Detlef Posted October 9, 2007 Report Share Posted October 9, 2007 Since I upgraded to PBX 2.1.0.xxxx my MOH is not working no more. Now after some testing I think it is the Aastra phone causing this. If I use an extension with a softphone X-Lite and call another extension with an Aastra 9133i phone and put the call on hold from the X-Lite softphone then the Aastra plays music on hold (from default file). Do I do it the oposite way and put the call on hold from the Aastra 9133i phone then its silent and no MOH plays. The same happens when I call in over the PSTN gateway. If I call the extension with the X-Lite softphone and put that call on hold from the softphone then the caller hears music on hold. If I call the extension with the Aastra phone and put it on hold then its silent to the caller over the PSTN line. Is that a setting in the Aastra phones that I need to correct? It was working with PBXnSIP 2.0.3.1715 and quit with the new 2.1.0.xxxx versions. Since the X-Lite softphone does it correctly I think it has to do with the Aastra phones themself. Currently I am running PBXnSIP 2.1.0.2114 and the Aastra 9133i is firmware 1.4.2.1081 Detlef Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 9, 2007 Report Share Posted October 9, 2007 Works here, using PBX 2.1.0.2115 and Aastra 57i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5. Maybe you need to upgrade the PBX to the 2115 build. Quote Link to comment Share on other sites More sharing options...
Detlef Posted October 9, 2007 Author Report Share Posted October 9, 2007 Works here, using PBX 2.1.0.2115 and Aastra 57i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5. Maybe you need to upgrade the PBX to the 2115 build. Will do tonight, but since I upgraded to any previous 2.1.0.xxxx version the 9133i phones here quit doing MOH. Going back to 2.0.3.1715 and the MOH always started working again. What I dont understand is that the X-Lite softphone does it correct. I went to the newst Aastra firmware available for download but no change - still no MOH. Below my MAC.CFG everything else on the Aastra settings is left to default: download protocol: TFTP tftp server: pbx.ims-va.com auto resync mode: 3 auto resync time: 03:00 time server disabled: 0 time server1: dc.ims-va.com sip digit timeout: 3 sip dial plan: "x+#|xx+*" sip mode: 0 sip proxy ip: pbx.ims-va.com sip proxy port: 5060 sip registrar ip: pbx.ims-va.com sip registrar port: 0 sip registration period: 3600 sip screen name: User Name sip user name: 101 sip display name: User Name sip auth name: 101 sip password: xxxxxx sip vmail: *97 directory 1: companylist.csv directory 2: externallist.csv Quote Link to comment Share on other sites More sharing options...
Detlef Posted October 9, 2007 Author Report Share Posted October 9, 2007 Works here, using PBX 2.1.0.2115 and Aastra 57i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5. Maybe you need to upgrade the PBX to the 2115 build. Ok done, upgraded to .2115 and the 9133i Aastra phones still do not initiate MOH when putting someone on hold or parking a call. Is someone using 9133i phones who has it working? Maybe we could exchange config files??? Would be really appreciated!! Detlef Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 10, 2007 Report Share Posted October 10, 2007 Do you have the SIP INVITE that tells the PBX to hold the call? Maybe the phone is using the old style with 0.0.0.0 (which was obsoleted in June 2002, see RFC 3264)? Quote Link to comment Share on other sites More sharing options...
Detlef Posted October 10, 2007 Author Report Share Posted October 10, 2007 Do you have the SIP INVITE that tells the PBX to hold the call? Maybe the phone is using the old style with 0.0.0.0 (which was obsoleted in June 2002, see RFC 3264)? This is the logfile I see as soon as I press the HOLD button on the Aastra phone during a connected call with the X-Lite softphone: 192.168.104.222 = Aastra Phone 192.168.104.129 = X-Lite Softphone 192.168.104.220 = PBXnSIP [9] 2007/10/10 08:55:20: 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=z9hG4bKdd007bf64 Max-Forwards: 70 Content-Length: 266 To: "X-Lite" <sip:150@localhost>;tag=43086 From: "Aastra 9133i" <sip:101@localhost>;tag=7fb09db3322e24a Call-ID: e4afabe4@pbx CSeq: 69380235 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: Aastra <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 1152796867 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] 2007/10/10 08:55:20: Resolve destination 2684: a udp 192.168.104.222 5060 [9] 2007/10/10 08:55:20: Resolve destination 2684: udp 192.168.104.222 5060 [9] 2007/10/10 08:55:20: SIP Tx udp:192.168.104.222:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bKdd007bf64 From: "Aastra 9133i" <sip:101@localhost>;tag=7fb09db3322e24a To: "X-Lite" <sip:150@localhost>;tag=43086 Call-ID: e4afabe4@pbx CSeq: 69380235 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/2.1.0.2115 Content-Type: application/sdp Content-Length: 275 v=0 o=- 34981 34981 IN IP4 192.168.104.220 s=- c=IN IP4 192.168.104.220 t=0 0 m=audio 64154 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] 2007/10/10 08:55:20: 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=z9hG4bK31b41b11a Max-Forwards: 70 Content-Length: 0 To: "X-Lite" <sip:150@localhost>;tag=43086 From: "Aastra 9133i" <sip:101@localhost>;tag=7fb09db3322e24a Call-ID: e4afabe4@pbx CSeq: 69380235 ACK Contact: Aastra <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 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 10, 2007 Report Share Posted October 10, 2007 Oh yea, they are using the 0.0.0.0. I am not sure if we should still support this more than 5 years old "workaround" - isn't there a SW upgrade for the phone available? Quote Link to comment Share on other sites More sharing options...
