similar to: CALLERID on pjsip doesn't work?

On 1 July 2016 at 17:41, Joshua Colp <jcolp at digium.com> wrote: > > >> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) >> same => n,Dial(PJSIP/phone123, 30) >> > > Your exten line has no priority, is that how it is in your dialplan? > Actually no, I stole that line from an earlier email to this list. Mine has a priority.

On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>> Just a guess (without knowing about your network), but are the two ends >>> points on public networks and visible to one another? If not the reinvite >>> may be passing an internal (nat'ed)

Hi folks, I was using the following featuremap: blindxfer => *1 disconnect => *9 atxfer => *2 parkcall => *7 automixmon => *0 and everything worked. Then the need arouse to use some features like automixmon during a conference, but MeetMet has the * key bound to the (admin) menu. Thus, in order to enable features like automon and transfers even during a conference, I

Thank you Joshua. I tried setting the from_domain for the endpoint, but it still sends the internal ip address for the INVITE's From field [acl1] type = acl deny = 0.0.0.0/0.0.0.0 permit = variousaddress permit = bluipaddress [transport1] type = transport bind = 0.0.0.0 protocol = udp [BLUIPIN] type = aor remove_existing = yes contact = sip:bluipaddress [auth7] type = auth username =

I'm doing an upgrade from Asterisk 11 to 13. I'm following the guide at https://wiki.asterisk.org/wiki/display/ast/setting+up+PJSIP+REaltime to setup realtime, as I use realtime on Asterisk 11 too. I'm getting the following error when trying to connect the peer to the server. Help? :) Thanks, Travis [Jun 15 16:20:03] NOTICE[5116] res_odbc.c: res_odbc: Connected to laf [laf] [Jun

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:

Hi Joshua > The chan_pjsip module doesn't prevent that. You'd need to provide the > full SUBSCRIBE now that it is actually finding the endpoint and coming > in. Ok, let's see if we can solve the mystery.. pjsip.conf [endpt-home](!) type=endpoint disallow=all allow=g722 allow=alaw allow=gsm ice_support=yes context=from-home allow_subscribe=yes

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 Mon, Apr 25, 2016 at 10:00 AM, George Joseph <gjoseph at digium.com> wrote: > > > On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4h07 at gmail.com> wrote: > >> Hello, >> >> I've just discovered PJSIP 's support of set_var setting in pjsip.conf. >> Is this setting also supported in pjsip_wizard.conf ? >> On a fresh 13.8.2, it

Hi list! I'm trying to configure Asterisk to record incoming calls, if the called press *3. I added in features.conf: automixmon => *3 then, in my dialplan: exten => 1,n,Dial(SIP/00493511111111,20,RcxX) Well, if I **CALL** a number I'm able to record the call, but if I'll be called, and press *3 nothing happens... In the console I can't see anything, too. Could you

Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any idea on how I can stop the MixMonitor application while it is running? [featuremap] automixmon =>

Using pjsip 2.7.2 on Asterisk 15.5 Really struggling to make sense of translating these old 1.8 SIP instructions into a neat pjsip_wizard conf suitable for 2018 http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 In pjsip_wizard.conf, I have the following, which seems to get me registered, and it responds to an incoming call, but I always get this: [Jul 28 18:32:29]

Hello List I am in the progress of migrating from chan_sip to pjsip. I fear I have missed something on how hints need to be specified for pjsip. For chan_sip I have configured sip.conf subscribecontext = localuser and in the dialplan I set: [localuser] exten => 11,hint,SIP/11 Now if a phone subscribes to '11' this works. Now I try to get the same working for pjsip. I understood

Hi all, So I have been reading through the docs available online and the different threads on this list, but I cannot seem to get this phone to work. I have configured the OS79XX.TXT and SIP/SEP*.cnf files (see attached), when I configure the phone to point to my tftp server and reboot it I get this message: Connection received from 10.6.0.224 on port 50608 [09/02 12:16:11.750] Read request

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..............................................>

On Mon, Apr 25, 2016 at 11:11 AM, Olivier <oza.4h07 at gmail.com> wrote: > > > > 2016-04-25 18:14 GMT+02:00 George Joseph <gjoseph at digium.com>: > >> >> >> On Mon, Apr 25, 2016 at 10:00 AM, George Joseph <gjoseph at digium.com> >> wrote: >> >>> >>> >>> On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4h07

Hello, I've just discovered PJSIP 's support of set_var setting in pjsip.conf. Is this setting also supported in pjsip_wizard.conf ? On a fresh 13.8.2, it doesn't seem but I may have missed somthing. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

On Fri, Sep 15, 2017, at 10:37 AM, Bryant Zimmerman wrote: > Joshua > > That is the interesting part of it. We took our configs and database > tables from our working 13.12.2 deployments and tried to use them with > our > new 13.17.1 deployments and we are having issues where the tables are not > working. On the new server asterisk keeps saying it can't find the

Hello everyone. I allow myself to submit a problem that I can not solve with my VOIP provider Orange in France [2023-06-08 13:19:03] ERROR[185091]: res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error configuring endpoint 'Biv_Sortie' - 'from_user' field contains invalid character '@' [2023-06-08 13:19:03] ERROR[185091]: config_options.c:798 aco_process_var: