Clients —> Astlinux (asterisk 18.x) —> VoIp Provider Romania (Oriunde)
—> VoIp Provider USA (Teliax)
—> VoIp Provider USA (Vitelity)
101/SIP - Cisco SPA-2102
200/SIP - Cisco IP Phone 7965
And some info from my sip.conf and extensions.conf
;---------------------------------------------------------
; USER 101 CHILA HOME LINE 1 SPA-2102
;---------------------------------------------------------
[101]
context=voip
type=friend
secret=*******
host=dynamic
canreinvite=yes
qualify=5000
qualifyfreq=60
;---------------------------------------------------------
; USER 200 Cisco Phone Office 7965
;---------------------------------------------------------
[200]
context=voip
type=friend
secret=********
host=dynamic
canreinvite=no
qualify=5000
qualifyfreq=60
transport=udp,tcp
mailbox=200
;
in extensions.conf:
; DIAL IN ROMANIA NUMBER
;
[oriunde]
exten => _338810603,1,Answer() ; Answer the call
exten => _338810603,2,DIAL(SIP/101&SIP/105&SIP/200&SIP/203,30)
exten => _338810603,3,Voicemail(101,u) ; Go to voicemail if no answer
exten => _338810603,n,Hangup() ; Hang up the call
; DIAL OUT ROMANIA NUMBERS
;
[voip]
exten => _XXXXXXXXXX,1,Set(CALLERid(NUmber)=0338810603)
exten => _XXXXXXXXXX,n,DIAL(SIP/oriunde/4${EXTEN}, 60)
VM-HOME-PBX asterisk # asterisk -rvv
Asterisk 18.22.0, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <mar...@di...>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 18.22.0 currently running on VM-HOME-PBX (pid = 1062)
VM-HOME-PBX*CLI>
VM-HOME-PBX*CLI>
Extensions 101:
[Aug 29 12:25:49] WARNING[1116][C-00000030]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX'
== Using SIP RTP CoS mark 5
> 0x151c20052820 -- Strict RTP learning after remote address set to: 192.168.0.99:16414
-- Executing [0338810606@voip:1] Set("SIP/101-00000081", "CALLERid(NUmber)=0338810600") in new stack
[Aug 29 12:25:50] ERROR[1413][C-00000030]: pbx_functions.c:700 ast_func_write: Function CALLERid not registered
-- Executing [0338810606@voip:2] Dial("SIP/101-00000081", "SIP/oriunde/40338810606, 60") in new stack
[Aug 29 12:25:50] ERROR[1413][C-00000030]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname
[Aug 29 12:25:50] WARNING[1413][C-00000030]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX'
== Using SIP RTP CoS mark 5
-- Called SIP/oriunde/40338810606
> 0x151c2400a2d0 -- Strict RTP learning after remote address set to: 3.233.63.160:44094
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> 0x151c2400a2d0 -- Strict RTP switching to RTP target address 3.233.63.160:44094 as source
> 0x151c20052820 -- Strict RTP switching to RTP target address 192.168.0.99:16414 as source
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
> 0x151c20052820 -- Strict RTP learning complete - Locking on source address 192.168.0.99:16414
-- SIP/oriunde-00000082 is making progress passing it to SIP/101-00000081
-- SIP/oriunde-00000082 requested media update control 26, passing it to SIP/101-00000081
> 0x151c2400a2d0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44094
== Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/101-00000081'
Extension 200:
[Aug 29 12:24:33] ERROR[1384][C-0000002e]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("VM-HOME-PBX", "(null)", ...): No address associated with hostname
[Aug 29 12:24:33] WARNING[1384][C-0000002e]: acl.c:890 resolve_first: Unable to lookup 'VM-HOME-PBX'
== Using SIP RTP CoS mark 5
-- Called SIP/oriunde/40338810606
> 0x151c080264a0 -- Strict RTP learning after remote address set to: 3.233.63.160:44040
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> 0x151c080264a0 -- Strict RTP switching to RTP target address 3.233.63.160:44040 as source
> 0x151c200547c0 -- Strict RTP switching to RTP target address 192.168.0.33:16390 as source
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> 0x151c200547c0 -- Strict RTP learning complete - Locking on source address 192.168.0.33:16390
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
> 0x151c080264a0 -- Strict RTP learning complete - Locking on source address 3.233.63.160:44040
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e is making progress passing it to SIP/200-0000007d
-- SIP/oriunde-0000007e requested media update control 26, passing it to SIP/200-0000007d
== Spawn extension (voip, 0338810606, 2) exited non-zero on 'SIP/200-0000007d’
|
|
From: Ionel C. <ion...@me...> - 2024-08-24 17:08:00
|
Thanks for your precious advise as always Lonnie. It is now down to 4 from 8 :) Cheers Ionel On Aug 24, 2024, at 11:01 AM, Lonnie Abelbeck <li...@lo...> wrote: Hi Ionel, Thanks for sharing your AstLinux bare metal -> UnRaid VM experiences. Great to hear! BTW, you may want to lower the RAM setting of the UnRaid AstLinux VM, it is currently 8 MB ... I would expect 4 MB or even 2 MB would not be noticed in the AstLinux VM. Granted you have 64 GB for the system, but in the future you might have use for the extra RAM not used in AstLinux. Lonnie On Aug 24, 2024, at 9:12 AM, Ionel Chila via Astlinux-users <ast...@li...> wrote: How long have I been running this system with this TDM410P 1FXO & 3FXS DAHDI card? I think 15+ years or even more? What a journey and what a rock solid Astlinux OS. Well my hardware died before the support for my DAHDI so I was forced to look for other options. Since I have some NAS boxes running UnRaid and supporting VM's I decided to give Astlinux a try and I am happy to report that Astlinux works beautifully as a VM under UnRaid. Manage to convert all my configs to this new vm instance replacing all my DAHDI card ports with the Cisco SPA-2102 device. I am gonna miss my old PBX box and I will put it next to my old Soekris box to collect dust :) If any of interested in this setup let me know and I will gladly go over. Cheers and THANKS for all the hard work guys <UNRAID-ASTLINUX-1.JPG><UNRAID-ASTLINUX-2.JPG> _______________________________________________ Astlinux-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pa...@kr... . _______________________________________________ Astlinux-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pa...@kr... . |
|
From: Lonnie A. <li...@lo...> - 2024-08-24 16:01:10
|
Hi Ionel, Thanks for sharing your AstLinux bare metal -> UnRaid VM experiences. Great to hear! BTW, you may want to lower the RAM setting of the UnRaid AstLinux VM, it is currently 8 MB ... I would expect 4 MB or even 2 MB would not be noticed in the AstLinux VM. Granted you have 64 GB for the system, but in the future you might have use for the extra RAM not used in AstLinux. Lonnie > On Aug 24, 2024, at 9:12 AM, Ionel Chila via Astlinux-users <ast...@li...> wrote: > > How long have I been running this system with this TDM410P 1FXO & 3FXS DAHDI card? I think 15+ years or even more? What a journey and what a rock solid Astlinux OS. Well my hardware died before the support for my DAHDI so I was forced to look for other options. > > Since I have some NAS boxes running UnRaid and supporting VM's I decided to give Astlinux a try and I am happy to report that Astlinux works beautifully as a VM under UnRaid. Manage to convert all my configs to this new vm instance replacing all my DAHDI card ports with the Cisco SPA-2102 device. > > I am gonna miss my old PBX box and I will put it next to my old Soekris box to collect dust :) > > If any of interested in this setup let me know and I will gladly go over. > > Cheers and THANKS for all the hard work guys > > <UNRAID-ASTLINUX-1.JPG><UNRAID-ASTLINUX-2.JPG> > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... |
|
From: Michael K. <li...@mk...> - 2024-08-24 15:53:10
|
> Am 24.08.2024 um 16:12 schrieb Ionel Chila via Astlinux-users <ast...@li...>: > > How long have I been running this system with this TDM410P 1FXO & 3FXS DAHDI card? I think 15+ years or even more? What a journey and what a rock solid Astlinux OS. Well my hardware died before the support for my DAHDI so I was forced to look for other options. > > Since I have some NAS boxes running UnRaid and supporting VM's I decided to give Astlinux a try and I am happy to report that Astlinux works beautifully as a VM under UnRaid. Manage to convert all my configs to this new vm instance replacing all my DAHDI card ports with the Cisco SPA-2102 device. > > I am gonna miss my old PBX box and I will put it next to my old Soekris box to collect dust :) > > If any of interested in this setup let me know and I will gladly go over. > > Cheers and THANKS for all the hard work guys > > <UNRAID-ASTLINUX-1.JPG><UNRAID-ASTLINUX-2.JPG> Hi Ionel, great that it works on UnRaid, and thanks for letting us know. Michael http://www.mksolutions.info |
|
From: Ionel C. <ion...@me...> - 2024-08-24 14:12:34
|
How long have I been running this system with this TDM410P 1FXO & 3FXS DAHDI card? I think 15+ years or even more? What a journey and what a rock solid Astlinux OS. Well my hardware died before the support for my DAHDI so I was forced to look for other options. Since I have some NAS boxes running UnRaid and supporting VM's I decided to give Astlinux a try and I am happy to report that Astlinux works beautifully as a VM under UnRaid. Manage to convert all my configs to this new vm instance replacing all my DAHDI card ports with the Cisco SPA-2102 device. I am gonna miss my old PBX box and I will put it next to my old Soekris box to collect dust :) If any of interested in this setup let me know and I will gladly go over. Cheers and THANKS for all the hard work guys |
|
From: Michael K. <mic...@ip...> - 2024-07-27 07:21:15
|
Hi David We actually use a Mail API (Mailgun) rather than SMTP for more reliable email transmission. Is a much better solution than the same account credentials used at many sites. Regards Michael Knill ________________________________ From: Lonnie Abelbeck <li...@lo...> Sent: Friday, 26 July 2024 11:44 PM To: AstLinux Users Mailing List <ast...@li...> Subject: Re: [Astlinux-users] OAuth for SMTP? Hi David, Thanks for testing. From what you describe, a Google App Password seems like a great solution, specific to an application so the master user/pass is not replicated over all your AstLinux boxes. Also relatively easy to revoke a lost App Password and regenerate a new one. Lonnie > On Jul 26, 2024, at 8:22 AM, David Kerr <Da...@Ke...> wrote: > > Lonnie, > I have tested google App Password and yes it works for us. The password is automatically generated, 16 characters in length (displayed in 4 blocks of 4, must exclude the spaces when entering it). The password is displayed only once, you cannot retrieve it again, only delete it. And I think I read that if you change your regular password then new app passwords have to be generated again. > > David. > > On Thu, Jul 25, 2024 at 5:24 PM Lonnie Abelbeck <li...@lo...> wrote: > Hi David, > > Generating a Google App Password seems like a reasonable solution. Trying to guess "long term" wrt Google is probably a fools errand. > > Does a Google App Password work properly with our msmtp implementation? > > Lonnie > > > > > > On Jul 25, 2024, at 2:49 PM, David Kerr <da...@ke...> wrote: > > > > It looks like I can use an App Password and that should survive Sept 30th. See... https://services.google.com/fh/files/emails/b_298235749.pdf > > and link on that leads to... https://support.google.com/mail/answer/185833 > > > > But it may be that long term we will need to support OAuth ?? > > > > David > > > > On Thu, Jul 25, 2024 at 3:35 PM David Kerr <da...@ke...> wrote: > > Google Workspace is turning off username/password access for all applications. I use this for my outbound SMTP Mail Relay on AstLinux. Effective Sept 30th this will no longer work. I do not know if this is just for Google Workspace or for all Google/Gmail accounts. > > > > I either need to be able to authenticate AstLinux to Google's SMTP server with OAuth or switch to some other SMTP mail relay service. > > > > What are my options? > > > > Thanks > > David > > _______________________________________________ > > Astlinux-users mailing list > > Ast...@li... > > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... > > > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... _______________________________________________ Astlinux-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... |
|
From: Lonnie A. <li...@lo...> - 2024-07-26 13:44:56
|
Hi David, Thanks for testing. From what you describe, a Google App Password seems like a great solution, specific to an application so the master user/pass is not replicated over all your AstLinux boxes. Also relatively easy to revoke a lost App Password and regenerate a new one. Lonnie > On Jul 26, 2024, at 8:22 AM, David Kerr <Da...@Ke...> wrote: > > Lonnie, > I have tested google App Password and yes it works for us. The password is automatically generated, 16 characters in length (displayed in 4 blocks of 4, must exclude the spaces when entering it). The password is displayed only once, you cannot retrieve it again, only delete it. And I think I read that if you change your regular password then new app passwords have to be generated again. > > David. > > On Thu, Jul 25, 2024 at 5:24 PM Lonnie Abelbeck <li...@lo...> wrote: > Hi David, > > Generating a Google App Password seems like a reasonable solution. Trying to guess "long term" wrt Google is probably a fools errand. > > Does a Google App Password work properly with our msmtp implementation? > > Lonnie > > > > > > On Jul 25, 2024, at 2:49 PM, David Kerr <da...@ke...> wrote: > > > > It looks like I can use an App Password and that should survive Sept 30th. See... https://services.google.com/fh/files/emails/b_298235749.pdf > > and link on that leads to... https://support.google.com/mail/answer/185833 > > > > But it may be that long term we will need to support OAuth ?? > > > > David > > > > On Thu, Jul 25, 2024 at 3:35 PM David Kerr <da...@ke...> wrote: > > Google Workspace is turning off username/password access for all applications. I use this for my outbound SMTP Mail Relay on AstLinux. Effective Sept 30th this will no longer work. I do not know if this is just for Google Workspace or for all Google/Gmail accounts. > > > > I either need to be able to authenticate AstLinux to Google's SMTP server with OAuth or switch to some other SMTP mail relay service. > > > > What are my options? > > > > Thanks > > David > > _______________________________________________ > > Astlinux-users mailing list > > Ast...@li... > > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... > > > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... |
|
From: David K. <da...@ke...> - 2024-07-26 13:22:34
|
Lonnie, I have tested google App Password and yes it works for us. The password is automatically generated, 16 characters in length (displayed in 4 blocks of 4, must exclude the spaces when entering it). The password is displayed only once, you cannot retrieve it again, only delete it. And I think I read that if you change your regular password then new app passwords have to be generated again. David. On Thu, Jul 25, 2024 at 5:24 PM Lonnie Abelbeck <li...@lo...> wrote: > Hi David, > > Generating a Google App Password seems like a reasonable solution. Trying > to guess "long term" wrt Google is probably a fools errand. > > Does a Google App Password work properly with our msmtp implementation? > > Lonnie > > > > > > On Jul 25, 2024, at 2:49 PM, David Kerr <da...@ke...> wrote: > > > > It looks like I can use an App Password and that should survive Sept > 30th. See... https://services.google.com/fh/files/emails/b_298235749.pdf > > and link on that leads to... > https://support.google.com/mail/answer/185833 > > > > But it may be that long term we will need to support OAuth ?? > > > > David > > > > On Thu, Jul 25, 2024 at 3:35 PM David Kerr <da...@ke...> wrote: > > Google Workspace is turning off username/password access for all > applications. I use this for my outbound SMTP Mail Relay on AstLinux. > Effective Sept 30th this will no longer work. I do not know if this is > just for Google Workspace or for all Google/Gmail accounts. > > > > I either need to be able to authenticate AstLinux to Google's SMTP > server with OAuth or switch to some other SMTP mail relay service. > > > > What are my options? > > > > Thanks > > David > > _______________________________________________ > > Astlinux-users mailing list > > Ast...@li... > > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > > > Donations to support AstLinux are graciously accepted via PayPal to > pa...@kr.... > > > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pa...@kr.... |
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From: Lonnie A. <li...@lo...> - 2024-07-25 21:24:46
