search for: callbackextension

Hello everybody, I am having a problem with realtime SIP peers. On Asterisk 1.8, I had SIP peers for external SIP providers configured in database and additional register lines in sip.conf so they would register. Now I upgraded to Asterisk 11.3.0, partly because of the promised callbackextension feature for realtime peers (https://reviewboard.asterisk.org/r/1717/). Removed the 'register' lines from sip.conf. My peers register correctly when Asterisk is started or if I do 'module unload chan_sip.so; module load chan_sip.so', but if I do 'sip reload', they stay in ...

Hi all, It looks like people commonly use this kind of dialplan when receiving faxes on Asterisk, with a jump to extension fax during the Wait() if a fax tone is detected: [start-here] exten => _X.,1,Answer() exten => _X.,n,Wait(n) exten => _X.,n,...do stuff... exten => _X.,n,Hangup() exten => fax,1,Goto(fax-rx,receive,1) [fax-rx] exten => receive,1,... exten =>

...e=de videosupport=yes callevents=yes caninvite=yes qualify=yes nat=force_rport,comedia faxdetect=yes t38pt_udptl=yes ... [abcde] type=peer insecure=invite defaultuser=12345678912 fromuser=12345678912 fromdomain=abcde.ab secret=guess-what host=abcde.ab qualify=yes context=from-sip dtmfmode=rfc2833 callbackextension=12345678912 but all i can see if i try to send a testfax is == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 -- Executing [12345678912 at from-sip:1] Answer("SIP/abcde-00000016", "") in new stack > 0x7fd11404cd00 -- Probation passed - setting RTP...

...11.8.1 and Kamailio 4.2.0-dev4 on CentOS 6.5, all are on the same rental virtual server. Clients are in my home network behind nat. In MySQL I have database asterisk with table sippeers, where I have clients added like this: INSERT INTO sippeers (name,defaultuser,host,sippasswd,fromuser,fromdomain,callbackextension,type) VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com ','660','friend'); In this message there are some outputs and a sip trace of a register: https://www.mail-archive.com/sr-users at lists.sip-router.org/msg18558.htm...

...172.16.1.2 5061 OK (1 ms) srv-pbx-in 172.16.1.7 5060 OK (1 ms) After `sip reload`: [Feb 16 14:30:20] DEBUG[1468]: res_config_mysql.c:497 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sipusers WHERE name LIKE '%' AND callbackextension LIKE '%' ORDER BY name [Feb 16 14:30:20] DEBUG[1468]: config.c:1650 config_text_file_load: Parsing /etc/asterisk/sip_notify.conf == Parsing '/etc/asterisk/sip_notify.conf': Found [Feb 16 14:30:20] DEBUG[1468]: chan_sip.c:32383 reload_config: SIP reload_config done...Runtime= 0...

...ms when a client with a From of 101864 attempts to make a call: SELECT * FROM ast_sipusers WHERE name = '101864' AND host = 'dynamic' SELECT * FROM ast_sipusers WHERE name = '101864' SELECT * FROM ast_sipusers WHERE host = '10.5.76.67' AND port = '5060' AND callbackextension = '014373500' OK, the above are fine. Asterisk looks for a user, and a callback exten. SELECT * FROM ast_sipusers WHERE host = '10.5.76.67' AND port = '5060' Now Asterisk looks for a peer. Still OK. SELECT * FROM ast_sipregs WHERE ipaddr = '10.5.76.67' AND port =...

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I

...s: 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 ad...

...s,redundancy,maxdatagram=400 directrtpsetup=yes disallow=all allow=ulaw allow=alaw and the corresponding Peer [sipcall.ch] type=peer insecure=invite defaultuser=123456789 fromuser=123456789 fromdomain=voipdomain.com secret=gueswhat host=voipdomain.com qualify=yes context=from-sip dtmfmode=rfc2833 callbackextension=123456789 the Dialplan [inbound] exten => _X.,1,Answer() exten => _X.,n,Set(DB(lastcaller/number)=${CALLERID(num)}) exten => _X.,n,GotoIf(${BLACKLIST()}?black,1) exten => _X.,n,Wait(2) exten => _X.,n,Dial(SIP/200&SIP/201,60,tToxX) exten => _X.,n,Goto(ausser-zeit,_X.,3) exten...

...nd verify asterisk works as expected; post your results to the existing mantis issue. Once each issue has been properly tested, we can continue triaging and step closer to merging the code. Let me know if you have questions / comments. --- [patch] Calls are not matched to correct peer when using callbackextension parameter https://issues.asterisk.org/view.php?id=14340 [patch] Strange nasty sound (Because Asterisk tryes to handle new voicemail, but there is no voicemails, voicemail isn't used) https://issues.asterisk.org/view.php?id=15999 [patch] default say.conf for new number method doesnt handle all...

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.

...sverify: no dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass sippasswd: a84a4ddcda13d13c9573d5294047b6a2 rpid: NULL domain: testers.com sippasswd2: 5c4671ae1043e6116118fed39bee091a callbackextension: NULL insecure: NULL cheers, Olli -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141205/ae825dbf/attachment.html>