search for: sipcallid

...d I feel like this should be a relatively simple task, but I just can't get it to work. I currently have a very basic asterisk v11.6 setup with a single extension (a Bria softphone) and a single sip trunk to my carrier. What I'm trying to accomplish is simply adding the asterisk generated SIPCALLID of the leg between asterisk and the carrier to the CDR when it's written. (This is in addition to the SIPCALLID of the original call leg, which I've already added.) I've been able to print the SIPCALLID for the carrier leg to the CLI if I jump into a Macro or Gosub from the Dial comman...

Hi I sent this 3 hours ago, seems not go through, so sent again. I have an asterisk php-agi application. It answer's call , then outdial to another number: $agi->exec_dial("SIP", 12345 at test.com , "20", $options); How can I get a SIPCALLID for this out-dialed call? The SIPCALLID seems the incoming call's SIPCALLID. Thanks. Mike -- Best Regards Mike

...> [outbound] ; this is called on the incoming (caller) channel > exten => _X.,1,Noop > same => n,Set(MASTER_CHANNEL(start_timestamp)=${STRFTIME(,,%s.%3q)}) > same => n,Set(CHANNEL(hangup_handler_push)=hangup_handler,s,1) > same => n,Set(MASTER_CHANNEL(callid_ingress)=${SIPCALLID}) > same => n, *** unrelated dialplan, AGIs, etc. *** > same => n,Dial(SIP/${EXTEN}@1.1.1.1,,U(answer_handler)b(pre_dial_ > handler^s^1)g > same => n, *** dialplan for the caller when the callee hangs up first, > not run when caller hangs up first. use it to try dialing...

Hi, Does anybody know how to get the SIP call ID of a "Dial" command? Thanks in advance. Ray -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.com! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071008/7f27548e/attachment.htm

Hi, To combine two sources of CDR's I want Asterisk to save the SIP callid for all calls. I know there's a variable that contains the SIP CallID value, but is this the callid value of the incoming INVITE message or the outgoing message? Are they the same? (I've not yet checked a trace, I'm sorry for that). I've tried to read chan_sip, but couldn't find something in the

Hi, Is there any way of telling Asteirsk to withhold answer subversion on a call till I call Answer. My DP looks like this: [incoming] Exten => 18005551212,1,Noop() same => n,Answer same => n,Mset(__uid=${SIPCALLID}) same => n,MixMonitor(/tmp/FROM_CALLER_${uid}-${START}.WAV) same => n,Dial(Local/1 at dial_call_center/n&Local/2 at dial_call_center /n&Local/3 at dial_call_center/n&Local/4 at dial_call_center /n&Local/5 at dial_call_center/n,,gm) same => n,Goto(2) [dial_call_center]...

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <- don't mind L, a quick hack for dtmf not working with sip exten => _...

...s of A have been merged into B2<ZOMBIE>. If there were duplicate names for variables, the channel now has those variables doubled. The DumpChan() application called from the h extension confirms this. In my case the channels are all SIP channels and in the h extension I want to access the SIPCALLID variable of the A channel. Every access to it gives me the wrong value namely that of channel B1. How do i access the _second_ variable named SIPCALLID in the channel? Extract from DumpChan() as an example: Dumping Info For Channel: SIP/sipout-00000055<ZOMBIE>: ===========================...

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I have 1 cdr record with only inbound call leg callid. I can see all the call legs with rasterisk -x "sip show channels...

...the outgoing number is dialed, no matter what I put for the timeout values. Anyone else using DISA that has run into this? exten => _2X,1,Answer exten => _2X,2,DigitTimeout(2) exten => _2X,3,ResponseTimeout(2) exten => _2X,4,SetAccount(1${EXTEN}) exten => _2X,5,SetCDRUserField(${SIPCALLID}) exten => _2X,6,DISA(no-password|<another context>) -- ----- Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 noc@isdn.net

Is there a way to retrieve the Call-ID from a call made using the 'Dial' command on a SIP channel without CDRs (i.e. variable) ? Thanks, - Darren

Hello friends, I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can access the caller Call ID (fbasename field in voipmonitor cdr) looking at the SIPCALLID variable in asterisk, but how can I access from within asterisk the Call ID of the second leg of the call (the one originating from asterisk to the destination peer)? is there a variable holding this value? Thank you Leandro -------------- next part -------------- An HTML attachment was scrubbed....

...-- PBX -- Context: XXXXXXXXXXX Extension: XXXXX Priority: XXXXXX Call Group: 0 Pickup Group: 0 Application: AgentLogin Data: (Empty) Blocking in: ast_waitfor_nandfds Variables: AVAILSTATUS=0 AVAILORIGCHAN=SIP/303 AVAILCHAN=SIP/303-0949f890 SIPCALLID=Y2MzOTc0NmExYjVkNDNjMzhhY2I1MDMwNTk0NTJkYzQ. SIPUSERAGENT=X-Lite release 1100l stamp 47546 SIPDOMAIN=XXXXXXXXX SIPURI=sip:303 at XXXXXXXXXXXXXXXXX CDR Variables: level 1: clid="Ext. 303" <303> level 1: src=303 level 1: dst=XXXXXXXXXX level 1: dcontext=XXXXXXXXXXX level 1: channel...

Hi, I have a dial plan where I need to notify an external system when a call was answered and when the call hung up. In both requests the start time needs to be the same. My Dialplan looks something like this: [outbound] Exten => _X.,1,Dial(SIP/${EXTEN}@1.1.1.1,,U(call-answer-from-carrier)) Exten => h,1,NoOp(ANSWERED_TIME: ${ANSWEREDTIME} >>> DIAL_TIME: ${DIALEDTIME}

Calling linphone from asterisk 13.9.1.: Dial(SIP/<user>@sip.linphone.org) And it works. But on the linphone side the caller is: <extno>@ipaddress or 2502 at 45.123.987.4 Is there any way to make it more descriptive, at least for the sip user name ? I tried setting SIPCALLID, which had no effect. Set(SIPCALLID=Office) Thanks, sean

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 19:58, Ludovic Gasc wrote: > Hi, > >

...Direct Bridge: <none> Indirect Bridge: <none> -- PBX -- Context: from-internal Extension: Priority: 1 Call Group: 0 Pickup Group: 0 Application: Playback Data: demo-congrats Blocking in: ast_waitfor_nandfds Variables: SIPCALLID=14a13ecb635daaed76e6ab905ba0cff1 at 192.168.5.193:5060 CDR Variables: level 1: dnid= level 1: clid="device" <1064> level 1: src=1064 level 1: dst=s level 1: dcontext=from-internal level 1: channel=SIP/1064-00000044 level 1: lastapp=Playback level 1: lastdata=demo-congrats level...

...Dialed Number Identifier ${RDNIS} Redirected Dial Number ID Service ${HANGUPCAUSE} Asterisk hangup cause ${ACCOUNTCODE} Account code (if specified) ${LANGUAGE} Current language ${SIPDOMAIN} SIP destination domain of an inbound call (if appropriate) ${SIPUSERAGENT} SIP user agent ${SIPCALLID} SIP Call-ID: header verbatim (for logging or CDR matching) Applications that works with variables * set your own variables with the setvar() and the setglobalvar() application. * the gotoif() app lets you can make conditional tests on variables and jump to various extensions or priorities o...

...in= 0 Framesout= 0 TimetoHangup= 0 ElapsedTime= 0h0m0s Context= voicepulse-in Extension= 12222222 Priority= 4 CallGroup= PickupGroup= Application= DumpChan Data= (Empty) Blocking_in= (Not Blocking) Variables: SIPCALLID=282e93ca78805a039fdf01729af52c at 64.62.94.171 SIPUSERAGENT=Asterisk PBX SIPDOMAIN=66.195.225.160 SIPURI=sip:3333333 at 64.62.94.171 <sip%3A3333333 at 64.62.94.171> ================================================================================ -- Executing [12222222 at voicepulse-in:5]...

...retrieve Call-ID from dialed SIP channel(w/o CDRs) Darren Sessions wrote: > Is there a way to retrieve the Call-ID from a call made using the 'Dial' > command on a SIP channel without CDRs (i.e. variable) ? (sometimes I wonder why we write documentation) doc/README.variables has ${SIPCALLID} documented to be exactly that. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users