Question: Capturing VoiceData received from Sagnoma E1 card
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 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 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKZcpS4tZeNddg3dwRAnVDAKCxXXkdbf0RKeeSMFYucCIno3tA9gCfUzbD 148BfuKavTtBoJNScRQDmSk= =JbtY -----END PGP SIGNATURE-----
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 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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Helmut Kuper