similar to: 2. Re: Does Asterisk support remove header from sip message?

Ding Peng ---------------------------------------------------------------------- Message: 1 Date: Mon, 21 Jan 2013 15:22:41 +0800 From: "Ding Peng" <roc.dingpeng at gmail.com> Subject: [asterisk-users] Does Asterisk support remove header from sip message? To: <asterisk-users at lists.digium.com> Message-ID: <000101cdf7a8$1c9e3870$55daa950$@gmail.com> Content-Type:

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, 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 everyone I’m building an Asterisk 16/PJSIP server and my dialplan uses SIPAddHeader & SIPRemoveHeader but the apps don’t appear to be installed in v16. Can anyone tell me where they went and how to get them installed please? Thanks Mark. Mark Farmer Senior UC Systems Architect Intercity Technology Limited HQ 101-114 Holloway Head, Birmingham, B1 1QP Tel: 0330 332 7933 / 07872542107 /

It seems that for local channels (asterisk 1.4.33) the variable Variable: SIPADDHEADER="Alert-Info: Ring Answer" (call polycom phones and ring then auto answer) Is ignored, Is this just an oversite or is there some reason? It works fine with I call the SIP phone directly - however - when I first call the Local channel - then Dial the SIP phone the SIPADDHEADER doesnt seem to do

Hello, when calling a group of SIP peers like this : Dial( "SIP/inno0&SIP/inno4&SIP/inno6,30") is it possible to have a SIP header added for just 1 of these SIP peers, like only for SIP/inno0 but not for SIP/inno4 and SIP/inno6 ?? I know the function SipAddHeader(), but when I use this in the dialplan before the Dial()-command, then the header is added for all the SIP

Hello, I have a situation where a call comes in to my Asterisk server B. This call comes from another Asterisk server A. I want to tell to this server A why my server B hangs up. So just before hanging up, I add a custom SIP-header : exten => s,n,SIPAddHeader(X-My-Hangup: MaxChan) exten => s,n,Hangup() But I notice that this extra SIP-header is not send within the SIP-reponse :

Hello list, I am trying to set a custom SIP header on all calls that are made by the app queue because I want to track a certain state at the SIP level. If I use the following code: exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID}) exten => s,n,Queue(myQueue) this works fine for the FIRST call made from the queue to an agent; but if that call does not go through, it's not repeated

I'm wondering if anyone has a solution to this before I begin looking at making some changes to the SIP channel. Basically when calling SIPAddHeader() twice from the Dialplan or an AGI script with the same header name it adds duplicate headers instead of overwriting the existing one. Here's a practical situation where this applies. A call is to be terminated via SIP and we have two

Prezado Fernando, Muito obrigado por sua complementa??o na resposta! Surgiram algumas d?vidas agora: A ?nica forma de retornar os dados num header field, como o Rafael dos Santos Saraiva sugeriu envolve criar outro channel? Ou seja, o que eu preciso ? que a mesma execu??o do dia plan obtenha um valor recebido do Sip Client, execute uma query num banco de dados e em seguida inclua a resposta

Hello! I want to thank everyone who helped me out with tips for load balancing asterisk machines in a cluster. I have encountered a new problem that is related to SIP diversion headers in the INVITE. I make calls through the manager interface and now want to add a SIP-Diversion header that changes the CallerID of a number that is not available on the trunk, the CallerID to be visible externally

Hey, Hi, All, We have a few dozen of the Kirk (ie: Polycom bought this European brand) out there & most all work very well & work very well with most all versions of Asterisk. But we have been tripped-up by one combination of firmware & version & configuration variables. We are running Asterisk 1.4.23.1 (TrixBox CE). We are running latest stable firmware on the handsets. Most

This is a problem that is completely stumping me, and my understanding of Asterisk dialplans tells me this should never be a problem. Moreover, this scenario works on Asterisk 1.4 but not 1.6. We have a customer with several Aastra 6731 phones. They want incoming calls from the PSTN to work and they also want to be able to call each other "internally" on a special non-DID number (like

Hello! * Version: 1.6.0.3-rc1 Scenario: * -> Proxy -> routed back to myself (The only thing changing is the Request URI) (And the Record-Route, Via that are added, of course). Outgoing Context is faxserver-out, incoming context is faxserver (at least should be). Outgoing context is straight forward: [faxserver-out] exten => _X.,1,NoOP(FAXOUT -- Connecting ${CALLERID(all)} ->

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

I'm scratching my head trying to work out a way of sending a pre-recorded message as a 'Page' to a list of phones ( "Oi! you muppets you've left the server room door open!" or somesuch message :-) controlled by an external trigger. I can do a normal page (phones auto-answer on speaker) with SipAddHeader but that doesn't let me play a pre-recorded message. Any

I came up with this a few days ago, mostly used the wiki examples, didn't have time to post on the wiki yet, maybe one of you guys with a few minutes can throw it up there, really, I forgot my logon. http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom The agi script didn't work for me, wouldn't call the active hint extensions, even though they were there, no

<div>Is there any chance to create feature request for that useful functionality?</div><div>š</div><div>02.07.2015, 14:03, "Rusty Newton" <rnewton@digium.com>:</div><blockquote type="cite"><div><div><div>On Wed, Jul 1, 2015 at 4:46 AM, <span><<a href="mailto:royj@yandex.ru"

I have a call coming in. I need to add a SIP Header to the channel. Then, I need to send the call to the Queue so it is sent to the Agent. The SIP header I added, I need to have appear in the INVITE sent to the Agent. It works in chan_sip. I send the call to a macro which does... n,SIPAddHeader(My-Header-Name:${MY-HEADER-VALUE}) n,Queue(${ARG2}) In PJSIP , this doesn't seem to work. Is

Hi all, I want to configure a SIP Channel to send Alert-Info with the INVITE. Right now I have added an extension like so: exten=>4444,n(ring5),SIPAddHeader("Alert-Info: R0") But there is no Alert-Info in the INVITE. Any idea how I can get this working? Seems a very basic error.. Thanks. Sandip -------------- next part -------------- An HTML attachment was scrubbed... URL: