VoIP Mailing List Archives

[maxim.tsvetov at gmail...]

I've tried to use "answer" command from outbound event socket and it's
working, but
the problem is that FS answering the call, but SIP Client (we tried this
with EyeBeam and CISCO 7960)
doesn't know that call was answered. So, as long as FS doesn't know what to
do with this number it then disconnects the call.

2009-08-12 11:25:07.599250 [NOTICE] mod_dptools.c:649 Channel
[sofia/internal/sip:1000@10.107.181.160:42840] has been answered
2009-08-12 11:25:07.599250 [NOTICE] switch_ivr_originate.c:2015 Channel
[sofia/internal/1003@10.107.249.12] has been answered
2009-08-12 11:25:07.614875 [ERR] switch_core_io.c:118
sofia/internal/sip:1000@10.107.181.160:42840 has no read codec.
2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:503 Hangup
sofia/internal/sip:1000@10.107.181.160:42840 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]
2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:1016 Hangup
sofia/internal/1003@10.107.249.12 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-12 11:25:07.630500 [NOTICE] switch_core_session.c:1086 Session 133
(sofia/internal/sip:1000@10.107.181.160:42840) Ended


Maybe there is the way to acknowledge SIP client that call was answered?

Regards,
Maxim Tsvetov

Diego Viola wrote:

[dujinfang at gmail.com]

It's not Eyebeam but FS hung up the call because it have nothing to do
after answer.

You should either playback a sound, do the echo command, record, hold
the call, bridge to another channel or transfer somewhere else.....

On Aug 12, 2009, at 4:54 PM, Maxim Tsvetov wrote:

[maxim.tsvetov at gmail...]

I will try to paraphrase my question.
Is there any possibility to answer call from CTI application and
synchronise answer with answer in SIP client?Maybe we can use SIP functions
in our CTI application instead of FS api commands?
I'm trying to find the way to make prototype of lineAnswer command in TAPI.


Seven Du wrote:

[brian at freeswitch.org]

Well you can only truly answer an inbound call to FS... you can't
force answer an outbound call.

/b

On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:

[maxim.tsvetov at gmail...]

If I have two FS extensions A and B. I'm calling from A to B and want to
answer from B-side in my CTI application and to make SIP phone to be
synchronised to my CTI application. Is it possible to do it?


Brian West-3 wrote:

[mike at jerris.com]

Sip does not support this functionality. The called device would have
to support this via some other mechanism such as ctsa which I have
seen recently someone was looking at for freeswitch. So the first
issue you must resolve is the called device needs to support some way
to do this.

Mike

On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov <maxim.tsvetov@gmail.com>
wrote:

[diego.viola at gmail.com]

Hey Michael,

Just wondering something, I have found that you added "conference_set_auto_outcall" on the dptools wiki, but I could not find that function in the mod_dptools.c, shouldn't that be part of the mod_conference wiki article? =D.

Best regards,

Diego

On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris <mike@jerris.com (mike@jerris.com)> wrote:

[mike at jerris.com]

It probably belongs there. It's a wiki, feel free to fix it. What does this have to do with this thread?

On Aug 13, 2009, at 4:03 AM, Diego Viola <diego.viola@gmail.com (diego.viola@gmail.com)> wrote:

[diego.viola at gmail.com]

Err, I asked if that was wrong to fix it.

On Thu, Aug 13, 2009 at 2:15 PM, Diego Viola <diego.viola@gmail.com (diego.viola@gmail.com)> wrote:

[diego.viola at gmail.com]

I was talking with Michael about fixing stuff in the wiki, so I just asked to fix that also.

On Thu, Aug 13, 2009 at 9:10 AM, Michael Jerris <mike@jerris.com (mike@jerris.com)> wrote: