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...namic 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 why you have issues using different names. I have multiple > AORs assigned to a single endpoint and it works fine. I have to admit that > my AORs do contain the endpoint name, though. For example, for endpoint > "myswitch" I have two...

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 "external" eth1. In pjsip.conf, I define the...

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 easy in chan_sip, too. If you are using...

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 protocol=udp bind=10.xx.yy.zz:5060 [untrusted] type=transport protocol=udp bind=12.4.aa.bb:5060 My internal endpoints use

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

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: &...

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:

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

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

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 >

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

...ed - setting RTP source address to > 217.195.xx.xx:46346 > -- SIP/FD-0000010e answered SIP/101-0000010d > > 0x1d08efa0 -- Probation passed - setting RTP source address to > 217.195.xx.xx:46346 > thanks and regards. > > 2015-03-20 17:15 GMT+00:00 Trey Hilyard <kctrey at gmail.com>: > >> I am making some assumptions, but assuming the 217.195.xx.xxx is your >> provider, you are getting this back from them: >> >> "Got SIP response 556 "No address found" back from 217.195.xx.xxx:5060" >> >> Are you sure...

hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xxxxxx at

So you are saying that it resolved the issue to activate voicemail on the device that sits past your trunk provider? That confuses me a little, but if your calls are working, that's great news. On Fri, Mar 20, 2015 at 1:44 PM Salaheddine Elharit < salah.elharit200 at gmail.com> wrote: > i noticed that when i active the voicemail in the IP-phone where the > number 0033149xxxxxx is

I don't know why you have issues using different names. I have multiple AORs assigned to a single endpoint and it works fine. I have to admit that my AORs do contain the endpoint name, though. For example, for endpoint "myswitch" I have two AORs, "myswitch_1" and "myswitch_2", and I assign them to the endpoint with aors=myswitch_1,myswitch_2. When you say that

I would especially look at the CHANUNAVAIL dial status Since it sounds like you are probably qualifying your IAX trunk, that status will be the quickest way to overflow from IAX to TDM. On Sat, May 30, 2015, 11:35 PM Ashwin Surendran < Ashwin.Surendran at now-health.com> wrote: > Hi Matt, > > > > I was a bit concerned on the delay if there might be any when my iax link

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 the INVITE as the extension in the dialplan. The INVITE R-URI looks like: INVITE sip:+19135041291;rn=+19136630000;npdi at 12.4.240.200:5060;user=phone;transport=udp SIP/2.0

I am making some assumptions, but assuming the 217.195.xx.xxx is your provider, you are getting this back from them: "Got SIP response 556 "No address found" back from 217.195.xx.xxx:5060" Are you sure that "0033149xxxxxx" is the format the provider is expecting? You might try enabling SIP debug on the 217.195.xx.xx IP and seeing what the INVITE looks like, but