Re: [asterisk-users] PRI hangup certain outgoing calls
Steve Totaro wrote: It's worth a shot, you should be running the latest 1.4.x code anyways right? Bugs can manifest themselves in different ways, or possibly the poster did not explain the issue accurately. I think the bug fits your problem more than any other explanation. Steve, the issue appears fixed after a upgrade and reboot, though I do not know if the remote party have changed their setup. Thanks for your help -- Alastair Battrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI hangup certain outgoing calls
I have a problem calling a certain number from our PRI line. Calling the number from a separate PSTN phone works fine. The remote number seems to have some funny call redivert setup, when you call it, it answers immediately, makes some kind of beep and then starts to ring. Our PRI is in the UK from Telewest/NTL/Virgin Media and most outgoing calls work without a problem. The server is trixbox based with a Digium TE410P PRI interface. Trixbox 2.6, Asterisk 1.4.18-3, FreeBPX 2.4. I'm no expert, but the PRI trace shows that the outgoing call is proceeding, and then seems to show that my end of the connection has disconnected the call, but I cannot work out why - it has already agreed to setup the call! What is causing this disconnect? Is there some tone detection going on behind the scenes? [zaptel] span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone= uk defaultzone = uk [zapata] [channels] language=en context=from-zaptel-custom usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=yes overlapdial=yes echocancel=64 echocancelwhenbridged=no echotraining=yes rxgain=0.0 txgain=0.0 immediate=no faxdetect=no busydetect=no callprogress=no pridialplan=local prilocaldialplan=local group=1 switchtype = euroisdn signalling = pri_cpe channel = 1-8 PRI Trace [ 02 01 01 2b ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 021 P/F: 1 0 bytes of data -- ACKing all packets from 20 to (but not including) 21 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Unsolicited RR with P/F bit, responding Sending Receiver Ready (23) [ 02 01 01 2f ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 023 P/F: 1 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter -- Accepting AUTHENTICATED call from 10.1.0.46: requested format = gsm, requested prefs = (), actual format = alaw, host prefs = (ulaw|alaw|gsm), priority = mine -- Executing [EMAIL PROTECTED]:1] Macro(IAX2/255-4, user-callerid|SKIPTTL|) in new stack -- Executing [EMAIL PROTECTED]:1] NoOp(IAX2/255-4, user-callerid: device 255) in new stack -- Executing [EMAIL PROTECTED]:2] Set(IAX2/255-4, AMPUSER=255) in new stack -- Executing [EMAIL PROTECTED]:3] GotoIf(IAX2/255-4, 0?report) in new stack -- Executing [EMAIL PROTECTED]:4] ExecIf(IAX2/255-4, 0|Set|REALCALLERIDNUM=255) in new stack -- Executing [EMAIL PROTECTED]:5] NoOp(IAX2/255-4, REALCALLERIDNUM is 255) in new stack -- Executing [EMAIL PROTECTED]:6] Set(IAX2/255-4, AMPUSER=255) in new stack -- Executing [EMAIL PROTECTED]:7] Set(IAX2/255-4, AMPUSERCIDNAME=Alastair Battrick) in new stack -- Executing [EMAIL PROTECTED]:8] GotoIf(IAX2/255-4, 0?report) in new stack -- Executing [EMAIL PROTECTED]:9] Set(IAX2/255-4, AMPUSERCID=255) in new stack -- Executing [EMAIL PROTECTED]:10] Set(IAX2/255-4, CALLERID(all)=Alastair Battrick 255) in new stack -- Executing [EMAIL PROTECTED]:11] Set(IAX2/255-4, REALCALLERIDNUM=255) in new stack -- Executing [EMAIL PROTECTED]:12] ExecIf(IAX2/255-4, 0|Set|CHANNEL(language)=) in new stack -- Executing [EMAIL PROTECTED]:13] NoOp(IAX2/255-4, TTL: ARG1: SKIPTTL) in new stack -- Executing [EMAIL PROTECTED]:14] GotoIf(IAX2/255-4, 1?continue) in new stack -- Goto (macro-user-callerid,s,23) -- Executing [EMAIL PROTECTED]:23] NoOp(IAX2/255-4, Using CallerID Alastair Battrick 255) in new stack -- Executing [EMAIL PROTECTED]:2] Set(IAX2/255-4, _NODEST=) in new stack -- Executing [EMAIL PROTECTED]:3] Macro(IAX2/255-4, record-enable|255|OUT|) in new stack -- Executing [EMAIL PROTECTED]:1] GotoIf(IAX2/255-4, 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing [EMAIL PROTECTED]:4] AGI(IAX2/255-4, recordingcheck|20080428-134020|1209386420.22) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20080428-134020|1209386420.22: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing [EMAIL PROTECTED]:5] NoOp(IAX2/255-4, No recording needed) in new stack -- Executing [EMAIL PROTECTED]:4] Macro(IAX2/255-4, dialout-trunk|1|02081237722||) in new stack -- Executing [EMAIL PROTECTED]:1] Set(IAX2/255-4, DIAL_TRUNK=1) in new stack -- Executing [EMAIL PROTECTED]:2] ExecIf(IAX2/255-4, 0|Authenticate|) in new stack -- Executing [EMAIL PROTECTED]:3] GotoIf(IAX2/255-4, 0?disabletrunk|1) in new stack -- Executing [EMAIL PROTECTED]:4] Set(IAX2/255-4, DIAL_NUMBER=02081237722) in new stack
Re: [asterisk-users] PRI hangup certain outgoing calls
On Mon, Apr 28, 2008 at 9:29 AM, Alastair Battrick [EMAIL PROTECTED] wrote: I have a problem calling a certain number from our PRI line. Calling the number from a separate PSTN phone works fine. The remote number seems to have some funny call redivert setup, when you call it, it answers immediately, makes some kind of beep and then starts to ring. Our PRI is in the UK from Telewest/NTL/Virgin Media and most outgoing calls work without a problem. The server is trixbox based with a Digium TE410P PRI interface. Trixbox 2.6, Asterisk 1.4.18-3, FreeBPX 2.4. I'm no expert, but the PRI trace shows that the outgoing call is proceeding, and then seems to show that my end of the connection has disconnected the call, but I cannot work out why - it has already agreed to setup the call! What is causing this disconnect? Is there some tone detection going on behind the scenes? [zaptel] span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone= uk defaultzone = uk [zapata] [channels] language=en context=from-zaptel-custom usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=yes overlapdial=yes echocancel=64 echocancelwhenbridged=no echotraining=yes rxgain=0.0 txgain=0.0 immediate=no faxdetect=no busydetect=no callprogress=no pridialplan=local prilocaldialplan=local group=1 switchtype = euroisdn signalling = pri_cpe channel = 1-8 PRI Trace [snipped] Location: Private network serving the local user (1) Ext: 1 Cause: Circuit/channel congestion (34), class = Network Congestion (resource unavailable) (2) ] -- Hungup 'Zap/1-1' Is this Trixbox/FreePBX? What version of Asterisk? http://www.trixbox.org/forums/trixbox-forums/help/inbound-calls-over-pri-get-fast-busy I believe the issue is a bug Mon, 04/14/2008 - 12:50pm I believe the issue is a bug in channel locking in Asterisk 1.4.18. I'm waiting for Asterisk 1.4.19 to be released through Trixbox, which has bugfixes applied to fix this exact issue. Last post on the webpage, seems to be the same issue. I believe the issue is a bug in channel locking in Asterisk 1.4.18. I'm waiting for Asterisk 1.4.19 to be released through Trixbox, which has bugfixes applied to fix this exact issue. This is the section to focus on. Sorry I cannot be of much help. Can you dial other numbers, it is just an issue with one in particular? http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause #define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34 #define AST_CAUSE_NO_ROUTE_TRANSIT_NET 2 (this is fishy) I would contact your service provider, they may be able to explain it better if not fix it. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI hangup certain outgoing calls
Steve Totaro wrote: On Mon, Apr 28, 2008 at 9:29 AM, Alastair Battrick [EMAIL PROTECTED] wrote: I have a problem calling a certain number from our PRI line. Calling the number from a separate PSTN phone works fine. The remote number seems to have some funny call redivert setup, when you call it, it answers immediately, makes some kind of beep and then starts to ring. Our PRI is in the UK from Telewest/NTL/Virgin Media and most outgoing calls work without a problem. The server is trixbox based with a Digium TE410P PRI interface. Trixbox 2.6, Asterisk 1.4.18-3, FreeBPX 2.4. Is this Trixbox/FreePBX? What version of Asterisk? Yes, Trixbox 2.6, Asterisk 1.4.18-3, FreeBPX 2.4. This is the section to focus on. Sorry I cannot be of much help. Can you dial other numbers, it is just an issue with one in particular? I can dial out to other numbers without issue. I don't think it is to do with the bug that you quoted as I do not need to reboot the server to start making calls to other numbers once this busy message has been experienced. -- Alastair Battrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI hangup certain outgoing calls
On Mon, Apr 28, 2008 at 10:08 AM, Alastair Battrick [EMAIL PROTECTED] wrote: Steve Totaro wrote: On Mon, Apr 28, 2008 at 9:29 AM, Alastair Battrick [EMAIL PROTECTED] wrote: I have a problem calling a certain number from our PRI line. Calling the number from a separate PSTN phone works fine. The remote number seems to have some funny call redivert setup, when you call it, it answers immediately, makes some kind of beep and then starts to ring. Our PRI is in the UK from Telewest/NTL/Virgin Media and most outgoing calls work without a problem. The server is trixbox based with a Digium TE410P PRI interface. Trixbox 2.6, Asterisk 1.4.18-3, FreeBPX 2.4. Is this Trixbox/FreePBX? What version of Asterisk? Yes, Trixbox 2.6, Asterisk 1.4.18-3, FreeBPX 2.4. This is the section to focus on. Sorry I cannot be of much help. Can you dial other numbers, it is just an issue with one in particular? I can dial out to other numbers without issue. I don't think it is to do with the bug that you quoted as I do not need to reboot the server to start making calls to other numbers once this busy message has been experienced. -- Alastair Battrick It's worth a shot, you should be running the latest 1.4.x code anyways right? Bugs can manifest themselves in different ways, or possibly the poster did not explain the issue accurately. I think the bug fits your problem more than any other explanation. If not, call your telco immediately to report the issue and also keep looking on your side. Burn the candle from both ends. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI hangup certain outgoing calls
I can dial out to other numbers without issue. Calling the number from a separate PSTN phone works fine. I have had such problem last week. from PRI interface (i get busy tone as fast as i finish typing last digit hit dial) from analog interface I can make the call without problem. my provider never could solve it (well so far) I myself never could understand what it could be. is this a special number (call center or something) ? because mine is a call center number ... so maybe they had a wrong setup on thier servers or something. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI hangup certain outgoing calls
On Mon, Apr 28, 2008 at 10:27 AM, Arthur [EMAIL PROTECTED] wrote: I can dial out to other numbers without issue. Calling the number from a separate PSTN phone works fine. I have had such problem last week. from PRI interface (i get busy tone as fast as i finish typing last digit hit dial) from analog interface I can make the call without problem. my provider never could solve it (well so far) I myself never could understand what it could be. is this a special number (call center or something) ? because mine is a call center number ... so maybe they had a wrong setup on thier servers or something. Curious, are you sending a toll free as your callerID? If so, try sending your BTN or some other regular toll number. I had a similar problem dialing a specific number (Gateway) when sending a toll free callerID, GXing said it was because other telcos are unsure how to bill the call so they send back busy. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
