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Not dial tones recognized - IVR not responding? lost call after 20 seconds? what could be?

csierra
Posts: 85
 Member Since:
2008-02-22
Submitted by csierra on Tue, 08/12/2008 - 12:42am.

Hi all,

At this point I can emulate an inbound call from any SIP peer (local or remote) with great success; IVR responds just fine, call queues, ring groups, etc. So I went to look for a DID to purchase and found DIDWW.COM, they have a 'Try Now' applet in their page, and they provided instructions as per how to configure the SIP Trunk, wich is:
PEER Details:
============
host=204.11.194.40
insecure=very
dtmf=rfc2833
dtmfmode=rfc2833
disallow=all
allow=ulaw&alaw
type=peer
============

User Context/Details:
============
secret=
type=user
context=from-trunk
============

Then I made my very first incoming call from a POTS; and worked (at half) It did picked up, I can here the announcement, the sound is pretty choppy, and what happens keeps me guessing... the call is cut and dropped at exactly 20 seconds after the pickup, and, and...
No dial tones (callee input) is detected by the IVR; this intrigues me more because when calling from a SIP peer I dont have this issues, IVR, callee input, dial tones, time outs, everything works fine. I have investigated with DIDWW.COM just to make sure this 20 secs cut out is not a demo limit; and it´s not, so, what should I look for to start debugging this? s this a bug? can this be related with a boot warning that I have -ZT_LOADZONE failed (I guess this is because I have a super novice purchase Tiger Chipset card from ZapMicro and it stills not configured...

Thanks again.

‹ Fritz PCI card Failover trunk taking 30 secs to work ›
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VOIP Newbie


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csierra
Posts: 85
 Member Since:
2008-02-22
It was not the firewall indeed
Sat, 08/30/2008 - 9:36pm

What were causing a conflic, and I don´t know if this is already documented but, this causes trixbox not to work properly; I read a post here http://pbxinaflash.com/forum/archive/index.php?t-267.html
where what is described actually causes a conflict and when those changes are done the conflic is gone.

It consist of a confusion when we use the same host as the server host and the same host name is indicated in the hosts files where Trixbox adds yourpseudo.dyndns.org pointing to 127.0.0.1 and at the same time, the sip_nat.conf uses externhost=yourpseudo.dyndns.org and that makes the remote SIP peers not to have audio, or dropping calls at 20 secs. I simply changed that entry in /etc/hosts from yourpseudo.dyndns.org to generichost or any other name but not the externhost name.

Hope this brings in some light to someone experiencing the same issues in the future.

As of all of you who guided and advised me, thank you again! pfSense not for now, my Thomson Router is doing like a charm (so far)

Thanks again

VOIP Newbie

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VOIP Newbie

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VOIP Newbie


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TDF
Posts: 483
 Member Since:
2006-12-19
So is this why changing
Sat, 08/30/2008 - 10:39pm

So is this why changing from a dyndns name to a no-ip one worked for my 20 sec audio problem. After changing my host file has the dyndns addy in it, sip.nat has the no-ip one and no more problem.


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AndyN
Posts: 16
 Member Since:
2006-08-16
Why is this so hard (SIP trunks and NAT) ??
Tue, 09/16/2008 - 8:19pm

I have a SIP provider that provided me with simple credentials.

I set my NAT to forward the SIP control/SIP audio port ranges to a private fixed IP address.

I have a "cheap" Grandstream Budgetone phone (set to the private fixed IP address) that i configured with the SIP provider credentials for Server IP/Domain, SIP user ID, Authenticate ID, and Authenticate Password; i also selected NAT Traversal and entered my Internet IP address into "Use NAT IP". Bottom line: the Grandstream makes calls thru the SIP provider reliably for as long as i want to call.

Move to Trixbox set to the same fixed IP address (phone is unplugged). I edited sip_nat.conf with appropriate externip & localnet values, with qualify=yes and canreinvite=no.

Outbound Settings for the SIP trunk are ..

fromuser=(credentials)
type=peer
secret=(credentials)
qualify=yes
insecure=very
host=(provider host domain)
fromdomain=(provider host domain - should this be my domain? but using the provider domain i can dial ... using my domain i can't)
nat=yes
canreinvite=no
disallow=all
allow=ulaw&alaw&ilbc&g729&gsm

i can dial out, the phone rings, and we all hear each other. BUT THE CALL DROPS after 15-20 seconds ...

per the above comments the contents of trixbox /etc/host is trivial, one line: "order hosts,bind" - so not much to change here.

thoughts? (why is trixbox so hard when the budgetone does this so easily?)

thanks - AndyN.


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SkykingOH
Posts: 9541
 Member Since:
2007-12-17
Don't use all of those
Tue, 09/16/2008 - 10:22pm

Don't use all of those CODEC's for one thing. The Budgetone probably does NAT a little better than Asterisk is why it is working better.

Do you have audio for the 20 seconds before it drops?

There is an RTP debug command from the Asterisk CLI (Asterisk -r) that is very useful in debugging these type of problems.

Scott

--

Scott

aka "Skyking"


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AndyN
Posts: 16
 Member Since:
2006-08-16
Possible solution
Wed, 09/17/2008 - 6:45am

... to my SIP trunk drops after 20 seconds problem ... my provider has given me an outbound proxy which i added to my Outgoing Settings PEER Details (outboundproxy=ip address). works for the moment ...

AndyN


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