Det

This version did the trick!! Yeah... my MoH is working again with the 9133i phones!! Thanks alot for the fast help adapting to the old style! Detlef

Really appreciate your fast workaround!!! (well, its not working yet - see earlier message with log)

Put the .2116 in this morning but there seems to be a bug - its still not working! It keeps asking for "Authentication Required" and reports "Password does not match". Below the logfile with a call from a X-Lite softphone to an Aastra 9133i when the Aastra was trying to put the X-Lite on hold: [9] 2007/10/15 07:43:51: SIP Rx udp:192.168.104.222:5060: INVITE sip:101@192.168.104.220:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bK24eefb4a0 Max-Forwards: 70 Content-Length: 265 To: "X-Lite Softphone" <sip:150@localhost>;tag=27498 From: "Aastra 9133i" <sip:101@localhost>;tag=48467520087b91d Call-ID: 1e63199b@pbx CSeq: 953138503 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: D.Schade <sip:101@192.168.104.222> Supported: replaces User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 v=0 o=MxSIP 0 369883472 IN IP4 192.168.104.222 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 3000 RTP/AVP 0 8 18 2 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [9] 2007/10/15 07:43:51: Resolve destination 190: a udp 192.168.104.222 5060 [9] 2007/10/15 07:43:51: Resolve destination 190: udp 192.168.104.222 5060 [9] 2007/10/15 07:43:51: SIP Tx udp:192.168.104.222:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bK24eefb4a0 From: "Aastra 9133i" <sip:101@localhost>;tag=48467520087b91d To: "X-Lite Softphone" <sip:150@localhost>;tag=27498 Call-ID: 1e63199b@pbx CSeq: 953138503 INVITE Contact: <sip:101@192.168.104.220:5060;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/2.1.0.2116 Content-Type: application/sdp Content-Length: 273 v=0 o=- 6791 6791 IN IP4 192.168.104.220 s=- c=IN IP4 192.168.104.220 t=0 0 m=audio 54610 RTP/AVP 0 8 18 2 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:18 g729/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [9] 2007/10/15 07:43:52: SIP Rx udp:192.168.104.222:5060: ACK sip:101@192.168.104.220:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bK00cb18415 Max-Forwards: 70 Content-Length: 0 To: "X-Lite Softphone" <sip:150@localhost>;tag=27498 From: "Aastra 9133i" <sip:101@localhost>;tag=48467520087b91d Call-ID: 1e63199b@pbx CSeq: 953138503 ACK Contact: D.Schade <sip:101@192.168.104.222> User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 [9] 2007/10/15 07:43:57: SIP Rx udp:192.168.104.129:1055: SUBSCRIBE sip:192.168.104.220 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport From: <sip:101@ims-va.com>;tag=2766 To: <sip:101@ims-va.com> Call-ID: zycdrj93q2jke09ah8rj CSeq: 10551 SUBSCRIBE Max-Forwards: 70 Contact: <sip:101@0.0.0.0:1055> Event: x-tapi Accept: application/x-tapi Expires: 3600 Content-Length: 0 [9] 2007/10/15 07:43:57: Last message repeated 2 times [9] 2007/10/15 07:43:57: Message repetition, packet dropped [9] 2007/10/15 07:43:57: Resolve destination 192: udp 192.168.104.129 1055 [9] 2007/10/15 07:43:57: SIP Tx udp:192.168.104.129:1055: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport=1055;received=192.168.104.129 From: <sip:101@ims-va.com>;tag=2766 To: <sip:101@ims-va.com>;tag=0b4a71dab2 Call-ID: zycdrj93q2jke09ah8rj CSeq: 10551 SUBSCRIBE User-Agent: pbxnsip-PBX/2.1.0.2116 WWW-Authenticate: Digest realm="ims-va.com",nonce="4f52fd692f715e2d6737406f9e800f2c",domain="sip:192.168.104.220",algorithm=MD5 Content-Length: 0 [9] 2007/10/15 07:43:57: SIP Rx udp:192.168.104.129:1055: SUBSCRIBE sip:192.168.104.220 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport From: <sip:101@ims-va.com>;tag=2766 To: <sip:101@ims-va.com> Call-ID: zycdrj93q2jke09ah8rj CSeq: 10552 SUBSCRIBE Max-Forwards: 70 Contact: <sip:101@0.0.0.0:1055> Event: x-tapi Accept: application/x-tapi Authorization: Digest realm="ims-va.com",nonce="4f52fd692f715e2d6737406f9e800f2c",response="067ab894d4b5db7e1cc860d426c2109d",username="101",uri="sip:192.168.104.220",algorithm=MD5 Expires: 3600 Content-Length: 0 [9] 2007/10/15 07:43:57: Resolve destination 193: udp 192.168.104.129 1055 [9] 2007/10/15 07:43:57: SIP Tx udp:192.168.104.129:1055: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport=1055;received=192.168.104.129 From: <sip:101@ims-va.com>;tag=2766 To: <sip:101@ims-va.com>;tag=0b4a71dab2 Call-ID: zycdrj93q2jke09ah8rj CSeq: 10552 SUBSCRIBE User-Agent: pbxnsip-PBX/2.1.0.2116 Warning: 399 ims-va.com Password does not match Content-Length: 0 [9] 2007/10/15 07:43:57: SIP Rx udp:192.168.104.129:1055: SUBSCRIBE sip:192.168.104.220 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport From: <sip:101@ims-va.com>;tag=2766 To: <sip:101@ims-va.com> Call-ID: zycdrj93q2jke09ah8rj CSeq: 10552 SUBSCRIBE Max-Forwards: 70 Contact: <sip:101@0.0.0.0:1055> Event: x-tapi Accept: application/x-tapi Authorization: Digest realm="ims-va.com",nonce="4f52fd692f715e2d6737406f9e800f2c",response="067ab894d4b5db7e1cc860d426c2109d",username="101",uri="sip:192.168.104.220",algorithm=MD5 Expires: 3600 Content-Length: 0 [9] 2007/10/15 07:43:57: SIP Tm udp:192.168.104.129:1055: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport=1055;received=192.168.104.129 From: <sip:101@ims-va.com>;tag=2766 To: <sip:101@ims-va.com>;tag=0b4a71dab2 Call-ID: zycdrj93q2jke09ah8rj CSeq: 10552 SUBSCRIBE User-Agent: pbxnsip-PBX/2.1.0.2116 Warning: 399 ims-va.com Password does not match Content-Length: 0 [9] 2007/10/15 07:43:57: Message repetition, packet dropped

Hello Specialists, I am looking to implement a VoIP PBX system for our world wide locations and I am wondering if and how pbxnsip would be used/recommended to accomplish this. General data: - 4 locations (3 US, 1 Mexico) that should use pbxnsip - 1 Headquater in Germany with Innovaphone IP305 as gateway (works with the pbxnsip demo already nicely) - each US/Mex location has about 4-8 PSTN lines and 20 extensions - all locations are connected via VPNs with each other - each location needs to maintain local PSTN numbers Questions: Anyone using a pbxnsip setup like that yet? Is it the best solution to place a pbxnsip at each location? How do multiple pbxnsip exchange data with each other (easy extension dialing)? How stable would be one central pbxnsip work? Thanks in advance Det