Re: [asterisk-users] Asterisk 15.6.1. Symbol pjsip_tls_transport_start2 not found
./configure --with-crypto --with-ssl --with-srtp --with-pjproject-bundled The culprit of this behavior is option --with-ssl Version 15.5 does not have this problem. 26.09.2018 16:46, George Joseph пишет: On Tue, Sep 25, 2018 at 2:18 PM Dmitriy Serov <mailto:serov@gmail.com>> wrote: Hello. After successful compilation 15.6.1 (bundled pjsip) and start asterisk i has error Symbol pjsip_tls_transport_start2 not found. /main/libasteriskpj.exports does not containg pjsip_tls_transport_start2 and pjsip_tls_transport_start. More: * All versions before (including 15.5) has not such error on this computer (ubuntu 18.04). * with 15.6.0, 15.6.1 has error on this computer I just tried building and running 15.6 on 18.04 and didn't have an issue. Can you double check that you don't have any libpj* files anywhere in /usr/lib or /usr/local/lib? You can also run "aptitude search pjsip pjproject" to make sure the packages aren't installed. * * Other server (ubuntu 16.04) has not this error with 15.6.0 What's wrong? Where to start searching for reasons? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *George Joseph* Digium - A Sangoma Company | Software Developer | Software Engineering 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct/fax: +1 256 428 6012 Check us out at: https://digium.com <https://digium.com/> · https://sangoma.com <https://sangoma.com/> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 15.6.1. Symbol pjsip_tls_transport_start2 not found
Hello. After successful compilation 15.6.1 (bundled pjsip) and start asterisk i has error Symbol pjsip_tls_transport_start2 not found. /main/libasteriskpj.exports does not containg pjsip_tls_transport_start2 and pjsip_tls_transport_start. More: * All versions before (including 15.5) has not such error on this computer (ubuntu 18.04). * with 15.6.0, 15.6.1 has error on this computer * Other server (ubuntu 16.04) has not this error with 15.6.0 What's wrong? Where to start searching for reasons? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 15.3.0-rc1 regression
Asterisk 15.2.0, pjsip Yesterday I installed 15.3.0-rc1 (after 15.2.0). Today I had to rollback to 15.2.0 (not 15.2.2). The reason is: the loaded server very often hung on the same places: SUBSCRIBE/NOTIFY processing. Unloading modules was the solution at that time. ; Off SUBSCRIBE + NOTIFY noload => res_pjsip_pubsub.so noload => res_pjsip_exten_state.so noload => res_pjsip_dialog_info_body_generator.so noload => res_pjsip_mwi_body_generator.so noload => res_pjsip_pidf_body_generator.so noload => res_pjsip_xpidf_body_generator.so At 15.3.0 asterisk stopped processing authorization, because it started to issue an error on the SUBSCRIBE packet. More precisely: the server believed that the client is authorized, and the client (PhonerLite) believed that authorization failed. Try to load the modules res_pjsip_pubsub and res_pjsip_exten_state has led to renewed hangs. Rollback to 15.2.2 did not bring success. Registering without res_pjsip_pubsub led to the client thinking that it was authorized, and the server did not think so. Whether the res_pjsip_pubsub module is mandatory for successful work? P.S. ASTERISK-27474. Bug not fixed in 15.3.0-rc1. This is very sad, because encryption should be mandatory in the modern world. Dmitriy Serov. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?
Hello.
A little sub from my dialplan:
[sub-Read]
exten => s,1,NoOp(Read)
same => n,Set(LOCAL(tmp_record_file)=/tmp/asterisk-in/${EPOCH})
same => n,Monitor(wav16,${tmp_record_file},o)
same => n,Read(tmp_ext,${ARG2},${ARG3},${ARG4},${ARG5},${ARG6})
same => n,StopMonitor()
same => n,NoOp(ReadStatus=${READSTATUS})
same => n,Gotoif($[ ${LEN(${tmp_ext})} > 0 ]?end)
same =>
n,AGI(agi-ruvoip-net.php,speeddial-voice,${ARG1},${tmp_record_file}-in.wav16)
same => n,NoOp(Voice recognition result: "${agi_result}")
same => n,Gotoif($[ "${agi_result}" != "found" ]?end)
same => n,Return(${agi_call_exten})
same => n(end),return(${tmp_ext})
21.01.2018 2:57, Jonathan H пишет:
On 20 January 2018 at 23:30, Tim S wrote:
I have seen this take over 2 seconds before on a sluggish machine.
Thanks - my host uses SSD and everything seems pretty quick, but I'll
give it a 1 second pause.
you'd need to pipe that to a Google Speech API tunnel.
That's probably not something you can hack away at with simple
Asterisk dialplan applications.
Funnily enough, I had just found an old reply from last year to
another similar question:
-- Forwarded message --
From: Matt Riddell
Date: 22 September 2017 at 16:01
Subject: Re: [asterisk-users] Asterisk 15, Jack, streams, speech recognition…
so many questions!
At least in older versions you can use EAGI to get a handle to the audio stream.
So I had a look and found this:
https://stackoverflow.com/questions/34026698/asterisk-write-plugin-to-catch-voice-stream
And read this:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_EAGI
There's a few knowledge gaps, but I think with a few days reading and
the great help here, we might have a solution :)
This is all very helpful - if anyone else feels like wading in, please do.
Many thanks!
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Re: [asterisk-users] Chan Local, Originate and slin
AMI action Originate has param "Codecs". I think it helps.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+ManagerAction_Originate
22.11.2017 13:24, Kseniya Blashchuk пишет:
Hi all!
Asterisk 13.1.0 Ubuntu 16.04, all latest.
Can anybody explain this to me - I run Originate command from dialplan:
same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum})
and I get crazy sound distortion in the conference, and I see that
transcoding takes place here:
NativeFormats: (slin192)
WriteFormat: slin
ReadFormat: slin192
WriteTranscode: Yes (slin@8000)->(slin@192000)
ReadTranscode: No
When I do the same from a call file like:
same => n,System(printf "Action: Originate\nActionID: 1\nChannel:
Local/${number}@mycontext\nApplication: Confbridge\nData:
${confnum}\n" > /var/spool/asterisk/outgoing/${number}-${confnum})
the sound is perfect and this is what I see on the channel params:
NativeFormats: (slin)
WriteFormat: slin
ReadFormat: slin
WriteTranscode: No
ReadTranscode: No
Can anybody explain what is going on?
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[asterisk-users] Return without Gosub: stack is empty
[sub-out-do-dial]
exten => s,1,NoOp(Dial)
same => n,NoOp(FirstChannel: ${CHANNEL})
same => n,Dial(,60,gF)
same => n,NoOp(SecondChannel: ${CHANNEL})
same => n,Return()
[some]
exten => s,1,GoSub(sub-out-do-dial,s,1)
In case of the destination channel hangs up in log i see:
ERROR[26570][C-00e7] app_stack.c: Return without Gosub: stack is empty
Yes. SecondChannel != FirstChannel.
Is it a wrong way to use Dial(g)+Gosub? Or any Dial in sub?
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Re: [asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity
Joshua, issue has been filed. Thank you! https://issues.asterisk.org/jira/browse/ASTERISK-26689 03.01.2017 20:58, Joshua Colp пишет: On Mon, Dec 19, 2016, at 06:36 AM, Dmitriy Serov wrote: Yes, this means the remote end was not sending any audio stream. But it shouldn't. According to [1] before remote end send OK or ACK there is one way SDP, no any audio stream. PJSIP channel (option rtp_timeout) does not take this one. Isn't it a mistake? What could be workarounds? Looking at the code we don't take that scenario into account it seems. Please file an issue[1] and we'll see if we can do something about it. [1] https://issues.asterisk.org/jira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity
Yes, this means the remote end was not sending any audio stream. But it shouldn't. According to [1] before remote end send OK or ACK there is one way SDP, no any audio stream. PJSIP channel (option rtp_timeout) does not take this one. Isn't it a mistake? What could be workarounds? 19.12.2016 11:33, Jean Aunis пишет: This means the remote end was not sending any audio stream, or the audio stream was not received by Asterisk. The problem may have many different reasons, but often it is a network-related issue. Le 16/12/2016 à 21:19, Dmitriy Serov a écrit : Today I faced a problem. Please help to solve this problem. Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware v2.06(AAGJ.9)C1 Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk). Call using early media (183 Session in progress) and rtp_timeout=10. After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/xxx-027b' for lack of RTP activity in 10 seconds SIP dump is attached. According to [1] before called user agent send OK or ACK there is one way SDP. In sip dump (attached) i didn't find such SIP packets. Whether that call was canceled due to RTP inactivity? Any help is welcome. [1] https://www.ietf.org/proceedings/46/I-D/draft-ietf-sip-183-00.txt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity
Today I faced a problem. Please help to solve this problem. Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware v2.06(AAGJ.9)C1 Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk). Call using early media (183 Session in progress) and rtp_timeout=10. After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/xxx-027b' for lack of RTP activity in 10 seconds SIP dump is attached. According to [1] before called user agent send OK or ACK there is one way SDP. In sip dump (attached) i didn't find such SIP packets. Whether that call was canceled due to RTP inactivity? Any help is welcome. [1] https://www.ietf.org/proceedings/46/I-D/draft-ietf-sip-183-00.txt INVITE sip:8xxx6yyy...@txxx37.ru SIP/2.0 Via: SIP/2.0/UDP 11.111.11.11:5060;rport;branch=z9hG4bKPjE3CIeR4UeZyQiqayKsBmGPtYTD7wM.Ql Max-Forwards: 70 From: "007" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC To: sip:8xxx6yyy...@txxx37.ru Contact: "007" Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg CSeq: 10072 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 90 Min-SE: 90 User-Agent: Keenetic Plus DECT Authorization: Digest username="login", realm="ruvoip.net", nonce="1481885583/bcb53e85a740689479f116a96fc7086b", uri="sip:8xxx6yyy...@txxx37.ru", response="843f8211896b5b05fcf3a633d6d8eedf", algori Content-Type: application/sdp Content-Length: 326 v=0 o=- 3690874445 3690874445 IN IP4 11.111.11.11 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4022 RTP/AVP 0 8 109 96 c=IN IP4 11.111.11.11 b=TIAS:64000 a=rtcp:4023 IN IP4 11.111.11.11 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:109 G726-32/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 [2016-12-16 13:53:03] VERBOSE[13985] netsock2.c: Using SIP RTP Audio TOS bits 184 [2016-12-16 13:53:03] VERBOSE[13985] netsock2.c: Using SIP RTP Audio CoS mark 5 [2016-12-16 13:53:03] VERBOSE[13985] res_pjsip_logger.c: <--- Transmitting SIP response (352 bytes) to UDP:11.111.11.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 11.111.11.11:5060;rport=5060;received=11.111.11.11;branch=z9hG4bKPjE3CIeR4UeZyQiqayKsBmGPtYTD7wM.Ql Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg From: "007" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC To: CSeq: 10072 INVITE Server: ruVoIP.net PBX Content-Length: 0 [2016-12-16 13:53:05] VERBOSE[8346] res_pjsip_logger.c: <--- Transmitting SIP response (849 bytes) to UDP:11.111.11.11:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 11.111.11.11:5060;rport=5060;received=11.111.11.11;branch=z9hG4bKPjE3CIeR4UeZyQiqayKsBmGPtYTD7wM.Ql Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg From: "007" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC To: ;tag=f30b688b-3358-4107-9992-fb6e3923bc15 CSeq: 10072 INVITE Server: ruVoIP.net PBX Contact: Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER Content-Type: application/sdp Content-Length: 266 v=0 o=- 3690874445 3690874447 IN IP4 222.222.222.22 s=ruVoIP.net PBX c=IN IP4 222.222.222.22 t=0 0 m=audio 25094 RTP/AVP 8 0 96 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=maxptime:150 a=sendrecv [2016-12-16 13:53:05] VERBOSE[6631] res_pjsip_logger.c: <--- Received SIP request (812 bytes) from UDP:11.111.11.11:5060 ---> UPDATE sip:222.222.222.22:5060 SIP/2.0 Via: SIP/2.0/UDP 11.111.11.11:5060;rport;branch=z9hG4bKPjL4elACGx8R4357HYMDC-eUWi5f5peNYk Max-Forwards: 70 From: "007" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC To: sip:8xxx6yyy...@txxx37.ru;tag=f30b688b-3358-4107-9992-fb6e3923bc15 Contact: "007" Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg CSeq: 10073 UPDATE Supported: replaces, 100rel, timer, norefersub Session-Expires: 90 Min-SE: 90 Content-Type: application/sdp Content-Length: 271 v=0 o=- 3690874445 3690874446 IN IP4 11.111.11.11 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4022 RTP/AVP 8 96 c=IN IP4 11.111.11.11 b=TIAS:64000 a=rtcp:4023 IN IP4 11.111.11.11 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=sendrecv [2016-12-16 13:53:05] VERBOSE[8346] res_pjsip_logger.c: <--- Transmitting SIP response (910 bytes) to UDP:11.111.11.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 11.111.11.11:5060;rport=5060;received=11.111.11.11;branch=z9hG4bKPjL4elACGx8R4357HYMDC-eUWi5f5peNYk Call-ID: 4dk-XDPyPZ1dW9DLnVB7fy1DgBIuDFRg From: "007" ;tag=I9oDGDVi9zj3Elh6MqcA6Oykb18S5QVC To: ;tag=f30b688b-3358-4107-9992-fb6e3923bc15 CSeq: 10073 UPDATE Session-Expires: 90;refresher=uac Require: timer Contact: Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER Supported: 100rel, timer, replaces, norefersub Server: ruVoIP.net PBX Content-Type: application/sdp Content-Length: 242 v=0 o=- 3690874445 3690874448 IN IP4 222.222.222.22 s=ruVoIP.net PBX c=IN IP4 222.222.222.22 t=0 0 m=audio 25094
Re: [asterisk-users] Upgrading asterisk 13.7 to 13.11. Segfaults
06.09.2016 17:08, George Joseph пишет: On Tue, Sep 6, 2016 at 7:58 AM, Dmitriy Serov <mailto:serov@gmail.com>> wrote: 06.09.2016 16:42, George Joseph пишет: On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov mailto:serov@gmail.com>> wrote: Hello. Several months server working on asterisk 13.7 and pjproject 2.5 (installed separately). Once a day the server crashes or hangs and is familiar sores that written watchdogs. Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5). Solved all the problems with compilation I started asterisk several times and each time after 5-7 seconds was seg fault. So I didn't get to use the new version of asterisk. And I really wanted to be able to find and remove the cause. I would be grateful for any help. All 3 of the backtraces are in different pjproject places which is weird. Makes me think there's still a library mismatch somewhere. Is the separately compiled pjproject still installed on both the build machine and the server? They /should/ be ignored if --with-pjproject-bundled is specified but you might want to remove them and try again. George, thank you that you responded. What I was hoping :) I was also very surprised that SF can happen so quickly and so consistently. With all this in completely different places. When installed separately pjproject source code with bundled not even compiled. Of course, I uninstalled and cleaned pjproject according to this instructions: https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject <https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject> At the moment I have restore separately pjproject and asterisk 13.7. But there is a directory with configured and compiled asterisk 13.11. Happy to provide the contents of the autoconfiguration files. config.log and makeopts would be useful if you can zip them up. Also, what distribution and version are you running? As Jonathan suggested, could there have been any left over or third-party modules left in the asterisk modules directory? https://ruvoip.net/_other/voip/2016-09-05/makeopts_configlogs.zip Slackware 14.1 x64 with some packages upgraded and some of libraries compiled from source. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrading asterisk 13.7 to 13.11. Segfaults
06.09.2016 16:51, Jonathan H пишет: All your libraries, kernel, headers and build tools up to date? I can't be sure because I do not know the required versions of these libraries. When you configure and build asterisk, no error detected. The server has not a new kernel 3.13.6 #4 SMP Wed Aug 20 17:52:10 MSK 2014 x86_64 The other thing that might be worth noting is the warning along the lines of "contains modules that were not installed by this version of Asterisk". Might be worth deleting anything that appears there, and then starting Asterisk. I don't use the autoloading of modules. All the necessary modules written manually in modules.conf. At the end of the installation it reported the extra modules (thanks) and they removed me as "garbage". On 6 September 2016 at 14:42, George Joseph wrote: On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov wrote: Hello. Several months server working on asterisk 13.7 and pjproject 2.5 (installed separately). Once a day the server crashes or hangs and is familiar sores that written watchdogs. Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5). Solved all the problems with compilation I started asterisk several times and each time after 5-7 seconds was seg fault. So I didn't get to use the new version of asterisk. And I really wanted to be able to find and remove the cause. I would be grateful for any help. All 3 of the backtraces are in different pjproject places which is weird. Makes me think there's still a library mismatch somewhere. Is the separately compiled pjproject still installed on both the build machine and the server? They should be ignored if --with-pjproject-bundled is specified but you might want to remove them and try again. first segfault: Program terminated with signal 11, Segmentation fault. #0 0x7fe19337357e in pj_atomic_dec_and_get (atomic_var=0x1a9a) at ../src/pj/os_core_unix.c:962 962pj_mutex_lock( atomic_var->mutex ); backtrace: https://ruvoip.net/_other/voip/2016-09-05-1/backtrace-threads.txt log (16 MB) : https://ruvoip.net/_other/voip/2016-09-05-1/full.txt second segfault: Program terminated with signal 11, Segmentation fault. #0 0x7f5777412788 in pjsip_auth_clt_reinit_req (sess=0x7f56f4088a30, rdata=0x7f56ec2b0c98, old_request=0x7f56f47775a8, new_request=0x7f56d5b38980) at ../src/pjsip/sip_auth_client.c:1144 1144PJ_ASSERT_RETURN(old_request->msg->type == PJSIP_REQUEST_MSG, backtrace: https://ruvoip.net/_other/voip/2016-09-05-2/backtrace-threads.txt log (15 MB) : https://ruvoip.net/_other/voip/2016-09-05-2/full.txt third segfault: Program terminated with signal 11, Segmentation fault. #0 0x7f592bcad53d in pj_pool_alloc (pool=0x7f58002c, size=80) at ../include/pj/pool_i.h:60 60void *ptr = pj_pool_alloc_from_block(pool->block_list.next, size); backtrace: https://ruvoip.net/_other/voip/2016-09-05-3/backtrace-threads.txt log (11 MB) : https://ruvoip.net/_other/voip/2016-09-05-3/full.txt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrading asterisk 13.7 to 13.11. Segfaults
