similar to: Question about SIP Dial

?I have the same exact issue. I cannot push any sounds or even Playtones to the caller, unless the channel is answered, which is not possible for billing reasons. I am also using the Local channel & Dial(PJSIP/...). I think this is a bug in Asterisk 13. The Dial function has not answered yet, so the Local channel should be able to play anything to the caller, without answering, in parallel

?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?

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

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

Hi, Is there any way I can get exact sip status from pjsip after a dial ? or all we can get is asterisk hangup causes ? Thanks in advance. KKh

Is this normal to create one thread per peer in Asterisk 12, chan_sip regular, not pjsip? What happens is I have 659 peers, and I get 682 tasks on ls /proc/15373/task | wc -l If this is normal then of course I can only get a few instances before my box collapses. Is it any different in pjsip? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I think HANGUPCAUSE is channel agnostic. See: core show function HANGUPCAUSE Some thing like this IIRC: Set(my_cause=${HANGUPCAUSE(${CHANNEL(name)},tech)}) Remember the incoming leg of the call and the outgoing leg of the call are different channels. Make sure you are giving HANGUPCAUSE the correct channel. On 06/09/2018 02:01 PM, Khalil Khamlichi wrote: > It seems very weird to me

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?

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:

I am migrating my app to Asterisk12 and pjsip, but I cannot find chan_local, what happened?

I cannot wait for the regular bug-patch process to play out. I am considering hiring a developer to fix bug 24015, and of course submit the patch for the bug. The issue is simple, the app Transfer does not transfer when using PJSIP.. I called Digium and they said that they do not do this kind of work. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear List It's probably been more than a year now I switched from chan_sip to pjsip. pjsip works much cleaner than chan_sip. But! I have come across a Problem I was not able to solve with Asterisk Dialplan Logic. With pjsip an endpoint can have multiple AOR, so you need to expand them with ${PJSIP_DIAL_CONTACTS()} to be able to Dial() all of them simultaneously. But there are also

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]:

I found an issue with how PJSIP handles a typo in the Dial application. If the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//xxxx...), the Dial applications fails (obviously), but it also kills the server. I put some code in my pbx_config to check for that string and not let the dialplan reload, but it seems like there should be a better way to handle in in the PJSIP stack or Dial

Hi, Am 16.02.2017 um 14:19 schrieb Annus Fictus: > And Microsip using PJSIP SIP stack :) Sorry (also, for off-topic), based on my latest experience with PJSIP, I'm not sure if this really is a sign of good quality. Maybe it's a problem with the implementation in Asterisk (I haven't tried PJSIP in other software), but after just five minutes of testing I found several bugs

Hi. I my dialplan I have : same = n,Dial(PJSIP/6001,10) same = n,Dial(PJSIP/6002,30) same = n,Hangup() The extension 6002 will not be invited until the called party 6001 hangs up or until 10 seconds if nobody answers the call in 6001. How to call 6001 and immediately call 6002, having 2 phones ringing at same time, but without doing something like this : same =