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From: Michael K. <li...@mk...> - 2022-04-24 14:54:22
|
Thanks for all your comments. I close this thread here, cause now I am in direct contact with the company who manufactures the analogue modems. And they offered me to debug their CallerID function with an Cisco SPA112. >> Am 20.04.2022 um 21:24 schrieb Michael Knill <mic...@ip...>: >> >> >> Ah I understand now. >> So do you know what the modem is expecting from a CLIP perspective or is this what you are trying to find? > > Exactly, this is what I am trying to find. CLIP works for me with an old analog DECT phone with 2 of the multiple variations. So the SPA112 are working in general. > >> What is the interface from the modem to the Emergency Console? Can you build something that connects directly to the console? > > No, I don‘t have access, it is a custom product from a company here in Germany, who are only producing elevation communication equipment (but it seems only analog). > >> It certainly seems like something that needs to be updated. >> >> Regards >> Michael Knill >> >> From: Michael Keuter <li...@mk...> >> Reply to: AstLinux List <ast...@li...> >> Date: Wednesday, 20 April 2022 at 9:47 pm >> To: AstLinux List <ast...@li...> >> Subject: Re: [Astlinux-users] Analogue CLIP (CallerID) >> >> >> >> >> Am 20.04.2022 um 01:36 schrieb Michael Knill <mic...@ip...>: >> >> Hi Michael >> >> I'm a little confused here. I was assuming this: >> elevator -> PSTN -> SPA112 -> Asterisk >> >> elevator -> PSTN -> Asterisk (via Voip to PSTN)) -> SPA112 -> Modem -> Emergency Support console (special construction modem with "handset") >> >> Update: I could simulate the internal part with an 20 year old Siemens analogue DECT phone at the SPA112. >> CallerID (CLIP) works fine after the first ring with "Bellcore (bell-202 or v.23)", "ETSI FSK (bell-202 or v.23)" and "ETSI FSK with PR (UK) (bell-202 + v.23)". Even the CallerID name is shown. >> >> When I set to e.g. "DTMF Denmark" the phone shows only "External Call". >> >> >> What am I missing? >> >> Regards >> Michael Knill >> >> From: Michael Keuter <li...@mk...> >> Reply to: AstLinux List <ast...@li...> >> Date: Wednesday, 20 April 2022 at 9:20 am >> To: AstLinux List <ast...@li...> >> Subject: Re: [Astlinux-users] Analogue CLIP (CallerID) >> >> Hi Michael, >> >> the „potential“ customer has these modems for a long time, and there are no alternatives (like SIP) at the market. >> >> And sure, in SIP I see the correct CallerID, but the modem don‘t see the „right“ CallerID generated by the ATA. >> >> Sent from a mobile device. >> >> Michael Keuter >> >> >> >> Am 20.04.2022 um 00:58 schrieb Michael Knill <mic...@ip...>: >> >> Hi Michael >> >> Just wondering why you need analogue modems to receive the calls from the elevators? >> Have you done a SIP Debug of the traffic coming from the SPA112’s to see if the number is anywhere in the SIP Invite? >> Do you know the particular standard for this in your country and is this supported by the SPA112’s? >> >> I have never used analogue FXO for anything sorry. >> >> Regards >> Michael Knill >> >> From: Michael Keuter <li...@mk...> >> Reply to: AstLinux List <ast...@li...> >> Date: Wednesday, 20 April 2022 at 2:23 am >> To: AstLinux List <ast...@li...> >> Subject: [Astlinux-users] Analogue CLIP (CallerID) >> >> Hi list, >> >> I am trying to install an Asterisk PBX for an elevator emergency central in Germany. >> The callees are 6 analogue modems who receive calls from the elevators. >> >> The main issue is that the modems need to identify the elevator via a so called analogue CLIP (Calling Line Identification Presentation) where the calling number is shown to the receiving modem. >> It is not the Asterisk CALLERID(num) but a special "message" which is transfered for analogue phones between the first 2 ringings (but I think this is generated from the CALLERID(num)). >> There are several different methods for that, that can be set in the ATA (Cisco SPA112, latest EOL firmware): >> >> <image001.png><image002.png> >> >> <image003.png><image004.png> >> >> I tried every possible combination of "Caller ID Method" and "Caller ID FSK Standard", but without success. >> >> Has anybody on the list made experiences with analogue CLIP? >> >> Michael Michael http://www.mksolutions.info |
|
From: Michael K. <li...@mk...> - 2022-04-20 21:07:12
|
Sent from a mobile device. Michael Keuter > Am 20.04.2022 um 21:24 schrieb Michael Knill <mic...@ip...>: > > > Ah I understand now. > So do you know what the modem is expecting from a CLIP perspective or is this what you are trying to find? Exactly, this is what I am trying to find. CLIP works for me with an old analog DECT phone with 2 of the multiple variations. So the SPA112 are working in general. > What is the interface from the modem to the Emergency Console? Can you build something that connects directly to the console? No, I don‘t have access, it is a custom product from a company here in Germany, who are only producing elevation communication equipment (but it seems only analog). > It certainly seems like something that needs to be updated. > > Regards > Michael Knill > > From: Michael Keuter <li...@mk...> > Reply to: AstLinux List <ast...@li...> > Date: Wednesday, 20 April 2022 at 9:47 pm > To: AstLinux List <ast...@li...> > Subject: Re: [Astlinux-users] Analogue CLIP (CallerID) > > > > > Am 20.04.2022 um 01:36 schrieb Michael Knill <mic...@ip...>: > > Hi Michael > > I'm a little confused here. I was assuming this: > elevator -> PSTN -> SPA112 -> Asterisk > > elevator -> PSTN -> Asterisk (via Voip to PSTN)) -> SPA112 -> Modem -> Emergency Support console (special construction modem with "handset") > > Update: I could simulate the internal part with an 20 year old Siemens analogue DECT phone at the SPA112. > CallerID (CLIP) works fine after the first ring with "Bellcore (bell-202 or v.23)", "ETSI FSK (bell-202 or v.23)" and "ETSI FSK with PR (UK) (bell-202 + v.23)". Even the CallerID name is shown. > > When I set to e.g. "DTMF Denmark" the phone shows only "External Call". > > > What am I missing? > > Regards > Michael Knill > > From: Michael Keuter <li...@mk...> > Reply to: AstLinux List <ast...@li...> > Date: Wednesday, 20 April 2022 at 9:20 am > To: AstLinux List <ast...@li...> > Subject: Re: [Astlinux-users] Analogue CLIP (CallerID) > > Hi Michael, > > the „potential“ customer has these modems for a long time, and there are no alternatives (like SIP) at the market. > > And sure, in SIP I see the correct CallerID, but the modem don‘t see the „right“ CallerID generated by the ATA. > > Sent from a mobile device. > > Michael Keuter > > > > Am 20.04.2022 um 00:58 schrieb Michael Knill <mic...@ip...>: > > Hi Michael > > Just wondering why you need analogue modems to receive the calls from the elevators? > Have you done a SIP Debug of the traffic coming from the SPA112’s to see if the number is anywhere in the SIP Invite? > Do you know the particular standard for this in your country and is this supported by the SPA112’s? > > I have never used analogue FXO for anything sorry. > > Regards > Michael Knill > > From: Michael Keuter <li...@mk...> > Reply to: AstLinux List <ast...@li...> > Date: Wednesday, 20 April 2022 at 2:23 am > To: AstLinux List <ast...@li...> > Subject: [Astlinux-users] Analogue CLIP (CallerID) > > Hi list, > > I am trying to install an Asterisk PBX for an elevator emergency central in Germany. > The callees are 6 analogue modems who receive calls from the elevators. > > The main issue is that the modems need to identify the elevator via a so called analogue CLIP (Calling Line Identification Presentation) where the calling number is shown to the receiving modem. > It is not the Asterisk CALLERID(num) but a special "message" which is transfered for analogue phones between the first 2 ringings (but I think this is generated from the CALLERID(num)). > There are several different methods for that, that can be set in the ATA (Cisco SPA112, latest EOL firmware): > > <image001.png><image002.png> > > <image003.png><image004.png> > > I tried every possible combination of "Caller ID Method" and "Caller ID FSK Standard", but without success. > > Has anybody on the list made experiences with analogue CLIP? > > Michael > > 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. <mic...@ip...> - 2022-04-20 19:24:02
|
