roger

I have upgraded a phone system from 2.0 to 3.1.1. It use to be that we place a call on hold and another extension will be able to pick it up.. Now on version 3.1.1 this cannot be done, is there some thing I'm missing? Or we have to use Park Orbit now? Thanks

Thanks, it works!!!

Yes I have added the number as an alias and it dont work. Yes it will forward all call to that extension. the log is below. Egix. thanks [7] 2008/12/16 21:23:01: SIP Rx udp:209.131.220.220:62328: INVITE sip:egix@66.158.175.178:5060;line=a87ff679;transport=udp SIP/2.0 Via:SIP/2.0/UDP 209.131.220.220;branch=z9hG4bK-BroadWorks.as01-66.158.175.178V5060-0-559967328-308845606-1229480581310- From:"ROGER WATSON CA"<sip:31734218@209.131.220.220;user=phone>;tag=308845606-1229480581310- To:"3176604804 3176604804"<sip:3176604804@egix.net;line=a87ff679> Call-ID:BW212301310161208-520487878@209.131.220.220 CSeq:559967328 INVITE Contact:<sip:209.131.220.220:5060> Supported:100rel Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Accept:multipart/mixed,application/media_control+xml,application/sdp Max-Forwards:10 Content-Type:application/sdp Content-Length:281 v=0 o=BroadWorks 77316568 1 IN IP4 209.131.222.201 s=- c=IN IP4 209.131.222.201 t=0 0 m=audio 16574 RTP/AVP 0 100 101 c=IN IP4 209.131.xxx.201 a=rtpmap:0 PCMU/8000 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 [5] 2008/12/16 21:23:01: Identify trunk (line match) 4 [9] 2008/12/16 21:23:01: Resolve 180: aaaa udp 209.131.xxx.220 5060 [9] 2008/12/16 21:23:01: Resolve 180: a udp 209.131.220.xxx 5060 [9] 2008/12/16 21:23:01: Resolve 180: udp 209.131.220.xxx 5060 [7] 2008/12/16 21:23:01: SIP Tx udp:209.131.xxx.220:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.131.xxx.220;branch=z9hG4bK-BroadWorks.as01-66.158.175.178V5060-0-559967328-308845606-1229480581310- From: "ROGER WATSON CA" <sip:3178334218@209.131.xxx.220;user=phone>;tag=308845606-1229480581310- To: "3176604804 3176604804" <sip:3176604804@egix.net;line=a87ff679>;tag=5706a1ee13 Call-ID: BW212301310161208-520487878@209.131.220.220 CSeq: 559967328 INVITE Content-Length: 0 [5] 2008/12/16 21:23:01: Trunk 3176604804 sends call to egix in domain localhost [5] 2008/12/16 21:23:01: Trunk call: Could not identify user [9] 2008/12/16 21:23:01: Resolve 181: aaaa udp 209.131.220.220 5060 [9] 2008/12/16 21:23:01: Resolve 181: a udp 209.131.220.220 5060 [9] 2008/12/16 21:23:01: Resolve 181: udp 209.131.220.220 5060 [7] 2008/12/16 21:23:01: SIP Tx udp:209.131.220.220:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 209.131.220.220;branch=z9hG4bK-BroadWorks.as01-66.158.xxx.178V5060-0-559967328-308845606-1229480581310- From: "ROGER WATSON CA" <sip:3178334218@209.131.220.220;user=phone>;tag=308845606-1229480581310- To: "3176604804 3176604804" <sip:3176604804@egix.net;line=a87ff679>;tag=5706a1ee13 Call-ID: BW212301310161208-520487878@209.131.220.220 CSeq: 559967328 INVITE Contact: <sip:egix@66.158.xxx.178:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.1.3023 Content-Length: 0

I Got a CS410, I have it configured to register againt a SIP trunk, every thing work normal, the PBX regiter fine... all good. when I call the Trunk ID number Example 317-8888888 it said you have reached a non working number, but If I go to the Trunk configuartion tab and scroll down to the option foward all call to and put and extesion number The PBX will forwad the incoming call to the assigned extension. Is like the PBX do not know what to do with the incoming call, Yes I have assigned the Number to the AA. I have traced the call and it arriving to the PBX but it not going any were. Please Help.

before I have updated my CS410 runing version 2 you could perform a factory restart.. that can be done on version 3?

Yes I have restarted the computer and phone. and yes I have used wireshark to check the packages, the CS410 is not reponding to the DHCP requests, is already configured on the lan interface.

I have tried to set up DCHP on my CS410 vesion3.01 and is not providing any ip address to any type of device... I have set it up on the wan interface and the lan interface and still nothing any Idea? Thanks

I have updated my CS410 to the latest firm ware version 3.01 and I want to do a factory restart how I do that. thanks.

I got a Polycom IP sound Station 4000 already register and it can make call but when I call this phone I cannot pick up the incoming call I'm running version3.0.2. I have tried several things to make this IP 4000 work with the PBX N SIP and it still doesn't work, I don't know what is the problem. I have down graded the firmware to 2.2.2 and it does not register to the PBX if I use the 3.0 it register fine. But I can (pick up) answer any calls. any thoughts? Roger Watson AllthingsIT