Detlef Posted October 10, 2007 Author Report Share Posted October 10, 2007 Oh yea, they are using the 0.0.0.0. I am not sure if we should still support this more than 5 years old "workaround" - isn't there a SW upgrade for the phone available? I just put the latest and greatest firmware on the phone that was available from their website... AASTRA TELECOM INC. June 2007 Generic SIP Firmware 1.4.2.1081 GA Release. FC-000032-01-11 480i FC-000040-00-11 480iCT FC-000046-01-11 9133i FC-000058-01-11 9112i Maybe I can have them change it if I contact their customer support and ask for the correct ON HOLD procedure? What would I need to tell them to see if they would change it to the up-to-date way? EDIT: I just sent them an email... lets see what they will say Quote Link to comment Share on other sites More sharing options...
Detlef Posted October 10, 2007 Author Report Share Posted October 10, 2007 Nice, I got a reply from Aastra. They have this problem on their feature request list... hopefully not since 5 years!! From: Layne Monson [mailto:layne.monson@aastra.com]Sent: Wednesday, October 10, 2007 5:33 PMTo: Detlef Schade Subject: IP11755: Aastra 9133i - ON HOLD Problem with 0.0.0.0 INVITE Detlef, This had already been submitted as a feature request for these phones . Other methods are not supported right now. Thanks, Layne Monson CCNA Customer Support Engineer II Aastra Intecom 2811 Internet Blvd. Frisco, TX 75034 layne.monson@aastra.com 469 365 3847 direct EDIT: Oh, just got an accurate estimate - its even worse than 5 years, they have no clue when they will update their phones: There is no way I can say how long it would be. It?s totally up to upper management to decide what features and fixes go into each load and allocate dev resources. Sorry , there is nothing I can do further. Thanks, Layne Monson CCNA Customer Support Engineer II Aastra Intecom 2811 Internet Blvd. Frisco, TX 75034 layne.monson@aastra.com 469 365 3847 direct Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 11, 2007 Report Share Posted October 11, 2007 Hmm. Maybe it is easier to program around it and support that old style as well. Quote Link to comment Share on other sites More sharing options...
Detlef Posted October 11, 2007 Author Report Share Posted October 11, 2007 Hmm. Maybe it is easier to program around it and support that old style as well. I would really appreciate that, currently I am stuck with the new PBX not supporting it any more and Aastra not knowing if or when they update their phones... the pain ist that it is silent to the caller if you put someone on hold with the Aastra phone and they think the line dropped and mostly hang up. I guess their latest firmware is used in all 4 of those phones listed in the release info? FC-000032-01-11 480i FC-000040-00-11 480iCT FC-000046-01-11 9133i FC-000058-01-11 9112i Quote Link to comment Share on other sites More sharing options...
Detlef Posted October 12, 2007 Author Report Share Posted October 12, 2007 Do you already know if you will have a PBX version 2.1.0.2116 that will support this historic on hold procedures again?? If I wouldn't have bought 40 of those stupid Aastra phones over the past couple months I wouldnt be so desperate Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 13, 2007 Report Share Posted October 13, 2007 FC-000032-01-11 480i FC-000040-00-11 480iCT FC-000046-01-11 9133i FC-000058-01-11 9112i Unfortunately, we have only a 57i here... Interestingly, this one uses 0.0.0.0 and a=sendonly in the same SDP. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 13, 2007 Report Share Posted October 13, 2007 Do you already know if you will have a PBX version 2.1.0.2116 that will support this historic on hold procedures again?? Okay there is a 2116 version (http://www.pbxnsip.com/download/pbxctrl-2.1.0.2116.exe), and we did a small change in the 0.0.0.0 detection, but we had no chance of trying it out. Please give it a try. Quote Link to comment Share on other sites More sharing options...
Det Posted October 15, 2007 Report Share Posted October 15, 2007 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 Quote Link to comment Share on other sites More sharing options...
Det Posted October 15, 2007 Report Share Posted October 15, 2007 Okay there is a 2116 version (http://www.pbxnsip.com/download/pbxctrl-2.1.0.2116.exe), and we did a small change in the 0.0.0.0 detection, but we had no chance of trying it out. Please give it a try. Really appreciate your fast workaround!!! (well, its not working yet - see earlier message with log) Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 16, 2007 Report Share Posted October 16, 2007 Okay, we did a simulation without the a= header on our phone here and maybe the following version fixes that problem: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2117.exe We found something that would explain why the 0.0.0.0 method would not work. Please verify. Quote Link to comment Share on other sites More sharing options...
Det Posted October 16, 2007 Report Share Posted October 16, 2007 Okay, we did a simulation without the a= header on our phone here and maybe the following version fixes that problem: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2117.exe We found something that would explain why the 0.0.0.0 method would not work. Please verify. 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 Quote Link to comment Share on other sites More sharing options...
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