06.09.2016 16:42, George Joseph пишет: On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov <mailto:serov@gmail.com>> wrote: Hello. Several months server working on asterisk 13.7 and pjproject 2.5 (installed separately). Once a day the server crashes or hangs and is familiar sores that written watchdogs. Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5). Solved all the problems with compilation I started asterisk several times and each time after 5-7 seconds was seg fault. So I didn't get to use the new version of asterisk. And I really wanted to be able to find and remove the cause. I would be grateful for any help. All 3 of the backtraces are in different pjproject places which is weird. Makes me think there's still a library mismatch somewhere. Is the separately compiled pjproject still installed on both the build machine and the server? They /should/ be ignored if --with-pjproject-bundled is specified but you might want to remove them and try again. George, thank you that you responded. What I was hoping :) I was also very surprised that SF can happen so quickly and so consistently. With all this in completely different places. When installed separately pjproject source code with bundled not even compiled. Of course, I uninstalled and cleaned pjproject according to this instructions: https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject At the moment I have restore separately pjproject and asterisk 13.7. But there is a directory with configured and compiled asterisk 13.11. Happy to provide the contents of the autoconfiguration files. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Upgrading asterisk 13.7 to 13.11. Segfaults
Hello. Several months server working on asterisk 13.7 and pjproject 2.5 (installed separately). Once a day the server crashes or hangs and is familiar sores that written watchdogs. Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5). Solved all the problems with compilation I started asterisk several times and each time after 5-7 seconds was seg fault. So I didn't get to use the new version of asterisk. And I really wanted to be able to find and remove the cause. I would be grateful for any help. first segfault: Program terminated with signal 11, Segmentation fault. #0 0x7fe19337357e in pj_atomic_dec_and_get (atomic_var=0x1a9a) at ../src/pj/os_core_unix.c:962 962pj_mutex_lock( atomic_var->mutex ); backtrace: https://ruvoip.net/_other/voip/2016-09-05-1/backtrace-threads.txt log (16 MB) : https://ruvoip.net/_other/voip/2016-09-05-1/full.txt second segfault: Program terminated with signal 11, Segmentation fault. #0 0x7f5777412788 in pjsip_auth_clt_reinit_req (sess=0x7f56f4088a30, rdata=0x7f56ec2b0c98, old_request=0x7f56f47775a8, new_request=0x7f56d5b38980) at ../src/pjsip/sip_auth_client.c:1144 1144PJ_ASSERT_RETURN(old_request->msg->type == PJSIP_REQUEST_MSG, backtrace: https://ruvoip.net/_other/voip/2016-09-05-2/backtrace-threads.txt log (15 MB) : https://ruvoip.net/_other/voip/2016-09-05-2/full.txt third segfault: Program terminated with signal 11, Segmentation fault. #0 0x7f592bcad53d in pj_pool_alloc (pool=0x7f58002c, size=80) at ../include/pj/pool_i.h:60 60void *ptr = pj_pool_alloc_from_block(pool->block_list.next, size); backtrace: https://ruvoip.net/_other/voip/2016-09-05-3/backtrace-threads.txt log (11 MB) : https://ruvoip.net/_other/voip/2016-09-05-3/full.txt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The combination of app_record and app_read
There is a separate app for recording voice (app_record) or dtmf input (app_read). But there is no way to allow the user to choose to enter by voice or by keypad in same time. app_record analyzes the dtmf input, but only the # and * (to quit). Nothing stored in variables :( Is there a workaround for recording dtmf events through AMI during the execution app_record. But this is a nasty path that is associated with the need to synchronize the different parts. Are there any other possibilities to assist the user enter the number? There may be developers who do it for the community, seemingly simple edits to app_record that should not break compatibility? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP outgoing INVITE and "contact" value
asterisk 13.8.7, PJSIP. One VoIP provider requires a specific value in the field "contact" of a INVITE. What setting does indicate the value will be in this field (instead "asterisk")? Thanks. currect settings (with templates): [srv_d22778](srv-auth) username=999 password=secret [srv_d22778](srv-aor) contact=sip:999x...@reg.dc-tc.com [srv_d22778](srv-endpoint) accountcode=5100 from_domain=reg.dc-tc.com aors=srv_d22778 outbound_auth=srv_d22778 [srv_d22778](srv-registration) outbound_auth=srv_d22778 client_uri=sip:999x...@reg.dc-tc.com contact_user=srv_d22778 endpoint=srv_d22778 server_uri=sip:reg.dc-tc.com INVITE sip:79265x...@reg.dc-tc.com SIP/2.0 Via: SIP/2.0/UDP 85.142.148.80:5060;rport;branch=z9hG4bKPj05fd6429-c1a3-4bfb-87f8-df094c6532e4 From: "999" ;tag=2b3456b9-8adc-4413-8012-98de85f949a8 To: Contact: -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Upgrading 13.7 (external pjproject) to 13.9 (bundled pjproject)
Today was another attempt to upgrade to version 13.9 (git). 1. The result was https://issues.asterisk.org/jira/browse/ASTERISK-25970 Had to temporarily block this contact and look forward to advice of how to fix it. 2. Also, an unpleasant surprise was the increase in CPU usage from 10-50% to 200-400% (4 cores). Stable CPU overusage with the same build options (DONT_OPTIMIZE, DEBUG_THREADS, BETTER_BACKTRACES, BUILD_NATIVE) 3. After 20 minutes of this work the server has ceased to respond promptly to registrations. "core show locks" is attached. Suggestions about solving the problem are very welcome. Dmitriy. Setting max files open to 30 === === 13.9.0-rc1 === Currently Held Locks === === === (): (times locked) === === Thread ID: 0x7fbaedb2c700 LWP:25598 (worker_start started at [ 1077] threadpool.c worker_thread_start()) === ---> Lock #0 (sorcery.c): RDLOCK 1883 ast_sorcery_retrieve_by_fields &(&object_type->wizards)->lock 0x2243420 (1) main/backtrace.c:59 __ast_bt_get_addresses() (0x466440+1D) main/lock.c:866 __ast_rwlock_rdlock() (0x53652a+BA) main/sorcery.c:1884 ast_sorcery_retrieve_by_fields() (0x5bf860+C2) res_pjsip/pjsip_options.c:290 find_an_endpoint() res_pjsip/pjsip_options.c:342 qualify_contact() res_pjsip/pjsip_options.c:443 qualify_contact_task() main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D) main/threadpool.c:1320 execute_tasks() main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D) main/threadpool.c:351 threadpool_execute() main/threadpool.c:1103 worker_active() main/threadpool.c:1024 worker_start() main/utils.c:1235 dummy_start() :0 start_thread() :0 __clone() (0x7fbb8bb68860+6D) === --- === === Thread ID: 0x7fbaedab0700 LWP:25596 (worker_start started at [ 1077] threadpool.c worker_thread_start()) === ---> Lock #0 (sorcery.c): RDLOCK 1883 ast_sorcery_retrieve_by_fields &(&object_type->wizards)->lock 0x2243420 (1) main/backtrace.c:59 __ast_bt_get_addresses() (0x466440+1D) main/lock.c:866 __ast_rwlock_rdlock() (0x53652a+BA) main/sorcery.c:1884 ast_sorcery_retrieve_by_fields() (0x5bf860+C2) res_pjsip/pjsip_options.c:290 find_an_endpoint() res_pjsip/pjsip_options.c:342 qualify_contact() res_pjsip/pjsip_options.c:443 qualify_contact_task() main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D) main/threadpool.c:1320 execute_tasks() main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D) main/threadpool.c:351 threadpool_execute() main/threadpool.c:1103 worker_active() main/threadpool.c:1024 worker_start() main/utils.c:1235 dummy_start() :0 start_thread() :0 __clone() (0x7fbb8bb68860+6D) === --- === === Thread ID: 0x7fbb25795700 LWP:25597 (worker_start started at [ 1077] threadpool.c worker_thread_start()) === ---> Lock #0 (sorcery.c): RDLOCK 1883 ast_sorcery_retrieve_by_fields &(&object_type->wizards)->lock 0x2243420 (1) main/backtrace.c:59 __ast_bt_get_addresses() (0x466440+1D) main/lock.c:866 __ast_rwlock_rdlock() (0x53652a+BA) main/sorcery.c:1884 ast_sorcery_retrieve_by_fields() (0x5bf860+C2) res_pjsip/pjsip_options.c:290 find_an_endpoint() res_pjsip/pjsip_options.c:342 qualify_contact() res_pjsip/pjsip_options.c:443 qualify_contact_task() main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D) main/threadpool.c:1320 execute_tasks() main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D) main/threadpool.c:351 threadpool_execute() main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+8CB8) main/threadpool.c:1024 worker_start() main/utils.c:1235 dummy_start() :0 start_thread() :0 __clone() (0x7fbb8bb68860+6D) === --- === === Thread ID: 0x7fbb254ad700 LWP:25895 (worker_start started at [ 1077] threadpool.c worker_thread_start()) === ---> Lock #0 (sorcery.c): RDLOCK 1883 ast_sorcery_retrieve_by_fields &(&object_type->wizards)->lock 0x2243420 (1) main/backtrace.c:59 __ast_bt_get_addresses() (0x466440+1D) main/lock.c:866 __ast_rwlock_rdlock() (0x53652a+BA) main/sorcery.c:1884 ast_sorcery_retrieve_by_fields() (0x5bf860+C2) res_pjsip/pjsip_options.c:290 find_an_endpoint() res_pjsip/pjsip_options.c:342 qualify_contact() res_pjsip/pjsip_options.c:443 qualify_contact_task() main/taskprocessor.c:852 ast_taskprocessor_
[asterisk-users] A few questions about bundled pjproject
At the moment I plan to migrate from asterisk 13.7 to 13.8. Because of relatively frequent updates I am building asterisk from a directory that is updated via git switch to the desired branch. 1. Will be updated pjproject patches with "git pull"? 2. Will update himself pjproject? 3. And what should be done to be sure that they are the most "fresh"? 4. What is the status of patches to pjproject? It is expected that they should eventually become part of pjproject (to be accepted by the maintainers)? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loss of devices registration (pjsip)
Thanks, George Joseph! Now a lot clearer the reasons for this behavior. It turns out that in the case of devices there are two ways to understand that they are "alive": 1. Registration from device to server 2. qualify from the server to the client And the second way does not seems superfluous. Since calling to the device this way will be used. if qualify doesn't working, then this call will not take place. Most likely the problem is that the device is behind two NAT (from your ISP and your own router). Can you advise how to configure the client in this case? Is it necessary to use a stun (did not seem to help, and it only works in the case of RTP) or proxy? Thanks. 21.03.2016 23:32, George Joseph пишет: On Mon, Mar 21, 2016 at 11:58 AM, Dmitriy Serov <mailto:serov@gmail.com>> wrote: Good day. Asterisk 13.7.2, res_pjsip. There is a problem of loss of registration of several devices. This happens not on all devices, but problem devices a lot. Below is the log of registration of a contact of one device. Is suspect two things: 1. delete a contact after the contact is added. But, like, it's a feature of code that may already be fixed. 2. deleting a contact much earlier than the 90 seconds specified during the registration Would be grateful for any clues. Dmitriy Serov. expiration settings: [common-aor](!) type=aor qualify_frequency=60 default_expiration=120 maximum_expiration=600 minimum_expiration=90 log: [2016-03-21 20:39:58] VERBOSE[30251] res_pjsip_registrar.c: Added contact 'sip:17367@46.39.229.18:37910 <http://sip:17367@46.39.229.18:37910>' to AOR '17367' with expiration of 90 seconds The client just registered [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:37910 <http://sip:17367@46.39.229.18:37910> has been created We added a new contact [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:27143 <http://sip:17367@46.39.229.18:27143> has been deleted We deleted the old contact [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:37910 <http://sip:17367@46.39.229.18:37910> is now Reachable. RTT: 41.882 msec We qualified the contact successfully [2016-03-21 20:41:01] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:37910 <http://sip:17367@46.39.229.18:37910> is now Unreachable. RTT: 0.000 msec At the next qualify, we couldn't reach the contact [2016-03-21 20:41:06] VERBOSE[3827] res_pjsip_registrar.c: Added contact 'sip:17367@46.39.229.18:60105 <http://sip:17367@46.39.229.18:60105>' to AOR '17367' with expiration of 90 seconds The client just registered (again) [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:60105 <http://sip:17367@46.39.229.18:60105> has been created We added a new contact [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:37910 <http://sip:17367@46.39.229.18:37910> has been deleted We deleted the old contact [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:60105 <http://sip:17367@46.39.229.18:60105> is now Reachable. RTT: 44.031 msec We qualified the contact successfully [2016-03-21 20:42:09] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:60105 <http://sip:17367@46.39.229.18:60105> is now Unreachable. RTT: 0.000 msec At the next qualify, we couldn't reach the contact This looks like a client that's going to sleep or a firewall that's timing out connections. Asterisk is only deleting the contact on the next successful register because it's replacing it. You need to figure out why the qualify is failing and why the client keeps registering. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Loss of devices registration (pjsip)
