Jump to content

Calls dropping during conversation


Tom Waterman
 Share

Recommended Posts

Ok in my office I have about 62 Snom M3 phones. We just went live today. I have received about 6 or 7 reports of calls dropping in midsentance. The calls are not on mute and I am running a permanent key. These calls go from the PBX to Audiocodes FXO gateways. It is very sporadic. Theres is also no NAT that goes on in this situation. My cpu usage is very low and I am working on getting a wiresharp capture the next time it is reported. I am running version 3.1.2.3120. Does anyone have any ideas?

 

Thank you for your help in advance.

Tom

Link to comment
Share on other sites

Ok in my office I have about 62 Snom M3 phones. We just went live today. I have received about 6 or 7 reports of calls dropping in midsentance. The calls are not on mute and I am running a permanent key. These calls go from the PBX to Audiocodes FXO gateways. It is very sporadic. Theres is also no NAT that goes on in this situation. My cpu usage is very low and I am working on getting a wiresharp capture the next time it is reported. I am running version 3.1.2.3120. Does anyone have any ideas?

 

You should definitevely turn on the email reporting of such events. Then you get an email when a gets dropped by the PBX.

 

Also, you better use version 3.2, 3.1 has a ugly bug in the web interface.

 

What you can do is writing a log file of the SIP traffic (only "other" message types) to the file system. Don't write all the other stuff, it just makes the system drown in messages. In the BYE message, the PBX reports how many packets have been sent and received. If there is something out of balance, then you get a hint.

 

In the 3.3 version we added sending of email messages when the user hangs up during a one-way audio situation. If the problem persists, you might have to (temporaily) move to version 3.3 to get better information. Make a backup so that you can move back to the 3.2 version after the problem has been identified!

Link to comment
Share on other sites

You should definitevely turn on the email reporting of such events. Then you get an email when a gets dropped by the PBX.

 

Also, you better use version 3.2, 3.1 has a ugly bug in the web interface.

 

What you can do is writing a log file of the SIP traffic (only "other" message types) to the file system. Don't write all the other stuff, it just makes the system drown in messages. In the BYE message, the PBX reports how many packets have been sent and received. If there is something out of balance, then you get a hint.

 

In the 3.3 version we added sending of email messages when the user hangs up during a one-way audio situation. If the problem persists, you might have to (temporaily) move to version 3.3 to get better information. Make a backup so that you can move back to the 3.2 version after the problem has been identified!

 

I have tried to turn on the email notification and this does not work. I have upgraded to version 3.2.0.3144. I did this a couple of days ago. Yesterday I had a couple reports of dropped calls in mid sentence and I have been in the office for 1.5 hours and have 10 reports already. If this does not get fixed soon we may need to pull the plug on this and revert back to our old system.

Link to comment
Share on other sites

I have tried to turn on the email notification and this does not work. I have upgraded to version 3.2.0.3144. I did this a couple of days ago. Yesterday I had a couple reports of dropped calls in mid sentence and I have been in the office for 1.5 hours and have 10 reports already. If this does not get fixed soon we may need to pull the plug on this and revert back to our old system.

 

Can I upgrade form 3.2 to 3.3 by just stopiing the service and then dropping in the the new controller? And which version should I use I have 3.3.0.3147 and 3.3.0.3152

 

Thank you

Link to comment
Share on other sites

Can I upgrade form 3.2 to 3.3 by just stopiing the service and then dropping in the the new controller? And which version should I use I have 3.3.0.3147 and 3.3.0.3152

 

Try http://pbxnsip.com/protect/pbxctrl-3.2.0.3139.exe. Make backup of the working directory before doing this!! If there should be trouble you can revert to the previous version any time.

 

Do other phones in that network work properly? Is the problem limited to the snom M3? What firmware is on those phones?

Link to comment
Share on other sites

Try http://pbxnsip.com/protect/pbxctrl-3.2.0.3139.exe. Make backup of the working directory before doing this!! If there should be trouble you can revert to the previous version any time.

 

Do other phones in that network work properly? Is the problem limited to the snom M3? What firmware is on those phones?