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Hi David, Generating a Google App Password seems like a reasonable solution. Trying to guess "long term" wrt Google is probably a fools errand. Does a Google App Password work properly with our msmtp implementation? Lonnie > On Jul 25, 2024, at 2:49 PM, David Kerr <da...@ke...> wrote: > > It looks like I can use an App Password and that should survive Sept 30th. See... https://services.google.com/fh/files/emails/b_298235749.pdf > and link on that leads to... https://support.google.com/mail/answer/185833 > > But it may be that long term we will need to support OAuth ?? > > David > > On Thu, Jul 25, 2024 at 3:35 PM David Kerr <da...@ke...> wrote: > Google Workspace is turning off username/password access for all applications. I use this for my outbound SMTP Mail Relay on AstLinux. Effective Sept 30th this will no longer work. I do not know if this is just for Google Workspace or for all Google/Gmail accounts. > > I either need to be able to authenticate AstLinux to Google's SMTP server with OAuth or switch to some other SMTP mail relay service. > > What are my options? > > Thanks > David > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... |
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From: David K. <da...@ke...> - 2024-07-25 20:04:08
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Google Workspace is turning off username/password access for all applications. I use this for my outbound SMTP Mail Relay on AstLinux. Effective Sept 30th this will no longer work. I do not know if this is just for Google Workspace or for all Google/Gmail accounts. I either need to be able to authenticate AstLinux to Google's SMTP server with OAuth or switch to some other SMTP mail relay service. What are my options? Thanks David |
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From: David K. <da...@ke...> - 2024-07-25 19:49:58
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It looks like I can use an App Password and that should survive Sept 30th. See... https://services.google.com/fh/files/emails/b_298235749.pdf and link on that leads to... https://support.google.com/mail/answer/185833 But it may be that long term we will need to support OAuth ?? David On Thu, Jul 25, 2024 at 3:35 PM David Kerr <da...@ke...> wrote: > Google Workspace is turning off username/password access for all > applications. I use this for my outbound SMTP Mail Relay on AstLinux. > Effective Sept 30th this will no longer work. I do not know if this is > just for Google Workspace or for all Google/Gmail accounts. > > I either need to be able to authenticate AstLinux to Google's SMTP server > with OAuth or switch to some other SMTP mail relay service. > > What are my options? > > Thanks > David > |
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From: Lonnie A. <li...@lo...> - 2024-07-06 13:44:11
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Greetings, Pre-Release Version: astlinux-1.5-6098-6ab1c5 AstLinux Project -> Development https://www.astlinux-project.org/dev.html Of particular note is a new feature to mount S3 Object Storage. -- S3 Object Storage Client (s3fs) https://doc.astlinux-project.org/userdoc:tt_s3_object_storage_client -- One interesting use case for s3fs is to provide a Read/Only mount to an S3 bucket containing a custom AstLinux firmware repository. -- https://doc.astlinux-project.org/userdoc:tt_s3_object_storage_client#custom_firmware_repository -- I'm sure there are many other interesting use cases. If you are so inclined, spin-up the beta Install ISO and give it a test. I personally tested it with: Linode (Akamai), Vultr, Backblaze B2, Cloudflare R2 (all but Vultr supports limited access keys) Complete Pre-Release ChangeLog: https://astlinux-project.org/beta/astlinux-changelog/ChangeLog.txt The "AstLinux Pre-Release ChangeLog" and "Pre-Release Repository URL" entries can be found under the "Development" tab of the AstLinux Project web site ... AstLinux Project -> Development https://www.astlinux-project.org/dev.html AstLinux Team |
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From: David K. <da...@ke...> - 2024-06-20 07:53:14
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I don't know about BLF, but I did find another change I needed to make in my dialplan. The CHANNEL() function has parameters that are unique to SIP / PJSIP. So I was using CHANNEL(recvip) and that failed because it is specific to SIP. I replaced it with CUT(CHANNEL(pjsip,remote_addr),":",1)... the CUT() is necessary because PJSIP returns both the IP address and the port. David. On Sun, Jun 16, 2024 at 6:03 PM The Cadillac Kid via Astlinux-users < ast...@li...> wrote: > has PJ fixed the issues they had with BLF caller ID ? I had tried it a > year or so ago and the NOTIFY messages for BLF were not sending the calling > party... I really need to be making the move in our next production > version to PJ since chan_sip is fully on its way out now.. > > havent tried compiling it on my RPI 5 yet to see if PJ will compile or > not. Chan_sip does without issue. > > On Sunday, June 16, 2024 at 05:40:06 PM EDT, Michael Knill < > mic...@ip...> wrote: > > > Thanks David > > Due to move to PJSIP in our next major release. > > Regards > > Michael Knill > ------------------------------ > *From:* David Kerr <da...@ke...> > *Sent:* Thursday, 13 June 2024 11:31 PM > *To:* AstLinux Users Mailing List <ast...@li...