Good day. Asterisk 13.7.2, res_pjsip. There is a problem of loss of registration of several devices. This happens not on all devices, but problem devices a lot. Below is the log of registration of a contact of one device. Is suspect two things: 1. delete a contact after the contact is added. But, like, it's a feature of code that may already be fixed. 2. deleting a contact much earlier than the 90 seconds specified during the registration Would be grateful for any clues. Dmitriy Serov. expiration settings: [common-aor](!) type=aor qualify_frequency=60 default_expiration=120 maximum_expiration=600 minimum_expiration=90 log: [2016-03-21 20:39:58] VERBOSE[30251] res_pjsip_registrar.c: Added contact 'sip:17367@46.39.229.18:37910' to AOR '17367' with expiration of 90 seconds [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:37910 has been created [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:27143 has been deleted [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:37910 is now Reachable. RTT: 41.882 msec [2016-03-21 20:41:01] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:37910 is now Unreachable. RTT: 0.000 msec [2016-03-21 20:41:06] VERBOSE[3827] res_pjsip_registrar.c: Added contact 'sip:17367@46.39.229.18:60105' to AOR '17367' with expiration of 90 seconds [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:60105 has been created [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:37910 has been deleted [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:60105 is now Reachable. RTT: 44.031 msec [2016-03-21 20:42:09] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:60105 is now Unreachable. RTT: 0.000 msec [2016-03-21 20:42:14] VERBOSE[3827] res_pjsip_registrar.c: Added contact 'sip:17367@46.39.229.18:52836' to AOR '17367' with expiration of 90 seconds [2016-03-21 20:42:14] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:52836 has been created [2016-03-21 20:42:14] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:60105 has been deleted [2016-03-21 20:42:14] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367@46.39.229.18:52836 is now Reachable. RTT: 40.032 msec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unexpected termination of the call when pick up (res_pjsip)
11.02.2016 18:50, Joshua Colp пишет: Dmitriy Serov wrote: The call initiated from internal extension. I have made two test call: Successful call from device on res_pjsip via endpoint on chan_sip: http://pastebin.com/LWeDYstj Unsuccessful call from device on res_pjsip via endpoint on res_pjsip: http://pastebin.com/hepVb6Nu And ones again i don't see anything that would make asterisk send BYE. I would be grateful for any ideas. Kia ora, I have a feeling it may be an off-nominal SDP negotiation issue, causing two paths to get triggered which both send a BYE. I'd suggest filing an issue[1] with the traces you've provided. We can potentially use them to construct a sipp scenario that reproduces the issue. The configuration would also be needed. You can also try to narrow it down slightly by disabling the video codecs and seeing if that changes things. If it does then it's with video involved. [1] https://issues.asterisk.org/jira Joshua, Thanks! Disabling all codecs except alaw (I guess video codecs) makes call successful. https://issues.asterisk.org/jira/browse/ASTERISK-25772 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unexpected termination of the call when pick up (res_pjsip)
The call initiated from internal extension. I have made two test call: Successful call from device on res_pjsip via endpoint on chan_sip: http://pastebin.com/LWeDYstj Unsuccessful call from device on res_pjsip via endpoint on res_pjsip: http://pastebin.com/hepVb6Nu And ones again i don't see anything that would make asterisk send BYE. I would be grateful for any ideas. 11.02.2016 1:47, Trey Hilyard пишет: How are you initiating the call out to that server? Are you dialing from an internal phone or doing it from the CLI? It looks like it is from an internal extension, if I were guessing, but that side of the call isn't in your log. If it is from an internal extension, I think a SIP trace on that side would help. On Wed, Feb 10, 2016, 3:20 PM Dmitriy Serov <mailto:serov@gmail.com>> wrote: Please help find the cause of strange behavior res_pjsip. Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up! Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4d No diagnostic messages why asterisk suddenly decided to hangup i don't found :( There are suggestions or strong belief about the reasons of such behavior? Thanks. Dmitriy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unexpected termination of the call when pick up (res_pjsip)
Please help find the cause of strange behavior res_pjsip. Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up! Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4d No diagnostic messages why asterisk suddenly decided to hangup i don't found :( There are suggestions or strong belief about the reasons of such behavior? Thanks. Dmitriy. [2016-02-10 22:58:17] VERBOSE[25442] res_pjsip_logger.c: <--- Received SIP response ( bytes) from UDP:83.143.192.141:5060 ---> SIP/2.0 183 Call progress Via: SIP/2.0/UDP 85.142.148.80:5060;rport=5060;branch=z9hG4bKPj1be0328e-fb97-426c-93fe-cf12e32f501a Record-Route: Record-Route: From: "admin" ;tag=0ea59f7e-817c-48a1-8e44-6e896322609a To: ;tag=B955C4E4-606476-16E1127B Call-ID: 5ac4642d-007b-4e29-908a-1b06417148c7 CSeq: 3711 INVITE Contact: Supported: 100rel,timer,replaces,histinfo,precondition User-Agent: CommuniGatePro-callLeg/5.4.10 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Content-Type: application/sdp Content-Length: 376 v=0 o=CGPLeg606476 1366433634 683216818 IN IP4 83.143.192.141 s=SIP Call c=IN IP4 83.143.192.141 t=0 0 m=audio 60132 RTP/AVP 8 101 c=IN IP4 83.143.192.141 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcpping:F:1253794:125379478 m=video 60136 RTP/AVP 100 c=IN IP4 83.143.192.141 a=inactive a=rtcpping:F:1253795:125379578 [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Function session_inv_on_state_changed called on event TSX_STATE [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The state change pertains to the endpoint 'srv_d15140(PJSIP/srv_d15140-0255)' [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The inv session still has an invite_tsx (0x7ffdf039dc58) [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: There is no transaction involved in this state change [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The current inv state is EARLY [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Source of transaction state change is RX_MSG [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Received response [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Response is 183 Call progress [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Function session_inv_on_tsx_state_changed called on event TSX_STATE [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The state change pertains to the endpoint 'srv_d15140(PJSIP/srv_d15140-0255)' [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The inv session still has an invite_tsx (0x7ffdf039dc58) [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The UAC INVITE transaction involved in this state change is 0x7ffdf039dc58 [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The current transaction state is Proceeding [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The transaction state change event is RX_MSG [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The current inv state is EARLY [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Received response [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Response is 183 Call progress [2016-02-10 22:58:17] VERBOSE[7404][C-018e] app_dial.c: PJSIP/srv_d15140-0255 is making progress passing it to PJSIP/admin-0254 [2016-02-10 22:58:17] DEBUG[31024] res_pjsip_session.c: Applying negotiated SDP media stream 'audio' using audio SDP handler [2016-02-10 22:58:17] DEBUG[31024] netsock2.c: Splitting '109.60.222.253' into... [2016-02-10 22:58:17] DEBUG[31024] netsock2.c: ...host '109.60.222.253' and port ''. [2016-02-10 22:58:17] DEBUG[31024] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7ffddc198f58' [2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7ffdc23b7320 [2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7ffdc23b7320 [2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Don't have a default tx payload type 96 format for m type on 0x7ffdc23b7320 [2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Setting tx payload type 97 based on m type on 0x7ffdc23b7320 [2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Don't have a default tx payload type 2 format for m type on 0x7ffdc23b7320 [2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Setting tx payload type 18 based on m type on 0x7ffdc23b7320 [2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7ffdc23b7320 [2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Copying payload 0 (0x7ffddd7ac7a0) from 0x7ffdc23b7320 to 0x7ffddc199120 [2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Copying payload 2 (0x7ffddc10bd20) from 0x7ffdc23b7320 to 0x7ffddc199120 [2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Copying payload 8 (0x7ffdde1b0ea0) from 0x7ffdc23b7320
[asterisk-users] Android native SIP client and 183 (Session Progress) call Declined
Good day. I have a problem when using android native sip client. When dialplan used Progress (sending 183 Session Progress) after some seconds android native sip client declines a call (the logs are at the end of). No ealry media be heard. In same call using Ringing (180) is everything ok. In same call using Progress and other SIP clients (any) is everything ok. Early media exists. Is it possible to make native SIP client works correctly? Should I send 180 and 183 at the same time? In what sequence? Standards and many discussions on the Internet claim that should be sent just one of them. Thanks. Dmitriy Serov. [2016-01-31 14:44:56] VERBOSE[1950] res_pjsip_logger.c: <--- Transmitting SIP response (874 bytes) to UDP:109.60.222.xxx:49912 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 109.60.222.xxx:49912;rport=49912;received=109.60.222.xxx;branch=z9hG4bKf6c1211a627c45cf6e255cffc3e9c9203738 Call-ID: 46c41cb03be20add7f1b3e3c5423ba30@192.168.1.98 From: "16006" ;tag=432079590 To: ;tag=f1a0e09e-f94b-4aca-84c6-d3d9af678852 CSeq: 6847 INVITE Server: ruVoIP.net PBX Contact: Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER Content-Type: application/sdp Content-Length: 292 v=0 o=- 2541746601 2541746605 IN IP4 85.142.148.xxx s=ruVoIP.net PBX c=IN IP4 85.142.148.xxx t=0 0 m=audio 25616 RTP/AVP 8 0 3 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=ptime:20 a=maxptime:150 a=sendrecv [2016-01-31 14:45:07] VERBOSE[1950] res_pjsip_logger.c: <--- Transmitting SIP response (527 bytes) to UDP:109.60.222.xxx:49912 ---> SIP/2.0 603 Decline Via: SIP/2.0/UDP 109.60.222.xxx:49912;rport=49912;received=109.60.222.xxx;branch=z9hG4bKf6c1211a627c45cf6e255cffc3e9c9203738 Call-ID: 46c41cb03be20add7f1b3e3c5423ba30@192.168.1.98 From: "16006" ;tag=432079590 To: ;tag=f1a0e09e-f94b-4aca-84c6-d3d9af678852 CSeq: 6847 INVITE Server: ruVoIP.net PBX Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER Reason: Q.850;cause=0 Content-Length: 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_pjsip/pjsip_configuration.c: Unable to create ast_sip_contact_status for contact
Hello.
