Wait a second, I thought in your original email that you said that
Asterisk was generating reinvites. It sounds now like you're saying
that the remote side is initiating reinvites instead.
My understanding is that the canreinvite/directmedia option only
influences Asterisk's behavior with regards
On 8/9/16 12:40 PM, Matt Fredrickson wrote:
> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly wrote:
>> Hi All,
>>
>> We have asterisk 11.23 running sip to vitelity and from there IAX trunks
>> split off to where they need to go. We are having a problem getting
>> chan_sip to quit ignoring re-inv
In 11 setting trustrpid sendrpid is enough that phone getting the tranfered
call shows the name and number of the caller and not the tranferer
In 13 the same shows the transferrs info
בתאריך 11 באוג׳ 2016 00:21, "Matt Fredrickson" כתב:
> How are you attempting to view the original CallerId?
>
>
How are you attempting to view the original CallerId?
Matthew Fredrickson
On Wed, Aug 10, 2016 at 2:59 PM, Israel Gottlieb wrote:
> Hi
> Is there any configuration change in asterisk 13.9.1 to show original
> callerid on a transfer
> In asterisk 11.21 it works as expected
>
> Thanks
>
>
> --
> _
My suggestion is to verify and debug against Asterisk 13 first, and
then you can try backing down versions, rather than reverse. WebRTC
is a rapidly moving target, and has required ongoing changes that may
not have made it into older and feature frozen versions of Asterisk.
Matthew Fredrickson
O
Hello
thank you for your answer.
I don't understand how there are many tutorials and examples on the web
where every time the outcome is a working setup. Very strange I feel now
after my personal experience with Asterisk 11 and webRTC.
You also say Asterisk 13. How about Asterisk 12 then ??
Hi
Is there any configuration change in asterisk 13.9.1 to show original
callerid on a transfer
In asterisk 11.21 it works as expected
Thanks
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Aster
I don't see an ice-ufrag or ice-pwd line in the response from
Asterisk, correlating with your suspicion that there is no ICE. Are
you sure that the stun server you're using (the google one) still
works? I haven't tried that server in a while, but I distantly seem
to recall that maybe they shut it
El 10/08/16 a las 12:06, Carlos Chavez escribió:
Anyone know a good replacement for phpagi? Unfortunately development
stalled long ago and it has not been updated. What is the best solution for
AMI and AGI on PHP? Thanks.
In the case of AMI, you could use the AMI client from the Elastix
I keep getting messages like these in the cli:
[Aug 10 12:20:17] WARNING[23411]: res_config_mysql.c:1162 require_mysql:
Realtime table general@ps_contacts: column 'qualify_timeout' cannot be
type 'int(10)' (need char)
[Aug 10 12:20:17] WARNING[23411]: res_config_mysql.c:1246 require_mysql:
What version of asterisk are you on?
Marlon Araujo
> On Aug 10, 2016, at 13:00, asterisk-users-requ...@lists.digium.com wrote:
>
> Calls are dropped after 15 minutes
--
_
-- Bandwidth and Colocation Provided by http://www.api-
Anyone know a good replacement for phpagi? Unfortunately
development stalled long ago and it has not been updated. What is the
best solution for AMI and AGI on PHP? Thanks.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
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__
On 2016-08-10 11:53, Joshua Colp wrote:
Jacek Konieczny wrote:
On 2016-08-09 10:06, Faheem Muhammad wrote:
trip time and Call Setup time of SIP Requests.
In case of GSM Network with high delay you need to set the T1 timer a
higher value like 1000ms (500 ms default). Similarly you can reduce the
On 10-08-16 08:52, Ludovic Gasc wrote:
For WebRTC, I recommend you to use Asterisk 13+.
Have a nice day.
Ludovic Gasc (GMLudo)
http://www.gmludo.eu/
Hello
then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??
This is no answer to my question.
So again : what am I m
Jacek Konieczny wrote:
On 2016-08-09 10:06, Faheem Muhammad wrote:
trip time and Call Setup time of SIP Requests.
In case of GSM Network with high delay you need to set the T1 timer a
higher value like 1000ms (500 ms default). Similarly you can reduce the
Call setup time by configuring 'T2' upto
On 2016-08-09 10:06, Faheem Muhammad wrote:
trip time and Call Setup time of SIP Requests.
In case of GSM Network with high delay you need to set the T1 timer a
higher value like 1000ms (500 ms default). Similarly you can reduce the
Call setup time by configuring 'T2' upto you choice as per you t
On 2016-08-09 10:06, Faheem Muhammad wrote:
Jacek,
This might be a bug or configuration issue, but you need to understand
the SIP Session Timers. With Session Timers you can control the round
trip time and Call Setup time of SIP Requests.
I don't think you really mean SIP Session Timers
(https
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