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

[Martin Joseph]

On Jun 8, 2006, at 6:59 AM, M.Hockings wrote:

> 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.
Ok,  but be warned, I am an asterisk light weight...

SIP generally uses one port for negotiating the connection (usually 
5060) and then several/many other ports (usually in the 10000-20000) 
range for actual voice data.

This means that calls can "go through" but that issues with the sounds 
can occur due to blocked or unavailable ports in the needed range.
>
> 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.
OK, so your old handset connected to a Zap channel works ok?
> Also the softphones are on the same side of the firewall as the 
> asterisk box.
Hmmm.  Do the softphones work when you do the echo test (ext 600)?
>
> 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?
Usually asterisk will remain in the middle of an SIP connection, but it 
is possible to set it up so that it attempts to connect the two 
devices...  That could be an issue for you.
>
> 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?
I like IAX because it uses only 1 port for everything (4569).  This 
means your port issues are much simpler to troubleshoot and NAT issues 
are much more minor or non-existant.
>
> Again, thanks for any thoughts or suggestions as to how to get this to 
> work right.
>
Also, I use 3 different Voip call termination services 
(teliax,nufone.net,sellvoip.net) and all of them allow me to use IAX 
for passing calls to them.

Marty