[Freeswitch-users] need hear solving a noise problem

[Juan Backson]

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 at 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 at 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 at 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
>
>
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