similar to: PJSIP Sends BYE with Wrong IP

On Thu, Apr 2, 2015 at 10:43 AM, Rusty Newton <rnewton at digium.com> wrote: > On Wed, Apr 1, 2015 at 9:08 AM, Trey Hilyard <kctrey at gmail.com> wrote: > >> Hello - >> >> I am trying to decide if I have stumbled across a bug in PJSIP or I am >> just missing something. My Asterisk has two interfaces, an "internal" eth0 >> and an

On Wed, Apr 1, 2015 at 9:08 AM, Trey Hilyard <kctrey at gmail.com> wrote: > Hello - > > I am trying to decide if I have stumbled across a bug in PJSIP or I am > just missing something. My Asterisk has two interfaces, an "internal" eth0 > and an "external" eth1. In pjsip.conf, I define the following transports: > > [trusted] > type=transport >

I just realized that you are asking about dynamic AORs, not static Contacts in an AOR. That may be the difference. I have never actually tried giving a dynamic AOR a different name. And you wouldn't want more than one dynamic AOR, you'd just use an AOR that allowed more than 1 contact. On Wed, Apr 1, 2015 at 2:59 PM Trey Hilyard <kctrey at gmail.com> wrote: > I don't know

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

Is there any "easy" way to add a custom subscribe handler? I have a set of users with Polycom phones that attempt to Events that Asterisk/PJSIP doesn't recognize, "call-info" and "as-feature-event". It just generates a warning, but it got me wondering if I could add my own handlers for those that didn't actually do anything but simply responded with a 200 OK.

I am running asterisk 13.1.0 In pjsip.conf, the endpoint section has an aors and an auth field. I can name the auth field anything I want. The key is to set the auth=field accordingly. However, when I try this with the aors field, it never works. It seems I have to name the aors=field to match the name of the endpoint section. Is this correct? Would there ever be a need for multiple aors to

Thanks for the reply Trey, should of said I'm using chan_sip. Regards Mick On 16 Feb 2016 18:03, "Trey Hilyard" <kctrey at gmail.com> wrote: > Are you using res_pjsip or chan_sip? > > For PJSIP, it's as easy as passing the parameters to the Dial. For example: > Dial(PJSIP/${ARG1}\;phone-context=mydomain.com at pjsippeer,60) > > I am pretty sure it was

Hi all, I am currently using asterisk 11, and I am trying to figure out how to set the uri parameter telephone-context. I need to set it for outbound calls for a specific carrier when making emergency calls and don't seem able to find the option to set it. Regards Impy aka Mick -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <admin at tootai.net> wrote: > Le 18/03/2016 16:20, Trey Hilyard a ?crit : > > I am trying to set up my Asterisk server so that it will recognize an > > incoming call to the Asterisk's own Location Routing Number (LRN), > > validating the "rn" in the INVITE and then using the Called Number from >

i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording SIP/101-0000010d -- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d > 0x2b393cfc2610 -- Probation passed

Hey all I am trying to configure all my pjsip transports form a database table. The issue I am running into is that pjsip is auto binding to 0.0.0.0:5060 before it reads my list of transports from the database. This means that my entries for port 5060 are already bound and the settings in the database are not loaded. When loading the transport form the .conf file it works as expected

Joshua So once a transport is pulled from the transports table in realtime during asterisk startup it can't get any updates? Can a new transport be added to the table and the associated endpoints be updated to use the new transport, or are transport types only read at startup across the board? Thanks Bryant ---------------------------------------- From: "Joshua

Bryant, I have the same problem with dynamic public IPs and PJSIP. What is your idea to solve the problem? My suggestion would be to write a script that monitors the change, pjsip.transports.conf updated and Asterisk restarts? Daniel > Am 26.01.2016 um 14:21 schrieb Joshua Colp <jcolp at digium.com>: > > Bryant Zimmerman wrote: >> Joshua >> So once a transport is

On Wednesday 17 Feb 2016, imperium broadcast wrote: > I kinda have it working with chan_sip. > > Dial(SIP/+${EXTEN}\;phone-context=+44 at 10.10.10.10;user=phone) > But it doesn't include the user=phone at the end when dialling out. > > "To: <sip:+4499999999999;phone-context=+44 at 10.10.10.10>". > > even adding > usereqphone=yes > to the

>>>>> "JC" == Joshua Colp <jcolp at digium.com> writes: JC> This stems from PJSIP not being dynamic with transports (it JC> doesn't like its environment changed to that degree while JC> in use). I'm afraid if your IP changes you'd have to restart JC> Asterisk when you are using PJSIP. Wow. I say this having voted for pjsip over the listed

On 2/15/16 12:50 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> Is it possible to use serveral protocols for a single transport section >> in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you >> cound use webrtc along with your phones but if I try: >> >> [transport-udp] >> type=transport >> protocol=udp,ws,wss >> bind=0.0.0.0 >

Hi, The asterisk wiki page says: "Sorcery.conf allows you to try to configure other PJSIP objects such as transport using realtime and it currently won't stop you from doing so. However, some of these object types should not be used with realtime and this can lead to errant behavior." Which objects and is this still true in 1.13.1 ? Thanks, Antonio. PS:

Joshua Since there is no automated way currently built in to update the external signaling and media address information. Does the realtime pjsip support having the transport contexts section being pulled from a database table? I was thinking a cron script updating the table and forcing a reload each time an IP address changed might a workable solution. Thanks Bryant

Hi Gang Server, two interfaces, routing to two different networks. Two transports defined, each bound to the corresponding ip assigned to the interface. But still, especially when an 183 message is sent, the Contact header does contain the wrong IP Address. Is this a known issue 13.18.3? Or is there a way to make absolutely sure the IP addresses within the Contact header is corresponding to

>>>>> "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