similar to: SIP 380 Alternative Service with PJSIP

I found that PJSIP allows only one asterisk per box. I tried to start several asterisks with the parameter -C and PJSIP only worked on the first process. In the other processes, the command "pjsip reload" was absent. Each pjsip transport in the second and subsequent processes was bound to a different IP in a multihomed box, something I routinely do with regular SIP. Am I wrong?

I am stuck in channel PJSIP trying to see the real flow of SIP messages, what in regular sip we used to type "sip set debug on" Also, is there an automated way to convert sip.conf options to pjsip.conf? Philip

A few months ago I started using and had to abandon PJSIP because my dialplan could not read the inbound signalling IP address, which I can read now in Asterisk11 using CHANNEL(recvip). My app relies on this information. The question is, is it possible now access the signalling IP of an incoming SIP call using PJSIP? Philip

I have been seeing errors saying the Asterisk cannot establish an RTP connection, so I did this: lsof -i -n -P | grep asterisk | wc -l 10483 but I have only Asterisk 11 has 1 open calls Asterisk 12 has 21 open calls Asterisk 14 has 19 open calls Asterisk 15 has 22 open calls Asterisk 16 has 15 open calls Asterisk 17 has 15 open calls Asterisk 30 has 71 open calls Total 164 active calls The

I did what we use to dim that is add a line to pjsip.conf like #include /etc/asterisk/pjpeers.conf but the file is not loaded. Am I doing something wrong this functionality is disabled?

I could not find a way to set a max on the calls allowed through a PJSIP endpoint. In case we decide to add it, the we need another reason for the call to fail in the Dial application, something like "limit reached" Am I missing this capability?

?Asterisk 13.11 rc1 ./configure LDFLAGS="-z muldefs" --libdir=/usr/lib64 --with-unixodbc=$(odbc_config --include-prefix)/ --disable-dev-mode --with-pjproject-bundled ?checking for pjsip_dlg_create_uas_and_inc_lock in -lpjsip... no checking for pjsip_tsx_create_uac2 in -lpjsip... no checking if "pjmedia_mod_offer_flag flag = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE" compiles using

In channel PJSIP I use this format Dial(PJSIP/endpoint/sip:${EXTEN}@ip.add.re.ss) what would be the equivalent of this format in old SIP? I tried Dial(SIP/peer/${EXTEN}@ip.add.re.ss) but it does not work. I just cannot embed the IP address in the peer's definition, but I need to use some other configuration features that are unique to each peer.

On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic > configuration works, and I am connected to a SIP trunk using SIP.US, and > have set up my inbound calling which works correctly (when I call my PBX > DID, the call does come into my PBX network). > > The

On Sun, Mar 15, 2015 at 1:33 PM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > Yes, I think the dial does get executed (sonny calling outbound > 202-555-1212): > > core set verbose 3 > Console verbose was OFF and is now 3. > -- Executing [912025551212 at from-internal:1] > Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from

On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > That was the issue, thanks. I now am able to get the caller ringing on an > outbound call, but an external phone number (E164) I am dialing does not > ring. > Any error messages? If you set 'core set verbose 3' and try it, does the Dial get executed? > > On Sun, Mar

Hello, I'm currently evaluating asterisk 13 (Currently on 11). We're testing the migration from SIP to PJSIP. Is there a way to alias the SIP channeltype to PJSIP when exlusively using pjsip? Matt Hoskins | NPG Corp | Systems Architect 816.749.2815 (Internal: ext. 10015) -------------- next part -------------- An HTML attachment was scrubbed... URL:

My phone is located behind a NAT, 172.16.0.0/21. Asterisk 16 is on a public IP. PJSIP has the config below: force_rport=yes direct_media=yes disable_direct_media_on_nat = yes direct_media_method=invite But when I send a call I see the RTP being sent to my private address, vs the public IP. This only happens when Asterisk has dialed the call to another carrier. If instead of Dial I choose

Hello, How should I configure Asterisk (13.7.0) to get persistently PJSIP SIP messages in a log file and not in console ? I would expect adding "debug=yes" in pjsip.conf to produce the same output as "pjsip set logger on". Am I understanding correctly ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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:

Hello, I have had numerous issues with PJSIP outbound calling in Asterisk 13.1.0 and SIP.US SIP trunk. My Asterisk server is on EC2 and I have opened up the appropriate ports. The SIP clients can be anywhere on the Internet, including behind NATs. I am able to get to my Asterisk server's internal extensions via the DID (and appropriate dialplans) but I am not able to make outbound calls to

Howdy, I'm trying to get my feet wet with pjsip using the conversion script mentioned on the Wiki on this page: https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip I'm using the copy of the script that's included with Asterisk 13 /usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip I assume I run it from /etc/asterisk with the input and output file as

That begs another interesting question...with analog phones picking up two extensions on the same "line" allow multiple people to participate on the call (without a "conference" feature) Does this become possible with multiple phones on the same PJSIP account? Or would the first phone answered need to somehow conference in the other phone? -----Original Message----- From:

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

Dear friends This is my simple dialplan [demopjsip] exten => _X.,1,Dial(PJSIP/${EXTEN}@10.10.10.2) exten => _X.,n,Hangup() I need to dial out via an IP address, not using an endpoint, as shown above. It fails with Executing [19544447408 at demopjsip:3] Dial("PJSIP/federico-00000002", "PJSIP/195XXX7408 at 10.10.10.2") in new stack [Jun 26 00:39:00] ERROR[10136]: