Re: [asterisk-users] outbound calls

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 0033149xx is the format the provider is expecting? You might try enabling SIP debug on

Re: [asterisk-users] outbound calls

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.elharit...@gmail.com wrote: i noticed that

[asterisk-users] PJSIP Sends BYE with Wrong IP

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]

[asterisk-users] Dial to PJSIP Channel with Typo PJSIP// Causes Asterisk Shutdown

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//...), 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

Re: [asterisk-users] PJSIP Sends BYE with Wrong IP

On Thu, Apr 2, 2015 at 10:43 AM, Rusty Newton rnew...@digium.com wrote: On Wed, Apr 1, 2015 at 9:08 AM, Trey Hilyard kct...@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

Re: [asterisk-users] PJSIP Endpoint AOR question

, Apr 1, 2015 at 2:59 PM Trey Hilyard kct...@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

Re: [asterisk-users] PJSIP Endpoint AOR question

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

Re: [asterisk-users] How to use TRUNK only if IAX fails?

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.surend...@now-health.com wrote: Hi Matt, I was a

Re: [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP

If you are using PJSIP, you should be able to define a different transport for each source IP that you want to use and simply tie the endpoint to your provider to the appropriate transport. Obviously, you'd need to define the IP on the interface as well. You could use the same interface with

[asterisk-users] PJSIP Returning 421 Extension Required

I am turning up a PJSIP Endpoint and am having problems when they send an INVITE to my server. Asterisk is returning a 421 Extenstion Required. Since "extension" means different things in the SIP stack versus Asterisk, I don't know what it is complaining about. I have attached the trace below.

Re: [asterisk-users] Unexpected termination of the call when pick up (res_pjsip)

How are you initiating the call out to that server? Are you dialing from an internal phone or doing it from the CLI? It looks like it is from an internal extension, if I were guessing, but that side of the call isn't in your log. If it is from an internal extension, I think a SIP trace on that

Re: [asterisk-users] Unexpected termination of the call when pick up (res_pjsip)

; Unsuccessful call from device on res_pjsip via endpoint on res_pjsip: > http://pastebin.com/hepVb6Nu > > And ones again i don't see anything that would make asterisk send BYE. > > I would be grateful for any ideas. > > 11.02.2016 1:47, Trey Hilyard пишет: > > How are

Re: [asterisk-users] SIP URI set 'telephone-context='

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@pjsippeer,60) I am pretty sure it was easy in chan_sip, too. If you are using chan_sip, I'll try and find an example. On Tue, Feb 16,

Re: [asterisk-users] SIP URI set 'telephone-context='

Agree. All you have to do is: Dial(SIP/+${EXTEN}\;phone-context=+44@10.10.10.10\;user=phone) I am actually surprised that the dialplan reload would work without it... On Wed, Feb 17, 2016 at 5:51 AM A J Stiles wrote: > On Wednesday 17 Feb 2016, imperium

Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

king the entire URI. I am working on a plan using a lot more CUTs than I think I should need, but we'll see if it works. On Fri, Mar 18, 2016 at 10:58 AM Trey Hilyard <kct...@gmail.com> wrote: > On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <ad...@tootai.net> > wrot

Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <ad...@tootai.net> wrote: > Le 18/03/2016 16:20, Trey Hilyard a écrit : > > 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), &

Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

On Mar 18, 2016 8:27 PM, "Steve Edwards" <asterisk@sedwards.com> wrote: >> >> On Fri, 18 Mar 2016, Trey Hilyard wrote: >> >>> I thought this would be as easy as >>> exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTE

[asterisk-users] Incoming INVITE with Portability Info and LRN

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

Re: [asterisk-users] PJSIP logging fails

On Wed, Apr 12, 2017, 4:14 PM Sree Harsha Totakura wrote: > Did you try setting the debug verbosity to a number > 3? > > Alternatively, if you want to see a register packet, try running > wireshark on the server and capture the request packets. > > Sree > On 04/12/2017

[asterisk-users] Adding Subscribe Handlers in PJSIP

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