Re: [asterisk-users] (solved) CPU Spikes in asterisk connected via IAX trunk
Hello, I had posted this mail some time back, Having got no responses I tried one suggestion I received in another thread and replaced all IAX trunks with SIP trunks. That has resolved this issue. Asterisk now does not hit more than 100% CPU and there is no call disturbance. CPU usage is now is more even. Thanks and regards, raj On Fri, Aug 14, 2009 at 12:31 PM, Rajkumar S rajkum...@gmail.com wrote: Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP clients connect. SIP clients can also dial outside and call goes like C - B - A - PSTN. I am facing some call disturbance for agents connected via SIP in C. While investigating I found that CPU usage hits 99% occasionally and in general CPU usage is very un even. Load average also goes up correspondingly some times till about 30. It has no correlation with number of calls. Some times even with about 29 calls the cpu is not much loaded (10%) but it hits 60% 70% with about 6 calls (12 channels) some other times. I am using ulaw through out, (disallow=all; allow=ulaw in iax.conf and sip.conf) One correlation I found was that when ever agents transfer calls to main menu (ie to server B) there is a load spike. This transfer again goes via same IAX trunk as the incoming. IAX conf in C is: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw jitterbuffer=no forcejitterbuffer=no [a16-q1] type=user auth=plaintext secret=password context=inbound-calls qualify=yes trunk=yes [a16-q1-a16-in1] type=peer host=192.168.79.177 auth=plaintext secret=password username=a16-in1 qualify=yes trunk=yes IAX conf in B is: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw jitterbuffer=no forcejitterbuffer=no transfer = no [a16-in1] type=user auth=plaintext secret=password context=inbound-calls qualify=yes trunk=yes [a16-in1-a16-q1] type=peer host=192.168.79.176 auth=plaintext secret=password username=a16-q1 qualify=yes trunk=yes I am pretty much stumped here. Could IAX trunk be the source of the problem? Should I switch to SIP ? thanks and regards, raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657
Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over SIP trunk from which calls get routed to third server (C) (1.6.0.9) via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP clients connect. SIP clients can also dial outside and call goes like C - B - A - PSTN. I have an occasional problem where DTMF is not recognized, ie if clients type a digit while in menu the system does not register it. In my C server I saw a log line like this today: DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657 Is the above message an indication of this problem? How can I fix it? with regards, raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue_log in mysql and file
Hi, I am using RT engine to log queue_log to a mysql database. My extconfig is [settings] queue_log = mysql,asterisk16_production Logging to mysql is working fine. But I find that the queue_log file now only has QUEUESTART lines for eg: 1250519094|NONE|NONE|NONE|QUEUESTART| 1250519186|NONE|NONE|NONE|QUEUESTART| How can I have queue_log in both db as well as in a file? thanks and regards, raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CPU Spikes in asterisk connected via IAX trunk
Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP clients connect. SIP clients can also dial outside and call goes like C - B - A - PSTN. I am facing some call disturbance for agents connected via SIP in C. While investigating I found that CPU usage hits 99% occasionally and in general CPU usage is very un even. Load average also goes up correspondingly some times till about 30. It has no correlation with number of calls. Some times even with about 29 calls the cpu is not much loaded (10%) but it hits 60% 70% with about 6 calls (12 channels) some other times. I am using ulaw through out, (disallow=all; allow=ulaw in iax.conf and sip.conf) One correlation I found was that when ever agents transfer calls to main menu (ie to server B) there is a load spike. This transfer again goes via same IAX trunk as the incoming. IAX conf in C is: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw jitterbuffer=no forcejitterbuffer=no [a16-q1] type=user auth=plaintext secret=password context=inbound-calls qualify=yes trunk=yes [a16-q1-a16-in1] type=peer host=192.168.79.177 auth=plaintext secret=password username=a16-in1 qualify=yes trunk=yes IAX conf in B is: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw jitterbuffer=no forcejitterbuffer=no transfer = no [a16-in1] type=user auth=plaintext secret=password context=inbound-calls qualify=yes trunk=yes [a16-in1-a16-q1] type=peer host=192.168.79.176 auth=plaintext secret=password username=a16-q1 qualify=yes trunk=yes I am pretty much stumped here. Could IAX trunk be the source of the problem? Should I switch to SIP ? thanks and regards, raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF is not working occasionally over IAX Trunk
Hi all, Did some more digging in. I changed the trunk from IAX to SIP and still there are not much difference. So I guess it's not an IAX problem. I have enabled DTMF logging and captured the DTMF logs for two servers. (A: where E1 card is connected asterisk-1.4.25, dahdi-linux-2.1.0.4) and B (v1.6.0.9) where IVR is running. I have just pressed * 3 3 but to my untrained eyes it seems asterisk is seeing * * 3 3 3 logs in A: [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/a16-in1-081af4c8] [ TYPE: DTMF Begin (12) SUBCLASS: * (42) ] [DAHDI/37-1] [ TYPE: DTMF Begin (12) SUBCLASS: * (42) ] [SIP/a16-in1-081af4c8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: * (42) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: * (42) ] [SIP/a16-in1-081af4c8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-081af4c8] [ TYPE: DTMF Begin (12) SUBCLASS: 3 (51) ] [DAHDI/37-1] [ TYPE: DTMF Begin (12) SUBCLASS: 3 (51) ] [SIP/a16-in1-081af4c8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: 3 (51) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: 3 (51) ] [SIP/a16-in1-081af4c8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-081af4c8] [ TYPE: DTMF Begin (12) SUBCLASS: 3 (51) ] [DAHDI/37-1] [ TYPE: DTMF Begin (12) SUBCLASS: 3 (51) ] [SIP/a16-in1-081af4c8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: 3 (51) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: 3 (51) ] [SIP/a16-in1-081af4c8] [ TYPE: DTMF Begin (12) SUBCLASS: 3 (51) ] [DAHDI/37-1] [ TYPE: DTMF Begin (12) SUBCLASS: 3 (51) ] [SIP/a16-in1-081af4c8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: 3 (51) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: 3 (51) ] [SIP/a16-in1-081af4c8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-081af4c8] logs in B: Over the SIP channel it seems B is getting * 3 3 3 [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: DTMF End (1) SUBCLASS: * (42) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: DTMF End (1) SUBCLASS: 3 (51) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5)
Re: [asterisk-users] DTMF is not working occasionally over IAX Trunk
Hi, The servers B C are running in a virtual machine (linux kvm) and uses ztdummy for timing. Server A has a digium card. I am not sure if this is the cause of the problems I am facing. raj On Fri, Jul 3, 2009 at 7:16 PM, Rajkumar Srajkum...@gmail.com wrote: Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digium card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP clients connect. SIP clients can also dial outside and call goes like C - B - A - PSTN. An IVR is implemented in B. extensions.conf looks like: exten = s, 1, SET(MENUFLOW=s) exten = s, n, Background(welcome) exten = s, n, WaitExten(30) exten = *, 1, Goto(menu-language,s,1) like this it goes couple of menus deep. A typical sequence is like * 3 2. Some times Background will continue to play even when I press *. It will go through. Some other times as soon as I press * 3 it will go to menu option of * 3 3. ie the 3 is repeated. I never had this problem on A. So I can rule out the DTMF problem in E1. So this has to be some thing with the way E1 is getting transmitted over IAX trunk. My iax.conf in A is like: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm jitterbuffer=no forcejitterbuffer=no [ccsrv-a16-in1] type=peer host=192.168.79.177 auth=plaintext secret=password username=a16-in1 qualify=yes trunk=yes and in B [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw jitterbuffer=no forcejitterbuffer=no transfer = no [a16-in1] type=user auth=plaintext secret=password context=inbound-calls qualify=yes trunk=yes I have also posted another mail with calls not terminated with same IAX trunk. I am not sure of they are related, but any help to resolve this would be very helpful with regards raj ___ -- 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] Some IAX calls do not disconnect.
Hi, The servers B C are running in a virtual machine (linux kvm) and uses ztdummy for timing. Server A has a digium card. I am not sure if this is the cause of the problems I am facing. raj On Fri, Jul 3, 2009 at 5:57 PM, Rajkumar Srajkum...@gmail.com wrote: On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote: I'd try adding transfer=no in the B iax.conf This does not help, I still have some ghost calls in B a16-in1*CLI core show channels Channel Location State Application(Data) IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-12174 outbo...@inbound-cal Up Dial(iax2/a16-in1-sangoma-flip IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-7161 outbo...@inbound-cal Up Dial(iax2/a16-in1-sangoma-flip IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-14813 s...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-4485 �...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-10115 s...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue 10 active channels 5 active calls raj ___ -- 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] Some IAX calls do not disconnect.
Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP clients connect. SIP clients can also dial outside and call goes like C - B - A - PSTN. Every day evening I find that there are about 30 calls in B which is not disconnected. This comprise of both calls from B - A as well as B - C. There are no such lingering calls in A or C. Every day I manually disconnect the calls, shown below are two example with first one from B - C and second B - A. a16-in1*CLI soft hangup IAX2/a16-in1-11080 Requested Hangup on channel 'IAX2/a16-in1-11080' -- Hungup 'IAX2/a16-in1-a16-q1-16420' == Spawn extension (queue, s, 20) exited non-zero on 'IAX2/a16-in1-11080' -- Hungup 'IAX2/a16-in1-11080' a16-in1*CLI soft hangup IAX2/a16-in1-903 Requested Hangup on channel 'IAX2/a16-in1-903' -- Hungup 'IAX2/a16-in1-sangoma-flip-outgoing-16393' == Spawn extension (inbound-calls, outbound, 1) exited non-zero on 'IAX2/a16-in1-903' -- Hungup 'IAX2/a16-in1-903' in iax.conf of B the entries are like: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw jitterbuffer=no forcejitterbuffer=no [a16-in1-a16-q1] type=peer host=192.168.79.176 auth=plaintext secret=password username=a16-q1 qualify=yes trunk=yes in C the corresponding entry is: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw jitterbuffer=no forcejitterbuffer=no [a16-q1] type=user auth=plaintext secret=password context=inbound-calls qualify=yes trunk=yes I do not know where even to start. Any idea to resolve this would be much appreciated. raj ___ -- 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] Some IAX calls do not disconnect.
On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote: I'd try adding transfer=no in the B iax.conf I'm guessing the box in the middle (B) is somehow transferring itself out of the call but retaining a ghost call entry. It would be interesting to know what state those ghost calls are in - iax2 show netstats on the CLI might tell you something interesting. Thanks, I will try these two suggestions also and let know the results. raj ___ -- 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] Some IAX calls do not disconnect.
On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote: iax2 show netstats The show netstats gives: a16-in1*CLI iax2 show netstats LOCAL - REMOTE ChannelRTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts IAX2/a16-in1-1869 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-4071 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-112621000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-124431000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-131071000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-145261000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-146771000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-a16-q1-16384 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-sangoma-flip 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-sangoma-flip 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-sangoma-flip 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-a16-q1-16388 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-a16-q1-16389 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-a16-q1-16391 1000 -10-1 -1 0 -1 0 00 0 0 00 0 I have added transfer=no also, watching for it's effect now. raj ___ -- 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] Some IAX calls do not disconnect.
On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote: I'd try adding transfer=no in the B iax.conf This does not help, I still have some ghost calls in B a16-in1*CLI core show channels Channel Location State Application(Data) IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-12174 outbo...@inbound-cal Up Dial(iax2/a16-in1-sangoma-flip IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-7161outbo...@inbound-cal Up Dial(iax2/a16-in1-sangoma-flip IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-14813 s...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-4485s...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-10115 s...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue 10 active channels 5 active calls raj ___ -- 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] DTMF is not working occasionally over IAX Trunk
Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digium card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP clients connect. SIP clients can also dial outside and call goes like C - B - A - PSTN. An IVR is implemented in B. extensions.conf looks like: exten = s, 1, SET(MENUFLOW=s) exten = s, n, Background(welcome) exten = s, n, WaitExten(30) exten = *, 1, Goto(menu-language,s,1) like this it goes couple of menus deep. A typical sequence is like * 3 2. Some times Background will continue to play even when I press *. It will go through. Some other times as soon as I press * 3 it will go to menu option of * 3 3. ie the 3 is repeated. I never had this problem on A. So I can rule out the DTMF problem in E1. So this has to be some thing with the way E1 is getting transmitted over IAX trunk. My iax.conf in A is like: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm jitterbuffer=no forcejitterbuffer=no [ccsrv-a16-in1] type=peer host=192.168.79.177 auth=plaintext secret=password username=a16-in1 qualify=yes trunk=yes and in B [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw jitterbuffer=no forcejitterbuffer=no transfer = no [a16-in1] type=user auth=plaintext secret=password context=inbound-calls qualify=yes trunk=yes I have also posted another mail with calls not terminated with same IAX trunk. I am not sure of they are related, but any help to resolve this would be very helpful with regards raj ___ -- 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] No CDR generated for calls to queues with no agents
Hi, I am using Asterisk 1.6.0.9. I have calls coming from another asterisk server via IAX and lands in a queue. I have noticed that if there are no agents logged in the queue no CDR is generated. If there is one agent logged in then the phone rings and a CDR is generated even if the call was pickedup or not. Looking at the bug db http://bugs.digium.com/view.php?id=13691 is similar to the problem I am facing. Btw queue_log is showing all entries as expected. My extensions.conf is as follows: [general] static=yes writeprotect=no [globals] [inbound-calls] exten = queue, 1, Queue(genenq) exten = queue, n, Hangup [sip] #include dialplan/1xxx.conf cdr.conf is [general] enable=yes unanswered = yes batch=no safeshutdown=yes [csv] usegmtime=no loguniqueid=yes loguserfield=yes [custom] loguniqueid=yes loguserfield=yes is this a bug or am I doing some thing stupid ? raj ___ -- 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] Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available
Hi, I am trying to log queue_log to odbc (MS SQL) I have res_odbc.conf configured and modules.conf have preload = res_odbc.so preload = res_config_odbc.so extconfig.conf has queue_log = odbc,asterisk. When I start asterisk I get the following messages. The important one being: Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available I can see that res_odbc is loading and registering Config Engine odbc, but that's after logger has started. Any clue what I am doing wrong? with regards, raj Asterisk 1.6.0.5, Copyright (C) 1999 - 2008 Digium, Inc. and others. snip = == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found == Binding queue_log to odbc/asterisk/queue_log Started Asterisk Event Logger Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available Asterisk Event Logger Started /var/log/asterisk/event_log 3 modules will be loaded. Connecting sqlserver Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': == Found == Parsing '/etc/asterisk/res_odbc.conf': == Found res_odbc: Connected to sqlserver [DSN_NAME] Registered ODBC class 'sqlserver' dsn-[DSN_NAME] res_odbc loaded. res_odbc.so = (ODBC resource) Registered Config Engine odbc ___ -- 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] [cdr_odbc] error: Cannot insert the value NULL into column 'calldate'
Hi,
I am trying to get * log to mssql server. I have odbc and freetds
configured, but my insert query is missing calldate which is a NOT
NULL field in database schema.
cdr_adaptive_odbc: Insert failed on 'sqlserver:cdr'. CDR failed:
INSERT INTO cdr
(clid,src,dst,dcontext,channel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid)
VALUES
('1000','1000','100','sip','SIP/1000-09388800','AddQueueMember','test,Local/1...@sip,5,,Agent/1000,Local/1...@sip',9,9,'ANSWERED',3,'1235681438.1')
Schema is:
My table structure is:
CREATE TABLE cdr (
[calldate] [datetime] NOT NULL ,
[clid] [varchar] (80) NOT NULL ,
[src] [varchar] (80) NOT NULL ,
[dst] [varchar] (80) NOT NULL ,
[dcontext] [varchar] (80) NOT NULL ,
[channel] [varchar] (80) NOT NULL ,
[dstchannel][varchar] (80) NOT NULL ,
[lastapp] [varchar] (80) NOT NULL ,
[lastdata] [varchar] (80) NOT NULL ,
[duration] [int] NOT NULL ,
[billsec] [int] NOT NULL ,
[disposition] [varchar] (45) NOT NULL ,
[amaflags] [int] NOT NULL ,
[accountcode] [varchar] (20) NOT NULL ,
[uniqueid] [varchar] (32) NOT NULL ,
[userfield] [varchar] (255) NOT NULL
)
How can I make sure that my insert commands reflects my database schema?
I have attached details of all my config files below.
I can connect using isql.
isql -v DSN_NAME sa password
+---+
| Connected!|
| |
| sql-statement |
| help [tablename] |
| quit |
| |
+---+
When I make a call I get the following error in console:
Unable to retrieve database handle. CDR failed.
SQL Execute returned an error -1: 23000: [FreeTDS][SQL Server]Cannot
insert the value NULL into column 'calldate', table
'production.dbo.cdr'; column does not allow nulls. INSERT fails. (153)
SQL Execute returned an error -1: 01000: [FreeTDS][SQL Server]The
statement has been terminated. (55)
SQL Execute error -1! Attempting a reconnect...
Connection is down attempting to reconnect...
Disconnected 0 from sqlserver [DSN_NAME]
Database handle deallocated
Connecting sqlserver
res_odbc: Connected to sqlserver [DSN_NAME]
SQL Execute returned an error -1: 23000: [FreeTDS][SQL Server]Cannot
insert the value NULL into column 'calldate', table
'production.dbo.cdr'; column does not allow nulls. INSERT fails. (153)
SQL Execute returned an error -1: 01000: [FreeTDS][SQL Server]The
statement has been terminated. (55)
SQL Execute error -1! Attempting a reconnect...
Connection is down attempting to reconnect...
Disconnected 0 from sqlserver [DSN_NAME]
Database handle deallocated
Connecting sqlserver
res_odbc: Connected to sqlserver [DSN_NAME]
cdr_adaptive_odbc: Insert failed on 'sqlserver:cdr'. CDR failed:
INSERT INTO cdr
(clid,src,dst,dcontext,channel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid)
VALUES
('1000','1000','100','sip','SIP/1000-09388800','AddQueueMember','test,Local/1...@sip,5,,Agent/1000,Local/1...@sip',9,9,'ANSWERED',3,'1235681438.1')
a16-q1:/etc/asterisk# cat cdr.conf
[general]
enable=yes
;unanswered = no
;batch=no
;size=100
;time=300
;scheduleronly=no
;safeshutdown=yes
;endbeforehexten=no
[csv]
usegmtime=yes; log date/time in GMT. Default is no
loguniqueid=yes ; log uniqueid. Default is no
loguserfield=yes ; log user field. Default is no
[odbc]
usegmtime=yes; log date/time in GMT. Default is no
loguniqueid=yes ; log uniqueid. Default is no
loguserfield=yes ; log user field. Default is no
;
; cdr_odbc.conf
;
[global]
username = sa
password = password
dsn = DSN_NAME
loguniqueid=yes
dispositionstring=yes
table=cdr
usegmtime=yes
a16-q1:/etc/asterisk# cat cdr_adaptive_odbc.conf
[first]
connection=sqlserver
table=cdr
alias calldate = start
a16-q1:/etc/asterisk# cat res_odbc.conf
[ENV]
[sqlserver]
enabled = yes
dsn = DSN_NAME
share_connections = no
limit = 5
username = sa
password = password
pre-connect = yes
sanitysql = select count(*) from systables
backslash_is_escape = no
a16-q1:/etc/asterisk# ls
asterisk.conf cdr.conf chan_dahdi.conf
extensions.conf iax.conf modules.conf queuerules.conf
res_odbc.conf
cdr_adaptive_odbc.conf cdr_odbc.conf dialplan
Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()
Resurrecting an old thread. On Fri, 9 May 2008, Russell Bryant wrote: Benoit Plessis wrote: So i'm wondering if someone already as made a dialplan function that could toggle the 'Use' flag of an agent ? or if this kind of function would be integrated into the core if i build it ? snip Alternatively, if you would like to control the usability of an agent through the dialplan, then you could use the DEVICE_STATE() function to create a custom device state. Then, you could list your custom device as what app_queue should look at before attempting to call the agent. How can this be done? I have looked much and there are no document/mail explaining how this can be done. An example especially in the context of queues-with-callback-members document would be very very helpful. with regards, raj ___ -- 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] Busy status of a snom connected to two asterisk servers?
