Hi, 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: API CALL [show(channels)] output: 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 0816684d-7a29-4814-93e4-104ffc2ed984,inbound,2009-05-23 11:25:30,1243092330,sofia/internal/1000@192.168.1.191 ,CS_EXECUTE,1000,1000,192.168.1.193,3000,bridge,{absolute_codec_string='GSM,PCMU'}sofia/ 192.168.1.191/454044009539026,XML,public,PCMU,8000,PCMU,8000 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 Thanks for any suggestion. Thanks, JB On Sat, May 23, 2009 at 3:11 PM, Juan Backson <juanbackson@gmail.com> wrote:
Hi,
I am getting problem when one UA is xlite and another UA is another sip application.
When I call from xlite to a sip application, I am getting noise:
I have tried these: <extension name="redial"> <condition field="destination_number" expression="^3000"> <action application="bridge" data="{absolute_codec_string='GSM,PCMU'}sofia/192.168.1.191/4540"/> </condition> </extension> <extension name="redial"> <condition field="destination_number" expression="^3000"> <action application="bridge" data="sofia/192.168.1.191/4540"/> </condition> </extension>
show channels give me the following:
c5f42dec-646a-4675-af40-c4d173c8a7c7,inbound,2009-05-23 10:36:30,1243089390,sofia/internal/1000@192.168.1.191 ,CS_EXECUTE,1000,1000,192.168.1.193,3000,bridge,sofia/ 192.168.1.191/4540,XML,public,GSM,8000,GSM,8000 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,,,,
The sip application and xlite is working fine ( voice is clear ) if I use Asterisk with the following line in sip.conf:
[4540] canreinvite=no type=friend context=sip-external allow=gsm host=dynamic
[1000] canreinvite=no type=friend context=sip-external allow=gsm host=dynamic
Does anyone know how to mimic the same behavior in Freeswitch?
Thanks, JB