T1 ISDN PRI pass through trunking?

[blanchae]

SOLVED - T1 ISDN PRI pass through trunking?

I have the following ISDN PRI configuration:

Cisco1 -> T1 -> PiaF -> T1 -> Cisco2

I can dial between either Cisco router to PiaF and from PiaF to Cisco with no problem. I can't dial through PiaF from Cisco1 to Cisco2. For each T1 span in /etc/asterisk/chan_dahdi_custom.conf, I have the context set to from-pstn. The response from PiaF is number not found.

I changed the context to from-internal and it appears that the trunks are being bridged but the line is dropped. Here's what a good call looks like when called from a SIP phone:

Dial("SIP/4032900001-09c8b1f0", "DAHDI/g1/5001,300,") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/5001

Here's what a call that is dropped looks like when sent from Cisco to Cisco and bridged:

Dial("DAHDI/12-1", "DAHDI/g1/5001,300,") in new stack
-- Requested transfer capability: 0x10 - - 3K1AUDIO
-- Called g1/5001

3k1AUDIO is for a fax call. How or where is the requested transfer capability being set for the ISDN PRI T1 channels?

 

[blanchae]

The correct term is Tandem not " pass through". I want to pass a telephone number from Cisco1 PBX through the Asterisk PBX (tandem) to Cisco2 PBX. How can I configure Asterisk to do that?

I have tried intra-company trunks and it didn't work either.

 

[blanchae]

I tried faxdetect=no in /etc/asterisk/chan_dahdi_custom.conf for both T1 configs and that didn't work either.

 

[blanchae]

Solved - Cisco misconfiguration!

It ended that when setting up the ISDN on the Cisco routers, I had configured the T1 channels as voice instead of modem.

Asterisk is smart enough to automatically transcode the incoming "voice" on the T1 line. Asterisk sends it out the T1 channels as a modem signal that allows auto negotiation of the calls which is the right thing to do for a T1 line. Basically, Asterisk is configured to use a type of "modem" channels and the Cisco Call Manager Express routers were configured for a type of "voice". When Asterisk forwarded the calls, it changed them to modem which Cisco said "What? I'm expecting voice" and dropped the call.

The offending Cisco router config line for the T1 was

isdn incoming-voice voice

It should be:

isdn incoming-voice modem

I also had to make each T1 line (there's 8 of them) to "from-internal" instead of "from-pstn". This way the incoming calls are treated as extensions as opposed to public trunks.

 

[blanchae]

For those that are interested.

I'm using a PiaF server with two quad T1 cards for a total of eight T1 lines. It is simulating the PSTN for the VoIP lab at SAIT Polytechnic. The students configure and connect their Cisco Call Manager Express routers to it to simulate the connection to the local phone system. We can have up to 96 (limited by the DSPs on our routers) simultaneous calls running between the 8 systems.

I have another PiaF server with just one T1 port connected to an Adtran channel bank with 16 FXS ports. We connected the Cisco CME routers to it to simulate connection to the PSTN using the router's FXO ports.

Later we'll connect to the PSTN simulations with eight PiaF servers. Then integrate Cisco Call Manager Express and PiaF together.

Currently,