Hi, I have a snom 360 connected to two asterisk servers(both 1.6.0.5), via two identities. Each asterisk server runs a queue and snom is a member of queue in both servers. Currently when snom is receiving call from one asterisk server, it can still receive a call from the other asterisk, because even though the snom is busy attending call from one asterisk the other server does not know this. Is there any way to share the actual busy status of snom to both asterisk server so that when snom is receiving a call from one server the other will see a busy status and will wait to become free before connecting the call? with warm regards, raj ___ -- 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] Busy status of a snom connected to two asterisk servers?
On Thu, Feb 19, 2009 at 8:28 PM, Philipp Kempgen philipp.kemp...@amooma.de wrote: Easy solution: Disable call waiting on the phone. But asterisk will attempt a call since it's status is idle, and will generate events which will confuse ADM I am using to display a url for call. Advanced solution: Use local channels as queue members and Custom hints. You could build a mechanism (outside of Asterisk) to sync the states of your Custom hints between both servers. I am already using local channels and will explore hints. I have not used it till now, any hints about writing custom hints are most welcome :) raj ___ -- 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] Busy status of a snom connected to two asterisk servers?
On Thu, 19 Feb 2009, Philipp Kempgen wrote: Rajkumar S schrieb: and will generate events which will confuse ADM I am using to display a url for call. ADM? Asterisk Desktop Manager. http://adm.hamnett.org/ core show function DEVICE_STATE (on 1.6) is a good start. Thanks. raj ___ -- 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] Stress Testing IVR
On Wed, Feb 18, 2009 at 3:51 AM, David Backeberg dbackeb...@gmail.com wrote: As for actually putting delays and pressing the right buttons, you're on your own. You would need to write a custom AGI script specific to your IVR, and call it from your call file, which you then put in a bash loop. In that case, DTMF is your friend. Thanks for the tip, I will work in this direction and post any results to the list. raj ___ -- 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] What is the purpose of membermacro in queues.conf
On Tue, 17 Feb 2009, Mark Michelson wrote: The purpose of exposing these values is to allow for an administrator to use these for any purpose he may desire. An example would be really great :) I am confused because these values are exported just before the call is connected and I am wondering how can I intervene at this point and do some thing? Finally, you asked about membermacro. This allows for a macro to execute on a queue member's channel when he answers the call. This is very similar to the 'M' option for the dial application. Thanks, Does this support some thing similar to MACRO_RESULT ? with regards, raj ___ -- 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] Distributed presence in 1.6
Hi, Russell's blog[1] is down and there are not much information about this any where else. Any one with more information about res_ais and how it is used? raj [1] http://www.russellbryant.net/blog/index.php/2008/06/10/asterisk-16-now-with-distributed-presence/ ___ -- 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] What is the purpose of membermacro in queues.conf
Hi, There are 3 new settings (setinterfacevar, setqueueentryvar, setqueuevar) and membermacro settings in 1.6 queues.conf. What is the potential use of these settings? The variables set are useful, but there is no indication of the purpose they could be used? Any one with some light on potential use case of these new features? raj ___ -- 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] Stress Testing IVR
Hi, How can I stress test an asterisk IVR? I am looking for some kind of sip phone which can be programmed to send out digits after specified time to simulate users pressing menu items. If it can originate large number of calls simultaneously then it's great! Does any one have any recommendations ? Any other method to stress test an IVR call flow? with regards, raj ___ -- 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] RTCP SR transmission error, rtcp halted
Hi, While looking for the cause of disturbance in call I found this error coming in console RTCP SR transmission error, rtcp halted Google search only shows some bug reports relating to MOH and Hold. What could cause this message? Could this be a symptom causing call disturbance? Where should I start digging to find out the reason for this error? I am using Asterisk 1.4.19 with zaptel 1.4.9.2 Thanks and regards, raj ___ -- 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] Call transfer using agi
Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password of caller. I have an agi to change password and can transfer call to agi, but I do not know how to transfer the call back to agent from agi. So basically how can an agi transfer a call to an extension? Thanks and regards, raj ___ -- 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] Conference with an AGI inside Queue for password change
Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password of caller. One possible solution to this is for the agent to call an agi into conference with the call after caller has been verified. The agi will prompt for the password which the caller will type in his keypad. Although the agent will hear the password prompt, he cannot overhear the DTMF digits typed by caller. Can this be implemented in asterisk? I have looked but did not find any hints. Is there a better solution to the problem I am having? Thanks for reading and any replies. raj ___ -- 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] VoIP phones supporting speex
Hi, Any one with any experience with VoIP hard phones or adapters supporting speex? I looked around google but could not find any phones supporting speex. raj ___ -- 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] Correlating queue_logs and cdr for abandoned calls
Hi, I am using asterisk 1.4.19, my requirement is to find out which agents were ringed by the queue when a call is abandoned (or connected) in a call center. While this information is available in parts in queue_logs and cdr, there is no way to correlate this information. For example this is the queue_log entries for a call that was abandoned 1207935049|1207935049.6|queue|NONE|ENTERQUEUE||_0_ 1207935060|1207935049.6|queue|Agent/3501|RINGNOANSWER|1 1207935078|1207935049.6|queue|Agent/3501|RINGNOANSWER|1 1207935090|1207935049.6|queue|Agent/3501|RINGNOANSWER|1 1207935093|1207935049.6|queue|NONE|ABANDON|1|1|44 and cdr during this time: ,3501,5501,sip,3501 3501,Local/[EMAIL PROTECTED],2,SIP/5501-082495a8,Dial,SIP/5501,2008-04-11 17:30:49,,2008-04-11 17:31:00,11,0,NO ANSWER,DOCUMENTATION,1207935049.8, ,3501,5501,sip,3501 3501,Local/[EMAIL PROTECTED],2,SIP/5501-0824b5f8,Dial,SIP/5501,2008-04-11 17:31:08,,2008-04-11 17:31:18,10,0,NO ANSWER,DOCUMENTATION,1207935068.11, ,3501,5501,sip,3501 3501,Local/[EMAIL PROTECTED],2,SIP/5501-0824b5f8,Dial,SIP/5501,2008-04-11 17:31:19,,2008-04-11 17:31:30,11,0,NO ANSWER,DOCUMENTATION,1207935079.14, ,3501,5501,sip,3501 3501,Local/[EMAIL PROTECTED],2,SIP/5501-0824b5f8,Dial,SIP/5501,2008-04-11 17:31:31,,2008-04-11 17:31:33,2,0,NO ANSWER,DOCUMENTATION,1207935091.17, there are no common fields in both logs to correlate them. Also missing in the cdr is the entry for the call coming in. ie record if the call from the customer to the callcenter. This entry is present when the call is completed. Just wondering how others are dealing with this requirement and if missing cdr entry is a bug? regards, raj ___ -- 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] Combined patch fixing queue-state and bug12127 for 1.4.x
On Thu, Apr 3, 2008 at 12:16 PM, Rajkumar S [EMAIL PROTECTED] wrote: If some one has a combined patch that addresses both this issues for 1.4.x series that would be great! Just caughtup with Atis in #asterisk and got the url of state_interface patch against 1.4.19, its at http://ftp.iq-labs.net/state_interface-1.4/ Since 1.4.19 merged in bug 12127, Atis's patch combines both the patches. raj ___ -- 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] Combined patch fixing queue-state and bug12127 for 1.4.x
Hi, I am using asterisk-1.4.15, and using AddQueueMember to add SIP interface to the queue. Each sip interface is member of multiple queues The queue does not recognize that an agent is busy and keeps trying to call the busy agent. I have identified two patches that can fix the problem, one at http://www.scopserv.com/download/asterisk-1.4.17-state_interface.diff in thread http://lists.digium.com/pipermail/asterisk-dev/2008-January/031579.html and bug no 12127. I have applied both of them in a test system running 1.4.17 and it looks okay. But one of the hunk in 12127 had failed and I had to manually edit the code. I am not sure if what I have done it correct or not. If some one has a combined patch that addresses both this issues for 1.4.x series that would be great! Thanks a lot for your help and time, regards, raj ___ -- 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] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?
