[Spce-user] 483 Too Many Hops

[Marc Rys]

ok I'll stop changing the subject, even though I regret not naming it SPCE Taqua Integration.. 

Anyways, I've heeded your advice and I normalized the patterns and I'm now routing the calls from my Taqua PSTN gateway to my IP phones.  Now the newest problem to reveal itself is the incoming calls to my Phones fail to setup completely.  It appears all the call routing is working properly now,  I can call my DID from the PSTN, and my IP phone rings, but when I pickup the call, I hear no audio, and it appears the phone never completely sets up the call.  The screen still shows the phone as ringing.

The wireshark cap shows RTP moving between my gateway & SPCE, and RTP moving between SPCE and my IP phone, but the call never appears to setup completely on the IP phone.

Any thoughts?

Marc Rys 
http://www.tri-lakes.net 
http://www.rystec.com 


----- Original Message -----
From: "Lorenzo Mangani" <lorenzo.mangani at gmail.com>
To: "Marc Rys" <m.rys at tri-lakes.net>
Cc: spce-user at lists.sipwise.com
Sent: Saturday, March 23, 2013 4:39:50 PM
Subject: Re: [Spce-user] 483 Too Many Hops

Marc, 


Please don't fork the messages by creating a new thread for each step of the discussion and consult the documentation. 
You need an inbound rewrite rule applied to strip the + from the INVITE in order to match the local user, as well described in the Handbook, actually this is exactly the example shown there: http://www.sipwise.com/doc/2.7/spce/ar01s06.html#dialplans 



Lorenzo Mangani 



HOMER DEV TEAM 
QXIP - Network Engineering 


On Sat, Mar 23, 2013 at 9:55 PM, Lorenzo Mangani < lorenzo.mangani at gmail.com > wrote: 


Marc, 


Your Tarqua invite has Max-Forwards set to 1. 
Try increasing the allowed hops and the call will terminate. 





Lorenzo Mangani 



HOMER DEV TEAM 
QXIP - Network Engineering 



On Sat, Mar 23, 2013 at 9:21 PM, Marc Rys < m.rys at tri-lakes.net > wrote: 




I've been evaluating SPCE over the last week. I've already setup a couple test subscribers and setup a peer with a provider we work with for SIP LD Term. All of those test have worked very successful. 

We also have our own media gateway which is interconnected with the local PSTN via TDM trunk, but I send incoming calls from the PSTN through our MediaGateway to SPCE, SPCE is responding back with 483 "Too Many Hops". Attached is the wireshark cap. 

Any help is appreciated. 

Marc Rys 
http://www.tri-lakes.net 
http://www.rystec.com 
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