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