similar to: [Asterisk-Dev] PRI Q.921 problem

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I've been trying lots of configurations now. And the problem that I can't solve is this: I have a Digium T205P card. I have connected one of the connections to our internal PBX (NEC 2000 IPS). The Asterisk is configured as pri_cpe, and the NEC is configured to be the network side of the connection. Both ends are using b-channels 1-15 and 17-31, the d-channel is on 16. When I start

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When inserting Ringing() before MeetMe()-conference picked up the call, everything works like a charm. I guess the PRI needed to see the ringing status before the call was answered. This is however never needed when dialing a SIP-extension or similar. I have also an update considering bad PRI b-channel numbering. It seems that only my first 15 channels actually work. Then our PBX tells Asterisk

Hello, if somebody using this scenario in production successfully, please send me info, which ISDN card for asterisk server is usefull for me (Digium, Sangoma)? my crucial requirement is "caller id name" transfer/display between ISDN (Siemens PBX) and IP phone connected to asterisk I'm using PRI interface and Q.SIG signaling. thank you PJ

We are in the process of installing a PRI line and we are going to connect it to an Asterisk Box. Verizon called us today to find out some information. I am surprised that they have never heard of Asterisk or Digium. But anyways, they needed some information in order to set up the circuit. Does the TE110P support NI1 or NI2? (I think the answer is both) What is the number of digits

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