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zgor
10-16-2010, 05:57 AM
Hello !

I m using a Epygi Quadro Dual E1/T1 directly connected to a Legacy PBX via E1 PRI.

In prior situations, we usually work using Epygi's device to make outbound calls using a SIP Provider, calls from the standard PSTN Provider (trunk2) are received and bridged directly to the trunk1 (Legacy PBX), working perfectly :)

But in this case, I also want to receive calls from SIP Provider but im missing something ...

For testing, I registered the Epygi to an external Asterisk box, o i can debug what SIP messages are sent. Epygi and Asterisk are with directly public IPs on the nic (no NAT at all).

I registered an extension (Users=>Extensions Management) to the Asterisk Box, register perfectly. Then, in Call Routing I setup => Route all Incoming SIP calls to routing table. In routing table I have only 1 route as: call type E1 and filter inbound type (From SIP, IP address of the Asterisk Box, and * as pattern).

When I call the Epygi i can confirm:
-the invite is sent to the same username name as register ("test") as normal.
-the Epygi always replys with a: 404 Not Found
=> Its not routing this incoming call the the E1 port.

¿Any ideas? ¿Im am making something really bad?

Really thanks for your help!


Gorka.

hrant
10-18-2010, 04:12 AM
Remove the registration of Quadro extension on the Asterisk and it should work as needed after that. Actually, you don't need any extension on Quadro, just send the INVITE from Asterisk to Quadro's IP address. With the call routing record configured as described above, Quadro will send all the calls from SIP to E1.

zgor
10-18-2010, 11:34 AM
Hello Hrant,

Thanks for your answer :) !

By the way, I need to register, in this test case at Asterisk i know the Epygi's IP address, but in the normal situation I need to register the Epygi GW with the ITSP SIP provider (so the provider know "where" i'am).

Any idea?

KSComs
10-18-2010, 04:49 PM
In some cases when I have had to work with the likes of Asterisk sip servers and a Quadro, I have had to go to the following page and put a check next to accept stray sip requests ...


http://epygiipaddress/generalconfig.cgi


And that usually sorts the registrations out ... the only other thing is the rtp ...


Hope this helps....


Kevin

PS - you may not have to do this with a sip registra - it really depends on which flavour it is.

zgor
10-19-2010, 06:35 AM
Hello,

Yes, I found on some thread here this option, i have this activated.
By the way, the Asterisk Box is only for testing, in the real scenario I register to an ITSP and they have an standard Kamailio based platform.

I cannot understand why if I register to the ITSP I cannot reveive calls from them :(

Its because its a gateway ?

Really thanks!

hrant
10-19-2010, 09:23 AM
Registration or it being a gateway are not the reasons of the failed calls. Probably something is wrong in configuration of the call routing. Enable the "developer logging" on Quadro, make a test inbound call and attach the logs to this thread. I'll try to help you to fix the issue.

zgor
10-20-2010, 03:18 AM
Hello Hrant,

Thanks for your help. We are building an small lab here with same hardware (dual e1 gw) and 2 boxes (sip provider, asterisk with tdm hardware), so we can debug on all points and confirm.

I will upgrade this thread with logs and info.


thanks again,

Gorka.

hrant
10-20-2010, 04:57 AM
that's fine, Gorka. Keep posting, thanks.

zgor
10-20-2010, 02:30 PM
Hello Hrant,

I finally found it :)

When using our SIP Provider, RTP Audio / SIP Proxy always come from / got to IP-A.

But, when receiving incoming calls from this ITSP, calls are efectly received by an TDM GW with IP-B and then (its a public PSTN DID to SIP service).

So, when Epygi its registered to the ITSP and we want to receive calls, SIP INVITE messages are sent from IP-Á (IP source of the packets) but, the sip header From: contains the IP-B.

The call routing table with inbound filter, looks for the From: @XXX header, not the ip source of the packet. I didnt know this before.

I dont know if its a bug/feature, the "Inbound host" looks like IP Address instead of SIP From.


Its working now, really thanks for your help.

Thanks for your help.