similar to: Asterisk 1.4.2 connection to Nortel CS1000M -followup with log

All, I am connecting to a CS 1000 nortel PBX. I can call out, I have limited success with call in. I get debug traffic that a call is coming in but I get the message "Unable to create/find channel". I was expecting that incoming calls over the trunk would be handled from my sip definition and goto the nortel context. It is not. Below is the actual incoming call debug information. I am

I am getting the following information and asterisk 1.0.7 is not answering the call. Any ideas? jerry ------------------ Sip read: INVITE sip:2828;phone-context=cdp.udp@qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:3173241052;phone-context=+1@qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed To:

Skipped content of type multipart/alternative-------------- next part -------------- Nov 21 14:17:47 VERBOSE[32580] logger.c: <-- SIP read from 172.25.103.222:5060: INVITE sip:1715;phone-context=exp_net.ascom@ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From:

Hello, thank you for the idea. I'm already behind the permission problem, which originally apeared, but I did overcome it by rules settings. this is the result for 'ls -l /dev/bus/usb/004/007': ---- crw-rw-rw- 1 root nut 189, 390 pro 23 13:23 /dev/bus/usb/004/007 ---- Os is Debian 6.0.5 Squeeze Regards Dee Dne 22.12.2014 v 13:01 nut-upsdev-request at lists.alioth.debian.org

Hi all, Following the thread from http://lists.llvm.org/pipermail/llvm-dev/2017-January/109029.html, I am sending this email to give a status on GlobalISel progress and situation. We are pushing GlobalISel from the state of prototype to a production quality framework. We welcome help with patches, reviews, feedbacks and so on. As explained during the last developer meeting, we are aiming at