search for: dialedpeername

Displaying 12 results from an estimated 12 matches for "dialedpeername".

2016 Sep 19
2
Ast 13.11.2 : bridgepeer variable empty ?
Hello I can confirm that the variable DIALEDPEERNAME contains the information that I would expect in the variable BRIDGEPEER. But I read nowhere that DIALEDPEERNAME has replaced BRIDGEPEER as of Asterisk version 13 ?! So if this is not the intention, then yes this is probably a bug and should be reported. Kind regards. Jonas. On 18-09-16...
2016 Sep 17
2
Ast 13.11.2 : bridgepeer variable empty ?
Hello a call goes out and is answered : [Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b is making progress passing it to SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b answered SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel SIP/myprovider-0000010b joined
2023 May 05
0
Calls running forever / CDRs inaccurate
...xxxxxxxx Priority: 26 Call Group: 0 Pickup Group: 0 Application: Dial Data: SIP/+49xxxxxxxx at provider Call Identifer: [C-000085c3] Variables: PROGRESSTIME_MS= PROGRESSTIME= RINGTIME_MS= RINGTIME= DIALEDTIME_MS= DIALEDTIME= ANSWEREDTIME_MS= ANSWEREDTIME= DIALEDPEERNAME= DIALEDPEERNUMBER= DIALSTATUS= SIPADDHEADER02=X-Something: something AUTO_MONITOR=wav,/var/spool/asterisk/monitor/20230505110016-customer-DE-EXTEN-49xxxxxxx-CLINUM-49xxxxxxxxx-CLINAME--PAICLEAN--CLICLEAN-49xxxxxxxxx-OCLINUM--OCLINAME-,mX MONITOR_EXEC=/var/lib/asterisk/2wav2mp3.sh CALLFILENAME=20230...
2014 May 07
1
asterisk12.2.0 PJSIP2.2.0 codec translation problem
...5f-896f-cea37599b148 -- PBX -- Context: outgoing-kamailio Extension: 7000 Priority: 1 Call Group: 0 Pickup Group: 0 Application: Dial Data: PJSIP/7000 Call Identifer: [C-00000000] Variables: BRIDGEPEER=PJSIP/7000-00000001 DIALEDPEERNUMBER=7000 DIALEDPEERNAME=PJSIP/7000-00000001 DIALSTATUS=ANSWER DIALEDTIME= ANSWEREDTIME= CDR Variables: level 1: calledsubaddr= level 1: callingsubaddr= level 1: dnid= level 1: clid="7001" level 1: src=7001 level 1: dst=7000 level 1: dcontext=outgoin level 1: channel=PJSIP/7 level 1: dstchannel=PJSIP/7 level 1:...
2012 Aug 01
2
Problem with callfile and CDR
...B1C0-0.0' with that of 'Local/21411615 at test_outgoing-cb92;2' [2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable DIALEDTIME. [2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable ANSWEREDTIME. [2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable DIALEDPEERNAME. [2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable DIALEDPEERNUMBER. [2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable DIALSTATUS. [2012-08-01 14:29:44] VERBOSE[6679] app_dial.c: -- Called khomp/gpstn/21411615 [2012-08-01 14:29:52] VERBOSE[6679] app_dial.c: -...
2010 Apr 06
2
polarity reverse
Hi, I have a problem with polarity reverse this my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1
2005 Jul 11
1
ASterisk@home + Broadvoice = Almost working installation...
...}) exten => s,5,Noop(CALLERIDNAME=${CALLERIDNAME}) exten => s,6,Noop(CALLERIDNUM=${CALLERIDNUM}) exten => s,7,Noop(CALLINGPRES=${CALLINGPRES}) exten => s,8,Noop(CHANNEL=${CHANNEL}) exten => s,9,Noop(CONTEXT=${CONTEXT}) exten => s,10,Noop(DATETIME=${DATETIME}) exten => s,11,Noop(DIALEDPEERNAME=${DIALEDPEERNAME}) exten => s,12,Noop(DIALEDPEERNUMBER=${DIALEDPEERNUMBER}) exten => s,13,Noop(DIALEDTIME=${DIALEDTIME}) exten => s,14,Noop(DIALSTATUS=${DIALSTATUS}) exten => s,15,Noop(DNID=${DNID}) exten => s,16,Noop(EPOCH=${EPOCH}) exten => s,17,Noop(EXTEN=${EXTEN}) exten =>...
2010 Mar 26
1
problem with polarity reverse
...[12577]: channel.c:3766 ast_channel_inherit_variables: Not copying variable DIALEDTIME. [Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. [Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. [Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. [Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables: Not copying variable DIALSTATUS. [Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_var...
2009 May 26
0
No Voice - only "noisy audio"
...21a568", "Mobile/Carlos/909037079681") in new stack 13:37:40 rtp.c: Channel 'Mobile/Carlos-0213' has no RTP, not doing anything 13:37:40 channel.c: Not copying variable DIALEDTIME. 13:37:40 channel.c: Not copying variable ANSWEREDTIME. 13:37:40 channel.c: Not copying variable DIALEDPEERNAME. 13:37:40 channel.c: Not copying variable DIALEDPEERNUMBER. 13:37:40 channel.c: Not copying variable DIALSTATUS. 13:37:40 channel.c: Not copying variable SIPCALLID. 13:37:40 channel.c: Not copying variable SIPDOMAIN. 13:37:40 channel.c: Not copying variable SIPURI. 13:37:40 chan_mobile.c: Calling C...
2005 Aug 02
12
WHat does it take
How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm
2015 Apr 08
1
Help debugging a possible SIP channel leak in 11.17.0, possible race condition
Have you tried Asterisk 13? The bridging mechanism has been completely rewritten on Asterisk 12, so there's no longer channel masquerading and zombie channels. Might be worth a try. 2015-04-07 20:33 GMT-03:00 Alex Villac??s Lasso <a_villacis at palosanto.com>: > El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?: > > I am trying to collect enough information about an
2012 Aug 17
2
How to test Websocket support in SIP in Asterisk trunk?
I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100