similar to: avoiding any media proxy with PJSIP

I cannot find in Asterisk 12, the channel variable ${CHANNEL(recvip)}, so if I use PJSIP, for scalability, how do I read what the signalling IP where the inbound call is coming from and what is the inbound codec? You would think that the new channel would set those variables up, isn't it? Philip Orleans

Asterisk is unique in terms that we can create new applications that talk to databases and generate any logic whatsoever. Asterisk is a development environment for anything telecom, not a PBX. I believe that we need to make PJSIP more efficient so Asterisk can expand its footprint. Please tell somebody to add a way to prohibit PJSIP from proxying RTP. I can help if you give me some directions, but

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 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

Dear Asterisk-Users, By means of Asterisk 11 and sip.conf, I got success implementing early media. That is, all information that come from callee (SIP 183 message/ SDP) is passed to the caller without any modification in the SDP body. However, in Asterisk 13 and using pjsip.conf I'm still failing to do the same thing. See: Softphojne1

?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

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?

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

I need a consultant to help me install the latest Kernel 3.2. I compiled it successfully under Centos 5.1 64 Bits, did "make world", "make install", end it did not show any error. I changed the grub.conf file as per the README, but the machine did not boot. There is must be something missing in the instructions. If somebody who can help me, I will pay him/her via paypal. Has

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

Hi there I have a pbx (v16.10) on AWS (Ubuntu 18.04) with Freepbx (14) that I am trying to set up the proxy reSIProcate on the same host as pbx. I can make it all work when the proxy is on a different host but when the proxy is on the same host asterisk sends the media address as 127.0.0.1 which the end user then happily sends media to 127.0.0.1 but it doesn’t get anywhere. Asterisk then

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?

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

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 issue is that I am not able to make outbound calls, because the call fails with the error:

The Call for Proposals for the 2013 Linux Plumbers Virtualization Microconference is now open. This uconf is being held as part of Linux Plumbers Conference in New Orleans, Louisiana, USA September 18-20th and is co-located with LinuxCon North America. For more information see: http://www.linuxplumbersconf.org/2013/ The tentative deadline for proposals is August 1st. To submit a topic please

The Call for Proposals for the 2013 Linux Plumbers Virtualization Microconference is now open. This uconf is being held as part of Linux Plumbers Conference in New Orleans, Louisiana, USA September 18-20th and is co-located with LinuxCon North America. For more information see: http://www.linuxplumbersconf.org/2013/ The tentative deadline for proposals is August 1st. To submit a topic please

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