Thanks Atis, On Tue, Mar 18, 2008 at 3:50 AM, Atis Lezdins [EMAIL PROTECTED] wrote: As for current problem - i suspect that device state don't get updated correctly for Queue application, so Queue tries to dial device, and call-limit blocks it from doing so. There's a patch, currently in testing (issue 12127), it should fix this, however if you intend to keep incominglimit to 1, and don't use local channels - there's nothing to worry about. I had gone through bug 12127. Currently I am testing with 1.4 Trunk, dated 20th. so the 12127 patch is applied. But even in trunk the behavior does not change. I still get the [Mar 21 18:18:59] ERROR[29689]: chan_sip.c:3266 update_call_counter: Call to peer '2501' rejected due to usage limit of 1 But some times, usually when I start testing, I get this new message, when a call is picked up by agent. [Mar 21 18:18:28] WARNING[29684]: app_queue.c:3002 try_calling: The device state of this queue member, Agent/2503, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. I had gone through the UPGRADE.txt and now my sip.conf is like the following: [general] context=sip bindport=5060 bindaddr=0.0.0.0 srvlookup=yes limitonpeer = yes [2501] type=friend username=2501 secret=2501 canreinvite=no host=dynamic dtmfmode=rfc2833 context = sip disallow=all allow=ulaw call-limit=1 nat=1 Also the queue show command shows that the agent is Not in use, though the call is being taken. Agent/2503 (dynamic) (Not in use) has taken 3 calls (last was 26 secs ago) sip show inuse command shows the following output for SIP/2501 (the phone of Agent/2503) asterisk:/etc/asterisk# asterisk -rx sip show inuse | grep 2501 2501 0 1 2501 1/0 1 To me it seems asterisk (or my configurations) is still not recognising the fact that SIP peers are busy when attending calls from queues. Thanks in advance for any assistance in resolving this, raj ___ -- 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] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?
Hi, I am using asterisk-1.4.15, My sip configs is like [2501] type=friend username=2501 secret=2501 canreinvite=no host=dynamic dtmfmode=rfc2833 context = sip disallow=all allow=ulaw incominglimit=1 nat=1 queue.conf is like [gen-enq] joinempty = yes musiconhold = default strategy = rrmemory servicelevel = 60 timeout = 60 retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes monitor-format = wav ringinuse = no I am using AddQueueMember to add SIP interface to the queue. Each sip interface is member of multiple queues. Occasionally I get messages like [Mar 17 11:33:01] ERROR[9253]: chan_sip.c:3232 update_call_counter: Call to peer '2505' rejected due to usage limit of 1 [Mar 17 11:33:01] ERROR[9254]: chan_sip.c:3232 update_call_counter: Call to peer '2509' rejected due to usage limit of 1 [Mar 17 11:33:01] ERROR[9255]: chan_sip.c:3232 update_call_counter: Call to peer '2502' rejected due to usage limit of 1 [Mar 17 11:33:01] ERROR[9256]: chan_sip.c:3232 update_call_counter: Call to peer '2506' rejected due to usage limit of 1 in my asterisk console. At this point the mentioned sip phones are busy. My understanding is that if ringinuse is set to no, queue should not try and ring phones that are busy, but some how it is trying. How can I disable this behavior? With regards, raj ___ -- 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] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?
On Mon, Mar 17, 2008 at 6:30 PM, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: Forgot to add: Multiple queues fo sip phone, it is normal that sometimes it is ringed, as reported busy for 1 queue and free for another. you limitited incoming call to max 1 ' incominglimit=1' so ;) My understanding was that if a SIP phone is busy, either due to a call from queue or a call from another sip phone or even making an out bound call, the queue application would detect that and skip trying that channel. Is this assumption wrong ? raj ___ -- 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] incoming call popup
On Tue, Mar 4, 2008 at 7:18 PM, marek cervenka [EMAIL PROTECTED] wrote: can you recommend cleansimplestable solution for incoming call popup (in browser)? ADM http://adm.hamnett.org/ can invoke browsers when a call arrives. raj ___ -- 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] Pass arguments from extensions.conf
On Thu, Feb 14, 2008 at 9:52 PM, Naveen Palani [EMAIL PROTECTED] wrote: How can i pass the arguments from my dialplan to the ruby file. Is there a way i can do it with the agi script? Set them as variables in your extensions.conf and use them inside your agi scripts. raj ___ -- 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] How to check if a local channel member of a queue?
