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On Fri, Apr 1, 2016 at 3:22 PM, George Joseph <george.joseph at fairview5.com> wrote: > > > On Fri, Apr 1, 2016 at 3:15 PM, Harley Peters <harley at thepetersclan.com> > wrote: > >> On 04/01/2016 04:06 PM, Joshua Colp wrote: >> >>> Harley Peters wrote: >>> >>>> I get the following error when trying to upd...

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.joseph at fairview5.com> wrote: > On Fri, Oct 16, 2015 at 4:00 AM, Bryant Zimmerman <BryantZ at zktech.com> > 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 &...

...n new stack -- Called PJSIP/12025551212 at sonnyGW1 the number 202-555-1212 does not ring. at hangup on caller (sonny): == Spawn extension (from-internal, 912025551212, 2) exited non-zero on 'PJSIP/sonny-00000031' On Sun, Mar 15, 2015 at 3:25 PM, George Joseph <george.joseph at fairview5.com> wrote: > 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 >...

On Thu, Mar 31, 2016 at 1:17 PM, Brian Wilson <brian at wildsong.biz> 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 I did -- I downloaded your diffs, unpacked a fresh copy of the

.... Vitelity chan_sip settings I have working, do not have a fromuser. sip.conf settings... [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.joseph at fairview5.com<mailto:george.joseph at fairview5.com>> wrote: On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote: I am not sure if I entered the correct settings for the transport information. For the local_net, I entered my local ip addres...

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. On Sun, Mar 15, 2015 at 12:19 PM, George Joseph <george.joseph at fairview5.com > wrote: > > > 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, a...

On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at amtelco.com> wrote: > > I am not sure if I entered the correct settings for the transport > information. > > For the local_net, I entered my local ip address, but no mask. I will > check with the network admin so he can verify the settings I entered. > > > You need the network and mask. For example if the ip

...01 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 sure ;-) On Thu, Feb 18, 2016 at 9:56 PM, George Joseph <george.joseph at fairview5.com> wrote: > > > On Thu, Feb 18, 2016 at 7:25 PM, Sonny Rajagopalan < > sonny.rajagopalan at 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 t...

...fy anything, it just has to be populated 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.joseph at fairview5.com<mailto:george.joseph at fairview5.com>> wrote: On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote: I am not sure if I entered the correct settings for the transport information. For the local_net, I entered my local ip address...

...iend > > 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.joseph at fairview5.com> > wrote: > > > > On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at amtelco.com> wrote: >> >> I am not sure if I entered the correct settings for the transport >> information. >> >> For the local_net, I entered my local ip address, but no m...

...nd look at the actual outbound INVITE and any response. > > at hangup on caller (sonny): > > == Spawn extension (from-internal, 912025551212, 2) exited non-zero on > 'PJSIP/sonny-00000031' > > On Sun, Mar 15, 2015 at 3:25 PM, George Joseph < > george.joseph at fairview5.com> wrote: > >> 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...

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 <https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-bundled> and in this email thread <http://lists.digium.com/pipermail/asterisk-users/2016-March/288685.html>. Since then I've fixed a few

On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <kevin.long at haloprivacy.com> 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? commands which can be useful for troubleshooting. A running tally of > #

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://10.10.10.10/24>>external_media_address=<your public ip address>external_signaling_address=<your public address>*

Is there a way to limit the items returned by pjsip show [type] using like chan_sip allowed for sip show peers like xxxx, but I can't seem to figure out how to lookup or limit my returns with pjsip Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

>>>>> "AS" == A J Stiles <asterisk_list at earthshod.co.uk> 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. Maybe you need to change your ISP? In some places (including here) static ip is not affordable. -JimC -- James Cloos <cloos at

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. >> Every previous update has always worked any idea what is wrong. >> >> OS=Debian Jessie, fully up to date. > > What version of Alembic is installed and how did

...r 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 15, 2015 at 12:19 PM, George Joseph < > george.joseph at fairview5.com> wrote: > >> >> >> 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...

...on a barebones Debian 8.x > 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.joseph at fairview5.com> wrote: > >> >> 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 >> <https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandIn...