[Freeswitch-users] Question: Capturing VoiceData received from Sagnoma E1 card
Helmut Kuper
helmut.kuper at ewetel.de
Mon Jul 27 02:34:49 PDT 2009
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Hello,
any ideas?
regards
Helmut
On 21.07.2009 16:01, Helmut Kuper wrote:
> Hello,
>
>
> For outgoing calls I'm hunting the cause for missing some 100ms of voice
> data send from remote right after pickup the remote phone (e.g. initial
> "Hello?" sound like "o?" or even nothing)
>
> On FreeSwitch server I captured the VoIP data to the called VoIP-Phone
> on the sofia interface. Using wireshark it also shows that the voice
> data from remote is missed. Using Mobil phones or ISDN phones calling
> the same remote party there is never a bit missed.
>
> This problem occurs rare - once or twice per day and per local voip
> phone, but it's quite anoying.
>
> So is there a way to capture the correspondig ISDN voice data FS
> receives before it is transmitted via RTP or just droped? I want to c
> whether FS drops the early RTP packets or whether FS never got the data
> from ISDN.
>
>
>
> Sofia Profile is using
> <param name="inbound-late-negotiation" value="true"/>
>
> The dialplan portion is:
> <extension name="outgoing-pstn">
> <condition field="destination_number"
> expression="^94([0-9]+)$" break="never">
> <action application="privacy" data="full"/>
> <action application="set"
> data="effective_caller_id_name=anonymous"/>
> <action application="set"
> data="effective_caller_id_number=anonymous"/>
> </condition>
> <condition field="destination_number"
> expression="^([0-9]+)$">
> <action application="set"
> data="ignore_early_media=true"/>
> <action application="set"
> data="absolute_codec_string=PCMA"/>
> <action application="set"
> data="continue_on_fail=true"/>
> <action application="bridge" data="openzap/1/a/$1"/>
> <action application="transfer"
> data="${destination_number} XML et_internal_error"/>
> </condition>
>
> Any ideas to refine my debugging?
>
> regard
> helmut
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