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From: OSS F. A. <ope...@op...> - 2008-09-30 02:58:13
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I am interested on developing this dialer using atlsip. Contact me thru my email. |
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From: OSS F. A. <ope...@op...> - 2008-09-27 06:11:27
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This is a bounty for any1 interested in developing a proper dialer rather than the single form ossphone. if any1 has done it already then that can also be good to go. requirements r: server variables, stun server, port and codec need to be hardcoded, xor encryption by default as well, basically remove those fields from interface so it cant be changed Windows XP, Vista compatible Using single codec of voiceage G729, later ill get u the licensed codec if everything works fine No hifi skins etc required, as sinple and easy to use as possible User must login before they can do anything such as dial, change settings etc |
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From: OSS F. A. <ope...@op...> - 2008-09-26 02:33:39
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Look for allcodecs.h under the opal headers and uncomment the codec registration macros you do not need.
> {quote:title=xbipin wrote:}{quote}
> ok on searching i found its not possble but can u tell me which code to modify so that i can display only specific codecs to choose from by the user rather than displaying all available codecs and is it possible to send all codecs as option while calling so as instead the softswitch failing the call due to codec not supported, it instead uses any other codec not selected in the 2 options of HBR and LBR to complete the call. basically if the 2 codecs select fail then fallover to any codec which the softswitch supports as well as whats in the ossphone
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From: OSS F. A. <ope...@op...> - 2008-09-25 20:37:58
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ok on searching i found its not possble but can u tell me which code to modify so that i can display only specific codecs to choose from by the user rather than displaying all available codecs and is it possible to send all codecs as option while calling so as instead the softswitch failing the call due to codec not supported, it instead uses any other codec not selected in the 2 options of HBR and LBR to complete the call. basically if the 2 codecs select fail then fallover to any codec which the softswitch supports as well as whats in the ossphone |
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From: OSS F. A. <ope...@op...> - 2008-09-25 14:17:06
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This question has been covered severa times in this forum. Please try searching for the term "codec". |
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From: OSS F. A. <ope...@op...> - 2008-09-24 20:24:11
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is it possible by any chance to remove some source file so as to be able to compile dialer with only some of the codecs as most of them my softswitch doesnt support and i dont need them also but cant figure out a way to avoid them from being compiled as well. im looking for only G729 and mayb GSM, G723, G711u, thats it |
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From: OSS F. A. <ope...@op...> - 2008-09-20 17:30:34
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i was wondering how is it possible to add the voiceage G729 codec to the ossphone as i have added it to opensbc as it searches and adds it but how to add it to the ossphone as im willing to buy the license for it if it works with ossphone as i want the xor encryption as that the only thing that works for me. |
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From: Yacine A. <yac...@ms...> - 2008-09-06 17:20:55
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Hello, is there any possible way to use auto answer feature? thanks _________________________________________________________________ Installez gratuitement les 20 émôticones Windows Live Messenger les plus fous ! Cliquez ici ! http://www.emoticones-messenger.fr/ |
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From: OSS F. A. <ope...@op...> - 2008-08-06 07:23:23
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Thanks for the quick answer - is there also any news regarding call conference ? |
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From: OSS F. A. <ope...@op...> - 2008-08-06 06:09:44
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> {quote:title=jhollerer wrote:}{quote}
>
> Sorry i am new - but does someone know about the "real" roadmap for the features Call Transfer and Call Conference - as i cant find them using the ATLSIP library ?
>
Current CVS code for ATLSIP already supports blind call transfer. You can find instructions here http://www.opensipstack.org/cvs.html
Joegen
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From: OSS F. A. <ope...@op...> - 2008-08-05 17:59:44
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Sorry i am new - but does someone know about the "real" roadmap for the features Call Transfer and Call Conference - as i cant find them using the ATLSIP library ? |
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From: OSS F. A. <ope...@op...> - 2008-07-23 18:21:21
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Can anyone tell me why various versions of OSSPhone and a separate test program I wrote using ATLSIP respond to incoming calls with
{quote}>>> SIP/2.0 404 Not Found DST: 198.65.166.131:5060:UDP SRC: 192.168.0.5:5060 enc=0 bytes=567{quote}?
Outgoing calls work.
I'm using a release ATLSIP.DLL built with current sources from CVS as of the morning of 2008-07-23.
Here is the log:
{quote}----------------424:48:10.259----------------
Querying STUN server at stun01.sipphone.com.
