<div>Hi, </div>
<div> </div>
<div>Just to follow up with this problem. If I set both xlite and sip application to use PCMU, I am still getting noise even channels show the same codec:</div>
<div> </div>
<div>API CALL [show(channels)] output:<br>uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate<br>0816684d-7a29-4814-93e4-104ffc2ed984,inbound,2009-05-23 11:25:30,1243092330,sofia/internal/<a href="mailto:1000@192.168.1.191">1000@192.168.1.191</a>,CS_EXECUTE,1000,1000,192.168.1.193,3000,bridge,{absolute_codec_string='GSM,PCMU'}sofia/<a href="http://192.168.1.191/454044009539026,XML,public,PCMU,8000,PCMU,8000">192.168.1.191/454044009539026,XML,public,PCMU,8000,PCMU,8000</a><br>
a6f1e90c-f6a9-4ac1-9f26-fe08c5c0dd74,outbound,2009-05-23 11:25:30,1243092330,sofia/internal/454044009539026,CS_EXCHANGE_MEDIA,1000,1000,192.168.1.193,454044009539026,,,XML,public,PCMU,8000,PCMU,8000<br></div>
<div>Thanks for any suggestion.</div>
<div> </div>
<div>Thanks,</div>
<div>JB<br><br></div>
<div class="gmail_quote">On Sat, May 23, 2009 at 3:11 PM, Juan Backson <span dir="ltr"><<a href="mailto:juanbackson@gmail.com">juanbackson@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div>Hi,</div>
<div> </div>
<div>I am getting problem when one UA is xlite and another UA is another sip application. </div>
<div> </div>
<div>When I call from xlite to a sip application, I am getting noise:</div>
<div> </div>
<div>I have tried these:</div>
<div> <extension name="redial"><br> <condition field="destination_number" expression="^3000"><br> <action application="bridge" data="{absolute_codec_string='GSM,PCMU'}sofia/<a href="http://192.168.1.191/4540" target="_blank">192.168.1.191/4540</a>"/><br>
</condition><br> </extension><br></div>
<div> <extension name="redial"><br> <condition field="destination_number" expression="^3000"><br> <action application="bridge" data="sofia/<a href="http://192.168.1.191/4540" target="_blank">192.168.1.191/4540</a>"/><br>
</condition><br> </extension><br></div>
<div> </div>
<div>show channels give me the following:</div>
<div> </div>
<div>c5f42dec-646a-4675-af40-c4d173c8a7c7,inbound,2009-05-23 10:36:30,1243089390,sofia/internal/<a href="mailto:1000@192.168.1.191" target="_blank">1000@192.168.1.191</a>,CS_EXECUTE,1000,1000,192.168.1.193,3000,bridge,sofia/<a href="http://192.168.1.191/4540,XML,public,GSM,8000,GSM,8000" target="_blank">192.168.1.191/4540,XML,public,GSM,8000,GSM,8000</a><br>
790d9b2a-88b9-4521-8934-31b059e04e7b,outbound,2009-05-23 10:36:30,1243089390,sofia/internal/4540,CS_CONSUME_MEDIA,1000,1000,192.168.1.193,4540,,,XML,public,,,,</div>
<div> </div>
<div>The sip application and xlite is working fine ( voice is clear ) if I use Asterisk with the following line in sip.conf:</div>
<div> </div>
<div>[4540]<br>canreinvite=no<br>type=friend<br>context=sip-external<br><font color="#ff6666">allow=gsm</font><br>host=dynamic</div>
<div><br>[1000]<br>canreinvite=no<br>type=friend <br>context=sip-external<br>allow=gsm <br>host=dynamic </div>
<div> </div>
<div> </div>
<div>Does anyone know how to mimic the same behavior in Freeswitch?</div>
<div> </div>
<div>Thanks,</div>
<div>JB</div><font color="#888888">
<div> </div></font></blockquote></div><br>