Ah I understand now. So do you know what the modem is expecting from a CLIP perspective or is this what you are trying to find? What is the interface from the modem to the Emergency Console? Can you build something that connects directly to the console? It certainly seems like something that needs to be updated. Regards Michael Knill From: Michael Keuter <li...@mk...> Reply to: AstLinux List <ast...@li...> Date: Wednesday, 20 April 2022 at 9:47 pm To: AstLinux List <ast...@li...> Subject: Re: [Astlinux-users] Analogue CLIP (CallerID) Am 20.04.2022 um 01:36 schrieb Michael Knill <mic...@ip...<mailto:mic...@ip...>>: Hi Michael I'm a little confused here. I was assuming this: elevator -> PSTN -> SPA112 -> Asterisk elevator -> PSTN -> Asterisk (via Voip to PSTN)) -> SPA112 -> Modem -> Emergency Support console (special construction modem with "handset") Update: I could simulate the internal part with an 20 year old Siemens analogue DECT phone at the SPA112. CallerID (CLIP) works fine after the first ring with "Bellcore (bell-202 or v.23)", "ETSI FSK (bell-202 or v.23)" and "ETSI FSK with PR (UK) (bell-202 + v.23)". Even the CallerID name is shown. When I set to e.g. "DTMF Denmark" the phone shows only "External Call". What am I missing? Regards Michael Knill From: Michael Keuter <li...@mk...<mailto:li...@mk...>> Reply to: AstLinux List <ast...@li...<mailto:ast...@li...>> Date: Wednesday, 20 April 2022 at 9:20 am To: AstLinux List <ast...@li...<mailto:ast...@li...>> Subject: Re: [Astlinux-users] Analogue CLIP (CallerID) Hi Michael, the „potential“ customer has these modems for a long time, and there are no alternatives (like SIP) at the market. And sure, in SIP I see the correct CallerID, but the modem don‘t see the „right“ CallerID generated by the ATA. Sent from a mobile device. Michael Keuter Am 20.04.2022 um 00:58 schrieb Michael Knill <mic...@ip...<mailto:mic...@ip...>>: Hi Michael Just wondering why you need analogue modems to receive the calls from the elevators? Have you done a SIP Debug of the traffic coming from the SPA112’s to see if the number is anywhere in the SIP Invite? Do you know the particular standard for this in your country and is this supported by the SPA112’s? I have never used analogue FXO for anything sorry. Regards Michael Knill From: Michael Keuter <li...@mk...<mailto:li...@mk...>> Reply to: AstLinux List <ast...@li...<mailto:ast...@li...>> Date: Wednesday, 20 April 2022 at 2:23 am To: AstLinux List <ast...@li...<mailto:ast...@li...>> Subject: [Astlinux-users] Analogue CLIP (CallerID) Hi list, I am trying to install an Asterisk PBX for an elevator emergency central in Germany. The callees are 6 analogue modems who receive calls from the elevators. The main issue is that the modems need to identify the elevator via a so called analogue CLIP (Calling Line Identification Presentation) where the calling number is shown to the receiving modem. It is not the Asterisk CALLERID(num) but a special "message" which is transfered for analogue phones between the first 2 ringings (but I think this is generated from the CALLERID(num)). There are several different methods for that, that can be set in the ATA (Cisco SPA112, latest EOL firmware): <image001.png><image002.png> <image003.png><image004.png> I tried every possible combination of "Caller ID Method" and "Caller ID FSK Standard", but without success. Has anybody on the list made experiences with analogue CLIP? Michael Michael http://www.mksolutions.info |
|
From: David K. <da...@ke...> - 2022-04-20 15:16:42
|
Google turned up a lengthy discussion of CallerID not working on SPA112's but it is from many years ago and according to the discussion was fixed by Cisco. But it is possible that you might learn something useful from reading that thread... https://community.cisco.com/t5/atas-gateways-and-accessories/spa112-not-allowing-caller-id-to-show/td-p/1826351 David On Wed, Apr 20, 2022 at 7:46 AM Michael Keuter <li...@mk...> wrote: > > > Am 20.04.2022 um 01:36 schrieb Michael Knill < > mic...@ip...>: > > Hi Michael > > I'm a little confused here. I was assuming this: > elevator -> PSTN -> SPA112 -> Asterisk > > > elevator -> PSTN -> Asterisk (via Voip to PSTN)) -> SPA112 -> Modem -> > Emergency Support console (special construction modem with "handset") > > Update: I could simulate the internal part with an 20 year old Siemens > analogue DECT phone at the SPA112. > CallerID (CLIP) works fine after the first ring with "Bellcore (bell-202 > or v.23)", "ETSI FSK (bell-202 or v.23)" and "ETSI FSK with PR (UK) > (bell-202 + v.23)". Even the CallerID name is shown. > > When I set to e.g. "DTMF Denmark" the phone shows only "External Call". > > What am I missing? > > Regards > Michael Knill > > From: Michael Keuter <li...@mk...> > Reply to: AstLinux List <ast...@li...> > Date: Wednesday, 20 April 2022 at 9:20 am > To: AstLinux List <ast...@li...> > Subject: Re: [Astlinux-users] Analogue CLIP (CallerID) > > Hi Michael, > > the „potential“ customer has these modems for a long time, and there are > no alternatives (like SIP) at the market. > > And sure, in SIP I see the correct CallerID, but the modem don‘t see the > „right“ CallerID generated by the ATA. > > Sent from a mobile device. > > Michael Keuter > > > Am 20.04.2022 um 00:58 schrieb Michael Knill < > mic...@ip...>: > > Hi Michael > > Just wondering why you need analogue modems to receive the calls from the > elevators? > Have you done a SIP Debug of the traffic coming from the SPA112’s to see > if the number is anywhere in the SIP Invite? > Do you know the particular standard for this in your country and is this > supported by the SPA112’s? > > I have never used analogue FXO for anything sorry. > > Regards > Michael Knill > > From: Michael Keuter <li...@mk...> > Reply to: AstLinux List <ast...@li...> > Date: Wednesday, 20 April 2022 at 2:23 am > To: AstLinux List <ast...@li...> > Subject: [Astlinux-users] Analogue CLIP (CallerID) > > Hi list, > > I am trying to install an Asterisk PBX for an elevator emergency central > in Germany. > The callees are 6 analogue modems who receive calls from the elevators. > > The main issue is that the modems need to identify the elevator via a so > called analogue CLIP (Calling Line Identification Presentation) where the > calling number is shown to the receiving modem. > It is not the Asterisk CALLERID(num) but a special "message" which is > transfered for analogue phones between the first 2 ringings (but I think > this is generated from the CALLERID(num)). > There are several different methods for that, that can be set in the ATA > (Cisco SPA112, latest EOL firmware): > > <image001.png><image002.png> > > <image003.png><image004.png> > > I tried every possible combination of "Caller ID Method" and "Caller ID > FSK Standard", but without success. > > Has anybody on the list made experiences with analogue CLIP? > > Michael > > > 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...> - 2022-04-20 11:46:27
|
> Am 20.04.2022 um 01:36 schrieb Michael Knill <mic...@ip... <mailto:mic...@ip...>>: > > Hi Michael > > I'm a little confused here. I was assuming this: > elevator -> PSTN -> SPA112 -> Asterisk elevator -> PSTN -> Asterisk (via Voip to PSTN)) -> SPA112 -> Modem -> Emergency Support console (special construction modem with "handset") Update: I could simulate the internal part with an 20 year old Siemens analogue DECT phone at the SPA112. CallerID (CLIP) works fine after the first ring with "Bellcore (bell-202 or v.23)", "ETSI FSK (bell-202 or v.23)" and "ETSI FSK with PR (UK) (bell-202 + v.23)". Even the CallerID name is shown. When I set to e.g. "DTMF Denmark" the phone shows only "External Call". > What am I missing? > > Regards > Michael Knill > > From: Michael Keuter <li...@mk...> > Reply to: AstLinux List <ast...@li...> > Date: Wednesday, 20 April 2022 at 9:20 am > To: AstLinux List <ast...@li...> > Subject: Re: [Astlinux-users] Analogue CLIP (CallerID) > > Hi Michael, > > the „potential“ customer has these modems for a long time, and there are no alternatives (like SIP) at the market. > > And sure, in SIP I see the correct CallerID, but the modem don‘t see the „right“ CallerID generated by the ATA. > > Sent from a mobile device. > > Michael Keuter > > >> Am 20.04.2022 um 00:58 schrieb Michael Knill <mic...@ip...>: >> >> Hi Michael >> >> Just wondering why you need analogue modems to receive the calls from the elevators? >> Have you done a SIP Debug of the traffic coming from the SPA112’s to see if the number is anywhere in the SIP Invite? >> Do you know the particular standard for this in your country and is this supported by the SPA112’s? >> >> I have never used analogue FXO for anything sorry. >> >> Regards >> Michael Knill >> >> From: Michael Keuter <li...@mk...> >> Reply to: AstLinux List <ast...@li...> >> Date: Wednesday, 20 April 2022 at 2:23 am >> To: AstLinux List <ast...@li...> >> Subject: [Astlinux-users] Analogue CLIP (CallerID) >> >> Hi list, >> >> I am trying to install an Asterisk PBX for an elevator emergency central in Germany. >> The callees are 6 analogue modems who receive calls from the elevators. >> >> The main issue is that the modems need to identify the elevator via a so called analogue CLIP (Calling Line Identification Presentation) where the calling number is shown to the receiving modem. >> It is not the Asterisk CALLERID(num) but a special "message" which is transfered for analogue phones between the first 2 ringings (but I think this is generated from the CALLERID(num)). >> There are several different methods for that, that can be set in the ATA (Cisco SPA112, latest EOL firmware): >> >> <image001.png><image002.png> >> >> <image003.png><image004.png> >> >> I tried every possible combination of "Caller ID Method" and "Caller ID FSK Standard", but without success. >> >> Has anybody on the list made experiences with analogue CLIP? >> >> Michael Michael http://www.mksolutions.info |
