with inband it is not identified. Using rtp debug I see
that the rtp is sent and received.
I did the same scenario with regular sip channel and the same happened.
If anyone has a clue please get back to me.
I will try to make the test with sipp.
Yaron
On Wed, Apr 1, 2015 at 6:34 PM, Yaron
Hi Everyone,
Sorry for all the questions.
Well I managed to understand the 488 issue - I had to add some codec
capabilities. Now the test works but only if I setup the dtmfmode to
rfc4733. If I set it to inband it fails - the Read on the receiver side
doesn't receive DTMF.
The following is the sc
Hi everyone,
I am still having problems with the testsuite. I made a simple scenario
that originates a call from the ami to a local channel, an then dials
through a PJSIP endpoint to another PJSIP endpoint.
The issue I am having is when I dial the other endpoint I receive 488 not
acceptable here.
Got it !!!
The testsuite was looking for these modules in /usr/lib64.
I recompiled the asterisk with --libdir=/usr/lib64 and it works.
Now the test is running
I will start working on it now.
Thank you.
On Tue, Mar 31, 2015 at 6:09 PM, Yaron Nachum
wrote:
> The following is the output
>
31, 2015 at 9:00 AM, Yaron Nachum
> wrote:
> > Thank you mathew,
> >
> > The pjproject was detected on the installation process. When I run
> Asterisk
> > I see that pjsip modules are running.
> >
>
> The dependency checking for Asterisk assumes that th
Thank you mathew,
The pjproject was detected on the installation process. When I run Asterisk
I see that pjsip modules are running.
Any idea?
Yaron
On Tue, Mar 31, 2015 at 4:12 PM, Matthew Jordan wrote:
> On Tue, Mar 31, 2015 at 8:04 AM, Yaron Nachum
> wrote:
> > Hi everyo
Hi everyone,
I am trying to add some tests for the PJSIP auto-dtmf support . Before I
start I just wanted to run some of the existing tests in order to
understand the process.
Whenever I try to run a test from the pjsip tests I get - --- -->
Dependency: res_pjsip - False. The following output was
Hello Everyone,
I am trying to run the rbt tool in order to get the patch up on
Reviewboard. I did the following steps:
1. Downloaded the latest trunk version.
2. Updated the code.
3. run - rbt post - command
4. I get the message - CRITICAL: 'Working Copy Root Path'.
Any idea what an I doing wrong
nstance is not handled in these functions.
The question is how do I get access to ast_sip_session_media instance? I
have in both functions access to ast_sip_session instance.
Thank you,
Yaron.
On Mon, Jan 19, 2015 at 5:33 PM, Matthew Jordan wrote:
>
>
> On Sat, Jan 17, 2015 at
On Wed, Apr 9, 2014 at 5:04 PM, Matthew Jordan wrote:
>
>
>
> On Wed, Apr 9, 2014 at 8:49 AM, Yaron Nachum wrote:
>
>> Hi,
>> Anyone has a workaround?
>>
>>
>> On Tue, Apr 8, 2014 at 9:17 PM, Yaron Nachum wrote:
>>
>>> Hi everyone,
>>&
Hi,
Anyone has a workaround?
On Tue, Apr 8, 2014 at 9:17 PM, Yaron Nachum wrote:
> 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
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 OP
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