This may take a while ...
----------------424:48:10.732----------------
*** LISTENER STARTED *** [OPAL] 127.0.0.1:5060
----------------424:48:10.733----------------
*** LISTENER STARTED *** [OPAL] 192.168.0.5:5060 [*** DEFAULT LISTENER ***]
SIP event:: SIPInitialized
----------------424:48:10.749----------------
>>> REGISTER sip:proxy01.sipphone.com SIP/2.0 DST: 198.65.166.131:5060:UDP SRC: 192.168.0.5:5060 enc=0 bytes=620
----------------424:48:10.841----------------
<<< SIP/2.0 401 Unauthorized SRC: 198.65.166.131:5060:UDP enc=0 bytes=587
----------------424:48:10.846----------------
>>> REGISTER sip:proxy01.sipphone.com SIP/2.0 DST: 198.65.166.131:5060:UDP SRC: 192.168.0.5:5060 enc=0 bytes=849
----------------424:48:10.944----------------
<<< SIP/2.0 200 OK SRC: 198.65.166.131:5060:UDP enc=0 bytes=548
SIP event:: LoginSuccessful
----------------424:48:32.422----------------
<<< INVITE sip:17476331234@72.240.235.122:21013;transport=udp SIP/2.0 SRC: 198.65.166.131:5060:UDP enc=0 bytes=1200
----------------424:48:32.424----------------
>>> SIP/2.0 100 Trying DST: 198.65.166.131:5060:UDP SRC: 192.168.0.5:5060 enc=0 bytes=452
SIP event:: IncomingCall: callerInfo="sip:174...@pr..."
----------------424:48:33.148----------------
>>> SIP/2.0 404 Not Found DST: 198.65.166.131:5060:UDP SRC: 192.168.0.5:5060 enc=0 bytes=567
----------------424:48:33.240----------------
<<< ACK sip:17476331234@72.240.235.122:21013;transport=udp SIP/2.0 SRC: 198.65.166.131:5060:UDP enc=0 bytes=353{quote}
Finest regards,
Bill Root
|
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From: OSS F. A. <ope...@op...> - 2008-06-21 18:48:34
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Great, thanks! I'll keep an eye on the ticket.
Bill
{quote}You are right. I have created a ticket for this
http://www.assembla.com/spaces/opensbc/tickets/24
Unfortunately there is currently no way of doing this in the appliation layer. But it will be there soon enough. Thanks for bringing it up.
Joegen{quote}
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From: OSS F. A. <ope...@op...> - 2008-06-20 00:54:59
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You are right. I have created a ticket for this http://www.assembla.com/spaces/opensbc/tickets/24 Unfortunately there is currently no way of doing this in the appliation layer. But it will be there soon enough. Thanks for bringing it up. Joegen > {quote:title=optotronic wrote:}{quote} > I'm using OSSPhone to test ATLSIP. It detects incoming calls, but it doesn't send out "180 Ringing" packets so caller doesn't get rings, and proxy eventually routes call to voicemail. > > Is there a way to automatically or manually send the "180 Ringing" messages with ATLSIP? > > Here is a log from OSSPhone: > {quote}----------------26:34:32.350---------------- > Querying STUN server at stun01.sipphone.com. > This may take a while ... > > ----------------26:34:42.772---------------- > STUN server "stun01.sipphone.com" replies Cone NAT, external IP 72.240.235.122 > ----------------26:34:47.820---------------- > *** LISTENER STARTED *** [OPAL] 127.0.0.1:5060 > ----------------26:34:47.880---------------- > *** LISTENER STARTED *** [OPAL] 192.168.0.20:5060 [*** DEFAULT LISTENER ***] > ----------------26:34:47.936---------------- > >>> REGISTER sip:proxy01.sipphone.com SIP/2.0 DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=629 > REGISTER sip:proxy01.sipphone.com SIP/2.0 > From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13 > To: sip:rec...@pr... > Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bKc3ce95d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport > CSeq: 1 REGISTER > Call-ID: c3c...@pr... > Contact: "recipient" <sip:recipient@72.240.235.122:22013;transport=udp> > User-Agent: OpenSIPStack v1.1.7-24 > Expires: 3600 > Max-Forwards: 70 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS, PRACK > Content-Length: 0 > > ----------------26:34:48.069---------------- > <<< SIP/2.0 401 Unauthorized SRC: 198.65.166.131:5060:UDP enc=0 bytes=548 > SIP/2.0 401 Unauthorized > From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13 > To: sip:rec...@pr...;tag=21a483426c2cd5d9b85bffe6bba40a2e.76c8 > Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bKc3ce95d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport=22013 > CSeq: 1 REGISTER > Call-ID: c3c...@pr... > WWW-Authenticate: Digest realm="proxy01.sipphone.com", nonce="4859131f88a465b9f177c8041e68d1ee6d8d51e1" > Content-Length: 0 > > ----------------26:34:48.125---------------- > >>> REGISTER sip:proxy01.sipphone.com SIP/2.0 DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=858 > REGISTER sip:proxy01.sipphone.com SIP/2.0 > From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13 > To: sip:rec...@pr... > Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bK2d1596d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport > CSeq: 2 REGISTER > Call-ID: c3c...