|
From: Michael K. <mic...@ip...> - 2022-04-19 23:37:12
|
Hi Michael I'm a little confused here. I was assuming this: elevator -> PSTN -> SPA112 -> Asterisk What am I missing? Regards Michael Knill From: Michael Keuter <li...@mk...> Reply to: AstLinux List <ast...@li...> Date: Wednesday, 20 April 2022 at 9:20 am To: AstLinux List <ast...@li...> Subject: Re: [Astlinux-users] Analogue CLIP (CallerID) Hi Michael, the „potential“ customer has these modems for a long time, and there are no alternatives (like SIP) at the market. And sure, in SIP I see the correct CallerID, but the modem don‘t see the „right“ CallerID generated by the ATA. Sent from a mobile device. Michael Keuter Am 20.04.2022 um 00:58 schrieb Michael Knill <mic...@ip...>: Hi Michael Just wondering why you need analogue modems to receive the calls from the elevators? Have you done a SIP Debug of the traffic coming from the SPA112’s to see if the number is anywhere in the SIP Invite? Do you know the particular standard for this in your country and is this supported by the SPA112’s? I have never used analogue FXO for anything sorry. Regards Michael Knill From: Michael Keuter <li...@mk...> Reply to: AstLinux List <ast...@li...> Date: Wednesday, 20 April 2022 at 2:23 am To: AstLinux List <ast...@li...> Subject: [Astlinux-users] Analogue CLIP (CallerID) Hi list, I am trying to install an Asterisk PBX for an elevator emergency central in Germany. The callees are 6 analogue modems who receive calls from the elevators. The main issue is that the modems need to identify the elevator via a so called analogue CLIP (Calling Line Identification Presentation) where the calling number is shown to the receiving modem. It is not the Asterisk CALLERID(num) but a special "message" which is transfered for analogue phones between the first 2 ringings (but I think this is generated from the CALLERID(num)). There are several different methods for that, that can be set in the ATA (Cisco SPA112, latest EOL firmware): [cid:image001.png@01D8549A.28246440][cid:image002.png@01D8549A.28246440] [cid:image003.png@01D8549A.28246440][cid:image004.png@01D8549A.28246440] I tried every possible combination of "Caller ID Method" and "Caller ID FSK Standard", but without success. Has anybody on the list made experiences with analogue CLIP? 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.... |
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From: Michael K. <li...@mk...> - 2022-04-19 23:20:21
|
Hi Michael, the „potential“ customer has these modems for a long time, and there are no alternatives (like SIP) at the market. And sure, in SIP I see the correct CallerID, but the modem don‘t see the „right“ CallerID generated by the ATA. Sent from a mobile device. Michael Keuter > Am 20.04.2022 um 00:58 schrieb Michael Knill <mic...@ip...>: > > > Hi Michael > > Just wondering why you need analogue modems to receive the calls from the elevators? > Have you done a SIP Debug of the traffic coming from the SPA112’s to see if the number is anywhere in the SIP Invite? > Do you know the particular standard for this in your country and is this supported by the SPA112’s? > > I have never used analogue FXO for anything sorry. > > Regards > Michael Knill > > From: Michael Keuter <li...@mk...> > Reply to: AstLinux List <ast...@li...> > Date: Wednesday, 20 April 2022 at 2:23 am > To: AstLinux List <ast...@li...> > Subject: [Astlinux-users] Analogue CLIP (CallerID) > > Hi list, > > I am trying to install an Asterisk PBX for an elevator emergency central in Germany. > The callees are 6 analogue modems who receive calls from the elevators. > > The main issue is that the modems need to identify the elevator via a so called analogue CLIP (Calling Line Identification Presentation) where the calling number is shown to the receiving modem. > It is not the Asterisk CALLERID(num) but a special "message" which is transfered for analogue phones between the first 2 ringings (but I think this is generated from the CALLERID(num)). > There are several different methods for that, that can be set in the ATA (Cisco SPA112, latest EOL firmware): > > > > > > I tried every possible combination of "Caller ID Method" and "Caller ID FSK Standard", but without success. > > Has anybody on the list made experiences with analogue CLIP? > > 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.... |
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From: Michael K. <mic...@ip...> - 2022-04-19 22:57:22
|
Hi Michael Just wondering why you need analogue modems to receive the calls from the elevators? Have you done a SIP Debug of the traffic coming from the SPA112’s to see if the number is anywhere in the SIP Invite? Do you know the particular standard for this in your country and is this supported by the SPA112’s? I have never used analogue FXO for anything sorry. Regards Michael Knill From: Michael Keuter <li...@mk...> Reply to: AstLinux List <ast...@li...> Date: Wednesday, 20 April 2022 at 2:23 am To: AstLinux List <ast...@li...> Subject: [Astlinux-users] Analogue CLIP (CallerID) Hi list, I am trying to install an Asterisk PBX for an elevator emergency central in Germany. The callees are 6 analogue modems who receive calls from the elevators. The main issue is that the modems need to identify the elevator via a so called analogue CLIP (Calling Line Identification Presentation) where the calling number is shown to the receiving modem. It is not the Asterisk CALLERID(num) but a special "message" which is transfered for analogue phones between the first 2 ringings (but I think this is generated from the CALLERID(num)). There are several different methods for that, that can be set in the ATA (Cisco SPA112, latest EOL firmware): [cid:image001.png@01D85494.98118E50][cid:image002.png@01D85494.98118E50] [cid:image003.png@01D85494.98118E50][cid:image004.png@01D85494.98118E50] I tried every possible combination of "Caller ID Method" and "Caller ID FSK Standard", but without success. Has anybody on the list made experiences with analogue CLIP? Michael http://www.mksolutions.info |
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From: Michael K. <li...@mk...> - 2022-04-19 16:22:38
|
Hi list, I am trying to install an Asterisk PBX for an elevator emergency central in Germany. The callees are 6 analogue modems who receive calls from the elevators. The main issue is that the modems need to identify the elevator via a so called analogue CLIP (Calling Line Identification Presentation) where the calling number is shown to the receiving modem. It is not the Asterisk CALLERID(num) but a special "message" which is transfered for analogue phones between the first 2 ringings (but I think this is generated from the CALLERID(num)). There are several different methods for that, that can be set in the ATA (Cisco SPA112, latest EOL firmware): I tried every possible combination of "Caller ID Method" and "Caller ID FSK Standard", but without success. Has anybody on the list made experiences with analogue CLIP? Michael http://www.mksolutions.info |
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From: Steve B. <ste...@gm...> - 2022-04-17 20:42:21
|
Problem solved – thanks Michael 👍
Your suggestion that by using SSH via WinSCP to re-start lighttpd by inputting
*/service lighttpd stop service lighttpd init/*
resulted in the following message:
*/2022-04-17 10:37:01: (mod_openssl.c.427) SSL:
BIO_read_filename('/etc/ssl/AstLinux.pem') failed
/*On investigation the file /etc/ssl/AstLinux.pem was not there on the
my file system
Then by using the /etc/ssl/default_https.pem file that was there, I
copied the “default_https.pem” file and renamed it to "AstLinux.pem" and
so it was there now.