 

OK I just got a Syslog and checked it out. It looks like there is a silent disconnect release message. Here is the log.

 

SIP/2.0 Via: SIP/2.0/UDP 172.31.3.11;branch=z9hG4bKac1025123861 Max-Forwards: 70 From: <sip:18776713355@172.31.3.11;user=phone>;tag=1c707751253 To: "Jessica Darling" <sip:807@pbx.mdgn.microdatagis.com>;tag=54922 Call-ID: 0abecac0@pbx CSeq: 1 BYE Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.40A.027.001 Reason: Q.850 ;cause=16 ;text="Silence Disconnect" Content-Length: 0 [Time: 8:49:40]

08:48:04.213 : 172.31.3.11 : NOTICE : ( sip_stack)(125741 ) UdpRtxMngr::Transmit 1 BYE Rtx Left: 6 Dest: 172.31.3.10:5060 CallID: (0abecac0@pbx) [Time: 8:49:40]

08:48:04.214 : 172.31.3.11 : NOTICE : ( sip_stack)(125742 ) SIPCall(#17) changes state from Connected to Disconnected [Time: 8:49:40]

08:48:04.214 : 172.31.3.11 : NOTICE : ( lgr_stk_ses)(125743 ) <SESSION #8> SendToCall - event: RELEASE_ACK m_Call = 31513336 [Time: 8:49:40]

08:48:04.215 : 172.31.3.11 : NOTICE : ( 08:48:04.202 : 172.31.3.11 : NOTICE : BYE lgr_flow)(125744 ) | | #8:RELEASE_ACK:(0abecac0@pbx) [Time: 8:49:40]

08:48:04.216 : 172.31.3.11 : NOTICE : ( sip_stack)(125745 ) AcSIPStackAPI::FreeCallAPI - #8 [Time: 8:49:40]

08:48:04.216 : 172.31.3.11 : NOTICE : ( sip_stack)(125746 ) Setting ApplicationCall of AcSIPCall #17 to NULL [Time: 8:49:40]

08:48:04.217 : 172.31.3.11 : NOTICE : ( lgr_stk_mngr)(125747 ) Resource StackSession <#8> Deleted [Time: 8:49:40]

08:48:04.217 : 172.31.3.11 : NOTICE : ( lgr_flow)(125748 ) | | #8:RELEASE:(0abecac0@pbx) [Time: 8:49:40]

08:48:04.218 : 172.31.3.11 : NOTICE : ( lgr_flow)(125749 ) | | #8:Call changing states from:DisconnectingState to:DisconnectingState [Time: 8:49:40]

08:48:04.218 : 172.31.3.11 : NOTICE : ( lgr_flow)(125750 ) | #6:RELEASE RELEASE_BECAUSE_SILENCE_DISC : (0abecac0@pbx) [Time: 8:49:40]

08:48:04.218 : 172.31.3.11 : NOTICE : ( lgr_flow)(125751 ) | #6:Close voice Channel [Time: 8:49:40]

08:48:04.219 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125752 ) #6:StopRTP_RTCP on channel 6 [Time: 8:49:40]

08:48:04.219 : 172.31.3.11 : NOTICE : ( lgr_flow)(125753 ) | #6:RELEASE_ACK (send) : (0abecac0@pbx) [Time: 8:49:40]

08:48:04.219 : 172.31.3.11 : NOTICE : ( lgr_flow)(125754 ) | | #8:RELEASE_ACK:(0abecac0@pbx) [Time: 8:49:40]

08:48:04.219 : 172.31.3.11 : NOTICE : ( lgr_callf)(125755 ) Call #8 deleted [Time: 8:49:40]

08:48:04.220 : 172.31.3.11 : NOTICE : ( lgr_psbrdex)(125756 ) InsertBoardEvent- event 105 inserted channel 6 [Time: 8:49:40]

08:48:04.220 : 172.31.3.11 : NOTICE : ( lgr_flow)(125757 ) #6:RELEASE_BECAUSE_IP_TIMER_EXPIRED_EV [Time: 8:49:40]

08:48:04.221 : 172.31.3.11 : NOTICE : ( lgr_flow)(125758 ) | #6:RELEASE_BECAUSE_IP_TIMER_EXPIRED_EV [Time: 8:49:40]

08:48:04.221 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125759 ) #6:cpDigitMapHndlr_Stop - Stoped (0) [Time: 8:49:40]

08:48:04.222 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125760 ) #6:CloseChannel: ChannelNum=6 [Time: 8:49:40]

08:48:04.223 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125761 ) Open channel: IsVoiceOn: 1, IsT38On: 0, IsVbdOn: 0, IsVideoOn: 0 [Time: 8:49:40]

08:48:04.223 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125762 ) #6:OpenChannel:on Trunk -1 BChannel:6 CID=6 with VoiceCoder: g711Ulaw64k20 VbdCoder: InvalidCoder255 DetectorSide: 0 FaxModemDet NO_FAX_MODEM_DETECTED [Time: 8:49:40]

08:48:04.224 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125763 ) #6:OpenChannel VoiceVolume= 1, DTMFVolume = -11, InputGain = 0, RTPRedundancyDepth = 0 FlashHookPeriod = 700 AgcCmd = 0x13180000 [Time: 8:49:40]

08:48:04.224 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125764 ) OpenChannel, CoderType = 1, Interval = 3, M = 1 [Time: 8:49:40]

08:48:04.225 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125765 ) #6:FAXTransportType = 1 [Time: 8:49:40]

08:48:04.225 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125766 ) #6:ConfigFaxModemChannelParams NSEMode=0, CNGDetMode=0, FAXTranType=1, VxxTranType=2, VoiceVol= 1, DTMFVol=-11, InGain=0, RTPRedDepth=0, ECE=1, SCE=1, ECNlpMode=0, DJBufMinDelay=10, DJBufOptFac=10) [Time: 8:49:40]

08:48:04.226 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125767 ) Detectors: Amd:0, Ans:0 En:0 IBScmd:0xa1 [Time: 8:49:40]

08:48:04.229 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125768 ) Turn ringer OFF for channel 6 [Time: 8:49:40]

08:48:04.230 : 172.31.3.11 : NOTICE : ( lgr_flow)(125769 ) | #6:FXO Release Line [Time: 8:49:40]

08:48:04.231 : 172.31.3.11 : NOTICE : ( lgr_psbrdif)(125770 ) #6:PSOSBoardInterface::StopPlayTone- Called [Time: 8:49:40]

08:48:04.232 : 172.31.3.11 : NOTICE : ( lgr_flow)(125771 ) ---- Incoming SIP Message from 172.31.3.10:5060 ---- [Time: 8:49:40]

Link to comment
Share on other sites

Oh can you turn the silence disconnect off on the gateway? That would solve the problem.

 

We just turned it off completly. We had "Disconnect call on silence detection" turned off already, I then went in and "SIlence Detection Method" to none on the audiocodes gateway and rebooted them. I will report back.

 

Tom

Link to comment
Share on other sites

We just turned it off completly. We had "Disconnect call on silence detection" turned off already, I then went in and "SIlence Detection Method" to none on the audiocodes gateway and rebooted them. I will report back.

 

Do you PnP the snom M3 phones? I guess it would also make sense to turn the silence detection off there as well.

Link to comment
Share on other sites

Do you PnP the snom M3 phones? I guess it would also make sense to turn the silence detection off there as well.

 

Where is that setting on the snom M3 I can't seem to find it. Also we are getting terrible feedback with the speaker on these M3s. I know it is the speaker feedbacking into the MIC and this causes the speaker to cut out. Any ideas?

 

Tom

Link to comment
Share on other sites

Where is that setting on the snom M3 I can't seem to find it. Also we are getting terrible feedback with the speaker on these M3s. I know it is the speaker feedbacking into the MIC and this causes the speaker to cut out. Any ideas?

 

On the M3,under "Management settings", set the "Configuration Address" to the IPv4-Address of the PBX.

 

If you use a later version of PBX 3.3, then you will even get the right time zone :) .

Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

 Share

×
×
  • Create New...