Re: [asterisk-users] Asterisk 13.10.0 just randomly got pjsip endpoint amnesia.

the 13th AstriCon, September 27-29, 2016 > http://www.asterisk.org/community/astricon-user-conference > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >

Re: [asterisk-users] max concurrent calls with bundled pjproject

ductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- George Joseph Digium, Inc. | Software Developer 445 Jan

Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol

On Wed, Aug 17, 2016 at 1:40 PM, Jonas Kellens <jonas.kell...@telenet.be> wrote: > On 16-08-16 17:45, George Joseph wrote: > > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens <jonas.kell...@telenet.be> > wrote: > >> On 16-08-16 04:38, George Joseph wrote

Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol

On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens <jonas.kell...@telenet.be> wrote: > On 16-08-16 04:38, George Joseph wrote: > > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens <jonas.kell...@telenet.be> > wrote: > >> Hello >> >> using pjp

Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol

ule > 'res_pjsip_pidf_eyebeam_body_supplement.so' could not be loaded. > [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error > loading module 'res_pjsip_rfc3326.so': > /usr/lib64/asterisk/modules/res_pjsip_rfc3326.so: > undefined symbol: ast_sip_session_u

Re: [asterisk-users] PJSIP is Ignored

On Fri, Aug 12, 2016 at 12:53 PM, George Joseph <gjos...@digium.com> wrote: > > On Fri, Aug 12, 2016 at 12:02 PM, Saint Michael <vene...@gmail.com> wrote: > >> ​Asterisk 13.11 rc1 >> >> ./configure LDFLAGS="-z muldefs" --libdir=/usr/lib64

Re: [asterisk-users] PJSIP is Ignored

gt; -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http:/

Re: [asterisk-users] PJSIP not detected

ttp://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users >

Re: [asterisk-users] PJSIP defaults for endpoints when using realtime

tion Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >

Re: [asterisk-users] Asterisk 13 MWI

On Wed, Jul 13, 2016 at 3:44 PM, Carlos Chavez <cur...@telecomabmex.com> wrote: > On 7/12/16 9:27 PM, George Joseph wrote: > > > > On Tue, Jul 12, 2016 at 11:55 AM, Carlos Chavez <cur...@telecomabmex.com> > wrote: > >> I am still a little confus

Re: [asterisk-users] Asterisk 13 MWI

__ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To

Re: [asterisk-users] Registration server with PJSIP

users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org --

Re: [asterisk-users] PJSIP does not qualify contacts after starting Asterisk

== > > Contact: pbx-node-1/sip:myurl.ch:5060771bf6a7d4 Avail > 8.833 > > > > > > > > The extconfig.conf file looks like this: > > > > [settings] > > ps_endpoints => odbc,asterisk > > ps_aut

[asterisk-users] Fedora GLIBC 2.22 warning

: https://gerrit.asterisk.org/#/c/2980/2 13: https://gerrit.asterisk.org/#/c/2979/2 11: https://gerrit.asterisk.org/#/c/2981/2 -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk

Re: [asterisk-users] Fwd: PJSIP subscribe

_ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list &

Re: [asterisk-users] PJSIP subscribe

s for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users &g

[asterisk-users] pjproject 2.5 update notes

.org/repos/changeset/5325 -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] Performance Note: Creating Local channels with ARI

associated with each codec. [1] https://gerrit.asterisk.org/2917 -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- __

Re: [asterisk-users] Asterisk PJSIP Multi-tenant

nt > 1...@sip.domain.com URI > sip:1001@95.250.29.3:50673;rinstance=1af959e7c0059fc4 > Unsolicited NOTIFY and OPTIONS both use the out-of-dialog path so I'm guessing we have an issue there. Open an issue at https://issues.asterisk.org > Regards > El 16/05/2016 a las 02:52, George Jos

Re: [asterisk-users] Asterisk PJSIP Multi-tenant

___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUB

Re: [asterisk-users] Asterisk (PJSIP) registers with sips Contact URI, but why?

com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- George Jo

Re: [asterisk-users] [asterisk-dev] Ubuntu 14 Warning

On Tue, May 3, 2016 at 1:34 AM, Tzafrir Cohen <tzafrir.co...@xorcom.com> wrote: > On Mon, May 02, 2016 at 10:04:54AM -0600, George Joseph wrote: > > This morning, 2 of us noticed that running contrib/scripts/install_prereq > > on a fresh Ubuntu 14 system actually removed c

[asterisk-users] Ubuntu 14 Warning

is unclear. To work around this, you can either run "apt-get update" and "apt-get upgrade" manually, or run "apt-get install libsnmp-dev" manually and say "no" to the first solution and "yes" to the second. Then run install_prereq. -- George Joseph Dig

Re: [asterisk-users] Upgrading 13.7 (external pjproject) to 13.9 (bundled pjproject)

To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _

Re: [asterisk-users] Is set_var allowed with pjsip_wizard.conf ?

On Mon, Apr 25, 2016 at 1:49 PM, Olivier <oza.4...@gmail.com> wrote: > > > 2016-04-25 21:16 GMT+02:00 George Joseph <gjos...@digium.com>: > >> >> >> On Mon, Apr 25, 2016 at 11:11 AM, Olivier <oza.4...@gmail.com> wrote: >> >>

Re: [asterisk-users] Is set_var allowed with pjsip_wizard.conf ?

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

Re: [asterisk-users] Is set_var allowed with pjsip_wizard.conf ?

On Mon, Apr 25, 2016 at 10:00 AM, George Joseph <gjos...@digium.com> wrote: > > > On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4...@gmail.com> wrote: > >> Hello, >> >> I've just discovered PJSIP 's support of set_var setting in pjsip.conf. >> Is

Re: [asterisk-users] Is set_var allowed with pjsip_wizard.conf ?

urs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, A

Re: [asterisk-users] Asterisk 13.8.0 alembic database update fails.

On Fri, Apr 1, 2016 at 3:30 PM, Harley Peters <har...@thepetersclan.com> wrote: > On 04/01/2016 04:26 PM, George Joseph wrote: > >> >> >> On Fri, Apr 1, 2016 at 3:22 PM, George Joseph >> <george.jos...@fairview5.com <mailto:george.jos...@fairview5.com>

Re: [asterisk-users] Asterisk 13.8.0 alembic database update fails.

On Fri, Apr 1, 2016 at 3:22 PM, George Joseph <george.jos...@fairview5.com> wrote: > > > On Fri, Apr 1, 2016 at 3:15 PM, Harley Peters <har...@thepetersclan.com> > wrote: > >> On 04/01/2016 04:06 PM, Joshua Colp wrote: >> >>> Harley Peters wrote

Re: [asterisk-users] Asterisk 13.8.0 alembic database update fails.

On Fri, Apr 1, 2016 at 3:15 PM, Harley Peters wrote: > On 04/01/2016 04:06 PM, Joshua Colp wrote: > >> Harley Peters wrote: >> >>> I get the following error when trying to update date the database via >>> contrib/ast-db-manage/alembic -c config.ini upgrade head. >>>

Re: [asterisk-users] Asterisk 13.8.0 alembic database update fails.

On Fri, Apr 1, 2016 at 3:06 PM, Joshua Colp wrote: > Harley Peters wrote: > >> I get the following error when trying to update date the database via >> contrib/ast-db-manage/alembic -c config.ini upgrade head. >> Every previous update has always worked any idea what is wrong.

Re: [asterisk-users] PJProject Bundled Update

d?​ If it's the former and everything otherwise works, then there's no harm done. Some of them may only apply to Ubuntu. If it's the latter and we're not installing everything needed to build, then open a ticket at issues.asterisk.org. > > > On Thu, Mar 31, 2016 at 12:28 PM, George

Re: [asterisk-users] PJProject Bundled Update

On Thu, Mar 31, 2016 at 1:17 PM, Brian Wilson wrote: > The way I got this build to succeed last night was by using a separate > pjproject, error I get with bundle is the same after applying your patches. > > First patch succeeds. > Second patch fails in 'configure'. > > What

Re: [asterisk-users] PJProject Bundled Update

t; virtual machine - not "older" unless you consider "stable" = "older". > ​Ha! No, it was just that the issues were being reported Debian 4 and 6. :)​ > > On Thu, Mar 31, 2016 at 8:57 AM, George Joseph < > george.jos...@fairview5.com> wrote: >

[asterisk-users] PJProject Bundled Update

As you know, the ability to use a bundled version of pjproject was introduced with Asterisk 13.8.0. More info on the Asterisk Wiki and in this email thread

Re: [asterisk-users] Loss of devices registration (pjsip)

Now do you mind if we get back to the original purpose of this thread before it was hijacked? Dmitriy... See my response further back. :) On Mon, Mar 21, 2016 at 8:42 PM, Pete Mundy wrote: > > Good result! Glad it worked for you :) > > Pete > > > On 22/03/2016, at 9:34

Re: [asterisk-users] Loss of devices registration (pjsip)

On Mon, Mar 21, 2016 at 11:58 AM, Dmitriy Serov wrote: > Good day. > > Asterisk 13.7.2, res_pjsip. > There is a problem of loss of registration of several devices. This > happens not on all devices, but problem devices a lot. > Below is the log of registration of a contact

Re: [asterisk-users] Asterisk now available with bundled pjproject!

On Sun, Mar 13, 2016 at 11:19 AM, Carlos Chavez <cur...@telecomabmex.com> wrote: > On 2016-03-13 02:30, Recursive wrote: > >> On 07.03.2016 20:28, George Joseph wrote: >> >>> The current Asterisk 13 and master git branches have a new feature that >>>

Re: [asterisk-users] Asterisk now available with bundled pjproject!

On Sun, Mar 13, 2016 at 1:30 AM, Recursive <li...@binarus.de> wrote: > On 07.03.2016 20:28, George Joseph wrote: > > The current Asterisk 13 and master git branches have a new feature that > will be included in 13.8.0: The ability to compile and run Asterisk with a > bundled

Re: [asterisk-users] Asterisk now available with bundled pjproject!

On Mon, Mar 7, 2016 at 2:53 PM, Jean-Denis Girard <jd.gir...@sysnux.pf> wrote: > Hi, > > Le 07/03/2016 09:28, George Joseph a écrit : > > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is released. > > I have tried GIT-master-ee5a944M on my Fedora 23 test serv

[asterisk-users] Asterisk now available with bundled pjproject!

The current Asterisk 13 and master git branches have a new feature that will be included in 13.8.0: The ability to compile and run Asterisk with a bundled version of pjproject. ​​ Why would you want to do this? Several reasons: - Predictability: When built with the ​bundled

Re: [asterisk-users] PJSIP signaling question

my iPhone > > On Mar 4, 2016, at 12:01 AM, George Joseph <george.jos...@fairview5.com> > wrote: > > > > On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <kevin.l...@haloprivacy.com> > wrote: > >> >> Thanks George I appreciate the info . Being able to

Re: [asterisk-users] PJSIP signaling question

On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long wrote: > > Thanks George I appreciate the info . Being able to see what codec is in > use for call in progress is very handy sometimes. > > As far as the RTP stats goes, I see there is some info with “rtp” and > “rtcp”

Re: [asterisk-users] PJSIP signaling question

On Tue, Mar 1, 2016 at 5:37 PM, Kevin Long wrote: > > > Interesting, thanks George. I pulled Asterisk 13 from git and the new > pjproject from the SVN and will test accordingly . > ​Yeah, actually you do need Asterisk 13 from git because pjproject deprecated an api

Re: [asterisk-users] Can't send 10 type frames with PJSIP

For right now, you could replace line 712 in channels/chan_pjsip.c with the following and recompile. ast_log(LOG_WARNING, "Can't send %u type frames with PJSIP. Endpoint: %s\n", frame->frametype, ast_sorcery_object_get_id(channel->session->endpoint)); You could also run Wireshark (or capture

Re: [asterisk-users] PJSIP signaling question

On Mon, Feb 29, 2016 at 2:04 PM, Kevin Long wrote: > > > Greetings. > > > I am using the PJSIP driver with TLS transport, and my endpoints are SIP > mobile apps operating in environments that I do not control. > > I would like Asterisk to default to sending INVITES

Re: [asterisk-users] No matching endpoint found for incoming call from SIP trunk

io, and > confirmed that the Asterisk server sends a 401 Unauthorized for the > initiation INVITE). > > Per the pjsip_wizard.conf samples, long ago, I removed pjsip.conf-based > Twilio config and placed it all in pjsip_wizard.conf. > > Thanks, re: wiki, I will be using it heavily, for s

Re: [asterisk-users] No matching endpoint found for incoming call from SIP trunk

On Thu, Feb 18, 2016 at 7:25 PM, Sonny Rajagopalan < sonny.rajagopa...@gmail.com> wrote: > Hello, > > I have an Asterisk 13.6.0 PBX using PJSIP connected to the Twilio gateway. > I am able to make calls outbound through the gateway, but I am not able to > make calls into the PBX from external

Re: [asterisk-users] Typo in http.conf sample file ?

On Thu, Feb 18, 2016 at 4:29 AM, Olivier wrote: > Hello, > > I'm having my first steps with WebRTC. > > I've found this line in http.conf.sample (asterisk 13.7.0): > ;tlsprivatekey=; path to private key file > (*.pem) only. > > > Is it a typo ? > ​Not really. The

Re: [asterisk-users] res_pjsip trunk between Asterisk servers

ystems Administration, 2014 > www.acritelli.com > (845) 283-4117 > > On Mon, Feb 8, 2016 at 10:08 PM, George Joseph < > george.jos...@fairview5.com> wrote: > >> >> >> On Mon, Feb 8, 2016 at 7:16 PM, Anthony Critelli <critel...@gmail.com> >> wrot

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

On Wed, Feb 17, 2016 at 12:13 PM, Sonny Rajagopalan < sonny.rajagopa...@gmail.com> wrote: > Wow. Incredible. That worked. The backslash is important there; I kept > trying with no backslash and followed the instructions in > pjsip_wizard.conf.sample (in configs/samples) and it says we have to say

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

On Wed, Feb 17, 2016 at 8:56 AM, Sonny Rajagopalan < sonny.rajagopa...@gmail.com> wrote: > I made some progress. The first thing I have realized is that it is my > Twilio configuration in pjsip_wizard.conf that was killing me. I have since > removed that entire file from /etc/asterisk and I am

Re: [asterisk-users] res_pjsip trunk between Asterisk servers

On Mon, Feb 8, 2016 at 7:16 PM, Anthony Critelli wrote: > Hi all, > > My goal is to trunk two Asterisk servers together using res_pjsip. I'm > really not familiar with res_pjsip, having only used chan_sip over a year > ago now. So, I apologize in advance if this is an overly

Re: [asterisk-users] sql schema without alembic

On Mon, Feb 8, 2016 at 2:54 AM, Marek Červenka wrote: > Dne 4.2.2016 v 12:17 A J Stiles napsal(a): > >> On Thursday 04 Feb 2016, Marek Červenka wrote: >> >>> hi, >>> >>> is there way to get SQL schema for Asterisk 13.7.0 without alembic? >>> thanks >>> >> Assuming you already

Re: [asterisk-users] PJSIP Stun/ICE

able to tie some kind of stun > support for updating the external media and signaling IP addresses. > > Thanks > > Bryant > > -- > *From*: "George Joseph" <george.jos...@fairview5.com> > *Sent*: Thursday, January 28, 2016 9:12 PM &g

Re: [asterisk-users] PJSIP Stun/ICE

On Thu, Jan 28, 2016 at 6:58 PM, James Cloos wrote: > > "AS" == A J Stiles writes: > > AS> If you are paying for a business-grade Internet connection, you > AS> should get a static IP address -- or a block of them -- as > AS> standard.

Re: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API

On Thu, Jan 28, 2016 at 5:34 PM, Sonny Rajagopalan < sonny.rajagopa...@gmail.com> wrote: > Hi, > > I am using Asterisk 13.6.0 and was wondering if I can programmatically add > users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk > server using API of some sort. > > ​You can use

Re: [asterisk-users] PJSIP TLS sometimes RTP, sometimes no RTP

With the exception of media_encryption_optimistic=yes and ice_support = no, my setup looks like yours and I'm not having any problems with CSipSimple, even with SRTP mode = mandatory. I assume your server has a public IP address and there's no NAT involved on the server side? Oddly enough, I

Re: [asterisk-users] res_pjsip/pjsip_configuration.c: Unable to create ast_sip_contact_status for contact

Hi Dmitriy, I think you're seeing ASTERISK-25675. If you check out the 13.7 branch, you should not see this issue. It's in the 13 branch only and I submitted review 1973 to back out an earlier patch that causes it. The fix should get merged into 13 in the next day or so. You can pull it in

Re: [asterisk-users] PJSIP Bind Issue

On Fri, Dec 18, 2015 at 7:14 AM, Dan Journo wrote: > Hi, > > > > I’ve got a test server with two IPs (one IP is virtual and moves to a > backup server if the first goes down). > > I’m trying out PJSIP and specified the virtual IP for the Bind address of > all