Rebooted and this solved the problem - https://192.168.1.141/status.php
now works fine
I’m not sure if any of that makes sense but this may help:
Before – when I couldn’t access the Web Interface GUI Management Page
After – problem solved
So – thanks once again
Best wishes
Steve
On 17/04/2022 11:10, Michael Keuter wrote:
> Let's wait for Lonnie to chime in - in a few hours.
>
>> Am 17.04.2022 um 11:48 schrieb Steve Barlow<ste...@gm...>:
>>
>> Thanks for your reply Michael
>>
>> I tried
>>
>> https://192.168.1.141/admin/status.php
>> But still no joy....
>>
>>
>> Please see below the response I got when I input your suggestions via putty
>>
>> pbx ~ #
>>
>> pbx ~ # ps | grep lighttpd
>>
>> 1848 root grep lighttpd
>>
>>
>> pbx ~ #
>>
>> pbx ~ # service lighttpd stop
>>
>> pbx ~ # service lighttpd init
>>
>> Starting lighttpd...
>>
>>
>>
>> WARNING WARNING WARNING
>>
>> YOU STILL HAVE NOT CHANGED YOUR HTTPS ADMIN PASSWORD
>>
>> ANYONE THAT KNOWS YOU ARE USING ASTLINUX CAN DESTROY YOUR
>>
>> SYSTEM. PLEASE CHANGE THIS OR DISABLE THE HTTPS ADMIN
>>
>> INTERFACE IMMEDIATELY!
>>
>> Example:
>>
>> htpasswd /var/www/admin/.htpasswd admin
>>
>> WARNING WARNING WARNING
>>
>>
>>
>> 2022-04-17 10:37:01: (mod_openssl.c.427) SSL: BIO_read_filename('/etc/ssl/AstLinux.pem') failed
>>
>> 2022-04-17 10:37:01: (server.c.1161) Initialization of plugins failed. Going down.
>>
>> pbx ~ #
>>
>>
>> Best wishes
>> Steve.
>>
>>
>>
>>
>>
>>
>> On 17/04/2022 10:13, Michael Keuter wrote:
>>>> Am 17.04.2022 um 00:25 schrieb Steve Barlow<ste...@gm...>
>>>> :
>>>>
>>>> Hello there everyone, I am a ‘very basic level user’ who has been successfully using Astlinux for approx. the last 5 years, for the interconnection of my heritage telephones and communication with other enthusiasts.
>>>>
>>>> I recently updated the firmware on my HP T5720 Thin Client PC to Astlinux 1.3.8 along with Asterisk 13.31.0
>>>>
>>>> However, now after rebooting/restarting I have ‘lost’ my Web Interface to the ‘Astlinux Management’ page.
>>>>
>>>> I used to access this by entering in my Google Chrome Browser:
>>>>
>>>>
>>>> https://192.168.1.141/status.php
>>> Hi Steve,
>>> do you have tried:
>>>
>>>
>>> https://192.168.1.141/admin/status.php
>>>
>>>
>>> If it fails, maybe lighttpd (the webserver) is not running. You can try via ssh:
>>>
>>> ps | grep lighttpd
>>> 1442 root lighttpd -f /etc/lighttpd.conf
>>> 5144 root grep lighttpd
>>>
>>> The first line shows here that lighttpd is running (1442), if not try this to start lighttpd:
>>>
>>> service lighttpd stop
>>> service lighttpd init
>>>
>>>
>>>> But I now get the message returned:
>>>>
>>>> This site can’t be reached
>>>> 192.168.1.141 took too long to respond.
>>>>
>>>> However, the good news is that I can still make and receive calls etc. etc. and can SSH access to all my folders/files via Putty or access them via WinSCP using SFTP
>>>>
>>>> Any ideas anyone on how I can access the ‘Astlinux Management’ GUI again without doing total reinstall?
>>>>
>>>> Best wishes
>>>> Steve - UK
>>>>
>>> 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....
>
> Michael
>
> http://www.mksolutions.info
>
>
> |
|
From: Steve B. <ste...@gm...> - 2022-04-17 09:48:53
|
Thanks for your reply Michael
I tried
*/https://192.168.1.141/admin/status.php /*
But still no joy....
Please see below the response I got when I input your suggestions via putty
*/pbx ~ #/*
*//*
*/pbx ~ # ps | grep lighttpd/*
*//*
*//**/1848 root/**//**/grep lighttpd/*
*//**/
/**//*
*/pbx ~ #/*
*//*
*/pbx ~ # service lighttpd stop/*
*//*
*/pbx ~ # service lighttpd init/*
*//*
*/Starting lighttpd.../*
*//*
*//*
*//*
*/WARNING WARNING WARNING/*
*//*
*/ YOU STILL HAVE NOT CHANGED YOUR HTTPS ADMIN PASSWORD/*
*//*
*/ANYONE THAT KNOWS YOU ARE USING ASTLINUX CAN DESTROY YOUR/*
*//*
*/SYSTEM. PLEASE CHANGE THIS OR DISABLE THE HTTPS ADMIN/*
*//*
*/INTERFACE IMMEDIATELY!/*
*//*
*/ Example:/*
*//*
*/ htpasswd /var/www/admin/.htpasswd admin/*
*//*
*/ WARNING WARNING WARNING/*
*//*
*//*
*//*
*/2022-04-17 10:37:01: (mod_openssl.c.427) SSL:
BIO_read_filename('/etc/ssl/AstLinux.pem') failed/*
*//*
*/2022-04-17 10:37:01: (server.c.1161) Initialization of plugins failed.
Going down./*
*//*
*/pbx ~ #/*
*//*
Best wishes
Steve.
On 17/04/2022 10:13, Michael Keuter wrote:
>> Am 17.04.2022 um 00:25 schrieb Steve Barlow<ste...@gm...>:
>>
>> Hello there everyone, I am a ‘very basic level user’ who has been successfully using Astlinux for approx. the last 5 years, for the interconnection of my heritage telephones and communication with other enthusiasts.
>>
>> I recently updated the firmware on my HP T5720 Thin Client PC to Astlinux 1.3.8 along with Asterisk 13.31.0
>>
>> However, now after rebooting/restarting I have ‘lost’ my Web Interface to the ‘Astlinux Management’ page.
>>
>> I used to access this by entering in my Google Chrome Browser:
>>
>> https://192.168.1.141/status.php
> Hi Steve,
> do you have tried:
>
> https://192.168.1.141/admin/status.php
>
> If it fails, maybe lighttpd (the webserver) is not running. You can try via ssh:
>
> ps | grep lighttpd
> 1442 root lighttpd -f /etc/lighttpd.conf
> 5144 root grep lighttpd
>
> The first line shows here that lighttpd is running (1442), if not try this to start lighttpd:
>
> service lighttpd stop
> service lighttpd init
>
>> But I now get the message returned:
>>
>> This site can’t be reached
>> 192.168.1.141 took too long to respond.
>>
>> However, the good news is that I can still make and receive calls etc. etc. and can SSH access to all my folders/files via Putty or access them via WinSCP using SFTP
>>
>> Any ideas anyone on how I can access the ‘Astlinux Management’ GUI again without doing total reinstall?
>>
>> Best wishes
>> Steve - UK
> 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 top...@kr.... |
|
From: Michael K. <li...@mk...> - 2022-04-17 09:13:41
|
> Am 17.04.2022 um 00:25 schrieb Steve Barlow <ste...@gm...>: > > Hello there everyone, I am a ‘very basic level user’ who has been successfully using Astlinux for approx. the last 5 years, for the interconnection of my heritage telephones and communication with other enthusiasts. > > I recently updated the firmware on my HP T5720 Thin Client PC to Astlinux 1.3.8 along with Asterisk 13.31.0 > > However, now after rebooting/restarting I have ‘lost’ my Web Interface to the ‘Astlinux Management’ page. > > I used to access this by entering in my Google Chrome Browser: > > https://192.168.1.141/status.php Hi Steve, do you have tried: https://192.168.1.141/admin/status.php If it fails, maybe lighttpd (the webserver) is not running. You can try via ssh: ps | grep lighttpd 1442 root lighttpd -f /etc/lighttpd.conf 5144 root grep lighttpd The first line shows here that lighttpd is running (1442), if not try this to start lighttpd: service lighttpd stop service lighttpd init > But I now get the message returned: > > This site can’t be reached > 192.168.1.141 took too long to respond. > > However, the good news is that I can still make and receive calls etc. etc. and can SSH access to all my folders/files via Putty or access them via WinSCP using SFTP > > Any ideas anyone on how I can access the ‘Astlinux Management’ GUI again without doing total reinstall? > > Best wishes > Steve - UK Michael http://www.mksolutions.info |
|
From: Steve B. <ste...@gm...> - 2022-04-16 22:25:16
|
Hello there everyone, I am a ‘very basic level user’ who has been successfully using Astlinux for approx. the last 5 years, for the interconnection of my heritage telephones and communication with other enthusiasts. I recently updated the firmware on my HP T5720 Thin Client PC to Astlinux 1.3.8 along with Asterisk 13.31.0 However, now after rebooting/restarting I have ‘lost’ my Web Interface to the ‘Astlinux Management’ page. I used to access this by entering in my Google Chrome Browser: https://192.168.1.141/status.php But I now get the message returned: /This site can’t be reached// //192.168.1.141 took too long to respond.// /// However, the good news is that I can still make and receive calls etc. etc. and can SSH access to all my folders/files via Putty or access them via WinSCP using SFTP Any ideas anyone on how I can access the ‘Astlinux Management’ GUI again without doing total reinstall? Best wishes Steve - UK |
|
From: Ionel C. <ion...@me...> - 2022-03-15 00:49:14
|
I even managed to upgrade to Asterisk 18. Man I LOVE this Astlinux. The most rock solid environment ever. Thanks all for all the hard and good work > On Mar 13, 2022, at 4:20 PM, Lonnie Abelbeck <li...@lo...> wrote: > > If you are at 1.4.5 with ast13se-firmware-1.x, change the Asterisk version via the Prefs tab (ast16-firmware-1.x or ast18-firmware-1.x), then go the System tab and first perform "Revert to Previous" then "Upgrade with New" as usual. > > Ref: https://doc.astlinux-project.org/userdoc:tt_asterisk_upgrade_version > > Lonnie > > >> On Mar 13, 2022, at 2:41 PM, Ionel Chila via Astlinux-users <ast...@li...> wrote: >> >> Thanks Michael. So I managed to update to 1.4.5 but on Asterisk 13. >> >> How do I switch to Asterisk 16 now? Or 18? >> >> >> >>> On Mar 13, 2022, at 1:19 PM, Michael Keuter <li...@mk...> wrote: >>> >>> Hi, >>> >>> as stated in the release info, the Asterisk 13 version is no more updated, but ast13se, ast16 + ast18. >>> >>>> Am 13.03.2022 um 19:15 schrieb Ionel Chila via Astlinux-users <ast...@li...>: >>>> >>>> Is strange, when I do an update from the GUI it tells me 1.4.4 is the latest version. It used to work upgrading from there just fine. Anything changed? >>>> >>>> >>>> <Screen Shot 2022-03-13 at 1.14.55 PM.png> >>>> >>>> >>>>> On Mar 2, 2022, at 7:49 AM, Lonnie Abelbeck <li...@lo...> wrote: >>>>> >>>>> Announcing AstLinux Release: 1.4.5 >>>>> >>>>> More Info: AstLinux Project >>>>> https://www.astlinux-project.org/ >>>>> >>>>> AstLinux 1.4.5 Highlights: >>>>> * Asterisk Versions: 13.38.3, 16.21.1, 18.10.0 >>>>> * Asterisk 18.x is now supported, along with Asterisk 16.x and Asterisk 13.x built --without-pjproject >>>>> * Previous ast13-firmware-1.x is no longer being updated, ast13-firmware-1.x users should either switch to ast16-firmware-1.x (recommended) or use ast13se-firmware-1.x if chan_pjsip is not used in your dialplan. >>>>> >>>>> * Linux Kernel 4.19.230, security and bug fixes >>>>> * RUNNIX, version bump to runnix-0.6.6 >>>>> * OpenSSL, version bump to 1.1.1m, security fixes: none >>>>> * WireGuard VPN, module 1.0.20211208 (version bump), tools 1.0.20210914 (no change) >>>>> * strongSwan, version 5.5.3, security fix: CVE-2021-45079 >>>>> * libcurl (curl) version bump to 7.81.0 >>>>> * chrony, version bump to 4.2 >>>>> * darkstat, version bump to 3.0.721 >>>>> * expat, version bump to 2.4.6, security fixes: many >>>>> * Monit, version bump to 5.31.0 >>>>> * msmtp, version bump to 1.8.19, 'msmtpd' security fix >>>>> * mtr, version bump to 0.95 >>>>> * prosody, version bump to 0.11.13 >>>>> * tarsnap, version bump to 1.0.40, "Trust No One" encrypted backups using the Tarsnap Backup service. >>>>> * vnStat, version bump to 2.9 >>>>> * zabbix, version bump to 4.0.38 >>>>> * Asterisk '13se' (stable edition) version 13.38.3 is the last Asterisk 13.x "Legacy" version, built --without-pjproject >>>>> * Package upgrades providing important security and bug fixes >>>>> >>>>> Full ChangeLog: >>>>> https://raw.githubusercontent.com/astlinux-project/astlinux/1.4.5/docs/ChangeLog.txt >>>>> >>>>> All users are encouraged to upgrade, read the ChangeLog for the details. >>>>> >>>>> AstLinux Team >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> 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.... >>> >>> >>> 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.... >> >> >> >> _______________________________________________ >> 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: Ionel C. <ion...@me...> - 2022-03-13 22:14:11
|
Hey THANKS much Lonnie. It worked Sir. Perfect migration :-) > On Mar 13, 2022, at 4:20 PM, Lonnie Abelbeck <li...@lo...> wrote: > > If you are at 1.4.5 with ast13se-firmware-1.x, change the Asterisk version via the Prefs tab (ast16-firmware-1.x or ast18-firmware-1.x), then go the System tab and first perform "Revert to Previous" then "Upgrade with New" as usual. > > Ref: https://doc.astlinux-project.org/userdoc:tt_asterisk_upgrade_version <https://doc.astlinux-project.org/userdoc:tt_asterisk_upgrade_version> > > Lonnie |
|
From: Lonnie A. <li...@lo...> - 2022-03-13 21:21:08
|
