search for: pjsip_header

Displaying 20 results from an estimated 62 matches for "pjsip_header".

2018 Dec 10
4
PJSIP_HEADER - Diversion header manipulation
...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...
2015 Jul 10
2
Can I use PJSIP_HEADER to read the SIP 183 message header?
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. RODRIGO...
2015 Oct 06
2
PJSIP: how to retrieve underlying SIP Call-ID
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: ${PJSIP_HEADER(read,Call-ID)}) same => n,Dial...
2015 Jul 10
2
RES: Can I use PJSIP_HEADER to read the SIP 183 message header?
...____ De: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] em Nome de Mark Michelson [mmichelson at digium.com] Enviado: sexta-feira, 10 de julho de 2015 15:14 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] Can I use PJSIP_HEADER to read the SIP 183 message header? On 07/10/2015 11:53 AM, Rodrigo Pimenta Carvalho wrote: > Hi. > > The ASTERISK wiki has a page showing the function PJSIP_HEADER(). However, it doesn't explain if such function works only over SIP INVITE messages or if it can be use, for example, to...
2018 Nov 27
2
PJSIP add header on forwarded call
Le 27/11/2018 à 12:13, Joshua C. Colp a écrit : > On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...] >> >> [TOOTAiAudio] >> ; >> ; Call our gateway >> >> exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${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 : ${DIALSTATUS}) >>  same = n,NoOp(X-TOOT...
2023 Jun 26
2
Get channel variables via ARI/AMI
It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the entire SIP header for a channel. I also read (on stackoverflow) that the PJSIP_HEADER function will only return the headers from the INVITE of the inbound channel. If that’s correct, how would I get the headers from the outbound channel (second leg of the bridged call) INVITE...
2015 Aug 24
3
PJSIP add
...thing works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs (verbose 999). In the SIP case, I see it. Does the function Set(PJSIP_HEADER(add, ..... not transfer over to the call when the Queue function is called? Am I calling the Set(PJSIP_Header(add portion incorrectly? Or is this a problem with the Asterisk PJSIP support? chan_sip... [macro-MY-SetDNID] exten => s,1,Verbose(X-MY-DNID:${MY_DNID}) exten => s,1,SIPAddHeader(X-...
2018 Nov 27
2
PJSIP add header on forwarded call
...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,Retu...
2023 Jun 17
1
Get SIP Call-ID from ARI
...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 better...
2023 Jun 26
1
Get channel variables via ARI/AMI
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 > (s...
2016 Mar 31
2
Asterisk 13 - Call Bridge issue.
...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(hold,...
2019 Nov 15
2
pre-dial handler, how to access variables from calling channel?
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]; Manipulate...
2023 Jun 17
1
Get SIP Call-ID from ARI
...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 i...
2023 Jun 17
1
Get SIP Call-ID from ARI
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>
2019 Mar 29
3
why doesn't extension "s" work ?
I'm using "s" extension in my dialplan: [gv-voice] exten => s,1,Verbose(callerid is "${CALLERID(all)}" or "${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)}) ; PJSIP_HEADER(read,To) same=>n,.... But when a call comes in to the gv-voice context, "s" doesn't match the extension: res_pjsip_session.c:2991 new_invite: Call from 'gv-voice' (UDP:10.10.10.80:5062) to extension '<xxxxxxxxxx>' rejected because extension not found...
2020 Jan 24
4
Perl AGI: read variable with quotes
Hi Gang I have stumbled of this problem. I need the P-Asserted-Identity header in an AGI scrip. In the Dial-Plan I do: same => n,Set(PAI=${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-I...
2023 Jun 26
2
Get channel variables via ARI/AMI
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
2023 Jun 26
2
Get channel variables via ARI/AMI
...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) INVITE...
2020 Aug 10
2
With ARI, is it possible to create (originate) a call and pass both the caller id name and number?
...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? >...
2020 Aug 10
0
With ARI, is it possible to create (originate) a call and pass both the caller id name and number?
...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 d...