similar to: PJSIP_HEADER - Diversion header manipulation

Hi All, I'm connecting to my carrier which requires setting of outboundproxy. There has been few cases where the proxy server failed due to network issues and required us to use a secondary one. Is there a timeout or qualify setting for outboundproxy setting in sip.conf? I do appreciate if anyone can help please. Thank you -Abeed -------------- next part -------------- An HTML attachment

Hi All, I cannot seem to find a resource that will allow me to RAID5 3 x 1tb drives on system install. Can this be done? -jason

We upgraded from asterisk 11 to asterisk 13. Recordings were working fine in 11 but nothing is being written on 13. Here is the dialplan segment same => n,ExecIF($["${TL_PHONE_CALL_RECORD}"="TRUE"]?SET(CONFBRIDGE(bridge,record_conference)=yes)) same =>

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:

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()

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 PIMENTA CARVALHO Inatel

I am trying to set add a SIP Header to a call before adding it to the Queue. The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything 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

Ok Mark Michelson. Thank you very much! You answer tells me that I was in the wrong path trying to access information from SIP 183 message. I need to find a way to let the callee pass information/data to the caller, even before accepting the call. That is, send data during the ringing time. And in my case, there will be more than one callee ringing at same time. As ASTERISK will not forward each

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 =

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. >

I have the following senerio. Call file calls 1st party. When connected give called party option to connect to second party. Issue Dial to second party. Caller answers and the two are bridged together. My issue is that 4 out of 5 calls fail to bridge the audio. Am I missing something or is there some kind of bug? Here is my test dialplan ;Dialer Base Code Files. ;Variables

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:

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

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 ? Or will PJSIP_HEADERS() in fact return the

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

Hi Dan, i would do something like this (it is not a copy of what we are doing but an example of how i would do it) Important here is the "--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki. curl -v -H "Content-Type: application/json" -u

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

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 /

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

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