Displaying 20 results from an estimated 3000 matches similar to: "PJSIP usereqphone setting in config file"
2014 Apr 08
1
PJSIP in dialog OPTIONS method handling
Hi everyone,
I am running asterisk with release 12.1.0.rc3 and PJSIP.
I have a peer which sends OPTIONS method for session keep-alive, and the
asterisk is not responding to it. That of course disconnects the call after
a few minutes.
Is there a settings in the PJSIP.conf to respond to in dialog OPTIONS
method? Looking at the documentation I haven't seen it. Does anybody know a
workaround?
2014 Mar 11
1
PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833
Hello,
I have installed the latest version 12 that has been released (12.1.0.rc3).
I have setup default dtmf mode (rfc47..) but when I am calling to a
endpoint that doesn't support it (no telephony event in the rtpmap) the
asterisk responds OK in the signalling but DTMF is not working.
Is it a known issue?
Below you can see the output of the asterisk monitor.
<--- Received SIP request
2015 May 31
2
Signaling incoming call
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Guenther Boelter <gboelter at gmail.com> schrieb:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA256
>
> On 05/31/2015 02:31 PM, Luca Bertoncello wrote:
> > Hi list!
> >
> > Now all works as expected, at least in the simulation I did with
> > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom
2015 Jan 17
1
Fwd: Asterisk pjsip auto dtmf mode
Hello Asterisk Users,
I have been looking for similar auto dtmf mode implementation on pjsip, but
didn't see it coming, so I decided to give it a try.
My basic plan was to do it as simple as possible with minimum changes
because I am not familiar with all Asterisk code. My idea is to use rfc4733
settings, but when going over the codecs check if telephone-event appear
and if not set the dtmf
2014 Mar 11
1
PJSIP - Using multiple AOR contacts when dialing through an endpoint
Hello everyone,
I have started testing the PJSIP stack.
I saw that it is possible to setup statically multiple AOR contacts, setup
qualify_timeout and attach it to an endpoint, and then dial using this
endpoint.
When I setup the configuration I used the cli in order to see the status of
the contacts, and it worked fine - whenever a contact is unreachable, the
status is updated to Unavailable.
2014 Nov 06
1
Function to get mailbox for a PJSIP Endpoint?
Howdy,
I'm trying to re-write my voicemail check extension.
I formerly used the SIPPEER function to get the mailbox for a peer with
${SIPPEER(${peer},mailbox)}
Is there a way to do this with PJSIP now that I've converted over?
I see a function PJSIP_ENDPOINT and it has a mailboxes subset but I'm not
retrieving any data from it when I query it.
--
A human being should be
2008 May 01
1
http://www.asteriskdocs.org/html/apas02.html
If one of the authors is listening:
http://www.asteriskdocs.org/html/apas02.html
lists usereqphone 2 times. One of the entries should really
be useragent. And the example for usereqphone is wrong.
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? ->
2015 Feb 18
2
Asterisk 13 - sorcery realtime for pjsip publish objects
Hello,
I am currently trying to set up pjsip realtime and would like to have
outbound-publish, inbound-publication, and asterisk-publication sorcery
object types in ODBC realtime. Is that currently supported? I know that
some object types are known working and others are not. I was curious
what the status of those objects are.
Thanks!
Matt Hoskins | NPG Corp | Systems Architect
2015 May 28
3
Peer is UNREACHABLE
Darryl Moore <darryl at moores.ca> schrieb:
> Ahh. Seen that before! That suggests to me that you don't have your
> sip.conf records setup right.
>
> What's your sip.conf look like?
Well, here what I wrote in my sip.conf:
register => 00493511111111:MYSECRET at pbxluca/00493511111111
register => 00493512222222:MYSECRET at pbxfax/00493512222222
register =>
2015 Jan 04
3
Confused by concepts behind pjsip: endpoint, aor, contact
Hello,
I am slightly confused by the difference between chan_sip and pjsip.
Especially the new (to me) objects aor and contact.
I am having trouble mapping them to the typical SIP configuration settings
on a phone.
Suppose I have a phone with two line buttons, for two extension numbers.
Now,
I think that means two 'endpoints' in pjsip right? But what exactly is the
difference
between
2014 Oct 26
1
DTMF behavior in asterisk 12 with PJSIP
Hello all,
We have recently upgraded some of our services to Asterisk 12 with PJSIP.
We have 2 issues related to DTMF:
1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMF
settings according to the incoming INVITE - RFC2833 or inband. The is no
such settings in PJSIP. Do you know is there is a plan to develop it?
2. When we setup 2 peers, one RFC4733 and the other inband,
2016 May 12
2
pjsip module reload problem
Hi!
Installing new asterisk server and decided to use chan_pjsip.
While module reload I get:
y 12 15:33:04] ERROR[21137]: config_options.c:715 aco_process_var: Could
not find option suitable for category '3567' named 'inband_progress' at
line 867 of
[May 12 15:33:04] ERROR[21137]: res_sorcery_config.c:317
sorcery_config_internal_load: Could not create an object of type
2006 Feb 23
6
username as extension
Is there a way to have extensions automatically created for
registered sip users ?
I did some investigation and found some hope in chan_sip with
relation to the somewhat undocumented usereqphone option but i may be
totally off track.
All i want to be able to do is send a call to number@ip_address where
the number is the username configured on the phone that has
registered with asterisk
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
2016 Jul 13
3
PJSIP defaults for endpoints when using realtime
Until Asterisk 11 I could use sip.conf to set defaults for all
phones (language, dtmf, vmexten, etc) and just leave many fields in the
database as NULL. What would be the proper way to do this for Asterisk
13 and PJSIP?
--
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez
+52 (55)9116-91161
2007 Oct 26
1
Nortel C15K <-> Asterisk
Has anyone had any luck getting an asterisk box to talk to a Nortel
C15K softswitch? Or any Nortel "sip" products? I've been playing with
it for several days and can't seem to pass calls either direction. I
know that whike the Nortel says the C15K speaks SIP, it really speaks
nortel's implementation of SIP, but I thought I could get it to at
least pass simple calls back
2020 Apr 06
2
Outgoing PJSIP using Kamailio
Hello,
We have a provider which is using Kamailio as front end. Our asterisk
13/chan_sip server has no problem to register and pass/receive calls
form this provider.
Now we want to move to asterisk 16/pjsip and face problem. Registration
is OK but when we pass a call our INVITE never receive answer from the
provider. We opened a ticket to their support but in the mean time we
want to know
2015 Feb 18
3
Asterisk 13 - sorcery realtime for pjsip publish objects
Excellent. I was using ast-13.1.0 with no luck. I upgraded to 13.2.0 and
have made it further, but am having a little difficulty. The
outbound-publish object types seems to be working in realtime now. But
the asterisk-publication object is only reading from sorcery.conf. I know
you said that it *should* work, with no guarantee, which I'm fine with. I
just want to make sure I don't
2014 Nov 12
1
Asterisk 12 crashes on CANCEL received during ANSWER handlingl
Hello Asterisk users and developers,
The last few weeks we had several crashes on live asterisks running
versions 12.2.0rc1 / 12.6.1 with PJPROJECT versions 2.1.0 / 2.2.1. We
opened a ticket - ASTERISK-24471.
After investigating the issue I can say that the scenario is a CANCEL being
received while handling ANSWER and before generating the 200OK response.
Looking at the core file we see that
2018 Feb 08
3
pjsip trunking configuration issue
Greetings !
My goal is to get Twilio trunking working, and with TLS/SRTP.
I see this concerning message in my log:
[Feb 7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an object of type 'endpoint' with id ?twilio' from configuration file ?pjsip.conf?
Thus, ?pjsip show endpoints? does not show the endpoint for the Twilio trunk.
Hoping for a sanity check of