Asterisk 13.7 (branch 13), res_pjsip
Log file contains a lot of lines:
[2016-01-11 15:33:00] ERROR[2862] res_pjsip/pjsip_configuration.c:
Unable to create ast_sip_contact_status for contact
17378/sip:17...@87.255.225.:5060
[2016-01-11 15:33:00] VERBOSE[2703] res_pjsip_registrar.c: Added contact
'sip:17...@87.255.225.:5060' to AOR '17378' with expiration of 5 seconds
[2016-01-11 15:33:00] VERBOSE[2862] res_pjsip/pjsip_configuration.c:
Contact 17378/sip:17...@87.255.225.:5060 has been created
/res/res_pjsip/pjsip_configuration.c
/*! \brief Function called when a contact is deleted */
static void persistent_endpoint_contact_deleted_observer(const void *object)
{
const struct ast_sip_contact *contact = object;
struct ast_sip_contact_status *contact_status;
contact_status =
ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), CONTACT_STATUS,
ast_sorcery_object_get_id(contact));
if (!contact_status) {
ast_log(LOG_ERROR, "Unable to create
ast_sip_contact_status for contact %s/%s\n",
contact->aor, contact->uri);
return;
}
1. Where is copy/paste error in log msg I guess.
2. And whether the log in this case? Think before adding the contact it
should remove it and logging msg because not finds. Or not so simple?
Why so many ERROR line in log?
Thanks.
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Re: [asterisk-users] Using external RTP proxy for res_pjsip
Bryant I would like to have only RTP proxy. SIP signaling to leave the existing server without duplicating the list of thousands of endpoints The desired behavior is very similar to the work of the STUN server, but only for RTP stream and without the real existence of NAT on the server side. Understand that without the registration data to the proxy will fully open proxy. Then maybe it makes sense to use the existing open STUN server? How to force asterisk (res_pjsip) and client to use common STUN for RTP stream without having the real NAT? Dmitriy. 02.11.2015 17:39, Bryant Zimmerman пишет: Dmitrity What kind of volume are you running? You can use asterisk as a proxy if you set it up correctly. The choice would fall on the volume and the operational needs. To use an external proxy you would either need to register to the proxy or have a trusted IP to IP relationship. If your carrier allows for endpoint registration then you could attached your asterisk server directly, and would not need a proxy. If you have an IP for a proxy you could also do a NAT translation from that IP directly to the Asterisk server and negate the need for a proxy all together. As far as connecting to a proxy. You would need a pjsip endpoint either with a trusted IP or with a registration to your proxy server. As far as exactly what should go in your pjsip.conf that depends on your final implementation. You have not given enough detail of your network situation, and the reasons for a proxy to adequately advise you any deeper. Good luck. I hope some of the above is helpful to you. Bryant *From*: "Dmitriy Serov" *Sent*: Monday, November 2, 2015 9:10 AM *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" *Subject*: [asterisk-users] Using external RTP proxy for res_pjsip The asterisk server has a permanent IP address, but the provider cannot ensure stable quality traffic for RTP. There is a desire to use an external server, the address of which shall be specified in the SDP, through which flowing media. I use asterisk 13.6 and res_pjsip. Prompt, please: 1. what software can be used on an external RTP proxy? 2. What settings need to be done in pjsip.conf to use this external RTP proxy? Preferably specifies the external RTP proxy to specify a specific endpoint, not globally. If only globally valid, the suit and the decision. I would be grateful for any clues. Dmitriy Serov. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using external RTP proxy for res_pjsip
The asterisk server has a permanent IP address, but the provider cannot ensure stable quality traffic for RTP. There is a desire to use an external server, the address of which shall be specified in the SDP, through which flowing media. I use asterisk 13.6 and res_pjsip. Prompt, please: 1. what software can be used on an external RTP proxy? 2. What settings need to be done in pjsip.conf to use this external RTP proxy? Preferably specifies the external RTP proxy to specify a specific endpoint, not globally. If only globally valid, the suit and the decision. I would be grateful for any clues. Dmitriy Serov. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk QoS stats
Hello. Recently I installed voipmonitor and voipmonitor-gui trial version. After examining it, I was amazed at the abundance of useful information that can and should be obtained from the work of Asterisk. 1. The cost voipmonitor-gui too expensive and not justified in my case. For this reason, I consider the implementation of a poor analogue: - based on the data collected from the QoS values obtained from CHANNEL and stored in the asterisk CDR - using data from tables of the free voipmonitor - something else? Which way would you choose and why? Or maybe you have chosen and share your experience? 2. Analyzing the data, I noticed the frequent discrepancy in the data between voipmonitor and asterisk channel qos. One of them shows packet loss, the other does not. Indicators PDV (Packet Delay Variation) and RTT are generally difficult to compare. Is there a reasonable and understandable explanation for such discrepancies? 3. PDD (Post dial delay), MOS - these values would be great to have in Channel QoS Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI event PeerStatus: Address option with res_pjsip
Hello. Continue to move from chan_sip to res_pjsip. For the work of my algorithms is very important to know the IP address of all trunks and endpoints (phones). In the case of chan_sip, I used PeerStatus AMI event through which was received the fact of online/offline and IP Address of endpoint. And requesting AMI SIPshowpeer I received UserAgent of this endpoint. res_pjsip. In the case of "ENDPOINT CONFIGURED FOR USE WITH A SIP PHONE" I can use AMI event "SuccessfulAuth" to retrieve IP Address. How can I get the IP address in the case of trunks? I will be grateful if tell me how to get the User agent. Thanks Dmitriy Serov. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] does res_pjsip support ZRTP?
06.10.2015 1:22, Joshua Colp пишет: On 15-10-05 05:58 PM, Dmitriy Serov wrote: 05.10.2015 23:24, Joshua Colp пишет: On 15-10-05 05:22 PM, Dmitriy Serov wrote: Hello. Do I understand correctly that the current implementation res_pjsip does not support ZRTP? http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html ZRTP is not supported in Asterisk itself. Nothing has changed since 2013? P.S. I greatly regret that moved from chan_sip to res_pjsip. Previously used very much lacking, and much of the promise failed. Dmitriy Serov. Any particular examples? - opus support. Ok... I know the reason why it is not supported fully this codec. But the existing foreign solution works fine with chan_sip and does not work with res_pjsip works. - endpoint specific ACL - No support for SIP message without authorization. For this reason, the previously working functionality of sending and receiving SMS from gateway GOIP had to rewrite their internal Protocol. Can you clarify what you mean here? There's an anonymous endpoint identifier which can be used for anonymous inbound messages basically. Something like auth_message_requests: http://lists.digium.com/pipermail/asterisk-users/2015-September/287516.html (ugg formating) In short: - GOIP gate (successfully registered as endpoint) send SIP MESSAGE - asterisk send registration request - nothing. I now understand that the reason may be exactly the same described below. - found hardphones and software phones that don't accept "long nonce" and refuse to register when using res_pjsip Have you filed an issue with this and details about the hardphones+softphones? Welltech WP589. Beautifully registered using chan_sip and res_pjsip not logged in. Analyzing the exchange of SIP packets I found a single difference: the format of the "nonce" field. When using a longer nonce (pjsip) this phone simply does not respond to the request packet authorization (as do many hardware and software encountering something incomprehensible). The same behavior was on the built-in nokia 95 SIP client. - enable icesupport also leads to problems of registration and cannot be "common solution" icesupport is only applied to calls, what happens for registration? Sorry. Not registration, but INVITE. The client software encounters an unfamiliar SDP headers and simply not responding to SIP packets. The specifics of my service is that I don't know what SIP client is on the other side. What it supports and what not. To give to configure to a user - not the best idea, because often they do not understand what they onoff and why stops working. - issue tracker now contains multiple error messages that arise every day and reboot my server (which cannot be called a production) - And watchdog logs SegFaults and Hangs including other stacks that are not yet documented in the issue tracker. Have you filed any issues for these with information? We can't make PJSIP better if we don't know about the problems people are having like this. Some of not fixed: https://issues.asterisk.org/jira/browse/ASTERISK-25439 https://issues.asterisk.org/jira/browse/ASTERISK-25435 https://issues.asterisk.org/jira/browse/ASTERISK-25421 https://issues.asterisk.org/jira/browse/ASTERISK-25378 https://issues.asterisk.org/jira/browse/ASTERISK-25279 Dmitriy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] does res_pjsip support ZRTP?
