search for: pjsip_head

...ewrite Diversion header when call forwarding is done on the phone. The phone sends "302 Moved Temporarily" response and sets Diversion header to a local number, but before Asterisk sends this call towards TSP provider I need to change Diversion header to a full PSTN number. I am using PJSIP_HEADER in a pre-dial handler (configuration is below). On the same call I can add some other custom headers (logs are below). Is there any chance I can rewrite Diversion header in this scenario with PJSIP_HEADER function? Asterisk version is 16.0.1 built from source on Debian 9. Thank you Davor...

Hi. The ASTERISK wiki has a page showing the function PJSIP_HEADER(). However, it doesn't explain if such function works only over SIP INVITE messages or if it can be use, for example, to read headers from others types of SIP messages too. So, can I use PJSIP_HEADER to read the SIP 183 message header? Any hint will be very helpful! Best regards. RODRIG...

Hello, I've started to play with PJSIP and got stuck at the following problem. I need to retrieve SIP Call-ID associated with PJSIP channel. For inbound channel I can use ${PJSIP_HEADER(read,Call-ID)}, but that doesn't work for outbound channel even in pre-dial or hangup handler. Whatever I do PJSIP_HEADER seem to be unable to read headers for outbound channel. Here's what I do: [xyz] exten => 999,1,NoOp(Call-ID: ${PJSIP_HEADER(read,Call-ID)}) same => n,Di...

...____ De: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] em Nome de Mark Michelson [mmichelson at digium.com] Enviado: sexta-feira, 10 de julho de 2015 15:14 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] Can I use PJSIP_HEADER to read the SIP 183 message header? On 07/10/2015 11:53 AM, Rodrigo Pimenta Carvalho wrote: > Hi. > > The ASTERISK wiki has a page showing the function PJSIP_HEADER(). However, it doesn't explain if such function works only over SIP INVITE messages or if it can be use, for example,...

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() >>  same = n,NoOp(Hangup Cause: ${HANGUPCAUSE}) >>  same = n,NoOp(Dial status : ${DIALSTATUS}) >>  same = n,NoOp(X-TO...

It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the entire SIP header for a channel. I also read (on stackoverflow) that the PJSIP_HEADER function will only return the headers from the INVITE of the inbound channel. If that’s correct, how would I get the headers from the outbound channel (second leg of the bridged call) INVIT...

...thing works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs (verbose 999). In the SIP case, I see it. Does the function Set(PJSIP_HEADER(add, ..... not transfer over to the call when the Queue function is called? Am I calling the Set(PJSIP_Header(add portion incorrectly? Or is this a problem with the Asterisk PJSIP support? chan_sip... [macro-MY-SetDNID] exten => s,1,Verbose(X-MY-DNID:${MY_DNID}) exten => s,1,SIPAddHeader(...

...ct 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 = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1})  same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T)  same = n,Return exten = h,1,NoOp()  same = n,NoOp(Hangup Cause: ${HANGUPCAUSE})  same = n,NoOp(Dial status : ${DIALSTATUS})  same = n,NoOp(X-TOOTAiAudio=${PJSIP_HEADER(read,X-TOOTAiAudio-CALLED)})  same = n,Re...

...3 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. > 'pjsip_header' is not a valid route. All possible routes are documented on the wiki, if it's not there then it doesn't exist. Instead you would use variable[1] to execute the PJSIP_HEADER dialplan function[2] or a bette...

On Mon, Jun 26, 2023 at 4:04 PM TTT <lists at telium.io> wrote: > It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the > entire SIP header for a channel. I also read (on stackoverflow) that the > PJSIP_HEADER function will only return the headers from the INVITE of the > *inbound* channel. > > > > If that’s correct, how would I get the headers from the outbound channel >...