@pr... > Contact: "recipient" <sip:recipient@72.240.235.122:22013;transport=udp> > User-Agent: OpenSIPStack v1.1.7-24 > Expires: 3600 > Max-Forwards: 70 > Authorization: Digest username="recipient", realm="proxy01.sipphone.com", nonce="4859131f88a465b9f177c8041e68d1ee6d8d51e1", uri="sip:proxy01.sipphone.com", response="4a78a8e0712b92b26d17ba786a89a4eb", opaque="", algorithm=MD5 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS, PRACK > Content-Length: 0 > > ----------------26:34:48.226---------------- > <<< SIP/2.0 200 OK SRC: 198.65.166.131:5060:UDP enc=0 bytes=509 > SIP/2.0 200 OK > From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13 > To: sip:rec...@pr...;tag=21a483426c2cd5d9b85bffe6bba40a2e.90ab > Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bK2d1596d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport=22013 > CSeq: 2 REGISTER > Call-ID: c3c...@pr... > Contact: <sip:recipient@72.240.235.122:22013;transport=udp>;expires=3600 > Content-Length: 0 > > ----------------26:34:58.666---------------- > <<< INVITE sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 SRC: 198.65.166.131:5060:UDP enc=0 bytes=1103 > INVITE sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 > From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c > To: <sip:rec...@pr...> > Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0 > Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060 > CSeq: 102 INVITE > Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46 > Contact: <sip:caller@66.54.140.46> > Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr> > Date: Wed, 18 Jun 2008 13:47:42 GMT > User-Agent: Asterisk PBX > Max-Forwards: 16 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > RemoteIP: 66.54.140.46 > Content-Type: application/sdp > Content-Length: 379 > > v=0 > o=root 16325 16325 IN IP4 66.54.140.46 > s=session > c=IN IP4 66.54.140.46 > t=0 0 > m=audio 14868 RTP/AVP 0 8 3 18 97 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=30 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > ----------------26:34:58.684---------------- > >>> SIP/2.0 100 Trying DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=411 > SIP/2.0 100 Trying > From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c > To: <sip:rec...@pr...> > Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0 > Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060 > CSeq: 102 INVITE > Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46 > Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr> > Content-Length: 0 > > ----------------26:35:24.111---------------- > <<< CANCEL sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 SRC: 198.65.166.131:5060:UDP enc=0 bytes=332 > CANCEL sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 > From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c > To: <sip:rec...@pr...> > Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0 > CSeq: 102 CANCEL > Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46 > Content-Length: 0 > > ----------------26:35:24.151---------------- > >>> SIP/2.0 200 OK DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=358 > SIP/2.0 200 OK > From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c > To: <sip:rec...@pr...> > Via: SIP/2.0/UDP 198.65.166.131;iid=2494;branch=z9hG4bK86dc.5c1c1044.0;rport=5060;received=198.65.166.131 > CSeq: 102 CANCEL > Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46 > Server: OpenSIPStack v1.1.7-24 > Content-Length: 0 > > ----------------26:35:24.163---------------- > >>> SIP/2.0 487 Request Cancelled DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=535 > SIP/2.0 487 Request Cancelled > From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c > To: <sip:rec...@pr...>;tag=a1f8ccd9d4fb1810991ba33f609baf13 > Via: SIP/2.0/UDP 198.65.166.131;iid=2494;branch=z9hG4bK86dc.5c1c1044.0;rport=5060;received=198.65.166.131 > Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060 > CSeq: 102 INVITE > Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46 > Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr> > Server: OpenSIPStack v1.1.7-24 > Content-Length: 0 > > ----------------26:35:24.263---------------- > <<< ACK sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 SRC: 198.65.166.131:5060:UDP enc=0 bytes=364 > ACK sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 > From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c > To: <sip:rec...@pr...>;tag=a1f8ccd9d4fb1810991ba33f609baf13 > Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0 > CSeq: 102 ACK > Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46 > Content-Length: 0{quote} > > Recipient OSSPhone was built using latest code in CVS on 2008-06-13. I tested by calling in through an ipKall phone number. No rings were heard from the caller phone system. Eventually the call was routed to voicemail. > > Finest regards, > Bill Root |
|
From: OSS F. A. <ope...@op...> - 2008-06-18 13:59:22
|
I'm using OSSPhone to test ATLSIP. It detects incoming calls, but it doesn't send out "180 Ringing" packets so caller doesn't get rings, and proxy eventually routes call to voicemail.
Is there a way to automatically or manually send the "180 Ringing" messages with ATLSIP?
Here is a log from OSSPhone:
{quote}----------------26:34:32.350----------------
Querying STUN server at stun01.sipphone.com.
This may take a while ...
----------------26:34:42.772----------------
STUN server "stun01.sipphone.com" replies Cone NAT, external IP 72.240.235.122
----------------26:34:47.820----------------
*** LISTENER STARTED *** [OPAL] 127.0.0.1:5060
----------------26:34:47.880----------------
*** LISTENER STARTED *** [OPAL] 192.168.0.20:5060 [*** DEFAULT LISTENER ***]
----------------26:34:47.936----------------
>>> REGISTER sip:proxy01.sipphone.com SIP/2.0 DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=629
REGISTER sip:proxy01.sipphone.com SIP/2.0
From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13
To: sip:rec...@pr...
Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bKc3ce95d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport
CSeq: 1 REGISTER
Call-ID: c3c...@pr...
Contact: "recipient" <sip:recipient@72.240.235.122:22013;transport=udp>
User-Agent: OpenSIPStack v1.1.7-24
Expires: 3600
Max-Forwards: 70
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS, PRACK
Content-Length: 0
----------------26:34:48.069----------------
<<< SIP/2.0 401 Unauthorized SRC: 198.65.166.131:5060:UDP enc=0 bytes=548
SIP/2.0 401 Unauthorized
From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13
To: sip:rec...@pr...;tag=21a483426c2cd5d9b85bffe6bba40a2e.76c8
Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bKc3ce95d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport=22013
CSeq: 1 REGISTER
Call-ID: c3c...@pr...
WWW-Authenticate: Digest realm="proxy01.sipphone.com", nonce="4859131f88a465b9f177c8041e68d1ee6d8d51e1"
Content-Length: 0
----------------26:34:48.125----------------
>>> REGISTER sip:proxy01.sipphone.com SIP/2.0 DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=858
REGISTER sip:proxy01.sipphone.com SIP/2.0
From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13
To: sip:rec...@pr...
Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bK2d1596d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport
CSeq: 2 REGISTER
Call-ID: c3c...@pr...
Contact: "recipient" <sip:recipient@72.240.235.122:22013;transport=udp>
User-Agent: OpenSIPStack v1.1.7-24
Expires: 3600
Max-Forwards: 70
Authorization: Digest username="recipient", realm="proxy01.sipphone.com", nonce="4859131f88a465b9f177c8041e68d1ee6d8d51e1", uri="sip:proxy01.sipphone.com", response="4a78a8e0712b92b26d17ba786a89a4eb", opaque="", algorithm=MD5
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS, PRACK
Content-Length: 0
----------------26:34:48.226----------------
<<< SIP/2.0 200 OK SRC: 198.65.166.131:5060:UDP enc=0 bytes=509
SIP/2.0 200 OK
From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13
To: sip:rec...@pr...;tag=21a483426c2cd5d9b85bffe6bba40a2e.90ab
Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bK2d1596d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport=22013
CSeq: 2 REGISTER
Call-ID: c3c...@pr...
Contact: <sip:recipient@72.240.235.122:22013;transport=udp>;expires=3600
Content-Length: 0
----------------26:34:58.666----------------
<<< INVITE sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 SRC: 198.65.166.131:5060:UDP enc=0 bytes=1103
INVITE sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0
From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c
To: <sip:rec...@pr...>
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060
CSeq: 102 INVITE
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Contact: <sip:caller@66.54.140.46>
Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr>
Date: Wed, 18 Jun 2008 13:47:42 GMT
User-Agent: Asterisk PBX
Max-Forwards: 16
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
RemoteIP: 66.54.140.46
Content-Type: application/sdp
Content-Length: 379
v=0
o=root 16325 16325 IN IP4 66.54.140.46
s=session
c=IN IP4 66.54.140.46
t=0 0
m=audio 14868 RTP/AVP 0 8 3 18 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
----------------26:34:58.684----------------
>>> SIP/2.0 100 Trying DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=411
SIP/2.0 100 Trying
From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c
To: <sip:rec...@pr...>
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060
CSeq: 102 INVITE
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr>
Content-Length: 0
----------------26:35:24.111----------------
<<< CANCEL sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 SRC: 198.65.166.131:5060:UDP enc=0 bytes=332
CANCEL sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0
From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c
To: <sip:rec...@pr...>
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0
CSeq: 102 CANCEL
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Content-Length: 0
----------------26:35:24.151----------------
>>> SIP/2.0 200 OK DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=358
SIP/2.0 200 OK
From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c
To: <sip:rec...@pr...>
Via: SIP/2.0/UDP 198.65.166.131;iid=2494;branch=z9hG4bK86dc.5c1c1044.0;rport=5060;received=198.65.166.131
CSeq: 102 CANCEL
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Server: OpenSIPStack v1.1.7-24
Content-Length: 0
----------------26:35:24.163----------------
>>> SIP/2.0 487 Request Cancelled DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=535
SIP/2.0 487 Request Cancelled
From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c
To: <sip:rec...@pr...>;tag=a1f8ccd9d4fb1810991ba33f609baf13
Via: SIP/2.0/UDP 198.65.166.131;iid=2494;branch=z9hG4bK86dc.5c1c1044.0;rport=5060;received=198.65.166.131
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060
CSeq: 102 INVITE
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr>
Server: OpenSIPStack v1.1.7-24
Content-Length: 0
----------------26:35:24.263----------------
<<< ACK sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 SRC: 198.65.166.131:5060:UDP enc=0 bytes=364
ACK sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0
From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c
To: <sip:rec...@pr...>;tag=a1f8ccd9d4fb1810991ba33f609baf13
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0
CSeq: 102 ACK
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Content-Length: 0{quote}
Recipient OSSPhone was built using latest code in CVS on 2008-06-13. I tested by calling in through an ipKall phone number. No rings were heard from the caller phone system. Eventually the call was routed to voicemail.
Finest regards,
Bill Root
|
|
From: OSS F. A. <ope...@op...> - 2008-06-17 02:01:48
|
I have patched CVS. Thanks for reporting it.
Joegen
> {quote:title=optotronic wrote:}{quote}
> A previous message in the OpenSipStack forum discussed re-enabling logging for ATLSIP:
> < http://www.opensourcesip.org:8080/clearspacex/message/4130#4130 >
>
> The proposed solution, as I read it, improves the log appearance by adding CRLF after ">>>" and "<<<" lines. It doesn't actually add any information to the log.
>
> To add CRLF and the incoming packet to the log, add
> traceStream << "\r\n";
> traceStream << msg->AsString();
> after the existing 'traceStream << "<<< "' statement in SIPTransportManager::OnProcessInbound in SIPTransportManager.cxx. The new code was added at line 1134 in my copy of the file from 2008-06-13.
>
> (Of course, you could always gang the new output onto the existing traceStream statement.)
>
> Similarly, you can add
> traceStream << "\r\n";
> traceStream << packet->AsString();
> after the existing 'traceStream << ">>> "' statement in SIPTransportManager::OnProcessOutbound. The new code was added at line 1233 in my copy.
>
> (The referenced file is part of the OpenSipStack project. You, will of course, need to be able to build OpenSipStack and ATLSIP.)
