Ah, a license thing. But yes the debugging is powerful. ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: Paul Scheepens <pscheep...@epo.org> To: Tony Graziano <tgrazi...@myitdepartment.net> Cc: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> Sent: Thu Nov 12 10:16:35 2009 Subject: Re: [sipx-users] Intermittent no-audio problem with Ericsson PBX Hi Tony, Just for info: I found the right debugging command and it helped..... debug mediagateway all gives a VERY nice "graphical" representation (ASCII drawing) of the complete call-setup process. Then I discovered calls were working on incoming channel 1-15 and not on 16 to 30. I bought the cheapest Patton 4960 and...it only supports 15 channels. One plus one is two, so I tested. The first calls worked. As soon as channel 16 was reached utter silence up to channel 30. Then channel 1 was used again and audio was OK. I have added the following to my Patton config: channel-range 0 15 That should do the job. As the debug was so beauti- and helpful I attached it. Best regards / Mit freundlichen Grüßen / Sincères salutations Paul Scheepens Tony Graziano <tgrazi...@myitdepartment.net> wrote on 11-11-2009 17:38:12: > From: > > Tony Graziano <tgrazi...@myitdepartment.net> > > To: > > Paul Scheepens <pscheep...@epo.org> > > Cc: > > sipx-users@list.sipfoundry.org > > Date: > > 11-11-2009 17:38 > > Subject: > > Re: [sipx-users] Intermittent no-audio problem with Ericsson PBX > > Normally dont see framing as crc4, but you are not here [image removed] . > > Guessing the only reason you have WAN is for radius and login from > your network, correct? I think I should assume you have no DNS SRV > issues, since it "has worked" for you thus far. > > I don't see anything "wrong" in the config. Have you checked to see > if your 5.x version is the most recent? > > (hint: timing can also be checked from the webgui pretty easily) > On Wed, Nov 11, 2009 at 11:21 AM, Paul Scheepens <pscheep...@epo.org> wrote: > Hi Tony, > > I know the debug option in the Patton and used it in the past. > I also used Patton support, very helpfull people. > The problem is that I don't know what to ask and whether it is a > Patton issue or an Ericsson issue or..... > > Because both the Ericsson phone and the softphone (X-lite/Bria/ > zoiper) think that > a call is established I am not sure what debugging to enable without > drawning in the output. > I 've got the feeling that Patton and Ericsson are not registering > any problems because the call stands. > > I would need a command (debug or not) to see that a voice-channel > from the E1 is transfered to an > RTP stream on the IP side (and vice-versa). > > I checked the timing, from the log: > > 2009-11-11T13:07:27 : LOGINFO : Link up on interface e1t1 0 0 0 > 2009-11-11T13:07:27 : LOGINFO : Link up on interface ISDN 0 0 > 2009-11-11T13:07:27 : LOGINFO : Warm start. > 2009-11-11T13:07:28 : LOGINFO : Link up on interface ethernet 0 0 1 > 2009-11-11T13:07:28 : LOGINFO : Link up on interface LAN > 2009-11-11T13:07:23 : LOGINFO : Time: set clock from > 2009-11-11T13:07:30 to 2009-11-11T13:07:23 > 2009-11-11T17:43:21 : LOGINFO : Link up on interface e1t1 0 0 0 > 2009-11-11T17:43:21 : LOGINFO : Link up on interface ISDN 0 0 > 2009-11-11T17:43:21 : LOGINFO : Warm start. > 2009-11-11T17:43:22 : LOGINFO : Link up on interface ethernet 0 0 1 > 2009-11-11T17:43:22 : LOGINFO : Link up on interface LAN > > After the first reboot the time was adjusted by 7 seconds. > Everything worked fine. > I tried to reproduce the problem. > I succeeded by calling from SIP to an Ericsson which was forwarded > (back) to SIP. > > After the second reboot the time was probably still OK. > Everything worked fine again. (Un)fortunately I still cant reproduce > the problem. > > Time seems to be synched according to the Patton: > > # show system-clock > > Current clock source > ==================== > > e1t1 0 0 0 > > Registered clock sources > ======================== > > Name Sync > e1t1 0 0 0 X > internal X > > I always see the X under Sync, also when I have the problem. > > The PBX is ours BTW, but it is managed by another department who > have outsourced the whole configuration part: not easy! > > Fortunately there is no NAT or FW in this setup, just IP and E1. > > I have attached the Patton cfg > > Best regards / Mit freundlichen Grüßen / Sincères salutations > > Paul > > Tony Graziano <tgrazi...@myitdepartment.net> wrote on 11-11-2009 16:32:45: > > > From: > > > > Tony Graziano <tgrazi...@myitdepartment.net> > > > > To: > > > > Paul Scheepens <pscheep...@epo.org> > > > > Cc: > > > > sipx-users@list.sipfoundry.org > > > > Date: > > > > 11-11-2009 16:32 > > > > Subject: > > > > Re: [sipx-users] Intermittent no-audio problem with Ericsson PBX > > > > The patton has a powerful debug system. > > > > I would suggest the next time it happens, you connect via telnet to > > the patton, and log your seeion to a file. Then i would do a debug > > of the call-router, call-control and sip (hint: type "debug" and tab > > to get the available options) at the highest level, then place a > > call which will fail. > > > > Then I would submit a case to patton and ask them to help you > > determine what they see as the problem. They will probably also want > > a copy of your config for the 4960. > > > > Patton will directly assist end users. > > > > The fact that you reset it it and it "sometimes" works makes me > > wonder when it DOES NOT work, whether you have checked to see if > > your link/timing is active with your carrier. I've always noticed > > that sometimes resetting it does not always resync the circuit so it > > is active. If it is active, then it will ring, it is not synced and > > acive, calls will fail altogether. > > > > Can it be assumed there is no NAT/firewall between the three devices? > > > > > > > On Wed, Nov 11, 2009 at 10:18 AM, Paul Scheepens <pscheep...@epo.org> wrote: > > ====Not directly SipX related, but maybe someone can shine a > light....===== > > > > I am testing with SipX 4.0.1, a Patton 4960 on R5.4 and an > Ericsson MD110 PBX. > > Everything was easy to set up and worked perfectly. > > > > All of a sudden calls from the PBX world to SIP would sometimes > haveno audio. > > > The first time I discovered this was when I called from outside to > > my office number (on the Ericsson) which was forwarded to my SIP softphone. > > The SIP phone rings, but after I pick up there is no audio on both ends. > > In a trace I do see RTP packets from and to the Patton. > > > > All calls directly from Ericsson to SIP worked. > > > > Now I even have problems directly from Ericsson to SIP. > > The softphone rings, a call is established, RTP traffic flows, but > > no audio on both ends. > > SIP to Ericsson is still OK. > > > > After a reset of the Patton it sometimes works, sometimes not. > > > > Any ideas where I should start looking? > > > > Best regards / Mit freundlichen Grüßen / Sincères salutations > > > > Paul Scheepens > > > > _______________________________________________ > > sipx-users mailing list sipx-users@list.sipfoundry.org > > List Archive: > http://list.sipfoundry.org/archive/sipx-users > > > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > > sipXecs IP PBX -- http://www.sipfoundry.org/ > > > > > > > > > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: tgrazi...@myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