06.10.2015 16:08, Matthew Jordan пишет: I know this is shocking to hear, but this is an open source project. That means anyone can fix something. Anyone can add something. Even you! You have all the power to affect your system. It also means that no one is under any obligation to do it for you. Surprising, right? I know, it's amazing to think that *YOU* have all the responsibility and power. We use PJSIP. We use it in a variety of settings. It works well for us. Does that mean it works well for you? I don't know. I'm not you. I don't have your use cases. Would I like it to work well for you? Of course! But if you don't participate by reporting issues, testing changes, and contributing code, there's not much I can do for you other than to note that the line is long, and feel free to stand in it until someone in the community gets around to what you'd like to have done. Matt I these words were repeatedly read and remember them well. That's why I haven't written any complaints to the developers. Where you saw them, what made again to write these words? I am a developer with more than two dozen years of experience. I have a hobby with a free service in which I don't owe anyone anything. And I understand your words. Wrote a lot of words, but erased everything. It was my subjective opinion that will not change anything, and therefore it is unnecessary. Now Why I wrote what I wrote. I feel the need to tell people my opinion on the difference in functionality between chan_sip and res_pjsip. They may be important for decision making. I this information was lacking in the past. Dmitriy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] does res_pjsip support ZRTP?
05.10.2015 23:24, Joshua Colp пишет: On 15-10-05 05:22 PM, Dmitriy Serov wrote: Hello. Do I understand correctly that the current implementation res_pjsip does not support ZRTP? http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html ZRTP is not supported in Asterisk itself. Nothing has changed since 2013? P.S. I greatly regret that moved from chan_sip to res_pjsip. Previously used very much lacking, and much of the promise failed. Dmitriy Serov. Any particular examples? - opus support. Ok... I know the reason why it is not supported fully this codec. But the existing foreign solution works fine with chan_sip and does not work with res_pjsip works. - endpoint specific ACL - No support for SIP message without authorization. For this reason, the previously working functionality of sending and receiving SMS from gateway GOIP had to rewrite their internal Protocol. - found hardphones and software phones that don't accept "long nonce" and refuse to register when using res_pjsip - enable icesupport also leads to problems of registration and cannot be "common solution" - issue tracker now contains multiple error messages that arise every day and reboot my server (which cannot be called a production) - And watchdog logs SegFaults and Hangs including other stacks that are not yet documented in the issue tracker. Be sure to have forgotten something, because it is not documented all meet and unsolved problems,workarounds. The transition to PJSIP was chosen as mainstream and full support for WebRTC. As a result, instead of developing a service I a few months I'm returning opportunities to which users are accustomed and expect to see. Having the knowledge and the overall picture a few months ago I would not have taken such a decision. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] does res_pjsip support ZRTP?
Hello. Do I understand correctly that the current implementation res_pjsip does not support ZRTP? http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html Nothing has changed since 2013? P.S. I greatly regret that moved from chan_sip to res_pjsip. Previously used very much lacking, and much of the promise failed. Dmitriy Serov. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_pjsip. Turn off the authorization request for an incoming MESSAGE
Hello. Continuea months-longstrugglethat is associatedwith the transfer from chan_sip to res_pjsip. Where are many gates (GSM gate) that do not supportauthentication whensendingMESSAGE. For example, 4goip when relay incoming SMS. Using chan_sip it was not a problem. Using res_pjsip is the problem :( Is any way to turn off the authorization request for an incoming MESSAGE using res_pjsip? Or any workaround? [2015-09-07 06:01:14] DEBUG[12947] pjsip: sip_endpoint.c Processing incoming message: Request msg MESSAGE/cseq=542 (rdata0x7f88642fdc28) [2015-09-07 06:01:14] VERBOSE[12947] res_pjsip_logger.c: <--- Received SIP request (447 bytes) from UDP:109.165.111.xx:5807 ---> MESSAGE sip:sm...@85.142.148.xx SIP/2.0 Via: SIP/2.0/UDP 109.165.111.xx:5807;branch=z9hG4bK837973400 Route: From: ;tag=284759743 To: Call-ID: 76603@192.168.1.100 CSeq: 542 MESSAGE Contact: Max-Forwards: 30 User-Agent: dble Content-Type: text/plain Content-Length: 35 111 Ваш баланс 68,08 rub. [2015-09-07 06:01:14] DEBUG[23059] pjsip: sip_endpoint.c Distributing rdata to modules: Request msg MESSAGE/cseq=542 (rdata0x7f88640a9288) [2015-09-07 06:01:14] DEBUG[23059] res_pjsip_endpoint_identifier_ip.c: No identify sections to match against [2015-09-07 06:01:14] DEBUG[23059] res_pjsip_endpoint_identifier_user.c: Retrieved endpoint srv_9185880046 [2015-09-07 06:01:14] DEBUG[23059] pjsip: endpoint .Response msg 401/MESSAGE/cseq=542 (tdta0x7f88717063b0) created [2015-09-07 06:01:14] VERBOSE[23059] res_pjsip_logger.c: <--- Transmitting SIP response (479 bytes) to UDP:109.165.111.xx:5807 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 109.165.111.xx:5807;rport=5807;received=109.165.111.xx;branch=z9hG4bK837973400 Call-ID: 76603@192.168.1.100 From: ;tag=284759743 To: ;tag=z9hG4bK837973400 CSeq: 542 MESSAGE WWW-Authenticate: Digest realm="ruvoip.net",nonce="1441594874/5741fb37496404a4aa5cf0e53a129867",opaque="7441b8c64eddc67a",algorithm=md5,qop="auth" Server: ruVoIP.net PBX Content-Length: 0 Dmitriy Serov. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
Hi, Andrew.
You are trying to solve two tasks: definition through what line the call
came and a beautiful display of this information.
1. definition through what line the call came. If the username and
password for inbound and outbound registration the same, then try the
following:
a) delete "register" lines.
b) add option "callbackextension=Company1" to Company1 friend section..
And in others with their names too.
or you can change "/s" to "/Company1" in register line.
2. beautiful display of this information
a) add option "setvar=fromCompany=Company1" to Company1 friend section..
b) In dialplan add
Set(CALLERID(name)=${fromCompany} ${CALLERID(name)})
Maybe this will help?
Dmitiy.
08.04.2015 2:48, Andrew Galdes пишет:
Hi Dmitriy and others and thanks for your help so far.
The option "match_auth_username=yes" seems to have had no effect. From
my reading, this option will try to match the username of the incoming
SIP account to a section heading. If that is how it must work then i
can see a big problem. I'm trying to present the receptionist with a
nice display of which line the call came in on. For example, the
receptionist answers calls for 8 different companies and would like
the phone to display the company name that she should announce to the
caller.
Here is a more complete output of an incoming call. I've changed the
SIP numbers to "Company1', etc, to hide the numbers.
Connected to Asterisk 10.12.4 currently running on asterisk (pid =
32267)
Verbosity is at least 12
asterisk*CLI>
asterisk*CLI>
asterisk*CLI>
== Using SIP RTP CoS mark 5
-- Executing [s@incoming:1] *Set*("*SIP/Company1-0797*",
"*thedid=""NodePhone"mailto:sip%3acompa...@sip.internode.on.net>>"*") in new stack
-- Executing [s@incoming:2]
*Set*("*SIP/**Company1**-0797*",
"*pseudodid="NodePhone"http://sip.internode.on.net>>*") in new stack
-- Executing [s@incoming:3]
*Set*("*SIP/**Company1**-0797*",
"*pseudodid="NodePhone" exit
And here is the "sip.conf":
[general]
match_auth_username=yes
register=081...:...@sip.internode.on.net/s
<http://081...:...@sip.internode.on.net/s>
register=082...:...@sip.internode.on.net/s
<http://082...:...@sip.internode.on.net/s>
register=083...:...@sip.internode.on.net:/s
register=084...:...@sip.internode.on.net:/s
register=085...:...@sip.internode.on.net/s
<http://085...:...@sip.internode.on.net/s>
register=086...:...@sip.internode.on.net/s
<http://086...:...@sip.internode.on.net/s>
register=087...:...@sip.internode.on.net/s
<http://087...:...@sip.internode.on.net/s>
register=088...:...@sip.internode.on.net/s
<http://088...:...@sip.internode.on.net/s>
[Company1]
username=081...
fromuser=081...
secret=...
canreinvite=no
qualify=yes
context=incoming
type=friend
insecure=invite,port
fromdomain=sip.internode.on.net <http://sip.internode.on.net>
host=sip.internode.on.net <http://sip.internode.on.net>
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
allow=g729
bindport=5060
bindaddr=0.0.0.0
nat=yes
registertimeout=5
allowoverlap=no
srvlookup=no
ubscribecontext=from-sip
callcounter=yes
[Company2]
...