...xy}) exten => s,n,NoOp(Carrier Trunk - ${l_Carrier}) exten => s,n,Set(_l_CallerIDnum=${CALLERID(num)}) exten => s,n,Set(CALLERID(num)=${g_SIPUser}) exten => s,n,Dial(PJSIP/${l_DialTo}@proxy_${l_Proxy},30,b(dialer-header^s^1)G(dialer- playmsg^s^1)) [dialer-header] exten => s,1,Set(PJSIP_HEADER(add,X-Carrier)=${l_Carrier}) same => n,Set(PJSIP_HEADER(add,X-CallerID)=${l_CallerIDnum}) same => n,NoOp(X-Carrier = ${PJSIP_HEADER(read,X-Carrier)}) same => n,Set(CONNECTEDLINE(number,i)=vap_002) same => n,DumpChan(1) same => n,Return() [dialer-playmsg] exten => s,1,Goto(hol...

Hi List Implementing screening and routing I have stumbled over this issue: [pbx-router] exten => s,1,NoOp(ROUTER FROM: ${CALLERID(Number)} TO: ${DESTINATION}) same => n,Set(SOURCE=${CHANNEL(name)}) same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) same => n,Set(FROM=${CALLERID(Number)}) same => n,Set(TO=${DESTINATION}) same => n,Set(DIVERSION=${PJSIP_HEADER(read,Diversion)}) same => n,AGI(router.agi) same => n,GoTo(dial-out,s,1) [predial-handler]; Manipulat...

...ble?variable=CHANNEL(pjsip,call-id) But it responds with "message": "Channel not in Stasis application" Since I want to get the call-id for a channel not in stasis I guess that won’t work. Similarly, I can’t force the channel through my own code in the dialplan, so the PJSIP_HEADER function won’t work. So it looks like I’ll have to upgrade my Asterisk test system to get the Call-ID from the ARI event. It looks like it was added in Ast 16. Out of curiosity, I see that call-id is returned in the “protocol_id” field of channel data structure. However, since all channels...

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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230617/ae7c536c/attachment.html>

I'm using "s" extension in my dialplan: [gv-voice] exten => s,1,Verbose(callerid is "${CALLERID(all)}" or "${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)}) ; PJSIP_HEADER(read,To) same=>n,.... But when a call comes in to the gv-voice context, "s" doesn't match the extension: res_pjsip_session.c:2991 new_invite: Call from 'gv-voice' (UDP:10.10.10.80:5062) to extension '<xxxxxxxxxx>' rejected because extension not foun...

Hi Gang I have stumbled of this problem. I need the P-Asserted-Identity header in an AGI scrip. In the Dial-Plan I do: same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) In the AGI I do: my $pai = $AGI->get_variable(PAI); This works fine, unless the PAI contains quotes: P-Asserted-Identity: <sip:1000 at 1.2.3.4:5060;user=phone> I get "<sip:1000 at 1.2.3.4:5060;user=phone>" in the variable $pai. P-Asserted...

On Mon, Jun 26, 2023 at 10:57 AM TTT <lists at telium.io> wrote: > I am connecting to the ARI with subscribe all, so I can see channels being > created. I now want to extract a variety of header variables (at the > moment the from and to tag). I tried to read them from the ARI but > Asterisk refuses since the channel is not in a stasis app. > > > > Is there a way

...s Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Get channel variables via ARI/AMI On Mon, Jun 26, 2023 at 4:04 PM TTT <lists at telium.io <mailto:lists at telium.io> > wrote: It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the entire SIP header for a channel. I also read (on stackoverflow) that the PJSIP_HEADER function will only return the headers from the INVITE of the inbound channel. If that’s correct, how would I get the headers from the outbound channel (second leg of the bridged call) INVIT...

...terisk -X POST " http://localhost:8088/ari/channels/newChannelId" <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" , "PJSIP_HEADER(add,P-Asserted-Identity":"foobar"} }' BR Jöran On Mon, Aug 10, 2020 at 3:43 PM Dan Cropp <dan at amtelco.com> wrote: > Hi Jöran, > > > > Would it be possible to see an example using curl of how you are passing > the PAI Header through ARI create? &gt...

...terisk -X POST " http://localhost:8088/ari/channels/newChannelId" <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" , "PJSIP_HEADER(add,P-Asserted-Identity)":"foobar"} }' there was a bracket missing after the function of PJSIP_HEADER BR On Mon, Aug 10, 2020 at 3:57 PM Jöran Vinzens <vinzens at sipgate.de> wrote: > Hi Dan, > > i would do something like this (it is not a copy of what we are...