similar to: PJSIP/SIPAddHeader etc

Hi, When implementing click2dial feature, I can trigger an Aastra phone to auto-answer using statement like : SIPAddHeader(Alert-Info: info=alert-autoanswer); This is very convenient when trying to reach a distant party (ie through PSTN) The trouble is when 2 Aastra are calling each other over the LAN, this single statement is memorized somehow and both phones (caller and callee) auto-answer.

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

Hi, I'm using the new SipAddHeader application on Asterisk 1.2.1, here's a snip of my extensions: exten => _9XXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM} exten => _9XXXXXXX,2,SipAddHeader(P-Asserted-Identity: tel:${CALLERIDNUM}) exten => _9XXXXXXX,3,Dial(SIP/${EXTEN}@${SIPTRUNK},,tT) exten => _9XXXXXXX,4,Congestion The problems is that Asterisk

Hi, I'm trying to use the SIPAddHeader application to add a header containing to semicolon separated strings like this: exten => 12, 1, SIPAddHeader(X-TestHeader:a=test1;b=test2) But in the resulting INVITE message only the first part (X-TestHeader:a=test1) is added. Setting into quotation mark doesn't change anything. exten => 12, 1,

Hi everyone, How can I add the equivalent of: exten => s,n,SIPAddHeader(Alert-Info: Ring Answer) in a .call file? This is to support paging to Polycom phones... Thanks for all info! Steve

Hi All, I'm doing some simple paging functions and using the SIPAddHeader cmd. * 1.2 branch. Using it in the extensions.conf file, it works fine: exten => _*2XX,1,SIPAddHeader(Call-Info: sip:\;answer-after=0) in * console: lab2*CLI> -- Executing SIPAddHeader("SIP/204-0818dcd0", "Call-Info: sip:;answer-after=0") in new stack When i put the same cmd in Realtime

Does SipAddHeader only allow headers to be added to INVITEs, or should it also allow headers to be added BYEs or SIP responses as well?

I use SIPAddHeader today to put some proprietary info into the SIP header of an outbound call. Now I'd like to add some proprietary info to the SDP portion of an outbound call. Can this be done with SIPAddHeader? Thanks in advance, Tom -------------- next part -------------- An HTML attachment was scrubbed... URL:

How to add SipAddHeader in outgoing call file. I am implementing a Callback scenario, in which a user makes a call to Local Access Number. The system have to callback to the user. During callback a call file is generated. All I want, is to add SipAddHeader("pchargingvector","val") in outgoing Invite. How can I achieve this? Please help me, where can I add SipAddHeader() in

Hi list, is there any existing IAX command to add information to a call like SipAddHeader? Another solution is sending text frame (0x07) frame type, but I don know how do it in a dialplan. Thanks for any hint. -- Daniel

How to add SipAddHeader in outgoing call file. I am implementing a Callback scenario, in which a user makes a call to Local Access Number. The system have to callback to the user. During callback a call file is generated. All I want, is to add SipAddHeader("pchargingvector","val") in outgoing Invite. How can I achieve this? regards, Asif

All other phones we work with will auto-answer when we do this: [macro-paging1way] exten => s,1,SIPAddHeader(Call-Info: answer-after=0) exten => s,n,Page(${PAGINGLIST}) exten => s,n, Hangup The SPA phones simply ring. I have verified that Auto Answer Page is set to yes (the default). We've tried a variety of firmware versions and phone ages, going back to an old 942 and new 504s.

Greetings: I'm in the situation where I'm trying to splash information picked off by an asterisk IVR into a Cisco call center environment. I'm under the impression that the ONLY way to do this is to setup socket connections with the Cisco "voice processor", or CVP, and send packets corresponding to GED-125. Cisco has a detailed 100+-page document detailing the internals of

Hi Guys I'm using asterisk 1.8.23.1 When I add a SIP Header from inside the extensions.conf (SIPAddHeader(Alert-Info:<http://www.notused.com>\;info=alert-internal\;x-line-id=0) ) it works fine. When I try to do the same thing from within a database table, all of the string apart from x-line-id=0 gets ignored. I've tried escaping the semicolon and not escaping it and the result is

Hi, I am using the XML-browser and Call-Info header features for some SIP phones. SIPAddHeader(Call-Info: ...) seems to work only in the outgoing direction. Does somebody know a way to send a Call-Info header to the originating SIP device by using only the dial plan? Currently, I am using the XML-browsers to update callee info, but I'd like to use the icon purpose to do that. It's

Le 27/11/2018 à 12:13, Joshua C. Colp a écrit : > On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...] >> >> [TOOTAiAudio] >> ; >> ; Call our gateway >> >> exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1}) >>  same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T) >>  same = n,Return >> >> exten = h,1,NoOp()

Thanks Scott. I was able to get the basic concept to run. However, it seems PJSIP INVITE for the Dial also does not support added headers. The Local channel dial plan did have the channel variable values. I added them as SIP headers, then Dial(PJSIP/Agent). The INVITE for the Dial on PJSIP continues to not include the SIP Headers I added. For chan_sip, I have no problem with this. Even the

Hi list, to manage an external queue agent the only solution I found is to connect a local account and redirect calls to this account using forward features from the phone (SNOM). The problem I face is that before calling the agent I would like to set extra header. Dialplan to call external agent is this one with (Gosub): [TOOTAiAudio] ; ; Call our gateway exten =

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

On Sat, Jun 17, 2023 at 2:55 PM TTT <lists at telium.io> wrote: > Based on postings it should be possible to get the SIP Call-ID header > value from the ARI. At what point is this value available ? As well, how > do I retrieve that value – something like > > > > GET /channels/{channelId}/pjsip_header?key=Call-Id > > > > But that doesn’t work. >