>
> The change might make the log too verbose once a connection is made. Since I haven't been able to get a connection working yet, I need the additional information to track down the problem.
|
|
From: OSS F. A. <ope...@op...> - 2008-06-16 22:09:35
|
A previous message in the OpenSipStack forum discussed re-enabling logging for ATLSIP: < http://www.opensourcesip.org:8080/clearspacex/message/4130#4130 > The proposed solution, as I read it, improves the log appearance by adding CRLF after ">>>" and "<<<" lines. It doesn't actually add any information to the log. To add CRLF and the incoming packet to the log, add traceStream << "\r\n"; traceStream << msg->AsString(); after the existing 'traceStream << "<<< "' statement in SIPTransportManager::OnProcessInbound in SIPTransportManager.cxx. The new code was added at line 1134 in my copy of the file from 2008-06-13. (Of course, you could always gang the new output onto the existing traceStream statement.) Similarly, you can add traceStream << "\r\n"; traceStream << packet->AsString(); after the existing 'traceStream << ">>> "' statement in SIPTransportManager::OnProcessOutbound. The new code was added at line 1233 in my copy. (The referenced file is part of the OpenSipStack project. You, will of course, need to be able to build OpenSipStack and ATLSIP.) The change might make the log too verbose once a connection is made. Since I haven't been able to get a connection working yet, I need the additional information to track down the problem. |
|
From: OSS F. A. <ope...@op...> - 2008-06-03 11:34:28
|
test 2 |
|
From: Raymund N. <ope...@op...> - 2008-04-28 09:25:33
|
This is a Test |
|
From: Ilian J. C. P. <ip...@so...> - 2008-04-18 09:44:24
|
Hi, You should be able to get the sparkplug code from CVS head. Check-out the ATLSIP project and look under the OSSPhoneSvc folder. You should be able to compile it against the Sparkplug kit. Take note, we are not actively maintaining the sparkplug as of the moment. So you're on your own now. Regards, Ilian ho wrote: > where are ossphone sparkplug cvs ? > > ------------------------------------------------------------------------- > This SF.net email is sponsored by the 2008 JavaOne(SM) Conference > Don't miss this year's exciting event. There's still time to save $100. > Use priority code J8TL2D2. > http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone > _______________________________________________ > Opensipstack-atlsipdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-atlsipdevel > > > |
|
From: ho <kev...@ho...> - 2008-04-18 00:18:24
|
where are ossphone sparkplug cvs ? |
|
From: Ilian J. C. P. <ip...@so...> - 2008-01-28 09:59:02
|
Hi, I'm currently implementing modifications for this. Testing is on the way. Thanks! - Ilian Joegen E. Baclor wrote: > This is a known bug. I have opened a ticket in Ilian's queue to provide > a bug fix ASAP. You should hear from him as soon as CVS is patched. > Thank you for the reminder. > > Joegen > > Jonas Gauffin wrote: > >> Hello >> >> I'm trying to use ATLSIP to authorize against FreeSWITCH. FreeSWITCH >> complains on an invalid authorization header, and it's because ATLSIP >> doesnt include "qop", "cnonce" and "nc" >> As I understand it, those parameters are used to add some extra >> protection. Can they be added in any way? >> >> Regards, >> Jonas >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Microsoft >> Defy all challenges. Microsoft(R) Visual Studio 2008. >> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >> _______________________________________________ >> Opensipstack-atlsipdevel mailing list >> Ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-atlsipdevel >> >> >> >> > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-atlsipdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-atlsipdevel > > > |
|
From: Joegen E. B. <joe...@gm...> - 2008-01-27 04:46:31
|
This is a known bug. I have opened a ticket in Ilian's queue to provide a bug fix ASAP. You should hear from him as soon as CVS is patched. Thank you for the reminder. Joegen Jonas Gauffin wrote: > Hello > > I'm trying to use ATLSIP to authorize against FreeSWITCH. FreeSWITCH > complains on an invalid authorization header, and it's because ATLSIP > doesnt include "qop", "cnonce" and "nc" > As I understand it, those parameters are used to add some extra > protection. Can they be added in any way? > > Regards, > Jonas > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-atlsipdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-atlsipdevel > > > |
|
From: Jonas G. <jon...@gm...> - 2008-01-25 08:49:15
|
Hello I'm trying to use ATLSIP to authorize against FreeSWITCH. FreeSWITCH complains on an invalid authorization header, and it's because ATLSIP doesnt include "qop", "cnonce" and "nc" As I understand it, those parameters are used to add some extra protection. Can they be added in any way? Regards, Jonas |
|
From: Ilian J. C. P. <ip...@so...> - 2007-11-19 09:09:20
|
Hi. I now have a rough implementation for call hold and call transfer (attended/unattended). The thing works but the implementation leaves a lot to be desired. I will try to iron out the kinks first before applying the changes to CVS. Expect this within this week or next week at the latest. - Ilian |
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