[Freeswitch-users] 4 second delay

[Sean Holt]

Hey MC,
Did you by chance get an opportunity to look over my configuration in the
pastebin?

Thanks
Sean


On 6/15/10 12:45 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> Pastebin, please.
> 
> On Tue, Jun 15, 2010 at 12:19 PM, Sean Holt <sean at obscuradigital.com> wrote:
>> Should I put the freeswitch log into the pastebin or attach it?
>> 
>> 
>> On 6/15/10 11:11 AM, "Michael Collins" <msc at freeswitch.org
>> <http://msc@freeswitch.org> > wrote:
>> 
>>> Sure. Put all this information into pastebin.freeswitch.org
>>> <http://pastebin.freeswitch.org>  <http://pastebin.freeswitch.org> :
>>> 
>>> 
>>> Your topology, including the SIP provider, any routers/firewalls (including
>>> make & model), phone make & model
>>> Your relevant dialplan
>>> Console log of the call, with SIP trace turned on
>>> 
>>> You can also email me the .pcap file, but I don't want to review it until
>>> you've got all the above information posted so that we can get a good
>>> picture of what's happening...
>>> 
>>> Thanks!
>>> -MC
>>> 
>>> 
>>> On Tue, Jun 15, 2010 at 10:03 AM, Sean Holt <sean at obscuradigital.com
>>> <http://sean@obscuradigital.com> > wrote:
>>>> Hey MC, would you be willing to look over my tcpdump to help determine
>>>> where the failure is happening?
>>>> 
>>>> Sean
>>>> 
>>>> 
>>>> 
>>>> On 6/14/10 12:32 PM, "Michael Collins" <msc at freeswitch.org
>>>> <http://msc@freeswitch.org>  <http://msc@freeswitch.org> > wrote:
>>>> 
>>>>> Hop on the console, turn on siptrace and watch the call flow for clues.
>>>>> You might need to do a tcpdump capturing both signaling and media and then
>>>>> analyze in Wireshark to see what exactly is happening.
>>>>> -MC
>>>>> 
>>>>> On Sun, Jun 13, 2010 at 6:14 PM, Sean Holt <sean at obscuradigital.com
>>>>> <http://sean@obscuradigital.com>  <http://sean@obscuradigital.com> >
>>>>> wrote:
>>>>>> Hello list,
>>>>>> 
>>>>>> I¹ve been dealing with a particular issue with in-coming calls.  
>>>>>> Leg A calls into the office, then Leg B (endpoint) picks up call but
>>>>>> hears nothing on other side.  Wait 4 sec call completes and Leg B can
>>>>>> hear the other person.  
>>>>>> 
>>>>>> I have Centos 5.4
>>>>>> Latest git build
>>>>>> Polycom phones
>>>>>> 
>>>>>> Calling out is not a problem.
>>>>>> 
>>>>>> Not sure how to troubleshoot this issue or maybe there¹s a delay setting
>>>>>> in the sip profile that waits for the channel to complete.
>>>>>> 
>>>>>> Thanks for the help
>>>>>> Sean 
>>>>>> 
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>>>>>> 
>>>>> 
>>>>> 
>>>>> 
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>>> 
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> 
> 
> 
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