similar to: PJSIP equivalent for SIPDtmfMode?

With chan_sip there is the variable SIP_MAX_FORWARDS to set Max-Forwards. This counter is persistant after a redirect. I can't find the equivalent for PJSIP, so I went the way of header manipulation. Only to find out that any headers added to the outbound leg are lost after a redirect (with redirect_method=uri_core (didn't try any other since in the past they didn't work for me)). Am

With pjsip (asterisk 13.14.1) I see the problem that an anonymous from header gets user=phone appendend to the URI if user_eq_phone=yes is specified: On the incoming leg: From: anonymous <sip:anonymous at anonymous.invalid:5060>;tag=Q5zBj7BMnvI6Fe6O2866fox3ZHmn-smt Get transformed to From: "Anonymous" <sip:anonymous at

Hey all! I recently tried the dtmf_mode "auto_info" on my setup to support endpoints that only understand SIP INFO as a fallback. My setup is the following: Endpoint A (RFC4733) --> Asterisk <-- Endpoint B (SIP INFO) Both are configured with "auto_info" dtmf_mode in pjsip.conf. What I ran into is, that DTMF sent from endpoint A to endpoint B is additionally sent via

Hi, when sending IMs from endpoint to endpoint with the MessageSend() application, I can check the MESSAGE_SEND_STATUS and send another message to the sender of the message to notify them that their message was not sent when the status indicates it. This works fine with chan_sip. With chan_pjsip, this works differently in that MESSAGE_SEND_STATUS is "SUCCESS" after sending the

Greetings. I'm having a bit of a problem using the sipdtmfmode app. I have two incoming paths to * from pstn via FWD that use differing dtmfmode. IPKall wants rfc2833, libretel wants inband. If I set dtmfmode= in the fwd peer config in sip.conf each works seperately, and I'm trying to use gotoif and sipdtmfmode to switch based on the CID calling. Output seems to indicate sipdtmfmode

While trying to use direct_media I'm seeing RTP payload mismatches after succesful reinvites. Initial INVITE from endpoint A to asterisk has rfc4733 DMTF m=audio 35648 RTP/AVP 9 8 111 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 >From asterisk to upstream U: m=audio 14338 RTP/AVP 9 8 111 18 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 So the payload types in the RTP

All, Here's a cool one.. I was attempting to call a retarded conferencing service, and was having problems with it picking up my DTMF.. after trying all the settings my Sipura SPA2000 offers, I found inband actually works.. unfortunately, I can't get anything else to pick up my inband DTMF (including asterisk's builtin voicemail! It just times out and says I never entered a login!).

I'm trying to figure out how to commit some code for review. Following: https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage Created a ssh alias. Cloned using: "git clone ssh://asterisk/asterisk" Set name and email. Installed the gerrit commit hook: "git review -s" Try to change to asterisk 13 for creating a patch: "git checkout 13" This fails with: error:

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:

On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <kevin.long at haloprivacy.com> wrote: > > Thanks George I appreciate the info . Being able to see what codec is in > use for call in progress is very handy sometimes. > > As far as the RTP stats goes, I see there is some info with ?rtp? and > ?rtcp? commands which can be useful for troubleshooting. A running tally of > #

Hello, I'd like share with you my tests about PJSIP channel with the aim of improving the functioning of the channel: * Multi domain support not work correctly: https://issues.asterisk.org/jira/browse/ASTERISK-26026 * Different context subscribe for each endpoint not possible: https://issues.asterisk.org/jira/browse/ASTERISK-25471 * BLF don't work correctly on my tests

Hi Richard > That could be possible and would be a bug in chan_sip. Ok, so I switched to PJSIP to see if this behaves differently So ip do a Transfer(PJSIP/${DESTNUMBER}@trunk) And this results in: Failed to parse destination URI '[destnumber scrubber]' for channel PJSIP/trunk-00000011 Do I have to specify the destination number differently when using Transfer with pjsip that I

I have multiple Asterisk instances set up in different locations and would like to modify the callerID of inbound calls to identify which instance the call is coming from.? I knew how to do that with the old sip format, but can't seem to figure it out with PJSip. For example: Currently Location A, extension 10 calls Location B, extension 20.? CallerID on Extension 20 displays

I'm trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures. For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <-> uplink RTP: enduser <-> SBC <-----------------> uplink SBC

Hello, In /etc/asterisk/users.conf, you can set hassip=yes to declare a chansip entity. Is there any equivalent for PJSIP ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20191230/50ece4fa/attachment.html>

Hi, I'm trying to port our configuration form sip to pjsip channel and have following issue. Sip.conf has a host parameter that sets the RURI to a given value. This functionality is needed in some of our scenarios where we need to send requests to specific IP address with specific domain in RURI. I did not found an equivalent to the host parameter in pjsip configuration. Did I

On Wed, 2020-10-28 at 14:40 -0300, Joshua C. Colp wrote: > This is not yet fixed, but is being worked on. I have it as a > security issue currently out of caution (although I don't think we'll > treat it as one after further investigation). Right OK, thanks. Do you have any idea of the sort of timescale, and whether it'll be available as a patch that we can apply to our

Hi, I'm using Asterisk to bridge the incoming call to another destination using the Dial command. However, when an anonymous call comes in then privacy information is not passed into the B Leg. For instance, the Privacy header and P-Asserted-Identity aren't copied to the B Leg. Is there an option to give to the Dial command, or another variable to set, to make Asterisk copy such

Hello, I am moving from realtime chan_sip to pjsip and one of the problem I am facing is the lack of "sipregs". With chan_sip, when an extension registers, the server where it has registered to is stored in sipregs. Is there something similar in pjsip? How can I find on which server the pjsip extension has registered to? Leandro -------------- next part -------------- An HTML

Hello, On a newly built Asterisk 13.13.1 system, I can't make HEP work with chan_sip (though I could make it work with PJSIP on another instance). Looking either at [1] or hep.conf, I can't see anything meaning HEP requires PJSIP. Before diging deeper, may I simply ask if HEP requires PJSIP or not ? What about a line mentioning the answer in above documents (to keep other from wondering