Hi, I am using asterisk-1.4.15 I have a queue with one agent added using AddQueueMember (FAO|Local/[EMAIL PROTECTED]|0||Agent/602). Once this command executes queue show FAO shows: FAO has 0 calls (max unlimited) in 'roundrobin' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 60s Members: Agent/602 (dynamic) (Not in use) has taken no calls yet There is no mention of the fact that which channel is used by Agent/602. I am adding agents to queue both from an agi using AddQueueMember, via phone, and manager command QueueAdd via web. In both cases I needs to find out whether the given sip channel has logged in to any queue previously, for proper validation. show agents command in previous version gave such details. How can I get the details given out by show agents in the new 1.4 scheme of things? With much regards, raj ___ -- 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] Transferring a call received by an agent in a queue
Hi, I have a queue with one agent added using AddQueueMember (FAO|Local/[EMAIL PROTECTED]|0||Agent/602). My extensions.conf is [general] static=yes writeprotect=yes autofallthrough=no clearglobalvars=no priorityjumping=no [from-sip] exten = 11000,1,Dial(SIP/11000,,t) exten = 1001,1,Dial(SIP/1001,,t) exten = 1002,1,Dial(SIP/1002,,t) exten = 1003,1,Dial(SIP/1003,,t) exten = 1004,1,Dial(SIP/1004,,t) exten = 2001,1,agi,login.php exten = 2002,1,Queue(FAO|tT) exten = 2004,1,MusicOnHold exten = 2004,2,Hangup When I call from 11000 to 1001, I can press # and type 2004 to transfer and 11000 gets MOH. When I dial 2002 (queue) from 11000, 1001 rings and I am able to talk both ways, but nothing happens when I press # at 1001. No logs appears at asterisk console in verbose 3 level. I am using asterisk 1.4.15. All the docs indicate that I just need to invoke Queue application with tT to enable call transfer. But that does not seems to work in my case. queues.conf [general] persistentmembers = no eventwhencalled = yes autofill = yes monitor-type = MixMonitor [FAO] musiconhold = default strategy = roundrobin servicelevel = 60 eventmemberstatus = yes eventwhencalled = yes timeout = 60 retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes monitor-format = gsm sip.conf [general] context=from-sip allowguest=no bindport=5060 bindaddr=192.168.3.36 srvlookup=yes [11000] host=dynamic type=friend dtmfmode=RFC2833 username=11000 secret=masked context=from-sip disallow=all allow=ulaw allow=alaw incominglimit=1 canreinvite=no [1001] host=dynamic type=friend dtmfmode=RFC2833 username=1001 secret=masked context=from-sip disallow=all allow=ulaw allow=alaw incominglimit=1 canreinvite=no Thanks and regards, raj ___ -- 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] createlink with out agents in 1.4
Hi, I am moving my call center to 1.4. Previously I was recording calls in agents.conf with the following config recordagentcalls=yes recordformat=wav createlink=yes So I had the filename in all calls which was *connected to agents*. I am looking for a similar functionality for 1.4. I am now recording calls using the following configuration. [general] persistentmembers = no eventwhencalled = yes autofill = yes monitor-type = MixMonitor [my-q] joinempty = yes musiconhold = default strategy = rrmemory servicelevel = 60 timeout = 60 retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes monitor-format = gsm The calls are being recorded, but no entry appears in cdr (obviously). I can add the filename to userfield using Set(CDR(userfield)=filename), just before calling Queue. But file name will be present in all calls that entered the queue, the previous behavior was that only those calls which was actually connected to agents had this entry. That field was one easy way to find out which calls were connected to agents by looking at the cdr alone, and I am using this feature in a home brew call analysis software. I would be very happy if this feature can be emulated in 1.4 with out using agents channel. thanks and regards, raj ___ -- 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] Limiting number of simultaneous calls in E1 line
Hi, I have a standard E1 line, but want to receive only 10 calls simultaneously. I want to give engaged tone to the 11th caller onwards. Can I configure E1 to do this? raj ___ --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] Agents and AddQueueMember
On Jan 4, 2008 4:21 PM, BJ Weschke [EMAIL PROTECTED] wrote: AddQueueMember(queuename[|interface[|penalty[|options[|membername): Thanks BJ Weschke and Alexandre Snarskii. Your mails together gives complete solution to my problem! raj ___ --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] Agents and AddQueueMember
Hi, I have callcenter running with v 1.2 with AgentCallbackLogin and now trying to move to 1.4 using the example doc, doc/queues-with-callback-members.txt. From what I understand the basic idea in the example is to 1. Authenticate a caller with VMAuthenticate 2. Get his SIP Channel number 3. Use ___ --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] Agents and AddQueueMember
Hi, I have callcenter running with v 1.2 with AgentCallbackLogin and now trying to move to 1.4 using the example doc, doc/queues-with-callback-members.txt. From what I understand the basic idea in the example is to 1. Authenticate a caller with VMAuthenticate 2. Get his SIP Channel number 3. Use AddQueueMember to add the SIP [Local] channel to queue. The queue logs come like 1197953879|1197953876.34|Auth-Enq|Local/[EMAIL PROTECTED]|CONNECT|3|1197953876.35 Previously it used to come like 1197013076|1197013055.27|Auth-Enq|Agent/1001|CONNECT|21|1197013055.30 Here the problem is that there is no way to find number of calls taken by a person, because there is no agent abstraction here. What is the recommended way to work around this problem ? raj PS: Apologies for my previous mail, that was sent when I clicked the wrong button. ___ --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] Interrupt rates and voip traffic
Hi, This is slightly off topic, but here I go any way... VoIP traffic has lot's of smaller packets, and since each packet can generate an interrupt, is there any way to determine the irq rates in a machine, and more importantly to know if I am hitting any of the limits in Linux or to determine how much interrupts per second can my box handle ? There seems to absolutely no information about his particular metric any where.. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fwd: [asterisk-users] Disconnect supervision in India?
On 1/1/07, ram [EMAIL PROTECTED] wrote: On 12/30/06, Rajkumar S [EMAIL PROTECTED] wrote: On 12/29/06, Chris Earle [EMAIL PROTECTED] wrote: anyone know the status of disconnect supervision on POTS lines in India? Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have disconnect supervision.. It does not work afaik, you may not get caller id also. I tested upto 1.4b3 and no luck. its all depends on the provider where you take from. Does any provider's land line works well with TDM Cards? raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnect supervision in India?
