Displaying 20 results from an estimated 10000 matches similar to: "Dial to PJSIP Channel with Typo "PJSIP//" Causes Asterisk Shutdown"
2015 Apr 01
1
PJSIP Endpoint AOR question
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
2015 Apr 01
2
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]
type=transport
protocol=udp
bind=12.4.aa.bb:5060
My internal endpoints use
2017 Mar 01
4
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 didn't actually do anything but simply responded with a 200 OK.
2015 Apr 02
1
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
2016 Feb 16
2
SIP URI set 'telephone-context='
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
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2016 Feb 16
2
SIP URI set 'telephone-context='
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/${ARG1}\;phone-context=mydomain.com at pjsippeer,60)
>
> I am pretty sure it was
2015 Apr 01
4
PJSIP Endpoint AOR question
I am running asterisk 13.1.0
In pjsip.conf, the endpoint section has an aors and an auth field.
I can name the auth field anything I want. The key is to set the auth=field accordingly.
However, when I try this with the aors field, it never works. It seems I have to name the aors=field to match the name of the endpoint section.
Is this correct?
Would there ever be a need for multiple aors to
2016 Mar 18
2
Incoming INVITE with Portability Info and LRN
On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <admin at tootai.net>
wrote:
> Le 18/03/2016 16:20, Trey Hilyard a ?crit :
> > I am trying to set up my Asterisk server so that it will recognize an
> > incoming call to the Asterisk's own Location Routing Number (LRN),
> > validating the "rn" in the INVITE and then using the Called Number from
>
2015 Mar 20
2
outbound calls
i noticed that when i active the voicemail in the IP-phone where the number
0033149xxxxxx is configured i can call this number without issue
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording
SIP/101-0000010d
-- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d
> 0x2b393cfc2610 -- Probation passed
2016 Feb 17
2
SIP URI set 'telephone-context='
On Wednesday 17 Feb 2016, imperium broadcast wrote:
> I kinda have it working with chan_sip.
>
> Dial(SIP/+${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
2014 Dec 21
3
PJSIP ports, multiple IP addresses and wrong owner
Dear list,
I am currently trying to send faxes via T.38 using PJSIP (newest version 2.3) with Asterisk 13.0.2. After having configured PJSIP, I have seen several things the cause of which I would like to know.
1) Ports and IP addresses which PJSIP bind to
I have configured one transport like that:
[tr_wZCMk5MvC2ATNzAr]
type = transport
protocol = udp
bind = 192.168.20.48
Nevertheless, PJSIP
2015 Apr 02
0
PJSIP Sends BYE with Wrong IP
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
>
2014 Jun 26
1
PJSIP Dial via IP fails
Dear friends
This is my simple dialplan
[demopjsip]
exten => _X.,1,Dial(PJSIP/${EXTEN}@10.10.10.2)
exten => _X.,n,Hangup()
I need to dial out via an IP address, not using an endpoint, as shown above.
It fails with
Executing [19544447408 at demopjsip:3] Dial("PJSIP/federico-00000002",
"PJSIP/195XXX7408 at 10.10.10.2") in new stack
[Jun 26 00:39:00] ERROR[10136]:
2017 Nov 14
2
Confbridge SFU for Asterisk 15
I am trying to get the "Mega Phone" demo working on my office PBX
but there seems to be a problem when trying to set the default bridge to
sfu mode. I have the following configuration in confbridge.conf in the
default_bridge section: video_mode = sfu but when I do a "confbridge
show profile bridge default_bridge" I see:
Video Mode: no video
I can change it
2016 Mar 18
3
Incoming INVITE with Portability Info and LRN
On Fri, 18 Mar 2016, Trey Hilyard wrote:
> I thought this would be as easy as
> exten => _XXXXXXXXXX\;rn=+19136630000,1,Goto(from_pstn,${EXTEN:0:10})
Have you tried the '_!.' pattern?
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
2003 Jul 07
1
Dial plan doesn't seem to save properly
When I first to the "add extension" the "show dialplan" has the lines that
say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and
restart the lines with "SIP/" are gone.
************************
"Show dialplan" before:
************************
asterisk01*CLI>
[ Context 'default' created by
2007 Mar 29
2
L options in Dial() dont seem to work....
Hello Asterisk users,
Can someone thwack me with a clue stick please?
I am following the Asterisk TFOT book Dial() example trying to get the limit
and announcements to work as per below.
These settings seem to have no effect.
There are no warning messages after 4 minutes or every 30 secs thereafter
and the call lasts longer than 5 minutes.
gunner*CLI> show dialplan
[ Context
2009 Jun 14
2
FXS - TDM400 - No dial tone
I have a TMD400 card installed in a PC with one fxs (installed in slot
2) and two fxos (installed in slots 3 & 4).? fxos work fine but I am
unable to get a dial tone for any devices connected to the fxs.? I
have correctly connected the power supply to the card and I have even
tried moving the card from slot 1 to 2 on the board.
Below is from the console when I try to route a call from FXO on
2016 Mar 18
2
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 sip:+19135041291;rn=+19136630000;npdi at 12.4.240.200:5060;user=phone;transport=udp
SIP/2.0
2011 Oct 27
5
Asterisk Executing outbound dial number twice
Hello,
I noticed Asterisk 1.8.4.1 execute number dial twice
Log
== Using SIP RTP CoS mark 5
-- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920",
"CALLERID(num)=2066604") in new stack
== Extension Changed 4773[sipphones] new state InUse for Notify User 4701
-- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",