[Asterisk-Users] Re: SIP to SIP connection problem

[M.Hockings]

Martin Joseph wrote:
> 
> On Jun 7, 2006, at 6:55 PM, M.Hockings wrote:
> 
>> I have a small asterisk setup here with one POTS line, one VOIP SIP 
>> connection an FXS connection to the house phones and a bunch of 
>> softphones.  Local calls are routed out through the POTS line and long 
>> distance through the VOIP line.  This works great for the old house 
>> phones but the softphones on the computers can only make local calls. 
>> That is any attempt to connect through the VOIP line end in silence as 
>> soon as the called party picks up and asterisk attempts to connect the 
>> VOIP SIP connection and the softphone SIP connection.  This is using 
>> xTen softphones on Linux and Windows.
>>
>> I was thinking that it might have to do with mismatched codecs or some 
>> such?  In the [general] section of the sip.conf I see that freePBX has 
>> put
>>
>> disallow=all
>> allow=ulaw
>> allow=alaw
>>
>> and none of the softphone definitions set any different requirements.
>>
>> If I connect a softphone directly to the VOIP provider it appears to 
>> use the g711u codec.
>>
>> This is all using asterisk 1.2.9.1 and freepbx 2.1.1 running on CentOS 
>> 4.3.
>>
>> Thanks for any suggestions.
>>
> Sounds more like a port issue to me.  Looking in the asterisk Console 
> and setting verbosity up when attempting these calls might give you more 
> info.
> 
> Also,  you might try using an IAX softphone instead, as these are much 
> less of a hassle in my opinion.  There are several available.
> 
> Marty

Hi Marty, can you expand on the "port issue" a bit.  I will admit that 
my understanding of sip connection handling is still a bit weak yet.

I can say that the VOIP provider is on the far side of a firewall from 
the asterisk box and seems to work OK when talking to an old phone on a 
Digium connection. Also the softphones are on the same side of the 
firewall as the asterisk box.

Is this a case that asterisk is trying to directly connect the VOIP sip 
connection and the softphone sip connection to each other or do both 
connect to asterisk and it manages the data flow between the two?

So, I'm not sure how an IAX softphone would help other than forcing 
asterisk to be between the voip sip and the softphone iax connection?

Again, thanks for any thoughts or suggestions as to how to get this to 
work right.

Mike