> > *Subject:* Re: [Astlinux-users] PJSIP > > Thanks for the pointer to the custom asterisk commands, for some reason my > eyes didn't pick up on those. So all is good, I used the show > registrations and contacts commands. > > Michael, I've been putting off moving to PJSIP for so long! I'm still on > Asterisk 16. But I decided I should move up to Asterisk 20 and before > doing that would make the shift to PJSIP (as old SIP is officially > deprecated). So I've made the move to PJSIP on 16 and if all is good, will > then move to 20. The process turned out to be easier than I expected. > > There is a python script in the Asterisk source tree that helps a lot. > It's not perfect, but it goes a long way towards creating a working > pjsip.conf file out of an existing sip.conf file. It includes a > commented-out section at the top which lists things it could not migrate, > somewhat surprisingly for me, that included "username" fields that need to > go into the authentication sections... I had to correct those manually. > And it did not convert "fullname" fields which I think need to go into a > callerid field, not done that yet. > > And then in extensions.conf you have to replace all Dial() destinations > that are "SIP/<number>" with "PJSIP/<number>" and if you have syntax that > looks like "SIP/<trunk>/<number>" then those need to change to > "PJSIP/<number>@<trunk>" > > And the last thing to note is don't have both SIP and PJSIP at the same > time... at least not using both your old sip.conf and new pjsip.conf > files. Either remove/rename your old sip.conf or do what I did and add a > noload statement to modules.conf for chan_sip. > > I still have to test all my esoteric paths in extensions.conf, but basic > ingoing and outgoing calls are working for me. > > David. > > > On Wed, Jun 12, 2024 at 8:52 PM Michael Knill < > mic...@ip...> wrote: > > Yes Im going to need to go down this path at some stage but Im not looking > forward to it 🙁 > > Regards > > Michael Knill > ------------------------------ > *From:* Home <da...@ry...> > *Sent:* Thursday, 13 June 2024 8:27 AM > *To:* AstLinux Users Mailing List <ast...@li...> > *Subject:* Re: [Astlinux-users] PJSIP > > For what it may be worth, I've found... > > pjsip show endpoints > > and > > pjsip show contacts > > ... to be useful. > > Dan > > > > -------- Original message -------- > From: Lonnie Abelbeck <li...@lo...> > Date: 6/12/24 6:05 PM (GMT-05:00) > To: AstLinux Users Mailing List <ast...@li...> > Subject: Re: [Astlinux-users] PJSIP > > Hi David, > > In the Prefs -> Status Tab Options:, there are 4 pairs of these: > -- > Custom Asterisk Name: > Custom Asterisk Command: > -- > Which you can label and call what commands you want. > > And uncheck: > -- > Show SIP Trunk Registrations > Show SIP Peer Status > -- > > Though I don't think there is an exact equivalent from chan_sip to > chan_pjsip for status. I still use chan_sip. > > > Lonnie > > > > > On Jun 12, 2024, at 4:17 PM, David Kerr <da...@ke...> wrote: > > > > It looks like > > > > pjsip list (or show) registrations > > pjsip list (or show) contacts > > > > Gets closest to the old versions? Should the prefs panel be updated to > allow a command to be provided? > > > > David > > > > On Wed, Jun 12, 2024 at 5:10 PM David Kerr <da...@ke...> wrote: > > I'm embarking on a long overdue conversion from SIP to PJSIP in my > Asterisk configuration. I think I have it mostly working now but I notice > that in the status page the commands to show SIP trunk and SIP peer status > no longer exist (I have noload for chan_sip.so). > > > > SIP Trunk Registrations:No such command 'sip show registry' (type 'core > show help sip show' for other possible commands) > > > > SIP Peer Status:No such command 'sip show peers' (type 'core show help > sip show' for other possible commands) > > > > > > Are there alternative commands we could use? > > > > Thanks > > David > > > > > > _______________________________________________ > > Astlinux-users mailing list > > Ast...@li... > > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > > > Donations to support AstLinux are graciously accepted via PayPal to > pa...@kr.... > > > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pa...@kr.... > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pa...@kr.... > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pa...@kr.... > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pa...@kr.... |
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From: The C. K. <eld...@ya...> - 2024-06-16 22:02:54
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has PJ fixed the issues they had with BLF caller ID ? I had tried it a year or so ago and the NOTIFY messages for BLF were not sending the calling party... I really need to be making the move in our next production version to PJ since chan_sip is fully on its way out now..
havent tried compiling it on my RPI 5 yet to see if PJ will compile or not. Chan_sip does without issue.
On Sunday, June 16, 2024 at 05:40:06 PM EDT, Michael Knill <mic...@ip...> wrote:
Thanks David
Due to move to PJSIP in our next major release.
Regards
Michael Knill
From: David Kerr <da...@ke...>
Sent: Thursday, 13 June 2024 11:31 PM
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] PJSIP Thanks for the pointer to the custom asterisk commands, for some reason my eyes didn't pick up on those. So all is good, I used the show registrations and contacts commands.
Michael, I've been putting off moving to PJSIP for so long! I'm still on Asterisk 16. But I decided I should move up to Asterisk 20 and before doing that would make the shift to PJSIP (as old SIP is officially deprecated). So I've made the move to PJSIP on 16 and if all is good, will then move to 20. The process turned out to be easier than I expected.
There is a python script in the Asterisk source tree that helps a lot. It's not perfect, but it goes a long way towards creating a working pjsip.conf file out of an existing sip.conf file. It includes a commented-out section at the top which lists things it could not migrate, somewhat surprisingly for me, that included "username" fields that need to go into the authentication sections... I had to correct those manually. And it did not convert "fullname" fields which I think need to go into a callerid field, not done that yet.
And then in extensions.conf you have to replace all Dial() destinations that are "SIP/<number>" with "PJSIP/<number>" and if you have syntax that looks like "SIP/<trunk>/<number>" then those need to change to "PJSIP/<number>@<trunk>"
And the last thing to note is don't have both SIP and PJSIP at the same time... at least not using both your old sip.conf and new pjsip.conf files. Either remove/rename your old sip.conf or do what I did and add a noload statement to modules.conf for chan_sip.
I still have to test all my esoteric paths in extensions.conf, but basic ingoing and outgoing calls are working for me.
David.
On Wed, Jun 12, 2024 at 8:52 PM Michael Knill <mic...@ip...> wrote:
Yes Im going to need to go down this path at some stage but Im not looking forward to it 🙁
Regards
Michael Knill
From: Home <da...@ry...>
Sent: Thursday, 13 June 2024 8:27 AM
To: AstLinux Users Mailing List <ast...@li...>
Subject: Re: [Astlinux-users] PJSIP For what it may be worth, I've found...
pjsip show endpoints
and
pjsip show contacts
... to be useful.
Dan
-------- Original message --------From: Lonnie Abelbeck <li...@lo...>Date: 6/12/24 6:05 PM (GMT-05:00) To: AstLinux Users Mailing List <ast...@li...>Subject: Re: [Astlinux-users] PJSIP
Hi David,
In the Prefs -> Status Tab Options:, there are 4 pairs of these:
--
Custom Asterisk Name:
Custom Asterisk Command:
--
Which you can label and call what commands you want.
And uncheck:
--
Show SIP Trunk Registrations
Show SIP Peer Status
--
Though I don't think there is an exact equivalent from chan_sip to chan_pjsip for status. I still use chan_sip.
Lonnie
> On Jun 12, 2024, at 4:17 PM, David Kerr <da...@ke...> wrote:
>
> It looks like
>
> pjsip list (or show) registrations
> pjsip list (or show) contacts
>
> Gets closest to the old versions? Should the prefs panel be updated to allow a command to be provided?
>
> David
>
> On Wed, Jun 12, 2024 at 5:10 PM David Kerr <da...@ke...> wrote:
> I'm embarking on a long overdue conversion from SIP to PJSIP in my Asterisk configuration. I think I have it mostly working now but I notice that in the status page the commands to show SIP trunk and SIP peer status no longer exist (I have noload for chan_sip.so).
>
> SIP Trunk Registrations:No such command 'sip show registry' (type 'core show help sip show' for other possible commands)
>
> SIP Peer Status:No such command 'sip show peers' (type 'core show help sip show' for other possible commands)
>
>
> Are there alternative commands we could use?
>
> Thanks
> David
>
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to pa...@kr...._______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... |
|
From: Michael K. <mic...@ip...> - 2024-06-16 21:39:34
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Thanks David Due to move to PJSIP in our next major release. Regards Michael Knill ________________________________ From: David Kerr <da...@ke...> Sent: Thursday, 13 June 2024 11:31 PM To: AstLinux Users Mailing List <ast...@li...> Subject: Re: [Astlinux-users] PJSIP Thanks for the pointer to the custom asterisk commands, for some reason my eyes didn't pick up on those. So all is good, I used the show registrations and contacts commands. Michael, I've been putting off moving to PJSIP for so long! I'm still on Asterisk 16. But I decided I should move up to Asterisk 20 and before doing that would make the shift to PJSIP (as old SIP is officially deprecated). So I've made the move to PJSIP on 16 and if all is good, will then move to 20. The process turned out to be easier than I expected. There is a python script in the Asterisk source tree that helps a lot. It's not perfect, but it goes a long way towards creating a working pjsip.conf file out of an existing sip.conf file. It includes a commented-out section at the top which lists things it could not migrate, somewhat surprisingly for me, that included "username" fields that need to go into the authentication sections... I had to correct those manually. And it did not convert "fullname" fields which I think need to go into a callerid field, not done that yet. And then in extensions.conf you have to replace all Dial() destinations that are "SIP/<number>" with "PJSIP/<number>" and if you have syntax that looks like "SIP/<trunk>/<number>" then those need to change to "PJSIP/<number>@<trunk>" And the last thing to note is don't have both SIP and PJSIP at the same time... at least not using both your old sip.conf and new pjsip.conf files. Either remove/rename your old sip.conf or do what I did and add a noload statement to modules.conf for chan_sip. I still have to test all my esoteric paths in extensions.conf, but basic ingoing and outgoing calls are working for me. David. On Wed, Jun 12, 2024 at 8:52 PM Michael Knill <mic...@ip...<mailto:mic...@ip...>> wrote: Yes Im going to need to go down this path at some stage but Im not looking forward to it 🙁 Regards Michael Knill ________________________________ From: Home <da...@ry...<mailto:da...@ry...>> Sent: Thursday, 13 June 2024 8:27 AM To: AstLinux Users Mailing List <ast...@li...<mailto:ast...@li...>> Subject: Re: [Astlinux-users] PJSIP For what it may be worth, I've found... pjsip show endpoints and pjsip show contacts ... to be useful. Dan -------- Original message -------- From: Lonnie Abelbeck <li...@lo...<mailto:li...@lo...>> Date: 6/12/24 6:05 PM (GMT-05:00) To: AstLinux Users Mailing List <ast...@li...<mailto:ast...@li...>> Subject: Re: [Astlinux-users] PJSIP Hi David, In the Prefs -> Status Tab Options:, there are 4 pairs of these: -- Custom Asterisk Name: Custom Asterisk Command: -- Which you can label and call what commands you want. And uncheck: -- Show SIP Trunk Registrations Show SIP Peer Status -- Though I don't think there is an exact equivalent from chan_sip to chan_pjsip for status. I still use chan_sip. Lonnie > On Jun 12, 2024, at 4:17 PM, David Kerr <da...