<br><br><div class="gmail_quote">On Tue, Mar 22, 2011 at 11:09 AM, Dmitry Bely <span dir="ltr"><<a href="mailto:dmitry.bely@gmail.com">dmitry.bely@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
On Tue, Mar 22, 2011 at 8:18 PM, Michael Collins <<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>> wrote:<br>
> How are you performing the attended transfer? What kind of phone is this?<br>
> -MC<br>
<br>
Yes, the attended transfer using phone's capabilities as explained<br>
here: <a href="http://www.youtube.com/watch?v=VpEVpr-4y-U" target="_blank">http://www.youtube.com/watch?v=VpEVpr-4y-U</a><br>
The phone is Grandstream GXP-2000.<br>
<br>
Here is an example. There was two active calls:<br>
<br>
gw <-> 1000 (PCMU)<br>
1000 <-> 1004 (G722)<br></blockquote><div><br></div><div>Is the gateway in the external profile? If so, try setting the disable-transcoding param on that profile. Or you could try FreeSWITCH's built in att-xfer using the *4 key combo. </div>
<div><br></div><div>-MC<br><br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<br>
Now operator transfers the fist call to 1004. The log:<br>
<br>
2011-03-18 17:08:27.221372 [DEBUG] sofia.c:4659 Channel<br>
sofia/internal/<a href="mailto:1000@192.168.121.66">1000@192.168.121.66</a> entering<br>
state [received][100]<br>
2011-03-18 17:08:27.222378 [DEBUG] sofia.c:4670 Remote SDP:<br>
v=0<br>
o=1000 8001 8002 IN IP4 192.168.121.153<br>
s=SIP Call<br>
c=IN IP4 192.168.121.153<br>
t=0 0<br>
m=audio 5030 RTP/AVP 9 0 8 18 4 99 3 2<br>
a=rtpmap:9 G722/8000<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:18 G729/8000<br>
a=rtpmap:4 G723/8000<br>
a=rtpmap:99 iLBC/8000<br>
a=fmtp:99 mode=20<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:2 G726-32/8000<br>
a=sendonly<br>
a=ptime:20<br>
<br>
2011-03-18 17:08:27.222378 [DEBUG] switch_channel.c:1377<br>
(sofia/internal/<a href="mailto:1000@192.168.121.66">1000@192.168.121.66</a>) Callstate Change ACTIVE -> HELD<br>
2011-03-18 17:08:27.222378 [DEBUG] switch_core_session.c:954 Send<br>
signal sofia/internal/<a href="http://sip:1004@192.168.121.136:5060" target="_blank">sip:1004@192.168.121.136:5060</a> [BREAK]<br>
2011-03-18 17:08:27.224386 [DEBUG] switch_core_session.c:709 Send<br>
signal sofia/internal/<a href="http://sip:1004@192.168.121.136:5060" target="_blank">sip:1004@192.168.121.136:5060</a> [BREAK]<br>
2011-03-18 17:08:27.364243 [DEBUG] switch_ivr.c:563<br>
sofia/internal/<a href="http://sip:1004@192.168.121.136:5060" target="_blank">sip:1004@192.168.121.136:5060</a> Command Execute<br>
playback(local_stream://moh)<br>
EXECUTE sofia/internal/<a href="http://sip:1004@192.168.121.136:5060" target="_blank">sip:1004@192.168.121.136:5060</a><br>
playback(local_stream://moh)<br>
2011-03-18 17:08:27.364243 [WARNING] mod_local_stream.c:393 Unknown<br>
source moh, trying 'default'<br>
2011-03-18 17:08:27.364243 [DEBUG] mod_local_stream.c:421 Opening<br>
Stream [default] 8000hz<br>
2011-03-18 17:08:27.364243 [DEBUG] switch_core_file.c:176 File moh<br>
sample rate 8000 doesn't match requested rate 16000<br>
2011-03-18 17:08:27.364243 [DEBUG] switch_ivr_play_say.c:1244 Codec<br>
Activated L16@16000hz 1 channels 20ms<br>
2011-03-18 17:08:27.472662 [DEBUG] sofia_glue.c:4474 Audio Codec<br>
Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000]<br>
2011-03-18 17:08:27.472662 [DEBUG] sofia_glue.c:2690 Already using G722<br>
2011-03-18 17:08:27.472662 [DEBUG] sofia_glue.c:2972 Audio params are<br>
unchanged for sofia/internal/<a href="mailto:1000@192.168.121.66">1000@192.168.121.66</a>.<br>
2011-03-18 17:08:27.472662 [DEBUG] sofia.c:5070 Processing updated SDP<br>
2011-03-18 17:08:27.473667 [DEBUG] sofia.c:4659 Channel<br>
sofia/internal/<a href="mailto:1000@192.168.121.66">1000@192.168.121.66</a> entering state [completed][200]<br>
2011-03-18 17:08:27.592069 [DEBUG] sofia.c:4659 Channel<br>
sofia/internal/<a href="mailto:1000@192.168.121.66">1000@192.168.121.66</a> entering state [ready][200]<br>
2011-03-18 17:08:27.769589 [DEBUG] sofia.c:5520 Process REFER to<br>
[<a href="mailto:1004@192.168.121.66">1004@192.168.121.66</a>]<br>
2011-03-18 17:08:27.769589 [DEBUG] sofia.c:5539 Replaces:<br>
[<a href="mailto:5cd372e1e81966ca@192.168.121.153">5cd372e1e81966ca@192.168.121.153</a>;from-tag=d55f7fbec6efbde7;to-tag=2N49SS5XNjUeB]<br>
...<br>
(no more codec negotiation)<br>
<br>
> On Tue, Mar 22, 2011 at 3:11 AM, Dmitry Bely <<a href="mailto:dmitry.bely@gmail.com">dmitry.bely@gmail.com</a>> wrote:<br>
>><br>
>> I have G722 and PCMU enabled for internal extensions and PCMU only for<br>
>> an external gateway. Inbound-late-negotiation parameter is set so then<br>
>> an incoming call arrives PCMU is used without transcoding. But it goes<br>
>> worse when transfer is involved:<br>
>><br>
>> call 1: gw <-- PCMU--> FreeSwitch <-- PCMU --> operator<br>
>> call 2: operator <-- G722 --> FreeSwitch <-- G722--> extension<br>
>> transfer (refer): gw <-- PCMU--> FreeSwitch <-- G722--> extension<br>
>><br>
>> Is it possible to renegotiate a codec during transfer to get rid of<br>
>> transcoding?<br>
<br>
- Dmitry Bely<br>
<br>
_______________________________________________<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
</blockquote></div><br>