VoIP Mailing List Archives

[helmut.kuper at ewetel.de]

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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 at ewetel.de]

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Hello,

any ideas?

regards
Helmut


On 21.07.2009 16:01, Helmut Kuper wrote: