asterisk users - Mar 2007 - Teliax problems, they say use SIP, more mature & better working than IAX

We have a Teliax IAX trunk that we use as an overflow for our four 
regular business lines into our local Asterisk PBX (Trixbox). We have 
our Teliax account set up so that it goes to a Teliax voicemail box if 
it cannot reach our Asterisk server, and we have the channel set up for 
5 simultaneous connections. Occasionally, calls are sent to the Teliax 
voicemail box for no apparent reason. In this case, we get this message 
in our local Asterisk log:

DEBUG[3129] chan_iax2.c: Raw Hangup 07.174.202.3:4569,src=0, dst=214

I've done tests on the weekend where I'm calling with one and two phones
continually. Once the local asterisk box picks up, I hang up. There is 
no other traffic at the time. One time I did 75 calls in about 25 
minutes, and 9 of them failed this way and went to voicemail. Another 
time, 88 calls went through fine, then 4 went straight to voicemail. 
Most of the time I was dialing from two phones simultaneously. In no 
case did one of the two calls go through while the other failed. However 
it was usually only seconds after a successful call, and seconds after 
calls would go through successfully again. During this period I was 
connected to the Asterisk box from a remote location (where I was 
placing the calls to the Teliax number), so I don't think the entire IP 
connection was going out. I was the only one placing any calls in or out 
of the Asterisk box during these tests.

The Teliax support person (note the IAX in the name) wrote: "You may 
want to try changing to SIP as it is a more mature protocol and seems to 
work better in general." Is it the general experience on the list that 
SIP is more mature and reliable than IAX? We like the fact that we don't 
have to open inbound ranges of ports for IAX to work. We are in Atlanta 
and we're connecting to a Teliax POP in Colorado. They've been talking 
about opening an Atlanta POP since at least April of last year. Do you 
think a provider with a closer point of presence would not have this 
problem so much? Considering the fact that we do see a message in the 
logs, and that we are able to keep up other IP streams, would you agree 
the problem is not our general IP connectivity? Any idea what in general 
causes the "Raw Hangup" message appearing in our logs and is it likely
to be on our end, their end, or either? 

Scott

On 3/19/07, Scott Plante <splante@insightsys.com>
wrote:
> work better in general." Is it the general experience on the list that
> SIP is more mature and reliable than IAX? We like the fact that we
don't
> have to open inbound ranges of ports for IAX to work. We are in Atlanta
I've switched to using SIP on our VoIP lines and noticed an overall
improvement in call quality, not necessarily based on any facts, just
general opinion.  We use Cisco firewalls and the Cisco takes care of
opening and closing the ports as required.

Scott Plante wrote:
> We have a Teliax IAX trunk that we use as an overflow for our four 
> regular business lines into our local Asterisk PBX (Trixbox). We have 
> our Teliax account set up so that it goes to a Teliax voicemail box if 
> it cannot reach our Asterisk server, and we have the channel set up 
> for 5 simultaneous connections. Occasionally, calls are sent to the 
> Teliax voicemail box for no apparent reason. In this case, we get this 
> message in our local Asterisk log:
>
> DEBUG[3129] chan_iax2.c: Raw Hangup 07.174.202.3:4569,src=0, dst=214
>
> I've done tests on the weekend where I'm calling with one and two 
> phones continually. Once the local asterisk box picks up, I hang up. 
> There is no other traffic at the time. One time I did 75 calls in 
> about 25 minutes, and 9 of them failed this way and went to voicemail. 
> Another time, 88 calls went through fine, then 4 went straight to 
> voicemail. Most of the time I was dialing from two phones 
> simultaneously. In no case did one of the two calls go through while 
> the other failed. However it was usually only seconds after a 
> successful call, and seconds after calls would go through successfully 
> again. During this period I was connected to the Asterisk box from a 
> remote location (where I was placing the calls to the Teliax number), 
> so I don't think the entire IP connection was going out. I was the 
> only one placing any calls in or out of the Asterisk box during these 
> tests.
>
> The Teliax support person (note the IAX in the name) wrote: "You may 
> want to try changing to SIP as it is a more mature protocol and seems 
> to work better in general." Is it the general experience on the list 
> that SIP is more mature and reliable than IAX? We like the fact that 
> we don't have to open inbound ranges of ports for IAX to work. We are 
> in Atlanta and we're connecting to a Teliax POP in Colorado.
They've
> been talking about opening an Atlanta POP since at least April of last 
> year. Do you think a provider with a closer point of presence would 
> not have this problem so much? Considering the fact that we do see a 
> message in the logs, and that we are able to keep up other IP streams, 
> would you agree the problem is not our general IP connectivity? Any 
> idea what in general causes the "Raw Hangup" message appearing in
our
> logs and is it likely to be on our end, their end, or either?
> Scott
>
I have come to the same conclusion about IAX a long time ago.

 I only use it when there is some sort of double NAT situation, where 
SIP absolutely will not work.  It really means something when TelIAX and 
ALSO IAX.CC both recommend using SIP.

Thanks,
Steve

On 19 Mar 2007, at 20:51, Scott Plante wrote:
> Raw Hangup
The code says:
                 /* A call arrived for a nonexistent destination.   
Unless it's an "inval"
                    frame, reply with an inval */

If you can, produce a packet dump of a failing call with ethereal - or
with iax2 debg, send it to me with your iax.conf  and I'll take a look.




Tim Panton

www.mexuar.net
www.westhawk.co.uk/

I really have lost loads of faith for IAX. "No authority found" and
"Rejected connect attempt" messages for no apparent reason. Sometimes
computability issues between asterisk versions. Fax/T.38 support?? But
I have no complaints about when it actually does work.

Not that Asterisk has the best SIP implementation either...