[Company3]
...
[Company4]
...
And here is some of the "extensions.conf" file:
[incoming]
; Get the DID number from the TO header.
exten => s,1,Set(thedid="${SIP_HEADER(TO)}")
exten => s,2,Set(pseudodid=${SIP_HEADER(To)})
exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
exten => s,4,Set(pseudodid=${CUT(pseudodid,:,2)})
; Direct the DID accordingly.
exten => s,5,GotoIf($["${pseudodid}" = "081"]?internal,33,1:6)
exten => s,6,GotoIf($["${pseudodid}" = "082"]?internal,88,1:7)
exten => s,7,GotoIf($["${pseudodid}" = "083"]?internal,36,1:8)
exten => s,8,GotoIf($["${pseudodid}" = "084"]?internal,36,1:9)
exten => s,9,GotoIf($["${pseudodid}" = "085"]?internal,36,1:10)
exten => s,10,GotoIf($["${pseudodid}" = "086"]?internal,89,1:11)
exten => s,11,GotoIf($["${pseudodid}" = "087"]?internal,36,1:12)
exten => s,12,GotoIf($["${pseudodid}" = "088"]?internal,13,1:13)
-Andrew Galdes
On Thu, Apr 2, 2015 at 3:46 PM, Dmitriy Serov <mailto:serov@gmail.com>> wrote:
This is one of the chronic problems. Try this option in sip.conf:
match_auth_username=yes
Carefully read the description,
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
This is one of the chronic problems. Try this option in sip.conf:
match_auth_username=yes
Carefully read the description, it is better to test in "after hours".
02.04.2015 2:50, Andrew Galdes пишет:
Hello all,
I have an Asterisk server (Asterisk 10.12.4) with multiple sip
accounts with the same service provides. We have 8 phone numbers in
total.
Incoming calls from the public are all correctly directed to
appropriate office handsets. However, the display on the reception
phone (the only one i care about) is always showing the same
"SIP/Account1_0843214321" rather than the account representing the
number dialed.
For-instance, if Sam on her mobile calls "*08*", Asterisk will
show a log entry like the following:
-- Executing [s@incoming:1] Set("SIP/*Account1_08*",
"thedid=""NodePhone">"") in new stack
But "Account1_*08*" (as the name suggests) has a phone number
of "*08*" and not "*08*".
So Sam's call will come through and be routed to the correct handset
as the business needs, but it seems that all incoming calls are being
labeled as though coming in on a different account. The effective
problem is that the calledID is now wrong.
I'm after some general advice on how to handle the problem.
Ta,
-Andrew
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[asterisk-users] Is there a way to escape text passwords in pjsip.conf?
Hello. I have plain text password for endpoint with outbound registration (someone else's server). My aim is to write it in pjsip.conf. md5 means that I know realm. I do not always know it. Is where any way? Dmitriy Serov. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 13. Writing call quality parameters to CDR. How?
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[asterisk-users] pjsip: outofcall_message_context
Hello. Is there an analog option "outofcall_message_context" for pjsip? or: how to determine that the "call" is an outbound text message? Dmitriy Serov. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP some AMI events is absent?
Hello. Asterisk 13.2, PJSIP. Problem: I do not get any AMI events when changing the status of the contact. When using chan_sip I got "peerstatus" event. When using res_pjsip and devices (endpoint configuration) I got "peerstatus" event. When using res_pjsip.so and OUTBOUND REGISTRATION WITH OUTBOUND AUTHENTICATION i got "registry" event. When using "ENDPOINT CONFIGURED AS A TRUNK, OUTBOUND AUTHENTICATION" and status on contact changed I do not get any AMI event. I missed something? Tell me how to determine the change in the status of the contact (or endpoint/trunk) through AMI? Simple config: [srv_dev] type=auth auth_type=userpass username=login password=secret [srv_dev] type=aor contact=sip:sip.example.com:5060 qualify_frequency=5 default_expiration=10 max_contacts=1 remove_existing=yes [srv_dev] type=endpoint from_domain=example.com aors=srv_dev outbound_auth=srv_dev rewrite_contact=yes allow=!all,alaw Dmitriy Serov -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_pjsip endpoint config object's 'identify_by' option needs new value "uri".
07.03.2015 1:21, Kevin Harwell пишет: On Fri, Mar 6, 2015 at 3:46 PM, Dmitriy Serov <mailto:serov@gmail.com>> wrote: 07.03.2015 0:24, Kevin Harwell пишет: On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov mailto:serov@gmail.com>> wrote: Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk] type=registration transport=udp-transport outbound_auth=siptrunk server_uri=sip:sip.example.com <http://sip.example.com> client_uri=sip:1234567...@sip.example.com <mailto:client_uri=sip:1234567...@sip.example.com> retry_interval=60 contact_user=siptrunk-in [siptrunk-in] type=endpoint transport=udp-transport context=from-trunk disallow=all allow=ulaw outbound_auth=siptrunk aors=siptrunk identify_by=uri Registration section has option "contact_user". Incoming call from this registration will be INVITE sip:siptrunk-in@ I offer to change res_pjsip_endpoint_identifier_user to realize endpoint identification by sip uri. I think it will be usefull. P.S. i hope issues will be rejected: https://issues.asterisk.org/jira/browse/ASTERISK-22306 and SWP-6069 Dmitriy Serov - I believe what you are looking for is already available. See the "identify" type (type=identify) section that is in the pjsip.conf file and the "identify" option for endpoints. These allow you to identify and endpoint by IP address. For more information see the pjsip.conf.sample file. Also take a look at configuring Asterisk for res_pjsip [1] specifically the part about configuring endpoint identification by IP address [2]. If you run into problems more information can also be found in the res_pjsip troubleshooting guide [3], specifically the section on "identify by IP address" [1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip_endpoint_identifier_ip [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide Hope that helps, Thank you for answer. But... ones again: I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? Simple Example. I have two trunks with their own credentials (and did) to the same sip server: - for home - for bussiness [home-example.com-endpoint] [bussiness-example.com-endpoint] [home-example.com-registration] contact_user=home-example.com-endpoint [bussiness-example.com-registration] contact_user=bussiness-example.com-endpoint ;and ok... i wrote identify by IP section [example.com-identify] type=identify match=example.com <http://example.com> endpoint= ??? It is very! important for me to know what trunk passes through the incoming call: home or bussiness. 1. Identify by IP. Do you have answer? 2. Identify by username. What? I can't make endpoints to all of my contacts. Ok. I can use contact_user in registraction and route incoming call by INVITE uri. Can i? Dmitriy Serov I don't think I fully understand the scenario, but if you have different named endpoints originating from the same address (or even different addresses) then these can be identified by the username portion of the sip uri. However, if you have endpoints for instance with the same name, but different addresses then these can be distinguished by using the identity type. Your scenario looks like the first option. The endpoint names are different, but the address is the same, so identification based on the username should be sufficient. However, if you have a mix of both types on your system, for instance multiple endpoints on two different systems (IP addresses) with the same names, then I am unsure how you would select the correct endpoint even while attempting to "identify_by" uri. -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:http://digium.com &http://asterisk.org I have different named endpoints originating from the same address. >>then these can be identified by the username portion of the sip uri ok. A little more examples. case 1. Device! we have: e
Re: [asterisk-users] res_pjsip endpoint config object's 'identify_by' option needs new value "uri".
07.03.2015 0:24, Kevin Harwell пишет: On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <mailto:serov@gmail.com>> wrote: Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk] type=registration transport=udp-transport outbound_auth=siptrunk server_uri=sip:sip.example.com <http://sip.example.com> client_uri=sip:1234567...@sip.example.com <mailto:client_uri=sip:1234567...@sip.example.com> retry_interval=60 contact_user=siptrunk-in [siptrunk-in] type=endpoint transport=udp-transport context=from-trunk disallow=all allow=ulaw outbound_auth=siptrunk aors=siptrunk identify_by=uri Registration section has option "contact_user". Incoming call from this registration will be INVITE sip:siptrunk-in@ I offer to change res_pjsip_endpoint_identifier_user to realize endpoint identification by sip uri. I think it will be usefull. P.S. i hope issues will be rejected: https://issues.asterisk.org/jira/browse/ASTERISK-22306 and SWP-6069 Dmitriy Serov - I believe what you are looking for is already available. See the "identify" type (type=identify) section that is in the pjsip.conf file and the "identify" option for endpoints. These allow you to identify and endpoint by IP address. For more information see the pjsip.conf.sample file. Also take a look at configuring Asterisk for res_pjsip [1] specifically the part about configuring endpoint identification by IP address [2]. If you run into problems more information can also be found in the res_pjsip troubleshooting guide [3], specifically the section on "identify by IP address" [1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip_endpoint_identifier_ip [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide Hope that helps, -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:http://digium.com &http://asterisk.org Thank you for answer. But... ones again: I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? Simple Example. I have two trunks with their own credentials (and did) to the same sip server: - for home - for bussiness [home-example.com-endpoint] [bussiness-example.com-endpoint] [home-example.com-registration] contact_user=home-example.com-endpoint [bussiness-example.com-registration] contact_user=bussiness-example.com-endpoint ;and ok... i wrote identify by IP section [example.com-identify] type=identify match=example.com endpoint= ??? It is very! important for me to know what trunk passes through the incoming call: home or bussiness. 1. Identify by IP. Do you have answer? 2. Identify by username. What? I can't make endpoints to all of my contacts. Ok. I can use contact_user in registraction and route incoming call by INVITE uri. Can i? Dmitriy Serov -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_pjsip ACL relation to endpoint
Hello. I continue to transfer chan_sip to pjsip. Friend in chan_sip can has options: deny=0.0.0.0/0.0.0.0 permit=192.168.0.1 pjsip offer to use "global" ACL without relation to any andpoint. My task is restriction via IP to registering in certain endpoint. Different rules to different endpoints. It will be better ACL has optional link to Endpoint. Or you can offer other solution? Dmitriy Serov -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_pjsip endpoint config object's 'identify_by' option needs new value "uri".
Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk] type=registration transport=udp-transport outbound_auth=siptrunk server_uri=sip:sip.example.com client_uri=sip:1234567...@sip.example.com retry_interval=60 contact_user=siptrunk-in [siptrunk-in] type=endpoint transport=udp-transport context=from-trunk disallow=all allow=ulaw outbound_auth=siptrunk aors=siptrunk identify_by=uri Registration section has option "contact_user". Incoming call from this registration will be INVITE sip:siptrunk-in@ I offer to change res_pjsip_endpoint_identifier_user to realize endpoint identification by sip uri. I think it will be usefull. P.S. i hope issues will be rejected: https://issues.asterisk.org/jira/browse/ASTERISK-22306 and SWP-6069 Dmitriy Serov -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP: Failed to create outgoing session to endpoint
Sorry, i found the source of problem. https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels dialing via pjsip have to change dialplan syntax :( May be it will be usefull sombody else. 04.03.2015 21:54, Dmitriy Serov пишет: Hello. I am using asterisk and chan_sip a lot of years. And newbie in chan_pjsip. Now i am transfering all from chan_sip to chan_pjsip. And have a lot of questions. First of... system: Asterisk 13.2 on slackware 14.1 Errors on outgoing call: [2015-03-03 00:18:58] ERROR[6794]: chan_pjsip.c:1778 request: Failed to create outgoing session to endpoint 'srv_d228' [2015-03-03 00:18:58] WARNING[24528][C-1bcc]: app_dial.c:2431 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) What settings has mistake? What logic to choose outgoing transport? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP: Failed to create outgoing session to endpoint
Hello. I am using asterisk and chan_sip a lot of years. And newbie in chan_pjsip. Now i am transfering all from chan_sip to chan_pjsip. And have a lot of questions. First of... system: Asterisk 13.2 on slackware 14.1 Errors on outgoing call: [2015-03-03 00:18:58] ERROR[6794]: chan_pjsip.c:1778 request: Failed to create outgoing session to endpoint 'srv_d228' [2015-03-03 00:18:58] WARNING[24528][C-1bcc]: app_dial.c:2431 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) What settings has mistake? What logic to choose outgoing transport? [transport-udp] type=transport bind=0.0.0.0:5070 protocol=udp [srv_d228] type=endpoint language=ru rtp_symmetric=yes force_rport=yes disable_direct_media_on_nat=yes rewrite_contact=yes ice_support=yes disallow=all allow=alaw context=ext-fromservers from_domain=sipnet.ru from_user=talk37.ru aors=srv_d228 auth=srv_d228 set_var=fromDeviceId=228 set_var=fromUserId=2 outbound_auth=srv_d228 ;outbound_proxy=sip:sipnet.ru:5060 transport=transport-udp [srv_d228] type=aor qualify_frequency=30 contact=sip:sipnet.ru:5060 ;outbound_proxy=sip:sipnet.ru:5060 max_contacts=10 remove_existing=yes [srv_d228] type=auth auth_type=userpass username=talk37.ru password=secret [srv_d228] type=registration transport=transport-udp outbound_auth=srv_d228 server_uri=sip:sipnet.ru client_uri=sip:talk37...@sipnet.ru retry_interval=60 ;auth_rejection_permanent=no contact_user=srv_d228 pjsip show registrations = srv_d228/sip:sipnet.ru srv_d228 Registered pjsip show endpoints Endpoint: srv_d228 Not in use0 of inf OutAuth: srv_d228/talk37.ru InAuth: srv_d228/talk37.ru Aor: srv_d228 10 Contact: srv_d228/sip:sipnet.ru:5060 Avail 9.858 Thanks! Dmitriy Serov -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Temporarily features (transfer) off during Read
Hello all.
Dialing with tT options and function Read (to prompt number) has a
trouble for me.
Can I temporarily features off during Read?
features.conf:
[featuremap]
blindxfer => ## ; Blind transfer (default is #)
atxfer => **; Attended transfer
I try:
exten => s,n,Set(LOCAL(tmp_atxfer)=${FEATUREMAP(atxfer)})
exten => s,n,Set(LOCAL(tmp_blindxfer)=${FEATUREMAP(blindxfer)})
exten => s,n,Set(FEATUREMAP(atxfer)=)
exten => s,n,Set(FEATUREMAP(blindxfer)=)
exten => s,n,Read(tmp_ext,${ARG1},${ARG2},${ARG3},${ARG4},${ARG5})
exten => s,n,Set(FEATUREMAP(atxfer)=${tmp_atxfer})
exten => s,n,Set(FEATUREMAP(blindxfer)=${tmp_blindxfer})
No success. FeatureMap is usefull only BEFORE Dial. It is no matter
during Dial :(
Remark: It's difficult to remove option (tT) in Dial.
Got any ideas?
Thanks.
Dmitriy.
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Re: [asterisk-users] Feature request: Need to INVITE to peer with other domain without peer domain addition
01.04.2013 23:52, Paul Belanger пишет: On 13-04-01 03:16 PM, Dmitriy Serov wrote: 31.03.2013 23:15, Barry Flanagan ?: On 31 March 2013 18:11, Dmitriy Serov mailto:serov@gmail.com>> wrote: Hi, asterisk admin and users. I need to SIP INVITE uri with domain via peer. And uri domain differ then peer domain. dialplan: exten => s,n,Dial(SIP/peer1/num...@domain2.com <mailto:num...@domain2.com>,60,r) [peer1] type=friend host=domain1.com <http://domain1.com> fromdomain=domain1.com <http://domain1.com> As a result in SIP packet uri: num...@domain2.com@domain1.com <http://domain1.com> I need: num...@domain2.com <mailto:num...@domain2.com> I can't use "SIP uri dial", i need authorization (peer1) I think asterisk can't do that. Is where work around? Would it work if you created a sip peer [domain2.com <http://domain2.com>] and set outboundproxy=domain1.com <http://domain1.com> then sent the call to SIP/num...@domain2.com <mailto:num...@domain2.com> ? -Barry does not matter. [skype.ippi.com](srv-options-common) type=friend secret=xxx host=ippi.fr fromdomain=ippi.fr outboundproxy=ippi.fr exten => 22,n,Dial(SIP/lo...@skype.ippi.com,60,rS(1200)) INVITE sip:lo...@ippi.fr SIP/2.0 Via: SIP/2.0/UDP 109.60.163.xx:5060;branch=z9hG4bK60e845b5;rport Max-Forwards: 70 From: "demon" ;tag=as518b59df To: and exten => 22,n,Dial(SIP/skype.ippi.com/lo...@skype.ippi.com,60,rS(1200)) do: INVITE sip:lo...@skype.ippi.com@ippi.fr SIP/2.0 I studied the source code and found no ways to implement it :( Dmitriy. How about: exten =>22,n,Dial(SIP/skype.ippi.com!lo...@skype.ippi.com,60,rS(1200)) Very strange syntax. I have never seen it. As a result: INVITE sip:usern...@ippi.fr SIP/2.0 Via: SIP/2.0/UDP 109.60.163.76:5060;branch=z9hG4bK1ba0386c;rport Max-Forwards: 70 From: "demon" ;tag=as4c2fa5f4 To: Field "To" is correct! But INVITE uri is incorrect. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feature request: Need to INVITE to peer with other domain without peer domain addition
31.03.2013 23:15, Barry Flanagan ?: On 31 March 2013 18:11, Dmitriy Serov <mailto:serov@gmail.com>> wrote: Hi, asterisk admin and users. I need to SIP INVITE uri with domain via peer. And uri domain differ then peer domain. dialplan: exten => s,n,Dial(SIP/peer1/num...@domain2.com <mailto:num...@domain2.com>,60,r) [peer1] type=friend host=domain1.com <http://domain1.com> fromdomain=domain1.com <http://domain1.com> As a result in SIP packet uri: num...@domain2.com@domain1.com <http://domain1.com> I need: num...@domain2.com <mailto:num...@domain2.com> I can't use "SIP uri dial", i need authorization (peer1) I think asterisk can't do that. Is where work around? Would it work if you created a sip peer [domain2.com <http://domain2.com>] and set outboundproxy=domain1.com <http://domain1.com> then sent the call to SIP/num...@domain2.com <mailto:num...@domain2.com> ? -Barry does not matter. [skype.ippi.com](srv-options-common) type=friend secret=xxx host=ippi.fr fromdomain=ippi.fr outboundproxy=ippi.fr exten => 22,n,Dial(SIP/lo...@skype.ippi.com,60,rS(1200)) INVITE sip:lo...@ippi.fr SIP/2.0 Via: SIP/2.0/UDP 109.60.163.xx:5060;branch=z9hG4bK60e845b5;rport Max-Forwards: 70 From: "demon" ;tag=as518b59df To: and exten => 22,n,Dial(SIP/skype.ippi.com/lo...@skype.ippi.com,60,rS(1200)) do: INVITE sip:lo...@skype.ippi.com@ippi.fr SIP/2.0 I studied the source code and found no ways to implement it :( Dmitriy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Feature request: Need to INVITE to peer with other domain without peer domain addition
Hi, asterisk admin and users. I need to SIP INVITE uri with domain via peer. And uri domain differ then peer domain. dialplan: exten => s,n,Dial(SIP/peer1/num...@domain2.com,60,r) [peer1] type=friend host=domain1.com fromdomain=domain1.com As a result in SIP packet uri: num...@domain2.com@domain1.com I need: num...@domain2.com I can't use "SIP uri dial", i need authorization (peer1) I think asterisk can't do that. Is where work around? Dmitriy Serov. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