If you are at 1.4.5 with ast13se-firmware-1.x, change the Asterisk version via the Prefs tab (ast16-firmware-1.x or ast18-firmware-1.x), then go the System tab and first perform "Revert to Previous" then "Upgrade with New" as usual. Ref: https://doc.astlinux-project.org/userdoc:tt_asterisk_upgrade_version Lonnie > On Mar 13, 2022, at 2:41 PM, Ionel Chila via Astlinux-users <ast...@li...> wrote: > > Thanks Michael. So I managed to update to 1.4.5 but on Asterisk 13. > > How do I switch to Asterisk 16 now? Or 18? > > > >> On Mar 13, 2022, at 1:19 PM, Michael Keuter <li...@mk...> wrote: >> >> Hi, >> >> as stated in the release info, the Asterisk 13 version is no more updated, but ast13se, ast16 + ast18. >> >>> Am 13.03.2022 um 19:15 schrieb Ionel Chila via Astlinux-users <ast...@li...>: >>> >>> Is strange, when I do an update from the GUI it tells me 1.4.4 is the latest version. It used to work upgrading from there just fine. Anything changed? >>> >>> >>> <Screen Shot 2022-03-13 at 1.14.55 PM.png> >>> >>> >>>> On Mar 2, 2022, at 7:49 AM, Lonnie Abelbeck <li...@lo...> wrote: >>>> >>>> Announcing AstLinux Release: 1.4.5 >>>> >>>> More Info: AstLinux Project >>>> https://www.astlinux-project.org/ >>>> >>>> AstLinux 1.4.5 Highlights: >>>> * Asterisk Versions: 13.38.3, 16.21.1, 18.10.0 >>>> * Asterisk 18.x is now supported, along with Asterisk 16.x and Asterisk 13.x built --without-pjproject >>>> * Previous ast13-firmware-1.x is no longer being updated, ast13-firmware-1.x users should either switch to ast16-firmware-1.x (recommended) or use ast13se-firmware-1.x if chan_pjsip is not used in your dialplan. >>>> >>>> * Linux Kernel 4.19.230, security and bug fixes >>>> * RUNNIX, version bump to runnix-0.6.6 >>>> * OpenSSL, version bump to 1.1.1m, security fixes: none >>>> * WireGuard VPN, module 1.0.20211208 (version bump), tools 1.0.20210914 (no change) >>>> * strongSwan, version 5.5.3, security fix: CVE-2021-45079 >>>> * libcurl (curl) version bump to 7.81.0 >>>> * chrony, version bump to 4.2 >>>> * darkstat, version bump to 3.0.721 >>>> * expat, version bump to 2.4.6, security fixes: many >>>> * Monit, version bump to 5.31.0 >>>> * msmtp, version bump to 1.8.19, 'msmtpd' security fix >>>> * mtr, version bump to 0.95 >>>> * prosody, version bump to 0.11.13 >>>> * tarsnap, version bump to 1.0.40, "Trust No One" encrypted backups using the Tarsnap Backup service. >>>> * vnStat, version bump to 2.9 >>>> * zabbix, version bump to 4.0.38 >>>> * Asterisk '13se' (stable edition) version 13.38.3 is the last Asterisk 13.x "Legacy" version, built --without-pjproject >>>> * Package upgrades providing important security and bug fixes >>>> >>>> Full ChangeLog: >>>> https://raw.githubusercontent.com/astlinux-project/astlinux/1.4.5/docs/ChangeLog.txt >>>> >>>> All users are encouraged to upgrade, read the ChangeLog for the details. >>>> >>>> AstLinux Team >>>> >>>> >>>> >>>> _______________________________________________ >>>> 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.... >> >> >> 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.... > > > > _______________________________________________ > 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: Ionel C. <ion...@me...> - 2022-03-13 19:41:20
|
Thanks Michael. So I managed to update to 1.4.5 but on Asterisk 13. How do I switch to Asterisk 16 now? Or 18? > On Mar 13, 2022, at 1:19 PM, Michael Keuter <li...@mk...> wrote: > > Hi, > > as stated in the release info, the Asterisk 13 version is no more updated, but ast13se, ast16 + ast18. > >> Am 13.03.2022 um 19:15 schrieb Ionel Chila via Astlinux-users <ast...@li...>: >> >> Is strange, when I do an update from the GUI it tells me 1.4.4 is the latest version. It used to work upgrading from there just fine. Anything changed? >> >> >> <Screen Shot 2022-03-13 at 1.14.55 PM.png> >> >> >>> On Mar 2, 2022, at 7:49 AM, Lonnie Abelbeck <li...@lo...> wrote: >>> >>> Announcing AstLinux Release: 1.4.5 >>> >>> More Info: AstLinux Project >>> https://www.astlinux-project.org/ >>> >>> AstLinux 1.4.5 Highlights: >>> * Asterisk Versions: 13.38.3, 16.21.1, 18.10.0 >>> * Asterisk 18.x is now supported, along with Asterisk 16.x and Asterisk 13.x built --without-pjproject >>> * Previous ast13-firmware-1.x is no longer being updated, ast13-firmware-1.x users should either switch to ast16-firmware-1.x (recommended) or use ast13se-firmware-1.x if chan_pjsip is not used in your dialplan. >>> >>> * Linux Kernel 4.19.230, security and bug fixes >>> * RUNNIX, version bump to runnix-0.6.6 >>> * OpenSSL, version bump to 1.1.1m, security fixes: none >>> * WireGuard VPN, module 1.0.20211208 (version bump), tools 1.0.20210914 (no change) >>> * strongSwan, version 5.5.3, security fix: CVE-2021-45079 >>> * libcurl (curl) version bump to 7.81.0 >>> * chrony, version bump to 4.2 >>> * darkstat, version bump to 3.0.721 >>> * expat, version bump to 2.4.6, security fixes: many >>> * Monit, version bump to 5.31.0 >>> * msmtp, version bump to 1.8.19, 'msmtpd' security fix >>> * mtr, version bump to 0.95 >>> * prosody, version bump to 0.11.13 >>> * tarsnap, version bump to 1.0.40, "Trust No One" encrypted backups using the Tarsnap Backup service. >>> * vnStat, version bump to 2.9 >>> * zabbix, version bump to 4.0.38 >>> * Asterisk '13se' (stable edition) version 13.38.3 is the last Asterisk 13.x "Legacy" version, built --without-pjproject >>> * Package upgrades providing important security and bug fixes >>> >>> Full ChangeLog: >>> https://raw.githubusercontent.com/astlinux-project/astlinux/1.4.5/docs/ChangeLog.txt >>> >>> All users are encouraged to upgrade, read the ChangeLog for the details. >>> >>> AstLinux Team >>> >>> >>> >>> _______________________________________________ >>> 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.... > > > 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.... |
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From: Michael K. <li...@mk...> - 2022-03-13 19:29:47
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Hi, as stated in the release info, the Asterisk 13 version is no more updated, but ast13se, ast16 + ast18. > Am 13.03.2022 um 19:15 schrieb Ionel Chila via Astlinux-users <ast...@li...>: > > Is strange, when I do an update from the GUI it tells me 1.4.4 is the latest version. It used to work upgrading from there just fine. Anything changed? > > > <Screen Shot 2022-03-13 at 1.14.55 PM.png> > > >> On Mar 2, 2022, at 7:49 AM, Lonnie Abelbeck <li...@lo...> wrote: >> >> Announcing AstLinux Release: 1.4.5 >> >> More Info: AstLinux Project >> https://www.astlinux-project.org/ >> >> AstLinux 1.4.5 Highlights: >> * Asterisk Versions: 13.38.3, 16.21.1, 18.10.0 >> * Asterisk 18.x is now supported, along with Asterisk 16.x and Asterisk 13.x built --without-pjproject >> * Previous ast13-firmware-1.x is no longer being updated, ast13-firmware-1.x users should either switch to ast16-firmware-1.x (recommended) or use ast13se-firmware-1.x if chan_pjsip is not used in your dialplan. >> >> * Linux Kernel 4.19.230, security and bug fixes >> * RUNNIX, version bump to runnix-0.6.6 >> * OpenSSL, version bump to 1.1.1m, security fixes: none >> * WireGuard VPN, module 1.0.20211208 (version bump), tools 1.0.20210914 (no change) >> * strongSwan, version 5.5.3, security fix: CVE-2021-45079 >> * libcurl (curl) version bump to 7.81.0 >> * chrony, version bump to 4.2 >> * darkstat, version bump to 3.0.721 >> * expat, version bump to 2.4.6, security fixes: many >> * Monit, version bump to 5.31.0 >> * msmtp, version bump to 1.8.19, 'msmtpd' security fix >> * mtr, version bump to 0.95 >> * prosody, version bump to 0.11.13 >> * tarsnap, version bump to 1.0.40, "Trust No One" encrypted backups using the Tarsnap Backup service. >> * vnStat, version bump to 2.9 >> * zabbix, version bump to 4.0.38 >> * Asterisk '13se' (stable edition) version 13.38.3 is the last Asterisk 13.x "Legacy" version, built --without-pjproject >> * Package upgrades providing important security and bug fixes >> >> Full ChangeLog: >> https://raw.githubusercontent.com/astlinux-project/astlinux/1.4.5/docs/ChangeLog.txt >> >> All users are encouraged to upgrade, read the ChangeLog for the details. >> >> AstLinux Team >> >> >> >> _______________________________________________ >> 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.... Michael http://www.mksolutions.info |
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From: Ionel C. <ion...@me...> - 2022-03-13 19:06:47
