similar to: PJSIP - State of the art

Hello, I'm trying to use presence with PJSIP and I have a "issue". I created correctly hint priorities like: exten => 1000,hint,PJSIP/1000 exten => 1001,hint,PJSIP/1001 Extension 1000 can subscribe extension 1001 y vice-versa. The problem is when the extension 1000 make or receive a call. In the softphone where the extension is present on buddy list, the extension appear

Hello, with qualify_frequency=0 I can't receive calls from others endpoints. Other strange think is if I set mailboxes parameter on the console, when the endpoint registering, i can see: ERROR[2208]: res_pjsip.c:2946 create_out_of_dialog_request: Unable to create outbound NOTIFY request to endpoint 1001 at sip.domain.com WARNING[2208]: res_pjsip_mwi.c:379

Hello, I think Device State for Agents don't work correctly My configuration: agents.conf [general] [agent](!) autologoff=15 ackcall=no acceptdtmf=# wrapuptime=5000 musiconhold=default recordagentcalls=no custom_beep=beep [2000](agent) fullname=Fulano [2001](agent) fullname=Zutano [2002](agent) fullname=Mengano queue.conf (Agents Related) member => Agent/2000 member =>

Hello List, following this thread: http://asterisk-users.digium.narkive.com/ulR5hd1M/same-pjsip-username-with-differents-domains I tried to configure on the pjsip.conf the same endpoint with different domains like: [1000 at sip.domain.com] type=endpoint [1000 at sip1.domain.com] type=endpoint I can register the two 1000 endpoints using different domain but on the Asterisk console:

Yes, I have tried that too (i.e, exten => +17775551212,1,Log(WARNING, TWILIO)). It does not work and NO error message in CLI. I have also tried http://orourketech.com/elastix-plus-sign-caller-id-messing-things/ since I first emailed this group, but that does not seem to work either. Here is my log: [Dec 2 15:09:28] NOTICE[26464]: res_pjsip_session.c:1920 new_invite: Call from

Humm whats is the diferent? Em 16/12/2015 14:19, "Annus Fictus" <annusfictus at gmail.com> escreveu: > CDR-STATS is for reporting. > > A2Billing is for billing... > > Regards > > El 16/12/2015 a las 11:15, Vitor Mazuco escribi?: > >> Hi everyone! >> >> I'm trying to install CDR-Stats (cdr-stats.org), but it very difficult. >>

Hello, I'd like to do some tests with the StatsD dialplan application but on the last version of Asterisk 13 (13.7.0) I can't find this application. New Features made in this release: ----------------------------------- * ASTERISK-25419 - Dialplan Application for Integration of StatsD (Reported by Ashley Sanders) res_statsd module are correctly compiled y loaded. Any hint?

Hello list, before to lost my time, I'd like know if someone have a WebRTC working configuration on Asterisk 13.11.0 SIP or PJSIP channel. Thank you Regards

Hi, Yes, we're implementing the dialplan in realtime too. Here the contents of sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints aor=realtime,ps_aors contact=realtime,ps_contacts [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips Cheers, Francisco. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf

Hi everyone! I'm trying to install CDR-Stats (cdr-stats.org), but it very difficult. Is there others optins for billing? Thanks

Hello, I am running Asterisk 13.6.0 in an AWS instance, and I set it up with Twilio SIP trunk using pjsip_wizard.conf (nice feature!). I see that the calls actually "reach" the PBX, but for some reason, they are not caught by any of my extensions context. Here's what I observe when I test this from a non-PBX connected E164 number (a landline), say 555-666-1212. My Twilio number is

Hi all, (sending this again from the correct address) I'm running Asterisk 13.8.0 (I need to check if that happens with 13.9.1 too when I have the time to build it) with PJSIP realtime config. I've defined several aors in the table ps_aors, like this (real url replaced by myurl): *CLI> pjsip show aor pbx-node-1 Aor: <Aor..............................................>

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

Has anyone successfully used Mysql realtime PJSIP with Asterisk 13.11? I have tried 13.11, 13.11.1 and 13.11.2 but I always get the following error now: Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'qualify_timeout' cannot be type 'int(10)' (need char) [Sep 12 14:42:35] WARNING[24498]:

Microsip (Windows) is free and small. 2.5Mb download, 10Mb RAM usage, does everything I need and configuring TLS is a doddle. http://www.microsip.org/ On 16 February 2017 at 13:04, Max Grobecker <max.grobecker at ml.grobecker.info> wrote: > Hello, > > I'm a big fan of PhonerLite. > It's more poplar in Germany, but also available in English language. > This client

hi, after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i have problem with segfault (centos 6) Program terminated with signal 11, Segmentation fault. #0 0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8, pkt_size=132, options=1, token=0xafecc2bc, parsed_len=0x0, src_addr=0xb0e47a20, src_addr_len=16) at ../src/pjnath/stun_session.c:1287 1287

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, Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plateform and I don't find any. Thanks by advance.

Hi all, I have problems with rxfax application. It seems to be ok but I don't receive the fax in my directory. My extension.conf is as follow: [macro-fax] exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/incoming/${UNIQUEID}.tif) exten => s,2,rxfax(${FAXFILE}) [fax] exten => 100,1,macro(fax) [reception] exten =>s,1,Answer() exten =>s,2,Background(00) exten

Hi, I am working on Trixbox. I want to create my own dial() function (named specificdial()) and I want to know how I can create a module and integrate the module in the trixbox plateform. thanks a lot Mickael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091006/b69d4103/attachment.htm