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
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 th
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
di
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
[untrust
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
t.
On Wed, Apr 1, 2015 at 2:59 PM Trey Hilyard 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
>
On Thu, Apr 2, 2015 at 10:43 AM, Rusty Newton wrote:
> On Wed, Apr 1, 2015 at 9:08 AM, Trey Hilyard 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 "
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
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. Not
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 sid
e 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 you initiating the call out
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/${ARG1}\;phone-context=mydomain.com@pjsippeer,60)
I am pretty sure it was easy in chan_sip, too. If you are using chan_sip,
I'll try and find an example.
On Tue, Feb 16, 20
Agree. All you have to do is:
Dial(SIP/+${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 broadcast wrote:
> > I kinda have it wor
without taking 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 wrote:
> On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI
> wrote:
>
>> Le 18/03/2016 16:20,
On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI
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),
> > validating
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:+19
On Mar 18, 2016 8:27 PM, "Steve Edwards" 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,${EXTEN:0:10})
>
>
> How abo
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 multip
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 d
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 08:55 PM, Saint Michael wro
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