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Thanks much as always. Dooh, who reads the instructions. I apologize :) > On Mar 13, 2022, at 1:49 PM, Lonnie Abelbeck <li...@lo...> wrote: > > So, if you are not using chan_pjsip you could change to ast13se-firmware-1.x (via Prefs Tab) or alternatively start using ast16-firmware-1.x ... either of those will get you to version 1.4.5 . |
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From: Lonnie A. <li...@lo...> - 2022-03-13 18:49:34
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Hi Ionel, Yes that is expected when using the ast13-firmware-1.x firmware. As stated below in the "AstLinux 1.4.5 Highlights:" -- * Asterisk 18.x is now supported, along with Asterisk 16.x and Asterisk 13.x built --without-pjproject * Previous ast13-firmware-1.x is no longer being updated, ast13-firmware-1.x users should either switch to ast16-firmware-1.x (recommended) or use ast13se-firmware-1.x if chan_pjsip is not used in your dialplan. -- So, if you are not using chan_pjsip you could change to ast13se-firmware-1.x (via Prefs Tab) or alternatively start using ast16-firmware-1.x ... either of those will get you to version 1.4.5 . Lonnie > On Mar 13, 2022, at 1:15 PM, Ionel Chila via Astlinux-users <ast...@li...> wrote: > > Is strange, when I do an update from the GUI it tells me 1.4.4 is the latest version. It used to work upgrading from there just fine. Anything changed? > > > <Screen Shot 2022-03-13 at 1.14.55 PM.png> > > >> On Mar 2, 2022, at 7:49 AM, Lonnie Abelbeck <li...@lo...> wrote: >> >> Announcing AstLinux Release: 1.4.5 >> >> More Info: AstLinux Project >> https://www.astlinux-project.org/ >> >> AstLinux 1.4.5 Highlights: >> * Asterisk Versions: 13.38.3, 16.21.1, 18.10.0 >> * Asterisk 18.x is now supported, along with Asterisk 16.x and Asterisk 13.x built --without-pjproject >> * Previous ast13-firmware-1.x is no longer being updated, ast13-firmware-1.x users should either switch to ast16-firmware-1.x (recommended) or use ast13se-firmware-1.x if chan_pjsip is not used in your dialplan. >> >> * Linux Kernel 4.19.230, security and bug fixes >> * RUNNIX, version bump to runnix-0.6.6 >> * OpenSSL, version bump to 1.1.1m, security fixes: none >> * WireGuard VPN, module 1.0.20211208 (version bump), tools 1.0.20210914 (no change) >> * strongSwan, version 5.5.3, security fix: CVE-2021-45079 >> * libcurl (curl) version bump to 7.81.0 >> * chrony, version bump to 4.2 >> * darkstat, version bump to 3.0.721 >> * expat, version bump to 2.4.6, security fixes: many >> * Monit, version bump to 5.31.0 >> * msmtp, version bump to 1.8.19, 'msmtpd' security fix >> * mtr, version bump to 0.95 >> * prosody, version bump to 0.11.13 >> * tarsnap, version bump to 1.0.40, "Trust No One" encrypted backups using the Tarsnap Backup service. >> * vnStat, version bump to 2.9 >> * zabbix, version bump to 4.0.38 >> * Asterisk '13se' (stable edition) version 13.38.3 is the last Asterisk 13.x "Legacy" version, built --without-pjproject >> * Package upgrades providing important security and bug fixes >> >> Full ChangeLog: >> https://raw.githubusercontent.com/astlinux-project/astlinux/1.4.5/docs/ChangeLog.txt >> >> All users are encouraged to upgrade, read the ChangeLog for the details. >> >> AstLinux Team >> >> >> >> _______________________________________________ >> 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: Ionel C. <ion...@me...> - 2022-03-13 18:15:56
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Is strange, when I do an update from the GUI it tells me 1.4.4 is the latest version. It used to work upgrading from there just fine. Anything changed? > On Mar 2, 2022, at 7:49 AM, Lonnie Abelbeck <li...@lo...> wrote: > > Announcing AstLinux Release: 1.4.5 > > More Info: AstLinux Project > https://www.astlinux-project.org/ > > AstLinux 1.4.5 Highlights: > * Asterisk Versions: 13.38.3, 16.21.1, 18.10.0 > * Asterisk 18.x is now supported, along with Asterisk 16.x and Asterisk 13.x built --without-pjproject > * Previous ast13-firmware-1.x is no longer being updated, ast13-firmware-1.x users should either switch to ast16-firmware-1.x (recommended) or use ast13se-firmware-1.x if chan_pjsip is not used in your dialplan. > > * Linux Kernel 4.19.230, security and bug fixes > * RUNNIX, version bump to runnix-0.6.6 > * OpenSSL, version bump to 1.1.1m, security fixes: none > * WireGuard VPN, module 1.0.20211208 (version bump), tools 1.0.20210914 (no change) > * strongSwan, version 5.5.3, security fix: CVE-2021-45079 > * libcurl (curl) version bump to 7.81.0 > * chrony, version bump to 4.2 > * darkstat, version bump to 3.0.721 > * expat, version bump to 2.4.6, security fixes: many > * Monit, version bump to 5.31.0 > * msmtp, version bump to 1.8.19, 'msmtpd' security fix > * mtr, version bump to 0.95 > * prosody, version bump to 0.11.13 > * tarsnap, version bump to 1.0.40, "Trust No One" encrypted backups using the Tarsnap Backup service. > * vnStat, version bump to 2.9 > * zabbix, version bump to 4.0.38 > * Asterisk '13se' (stable edition) version 13.38.3 is the last Asterisk 13.x "Legacy" version, built --without-pjproject > * Package upgrades providing important security and bug fixes > > Full ChangeLog: > https://raw.githubusercontent.com/astlinux-project/astlinux/1.4.5/docs/ChangeLog.txt > > All users are encouraged to upgrade, read the ChangeLog for the details. > > AstLinux Team > > > > _______________________________________________ > 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...> - 2022-03-02 13:49:22
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Announcing AstLinux Release: 1.4.5 More Info: AstLinux Project https://www.astlinux-project.org/ AstLinux 1.4.5 Highlights: * Asterisk Versions: 13.38.3, 16.21.1, 18.10.0 * Asterisk 18.x is now supported, along with Asterisk 16.x and Asterisk 13.x built --without-pjproject * Previous ast13-firmware-1.x is no longer being updated, ast13-firmware-1.x users should either switch to ast16-firmware-1.x (recommended) or use ast13se-firmware-1.x if chan_pjsip is not used in your dialplan. * Linux Kernel 4.19.230, security and bug fixes * RUNNIX, version bump to runnix-0.6.6 * OpenSSL, version bump to 1.1.1m, security fixes: none * WireGuard VPN, module 1.0.20211208 (version bump), tools 1.0.20210914 (no change) * strongSwan, version 5.5.3, security fix: CVE-2021-45079 * libcurl (curl) version bump to 7.81.0 * chrony, version bump to 4.2 * darkstat, version bump to 3.0.721 * expat, version bump to 2.4.6, security fixes: many * Monit, version bump to 5.31.0 * msmtp, version bump to 1.8.19, 'msmtpd' security fix * mtr, version bump to 0.95 * prosody, version bump to 0.11.13 * tarsnap, version bump to 1.0.40, "Trust No One" encrypted backups using the Tarsnap Backup service. * vnStat, version bump to 2.9 * zabbix, version bump to 4.0.38 * Asterisk '13se' (stable edition) version 13.38.3 is the last Asterisk 13.x "Legacy" version, built --without-pjproject * Package upgrades providing important security and bug fixes Full ChangeLog: https://raw.githubusercontent.com/astlinux-project/astlinux/1.4.5/docs/ChangeLog.txt All users are encouraged to upgrade, read the ChangeLog for the details. AstLinux Team |
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From: Lonnie A. <li...@lo...> - 2022-02-17 19:51:58
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Announcing AstLinux Pre-Release: astlinux-1.4-5380-6e3fb2
Key new features:
-- Asterisk 18.x is now supported, along with Asterisk 16.x and Asterisk 13.x built --without-pjproject
-- Previous ast13-firmware-1.x is no longer being updated, ast13-firmware-1.x users should either switch to ast16-firmware-1.x (recommended)
or use ast13se-firmware-1.x if chan_pjsip is not used in your dialplan.
** The AstLinux Team is regularly upgrading packages containing security and bug fixes as well as adding new features of our own.
-- Linux Kernel 4.19.230 (version bump), security and bug fixes
-- OpenSSL, version bump to 1.1.1m, security fixes: none
-- WireGuard VPN, module 1.0.20211208 (version bump), tools 1.0.20210914 (no change)
-- strongSwan, version 5.5.3, security fix: CVE-2021-45079
-- expat, version bump to 2.4.4, security fixes: many, many
-- libcurl (curl) version bump to 7.81.0
-- LibreTLS, version bump to 3.4.2
-- Monit, version bump to 5.31.0
-- msmtp, version bump to 1.8.19, 'msmtpd' security fix
-- nano, version 2.7.5, fix issue where not saving a file could still copy the file to /mnt/asturw/
-- tarsnap, version bump to 1.0.40, "Trust No One" encrypted backups using the Tarsnap Backup service.
-- zabbix, version bump to 4.0.38
-- Network tab, Non-ACME Self-Signed HTTPS Certificate, use 2048 key length.
-- Asterisk 13.38.3 ('13se' no change)
Last Asterisk 13.x "Legacy" version, built --without-pjproject
-- Asterisk 16.21.1 (no change) and 18.10.0 (new version)
Note: Asterisk 16.23.0 has issues with high call usage, reverting to 16.21.1
-- 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: Michael K. <mic...@ip...> - 2022-02-09 22:24:08
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Awesome thanks Lonnie. Some great options there.
Not at 1.4 yet (coming soon) so might try the iPoE option initially.
The PPPoE options look very interesting. Think I may do some fine tuning in my 1.4.4 release. Would be interesting to see if CAKE improves anything too.
Regards
Michael Knill
On 10/2/22, 12:57 am, "Lonnie Abelbeck" <li...@lo...> wrote:
Hi Michael,
Nicely described issue.
1) Adjust lcp-echo-* settings (requiring AstLinux 1.4.1 or later)
By default the pppoe ppp peer options include:
--
lcp-echo-interval 20
lcp-echo-failure 3
--
Try adding a PPPOE_PPP_OPTIONS variable in your /mnt/kd/rc.conf.d/user.conf file:
--
PPPOE_PPP_OPTIONS="lcp-echo-interval 5 lcp-echo-failure 10"
--
or also add lcp-echo-adaptive
--
PPPOE_PPP_OPTIONS="lcp-echo-interval 5 lcp-echo-failure 10 lcp-echo-adaptive"
--
Test and adjust values accordingly.
2) Adjust QoS
Possibly (AstLinux 1.4.4 or later) CAKE support in the traffic shaper would help, but no evidence it would.
3) Changing the service to IPoE
I have always thought to avoid PPPoE if possible, so if IPoE is an available choice, that may be a good idea.
Lonnie
> On Feb 8, 2022, at 10:49 PM, Michael Knill <mic...@ip...> wrote:
>
> Hi Group
>
> I have a site that for years intermittently has periods where it loses PPPoE connectivity on a regular basis. After further investigation by one of my techs, it appears that when this is happening there is significant upstream congestion on the service due to a Veeam backup in progress.
> Note that I have set traffic shaping and the voice is not affected however it is when the PPPoE drops the connection e.g.
> Feb 9 12:40:33 3060-ETS_Ref-CM1 daemon.info pppd[362]: No response to 3 echo-requests
>
> We have always blamed the access provider but have not been able to pinpoint the issue. I'm now thinking that possibly during this high congestion, LCP Echo Request/Reply are being delayed and/or dropped meaning that Astlinux thinks connectivity is lost and it resets the connection.
>
> So my questions are:
> • Is this possible?
> • If so, how can I fix it? Something in QoS? Can I change the PPPoE parameters for LCP echos maybe?
> • Would changing the service to IPoE fix the problem e.g. only DHCP then?
>
> Thanks all.
> Regards
>
> Michael Knill
> Managing Director
>
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>
> <image001.png>
> Smarter Business Communications
>
> _______________________________________________
> 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...> - 2022-02-09 13:57:07
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Hi Michael, Nicely described issue. 1) Adjust lcp-echo-* settings (requiring AstLinux 1.4.1 or later) By default the pppoe ppp peer options include: -- lcp-echo-interval 20 lcp-echo-failure 3 -- Try adding a PPPOE_PPP_OPTIONS variable in your /mnt/kd/rc.conf.d/user.conf file: -- PPPOE_PPP_OPTIONS="lcp-echo-interval 5 lcp-echo-failure 10" -- or also add lcp-echo-adaptive -- PPPOE_PPP_OPTIONS="lcp-echo-interval 5 lcp-echo-failure 10 lcp-echo-adaptive" -- Test and adjust values accordingly. 2) Adjust QoS Possibly (AstLinux 1.4.4 or later) CAKE support in the traffic shaper would help, but no evidence it would. 3) Changing the service to IPoE I have always thought to avoid PPPoE if possible, so if IPoE is an available choice, that may be a good idea. Lonnie > On Feb 8, 2022, at 10:49 PM, Michael Knill <mic...@ip...> wrote: > > Hi Group > > I have a site that for years intermittently has periods where it loses PPPoE connectivity on a regular basis. After further investigation by one of my techs, it appears that when this is happening there is significant upstream congestion on the service due to a Veeam backup in progress. > Note that I have set traffic shaping and the voice is not affected however it is when the PPPoE drops the connection e.g. > Feb 9 12:40:33 3060-ETS_Ref-CM1 daemon.info pppd[362]: No response to 3 echo-requests > > We have always blamed the access provider but have not been able to pinpoint the issue. I'm now thinking that possibly during this high congestion, LCP Echo Request/Reply are being delayed and/or dropped meaning that Astlinux thinks connectivity is lost and it resets the connection. > > So my questions are: > • Is this possible? > • If so, how can I fix it? Something in QoS? Can I change the PPPoE parameters for LCP echos maybe? > • Would changing the service to IPoE fix the problem e.g. only DHCP then? > > Thanks all. > Regards > > Michael Knill > Managing Director > > D: +61 2 6189 1360 > P: +61 2 6140 4656 > E: mic...@ip... > W: ipcsolutions.com.au > > <image001.png> > Smarter Business Communications > > _______________________________________________ > 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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