@ke...<mailto:da...@ke...>> wrote: > > It looks like > > pjsip list (or show) registrations > pjsip list (or show) contacts > > Gets closest to the old versions? Should the prefs panel be updated to allow a command to be provided? > > David > > On Wed, Jun 12, 2024 at 5:10 PM David Kerr <da...@ke...<mailto:da...@ke...>> wrote: > I'm embarking on a long overdue conversion from SIP to PJSIP in my Asterisk configuration. I think I have it mostly working now but I notice that in the status page the commands to show SIP trunk and SIP peer status no longer exist (I have noload for chan_sip.so). > > SIP Trunk Registrations:No such command 'sip show registry' (type 'core show help sip show' for other possible commands) > > SIP Peer Status:No such command 'sip show peers' (type 'core show help sip show' for other possible commands) > > > Are there alternative commands we could use? > > Thanks > David > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li...<mailto:Ast...@li...> > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr...<mailto:pa...@kr...>. _______________________________________________ Astlinux-users mailing list Ast...@li...<mailto:Ast...@li...> https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pa...@kr...<mailto:pa...@kr...>. _______________________________________________ Astlinux-users mailing list Ast...@li...<mailto:Ast...@li...> https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pa...@kr...<mailto:pa...@kr...>. |
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From: David K. <da...@ke...> - 2024-06-13 14:54:36
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I first stumbled on AstLinux in 2008. Sixteen years later I am still using it as my router, firewall and land-line phone system in my house. Who would have thought. I love that I have full control and can add features that improve how it works for me. I'm writing this because I confess there are times when I think I should use something more mainstream... maybe I should separate the router/firewall from the phone system (which over the years has become less and less important). E.g. maybe I should just go with one of Unifi's gateways. Or use pfSense or OPNSense and a separate FreePBX or Asterisk. But every time I look at it, I land back at AstLinux. My home network is non-trivial. I have a mix of 1Gbps, 2.5Gbps and 10Gbps attached systems. I have a proliferation of IoT devices on multiple VLANs, I have Raspberry Pi's, A humongous NAS, and a Proxmox server on which I have dozen+ containers and multiple VMs... one of which is AstLinux at the center of it all. A few months ago fiber-based internet finally arrived, so multi-gig internet is now possible for me. I started looking for a system with 10Gbps NBase-T or SFP+ ports that could become my Proxmox server. I have not bought anything yet (because I don't *really* need it) but it got me looking again at pfSense/OPNSense and I've again discovered how good AstLinux is. The core issue for those is that they are based on FreeBSD, and it has very poor network support when running as a guest VM. The maximum throughput on a VirtIO network (to a Proxmox hosted Linux container or VM) is 2-3Gbps. Which may sound okay, but is in fact lousy... AstLinux can achieve 10x that, comfortably 25Gbps. AstLinux can route between two subnets (different VLANs) at >10Gbps... and may be able to do better if I separate the VLANs into different interfaces (rather than VLAN tags on the same interface). I could pass through the SFP+ PCI h/w to FreeBSD, but then the rest of my containers and VMs need to route through that to my network... and run into the VirtIO limitation. SFP+ devices support SR-IOV that virtualizes the network device at the h/w layer... but no one appears to have got that working with FreeBSD guests on Proxmox (whereas it works fine with Linux). Which left me looking for a Linux-based router/firewall of which only OpenWRT comes close to being mainstream, but again it isn't a good fit either... it's really intended to run directly on the h/w. Support for VM guest or even running in a container exists, but on investigation I concluded it may be okay for dev/test, but really not for production (main issue is how updates would be applied). So... AstLinux turns out to be (in my opinion) best-of-breed. It's a shame that it is not more widely known. Now it's not perfect... I would love it if my enhancements were merged into the mainline, but that aside there are a few things I would really like to see done... 1) Bring our build environment up-to-date with buildroot, so it is much easier to keep in sync with updates contributed from the wider user base that has. 2) Add a package manager (is that even possible with buildroot). 3) Modernize the user interface Of course, that is a very large project for which none of us have the time. But some of the design decisions / constraints that applied to AstLinux 16 years ago are really not relevant any more. So there you have it... sixteen years in and AstLinux still rules for me. David. |
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From: David K. <da...@ke...> - 2024-06-13 13:31:37
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Thanks for the pointer to the custom asterisk commands, for some reason my eyes didn't pick up on those. So all is good, I used the show registrations and contacts commands. Michael, I've been putting off moving to PJSIP for so long! I'm still on Asterisk 16. But I decided I should move up to Asterisk 20 and before doing that would make the shift to PJSIP (as old SIP is officially deprecated). So I've made the move to PJSIP on 16 and if all is good, will then move to 20. The process turned out to be easier than I expected. There is a python script in the Asterisk source tree that helps a lot. It's not perfect, but it goes a long way towards creating a working pjsip.conf file out of an existing sip.conf file. It includes a commented-out section at the top which lists things it could not migrate, somewhat surprisingly for me, that included "username" fields that need to go into the authentication sections... I had to correct those manually. And it did not convert "fullname" fields which I think need to go into a callerid field, not done that yet. And then in extensions.conf you have to replace all Dial() destinations that are "SIP/<number>" with "PJSIP/<number>" and if you have syntax that looks like "SIP/<trunk>/<number>" then those need to change to "PJSIP/<number>@<trunk>" And the last thing to note is don't have both SIP and PJSIP at the same time... at least not using both your old sip.conf and new pjsip.conf files. Either remove/rename your old sip.conf or do what I did and add a noload statement to modules.conf for chan_sip. I still have to test all my esoteric paths in extensions.conf, but basic ingoing and outgoing calls are working for me. David. On Wed, Jun 12, 2024 at 8:52 PM Michael Knill < mic...@ip...> wrote: > Yes Im going to need to go down this path at some stage but Im not looking > forward to it 🙁 > > Regards > > Michael Knill > ------------------------------ > *From:* Home <da...@ry...> > *Sent:* Thursday, 13 June 2024 8:27 AM > *To:* AstLinux Users Mailing List <ast...@li...> > *Subject:* Re: [Astlinux-users] PJSIP > > For what it may be worth, I've found... > > pjsip show endpoints > > and > > pjsip show contacts > > ... to be useful. > > Dan > > > > -------- Original message -------- > From: Lonnie Abelbeck <li...@lo...> > Date: 6/12/24 6:05 PM (GMT-05:00) > To: AstLinux Users Mailing List <ast...@li...> > Subject: Re: [Astlinux-users] PJSIP > > Hi David, > > In the Prefs -> Status Tab Options:, there are 4 pairs of these: > -- > Custom Asterisk Name: > Custom Asterisk Command: > -- > Which you can label and call what commands you want. > > And uncheck: > -- > Show SIP Trunk Registrations > Show SIP Peer Status > -- > > Though I don't think there is an exact equivalent from chan_sip to > chan_pjsip for status. I still use chan_sip. > > > Lonnie > > > > > On Jun 12, 2024, at 4:17 PM, David Kerr <da...@ke...> wrote: > > > > It looks like > > > > pjsip list (or show) registrations > > pjsip list (or show) contacts > > > > Gets closest to the old versions? Should the prefs panel be updated to > allow a command to be provided? > > > > David > > > > On Wed, Jun 12, 2024 at 5:10 PM David Kerr <da...@ke...> wrote: > > I'm embarking on a long overdue conversion from SIP to PJSIP in my > Asterisk configuration. I think I have it mostly working now but I notice > that in the status page the commands to show SIP trunk and SIP peer status > no longer exist (I have noload for chan_sip.so). > > > > SIP Trunk Registrations:No such command 'sip show registry' (type 'core > show help sip show' for other possible commands) > > > > SIP Peer Status:No such command 'sip show peers' (type 'core show help > sip show' for other possible commands) > > > > > > Are there alternative commands we could use? > > > > Thanks > > David > > > > > > _______________________________________________ > > Astlinux-users mailing list > > Ast...@li... > > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > > > Donations to support AstLinux are graciously accepted via PayPal to > pa...@kr.... > > > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pa...@kr.... > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pa...@kr.... > |
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From: Michael K. <mic...@ip...> - 2024-06-13 00:51:49
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Yes Im going to need to go down this path at some stage but Im not looking forward to it 🙁 Regards Michael Knill ________________________________ From: Home <da...@ry...> Sent: Thursday, 13 June 2024 8:27 AM To: AstLinux Users Mailing List <ast...@li...> Subject: Re: [Astlinux-users] PJSIP For what it may be worth, I've found... pjsip show endpoints and pjsip show contacts ... to be useful. Dan -------- Original message -------- From: Lonnie Abelbeck <li...@lo...> Date: 6/12/24 6:05 PM (GMT-05:00) To: AstLinux Users Mailing List <ast...@li...> Subject: Re: [Astlinux-users] PJSIP Hi David, In the Prefs -> Status Tab Options:, there are 4 pairs of these: -- Custom Asterisk Name: Custom Asterisk Command: -- Which you can label and call what commands you want. And uncheck: -- Show SIP Trunk Registrations Show SIP Peer Status -- Though I don't think there is an exact equivalent from chan_sip to chan_pjsip for status. I still use chan_sip. Lonnie > On Jun 12, 2024, at 4:17 PM, David Kerr <da...@ke...> wrote: > > It looks like > > pjsip list (or show) registrations > pjsip list (or show) contacts > > Gets closest to the old versions? Should the prefs panel be updated to allow a command to be provided? > > David > > On Wed, Jun 12, 2024 at 5:10 PM David Kerr <da...@ke...> wrote: > I'm embarking on a long overdue conversion from SIP to PJSIP in my Asterisk configuration. I think I have it mostly working now but I notice that in the status page the commands to show SIP trunk and SIP peer status no longer exist (I have noload for chan_sip.so). > > SIP Trunk Registrations:No such command 'sip show registry' (type 'core show help sip show' for other possible commands) > > SIP Peer Status:No such command 'sip show peers' (type 'core show help sip show' for other possible commands) > > > Are there alternative commands we could use? > > Thanks > David > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... _______________________________________________ Astlinux-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... |
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