Re: [asterisk-users] hep_queue_cb: Error [1] while sending packet to HEPv3 server: Operation not permitted

Are you actually using a HEP server to capture sip traffic? If not, just disable hep in hep.conf or add noload statements in modules.conf for res_hep, res_hep_pjsip and res_hep_rtcp. On Sun, Dec 6, 2015 at 2:30 AM, Julien Sansonnens wrote: > Hello, > > I upgraded to

Re: [asterisk-users] asterisk 13 systemd

I deploy on Fedora exclusively (not using the Fedora packages) and this is the service file I use. It's different from the packaged version. [Unit] Description=Asterisk PBX and telephony daemon. Requires=network-online.target After=network-online.target [Service] Type=forking

Re: [asterisk-users] pjsip show xxxx like endpoint?

On Sun, Oct 18, 2015 at 5:07 PM, Matthew Jordan <mjor...@digium.com> wrote: > On Sun, Oct 18, 2015 at 12:39 PM, George Joseph > <george.jos...@fairview5.com> wrote: > > Did you open a Jira issue for this yet? I can actually work on this this > > week. >

Re: [asterisk-users] pjsip show xxxx like endpoint?

Did you open a Jira issue for this yet? I can actually work on this this week. On Fri, Oct 16, 2015 at 9:44 AM, George Joseph <george.jos...@fairview5.com> wrote: > On Fri, Oct 16, 2015 at 4:00 AM, Bryant Zimmerman <brya...@zktech.com> > wrote: > >> Is there a way

Re: [asterisk-users] pjsip show xxxx like endpoint?

On Fri, Oct 16, 2015 at 4:00 AM, Bryant Zimmerman wrote: > Is there a way to limit the items returned by pjsip show [type] using like > There isn't but there could be. Open an issue and reply with the id and I'll take a look. > chan_sip allowed for sip show peers like

Re: [asterisk-users] CLI for pjsip registrations in Asterisk v13.1.0?

On Sun, Mar 22, 2015 at 7:59 AM, Sonny Rajagopalan sonny.rajagopa...@gmail.com wrote: Hello, I am trying to force a registration and unregistration with my SIP trunks, but I see pjsip send unregister, but no register. I.e., I am looking for pjsip send register. Is there any such command?

Re: [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan sonny.rajagopa...@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

Re: [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

. at hangup on caller (sonny): == Spawn extension (from-internal, 912025551212, 2) exited non-zero on 'PJSIP/sonny-0031' On Sun, Mar 15, 2015 at 3:25 PM, George Joseph george.jos...@fairview5.com wrote: On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan sonny.rajagopa...@gmail.com wrote

Re: [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

' and try it, does the Dial get executed? On Sun, Mar 15, 2015 at 12:19 PM, George Joseph george.jos...@fairview5.com wrote: On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan sonny.rajagopa...@gmail.com wrote: I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic

Re: [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

On Sun, Mar 15, 2015 at 5:56 PM, Sonny Rajagopalan sonny.rajagopa...@gmail.com wrote: George, I have the detailed log below. (Resending after trimming the email to 40KB.) The sequence below just repeats ad-nauseam. Is this a SIP trunk issue? Thanks! I don't see anything obvious. The

Re: [asterisk-users] Asterisk 13 - sorcery realtime for pjsip publish objects

On Thu, Feb 19, 2015 at 9:15 AM, Joshua Colp jc...@digium.com wrote: Matt Hoskins wrote: Good Morning, After further investigation, I found that the res_pjsip_publish_asterisk module does not use the realtime sorcery wizard, but instead only reads from the configuration files. I've been

Re: [asterisk-users] Confused by concepts behind pjsip: endpoint, aor, contact

On Sun, Jan 4, 2015 at 3:29 PM, Antonio Gómez Soto antonio.gomez.s...@gmail.com wrote: Hello, I am slightly confused by the difference between chan_sip and pjsip. Especially the new (to me) objects aor and contact. I am having trouble mapping them to the typical SIP configuration settings

Re: [asterisk-users] PJSIP configuration question

in the packet. If it is, all bets are off.* Second. Can you try making a call from a phone instead of from an AMI originate? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *George Joseph *Sent:* Monday, December 15, 2014 11

Re: [asterisk-users] PJSIP configuration question

On Tue, Dec 16, 2014 at 11:45 AM, Dan Cropp d...@amtelco.com wrote: Here's an update... My network admin would not turn off the ALG because it would cause several other problems to other phone systems we have. He looked at the sip trace. What he found is the PJSIP trace showed a different

Re: [asterisk-users] PJSIP configuration question

On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp d...@amtelco.com wrote: Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0. Same problem is happening with both of them. Could this be caused by PJPROJECT 2.3? Anyone have any suggestions for what I can try? My boss

Re: [asterisk-users] PJSIP configuration question

*From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *George Joseph *Sent:* Monday, December 15, 2014 3:40 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] PJSIP configuration question

Re: [asterisk-users] PJSIP configuration question

control the call through AMI to perform the work we need. And it's outbound calls that aren't working right? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *George Joseph *Sent:* Monday, December 15, 2014 4:42 PM

Re: [asterisk-users] PJSIP configuration question

Ok Dan, try this... I was able to get this to work behind a NAT and with ip address authentication. [global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol = udp *local_net=yourlocalnet I.E. 10.10.10.10/24 http://10.10.10.10/24external_media_address=your

Re: [asterisk-users] PJSIP configuration question

the 'fromuser' from sip.conf and user is from_user. Have a great day! Da *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *George Joseph *Sent:* Monday, December 15, 2014 7:27 PM *To:* Asterisk Users Mailing List - Non

Re: [asterisk-users] PJSIP configuration question

with something other than a sub-account username. [HVout] type=friend dtmfmode=auto host=64.2.142.93 disallow=all allow=ulaw canreinvite=no trustrpid=yes sendrpid=yes nat=yes context=TestApp On Dec 15, 2014, at 9:32 PM, George Joseph george.jos...@fairview5.com wrote

Re: [asterisk-users] PJSIP configuration question

On Wed, Dec 10, 2014 at 1:27 PM, Dan Cropp d...@amtelco.com wrote: Not sure why, but Vitelity changed the settings to IP based authentication on me. Here's the new sip.conf settings they sent me. type=friend dtmfmode=auto host=64.2.142.93 allow=all nat=yes canreinvite=no trustrpid=yes

Re: [asterisk-users] PJSIP configuration question

On Wed, Dec 10, 2014 at 2:03 PM, Dan Cropp d...@amtelco.com wrote: Thanks George. That was the ip address I was given. Unfortunately, my contact at Vitelity is gone for the day so I can’t verify it with him. I added the qualify_frequency as you suggested and it does appear that I have

Re: [asterisk-users] Get the status of a PJSIP endpoint?

On Tue, Nov 18, 2014 at 10:55 AM, George Joseph george.jos...@fairview5.com wrote: On Mon, Nov 17, 2014 at 1:52 PM, John Kiniston johnkinis...@gmail.com wrote: Is there an equivalent to ${SIPPEER(${peer},status)} for PJSIP? I could swear there was a non-obvious way to get state but I

Re: [asterisk-users] pjsip phoneprov realtime?

On Thu, Nov 13, 2014 at 12:11 PM, John Kiniston johnkinis...@gmail.com wrote: Howdy, Is there a way to use realtime with phoneprov.com and pjsip? Not yet. I forgot that bit in the initial version of the res_pjsip_phoneprov_provider module. I have a patch ready but it's tangled up in other

Re: [asterisk-users] pjsip phoneprov realtime?

On Thu, Nov 13, 2014 at 4:33 PM, George Joseph george.jos...@fairview5.com wrote: On Thu, Nov 13, 2014 at 12:11 PM, John Kiniston johnkinis...@gmail.com wrote: Howdy, Is there a way to use realtime with phoneprov.com and pjsip? Not yet. I forgot that bit in the initial version

Re: [asterisk-users] pjsip phoneprov realtime?

that pulls the data out of my database but it still requires me to do a reload every time I make a change, Full realtime support will be nice. Thank you for the quick response. No worries. We'll use your issue and I'll scrap mine. On Thu, Nov 13, 2014 at 5:16 PM, George Joseph

Re: [asterisk-users] Function to get mailbox for a PJSIP Endpoint?

On Fri, Nov 7, 2014 at 6:20 AM, Joshua Colp jc...@digium.com wrote: John Kiniston wrote: Here's my config, I am configuring the mailboxes as you see below in the aor. It looked like that was the recommended place to configure it? There's no recommendation one way or the other really - it

Re: [asterisk-users] unidata incom ICW-1000G - On asterisk

On Fri, Sep 5, 2014 at 7:26 AM, Bryant Zimmerman brya...@zktech.com wrote: I am trying to use an ICW-1000G wireless handset connected to an asterisk server remotely The user is working from an offsite location and it appears that the device is not sending out keep-alives or stun. The

Re: [asterisk-users] DPMA: No provider found for label CustomPresence

Make sure the func_presencestate.so module is being loaded. Did you compile Asterisk yourself or are you using a pre-built from a distro? On Thu, Aug 21, 2014 at 5:34 PM, Mitch Claborn mitch...@claborn.net wrote: Asterisk 12.5.0 DPMA 12.0_2.0.0 Ubuntu 12.04 64 bit WARNING[5797]:

Re: [asterisk-users] DPMA: No provider found for label CustomPresence

On Thu, Aug 21, 2014 at 6:15 PM, Mitch Claborn mitch...@claborn.net wrote: It appears to be loaded. This is a fresh build of Asterisk 12.5 from source. *CLI module show like func_presencestate.so Module Description Use Count Status func_presencestate.so

Re: [asterisk-users] Asterisk 12 and DPMA

On Fri, Aug 1, 2014 at 9:32 AM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: I read somewhere that DPMA is not supported for Asterisk 12. Can anyone confirm or deny that? If not supported yet, will it be? If so, when? Per this link:

Re: [asterisk-users] PJSIP endpoint max-calls limit missing

On Fri, Jun 27, 2014 at 9:28 PM, CDR vene...@gmail.com wrote: 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

Re: [asterisk-users] Using macros in extensions.lua?

On Fri, Jun 6, 2014 at 1:48 AM, Dennis Guse dennis.g...@alumni.tu-berlin.de wrote: Hi, I have defined a dialplan in lua and now would like to use dial with the macro M to implement some logic, when the callee-channel gets created. Working old style would be (extensions.conf) [default]

Re: [asterisk-users] Problem building Asterisk-12.2.0

On Sat, Apr 26, 2014 at 3:37 PM, Richard Kenner ken...@gnat.com wrote: When I run ./configure, it aborts with: checking for uuid_generate_random in -luuid... no checking for uuid_generate_random in -le2fs-uuid... no checking for uuid_generate_random... no configure: error: *** uuid support

Re: [asterisk-users] How does extensions.lua compares to extensions.conf ?

On Thu, Feb 13, 2014 at 8:39 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Thu, Feb 13, 2014 at 1:04 AM, George Joseph george.jos...@fairview5.com wrote: On Wed, Feb 12, 2014 at 6:26 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Feb 12, 2014 at 12:50 PM

Re: [asterisk-users] How does extensions.lua compares to extensions.conf ?

On Wed, Feb 12, 2014 at 6:26 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Feb 12, 2014 at 12:50 PM, Olivier oza.4...@gmail.com wrote: Hello, How does extensions.lua compares to extensions.conf or extensions.ael on stability, performance and features ? Would you

Re: [asterisk-users] grp_lock error when compiling against pjproject

On Tue, Jan 28, 2014 at 6:29 AM, Matthew Jordan mjor...@digium.com wrote: On Tue, Jan 28, 2014 at 2:40 AM, Ira i...@extrasensory.com wrote: Hello Matthew, Monday, January 27, 2014, 1:49:44 PM, you wrote: Do you have the exact error message that pjproject gave when you ran into this

[asterisk-users] DPMA for Asterisk 12?

Looks like res_digium_phone will need some work for Asterisk 12... WARNING[9372]: loader.c:561 load_dynamic_module: Error loading module 'res_digium_phone.so': /usr/lib64/asterisk/modules/res_digium_phone.so: undefined symbol: __ao2_container_alloc --

Re: [asterisk-users] How do I remotely force an *unconfigured* Digium DPMA phone to re-query the network for the DPMA server?

On Fri, Sep 6, 2013 at 10:41 AM, Alex Villací­s Lasso a_villa...@palosanto.com wrote: Consider the following scenario: 1) One or more Digium DPMA phones are plugged into the network. I know their IP addresses and MACs. 2) The Asterisk I want to use as the telephony server starts without the

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