On 12/29/06, Chris Earle [EMAIL PROTECTED] wrote: anyone know the status of disconnect supervision on POTS lines in India? Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have disconnect supervision.. It does not work afaik, you may not get caller id also. I tested upto 1.4b3 and no luck. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ChanSpy * and 1234# not working
Hi, I am using ChanSpy with Asterisk 1.2.12.1. My extensions.conf has the following lines for ChanSpy exten = 1234,1,ChanSpy(Agent) exten = 1234,2,Hangup When I dial 1234 I can listen to one agent talking, but nothing happens if I press * or another agent number followed by #. Also archives are full of horror stories of asterisk 1.2.12.1 crashing with chanspy, does 1.2.13 have any patches related to that? regards, raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Condensing queue CDRs into single entry
Hi, When a call is made to a queue and picked up by agents at least 2 CDR entries are made, one from local to the agent's (sip) phone, and from incoming line to Agent. There are other entries generated when other conditions happen, like agent do not pickup phones and so on. Going through the cdr entries, there seems to be no common filelds by which all entries belonging to a single call can be picked up, so how can I extract all entries belonging to a single call, say if I have the CDR in a database. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fwd: [asterisk-users] Asterisk IVR functionality
On 11/13/06, nik600 [EMAIL PROTECTED] wrote: i have an application developed with bayonne. I would like to know if i can do these things whit asterisk: - IVR integration with database (mysql, insert,delete,update,select) Yes, you have to write AGI scripts to do this. - TTS No idea. - record exploration (for example, check if some resources are available in the database, and list them to the user (via TTS)) Should be possible via AGI. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extending a call limited by L in Dial app
Hi, If I use L(x[:y][:z]) in Dial app the call is limited to x milliseconds, Is it possible for the callee to extend the call past x milliseconds? raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager and Ruby
Hi, Any one using Rubi asterisk manager interface http://rubyforge.org/projects/rami/ ? How stable/usable it is? raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] S(x) - Hang up the call after 'x' seconds - Not working from queue
On 10/31/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Rajkumar S wrote: -- Executing Queue(SIP/1002-74e9, Auth-Enq|t) in new stack -- Started music on hold, class 'default', on channel 'SIP/1002-74e9' -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1' -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1001||tS(30)) in new stack -- Setting call duration limit to 30 seconds. -- Called 1001 -- Called Agent/1001 -- SIP/1001-d43c is ringing -- Agent/1001 is ringing -- SIP/1001-d43c answered Local/[EMAIL PROTECTED],2 -- Agent/1001 answered SIP/1002-74e9 -- Stopped music on hold on SIP/1002-74e9 == Spawn extension (from-sip, 1001, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Spawn extension (from-sip, 99, 1) exited non-zero on 'SIP/1002-74e9' Someone correct me if I'm wrong: The Dial string is missing a '/n' parameter for the Local channel. Without /n, Asterisk will do a native transfer to SIP/1001 and lose the timeout value defined earlier. What does '/n' refer here? There is no mention about this in the wiki. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] S(x) - Hang up the call after 'x' seconds - Not working from queue
On 11/1/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Rajkumar S wrote: On 10/31/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Someone correct me if I'm wrong: The Dial string is missing a '/n' parameter for the Local channel. Without /n, Asterisk will do a native transfer to SIP/1001 and lose the timeout value defined earlier. What does '/n' refer here? There is no mention about this in the wiki. It's in the wiki, see this: http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels Thanks Leo. I went though the code of the app_queue to find out if the cutoff value I gave in the dialplan is indeed being passed though when bridging is happening and it's not. The actual line where bridging is happening is bridge = ast_bridge_call(qe-chan,peer, bridge_config); The bridge_config is of type ast_bridge_config and holds the options to use for this bridging and it has a field called timelimit, which holds the timelimit of the call. This variable is not set in app_queue. This is the reason why the timelimit was not working when called from queue. I edited the code and put a sample value (in milliseconds) and the call cutoff is working fine. I am not sure if this introduces any side effects, but it's so far so good. Another advantage of this method is that the call cutoff will work only when the call is bridged from queue and not from directly called calls. Thanks for your help, Leo and Lenz. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] S(x) - Hang up the call after 'x' seconds - Not working from queue
Hi, I have a requirement to limit the calls to our agents via a queue to 5 minutes. I had posted this to a previous thread by name Maximum talktime in a queue? One work around that was suggested was to use the S(x) in the dial command to the agents, so that all calls to that extension would be terminated after x seconds. So I modified the dial command to the agent as: exten = 1001,1,Dial(SIP/1001,,tS(30)) Now when I call 1001 from another sip phone in the same context it get's disconnected after 30 seconds. -- Executing Dial(SIP/1002-b119, SIP/1001||tS(30)) in new stack -- Setting call duration limit to 30 seconds. -- Called 1001 -- SIP/1001-b605 is ringing -- SIP/1001-b605 answered SIP/1002-b119 -- Attempting native bridge of SIP/1002-b119 and SIP/1001-b605 == Spawn extension (from-sip, 1001, 1) exited non-zero on 'SIP/1002-b119' All is fine so far and it works as advertised. Now I am attempting a call via queue: -- Executing Queue(SIP/1002-74e9, Auth-Enq|t) in new stack -- Started music on hold, class 'default', on channel 'SIP/1002-74e9' -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1' -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1001||tS(30)) in new stack -- Setting call duration limit to 30 seconds. -- Called 1001 -- Called Agent/1001 -- SIP/1001-d43c is ringing -- Agent/1001 is ringing -- SIP/1001-d43c answered Local/[EMAIL PROTECTED],2 -- Agent/1001 answered SIP/1002-74e9 -- Stopped music on hold on SIP/1002-74e9 == Spawn extension (from-sip, 1001, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Spawn extension (from-sip, 99, 1) exited non-zero on 'SIP/1002-74e9' This call does not terminate after 30 seconds. I hope I have currently followed the tip from Lenz in my previous tip. with warm regards, raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Maximum talktime in a queue?
On 10/26/06, Lenz [EMAIL PROTECTED] wrote:
When you log in a callback agent, you enter first the agent code, and then
the extension he's sitting at. The context is usually specified in the
dialplan command, but the result is that asterisk knows that agent 103 is
sitting at [EMAIL PROTECTED] [EMAIL PROTECTED] is a working extension in a
working
context, where you do something on the lines of:
[agents]
exten = _2XX,1,Dial(SIP/${EXTEN})
In this dial command you're free to add whatever option you may like,
including the ones to limit call length.
I have
[sip]
exten = 605,1,Dial(SIP/605||S(30))
in my sip context (where agents are located) When I call this from
another sip phone, it works ie call gets terminated in 30 seconds, but
when this extension is called from queue it does not work.
The asterisk cli gives the following output
-- outgoing agentcall, to agent '605', on 'Local/[EMAIL PROTECTED],1'
-- Called Agent/605
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/605||S(30)) in new stack
-- Setting call duration limit to 30 seconds.
-- Called 605
-- SIP/605-082f56c0 is ringing
-- Agent/605 is ringing
As you can see the call duration is properly set in the logs.
The queue.conf has the following entries for the queue
musiconhold = default
strategy = roundrobin
servicelevel = 60
timeout = 60
retry = 5
wrapuptime=5
announce-frequency = 90
announce-holdtime = yes
member = Agent/605
Any help to get this working will be much appreciated.
raj
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Re: [asterisk-users] Maximum talktime in a queue?
Hi Lenz,
On 10/26/06, Lenz [EMAIL PROTECTED] wrote:
[agents]
exten = _2XX,1,Dial(SIP/${EXTEN})
In this dial command you're free to add whatever option you may like,
including the ones to limit call length.
I hope this helps
That did help. Thanks a lot!!
raj
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[asterisk-users] Maximum talktime in a queue?
Hi, Is it possible to define maximum talk time in a queue? ie any one who joins a queue should not be able to talk more than say 5 minutes to the agent. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum talktime in a queue?
Hi Lenz, On 10/25/06, Lenz [EMAIL PROTECTED] wrote: if you use Local channels for agents (or callback agents), you can easily do this in the Dial() command after the Local channel is called. I am using call back agents. Pardon me if this is obvious, but the dial is performed by the queue app, so how do I control the dial command? A bit more elaboration will be of great help. Of course your clients may get a bit angry at being disconnected, it is usually better to jave each agent stay aware od the call length and occasionally tolerate longer calls :) Just my $0.02 I am replacing an old call center system with asterisk, which had this facility. So the clients are pretty used to getting cut at 5 minutes. So every one tries to make the calls short and sweet :) thanks a lot, raj On Wed, 25 Oct 2006 15:06:35 +0200, Rajkumar S [EMAIL PROTECTED] wrote: Hi, Is it possible to define maximum talk time in a queue? ie any one who joins a queue should not be able to talk more than say 5 minutes to the agent. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes MP-20x
On 10/23/06, Andrew Nowrot [EMAIL PROTECTED] wrote: I've been testing this for 3 weeks now. No problems so far. This gateway has many features including IPSec and is not that expensive. Appreciate if you can post the sample configs to wiki or to the list. There is no information about configuring Audiocodes with asterisk. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes MP-20x
On 10/23/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf Seems like a good device, but I can't seem to find anyone actually using them... I am using an AudioCodes Mediant1000 and now trying to configure MP-118. The mediant1000 works well, and I will update the wiki some time soon with the exact configurations to get it working. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP trunk from an Audiocodes mediant 1000
Hi, I am configuring an audiocodes Medant1000 to talk to my asterisk box. So far I have successfull in landing a single call from mediant to my *box. my sip conf is as follows: [general] context=sip bindport=5060 bindaddr=0.0.0.0 srvlookup=yes [3911700] type=friend host=dynamic dtmfmode=info secret=blah context=sip where 3911700 is my E1 telephone no. in my extensions.conf I have exten = 3911700,1,Dial(SIP/100) When I dial from outside to my E1 number calls are coming like the following: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.9.230;branch=z9hG4bKac806223297 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];tag=1c806218385 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED] Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Remote-Party-ID: sip:[EMAIL PROTECTED];party=called;npi=1;ton=4 Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;privacy=off;screen=yes;screen-ind=1;npi=1;ton=0 User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.4.60A.016.003 Content-Type: application/sdp Content-Length: 348 and the call get's connected to SIP/100 via the line in extensions.conf But what I am expecting is that the calls to come to the context's 's' extension. I am not sure if the changes are to be done in Asterisk or to Mediant. Any help in this will be much appreciated. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] India:Reliance - E1configuration using TE110P
Hi, I bought an asterisk TE110P to connect to our Reliance Infocomm E1 line to asterisk, I have loaded the driver, but looking for an appropriate zaptel.conf and zapata.conf. I googled a lot but there does not seems to be any india specific configuration. If any one has successfully configured this on a Reliance E1 line, I would be very grateful if you can share the appropriate entries in zaptel and zapata conf. with warm regards, raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No voice for when using Playback and background
Hi, I am using 1.2.12.1 (actually was using 1.2.11, and upgraded) it's connected to a Cisco ATA 188. The phones connected to ATA can register to * and two phones connected to ATA can call each other. I can hear Music On Hold, when called using the following fragment exten = 6000,1,Answer exten = 6000,2,MusicOnHold() But the Playback and Background does not work, ie I cannot hear any thing. exten = 200,1,Playback(tt-allbusy) exten = 200,n,Playback(moo2) The sip.conf fragment for ATA Phone is [100] type=friend username=100 secret=password canreinvite=no host=dynamic dtmfmode=rfc2833 context = sip nat=1 Actually this was working couple of days back, the last modification done was to install zaptel and libpri. I have looked far and wide in google,but nothing came up. Any help to fix this will be much appreciated. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fwd: [asterisk-users] No voice for when using Playback and background
On 10/5/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: See if adding an answer line helps: Rajkumar S wrote: exten = 200,1,Playback(tt-allbusy) exten = 200,n,Playback(moo2) change to: exten = 200,1,Answer exten = 200,n,Playback(tt-allbusy) exten = 200,n,Playback(moo2) Nope, Infact I had tried this before posting to the list. The full sip debug is: -- SIP read from 192.168.9.230:5060: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.9.230:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3810654101 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp User-Agent: Cisco ATA 188 v2.16.1 ata18x (030709a) Expires: 300 Content-Length: 246 Content-Type: application/sdp v=0 o=100 8904 8904 IN IP4 192.168.9.230 s=ATA186 Call c=IN IP4 192.168.9.230 t=0 0 m=audio 16384 RTP/AVP 0 4 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (11 headers 11 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.9.230 : 5060 (non-NAT) Reliably Transmitting (NAT) to 192.168.9.230:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.9.230:5060;received=192.168.9.230 From: sip:[EMAIL PROTECTED];user=phone;tag=3810654101 To: sip:[EMAIL PROTECTED];user=phone;tag=as43f3d7b7 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=453e6aef Content-Length: 0 -- SIP read from 192.168.9.230:5060: ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.9.230:5060;received=192.168.9.230 From: sip:[EMAIL PROTECTED];user=phone;tag=3810654101 To: sip:[EMAIL PROTECTED];user=phone;tag=as43f3d7b7 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK User-Agent: Cisco ATA 188 v2.16.1 ata18x (030709a) Content-Length: 0 --- (8 headers 0 lines)--- -- SIP read from 192.168.9.230:5060: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.9.230:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3810654101 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp User-Agent: Cisco ATA 188 v2.16.1 ata18x (030709a) Proxy-Authorization: Digest username=100,realm=asterisk,nonce=453e6aef,uri=sip:[EMAIL PROTECTED],response=4f0cfbdda408c879f8ac15bd27bcc02c Expires: 300 Content-Length: 246 Content-Type: application/sdp v=0 o=100 8906 8906 IN IP4 192.168.9.230 s=ATA186 Call c=IN IP4 192.168.9.230 t=0 0 m=audio 16384 RTP/AVP 0 4 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (12 headers 11 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.9.230 : 5060 (NAT) Found user '100' Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.9.230:16384 Found description format PCMU Found description format G723 Found description format PCMA Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 200 in sip (domain 192.168.9.224;user=phone) list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Transmitting (NAT) to 192.168.9.230:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.9.230:5060;received=192.168.9.230 From: sip:[EMAIL PROTECTED];user=phone;tag=3810654101 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Executing Playback(SIP/100-081b28b8, tt-allbusy) in new stack We're at 192.168.9.224 port 14652 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.9.230:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.9.230:5060;received=192.168.9.230 From: sip:[EMAIL PROTECTED];user=phone;tag=3810654101 To: sip:[EMAIL PROTECTED];user=phone;tag=as35a40f82 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 218 v=0 o=root 14808 14808 IN IP4 192.168.9.224 s=session c=IN IP4 192.168.9.224 t=0 0 m=audio 14652 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Playing 'tt-allbusy' (language 'en') -- SIP read from 192.168.9.230:5060: ACK sip:[EMAIL PROTECTED] SIP
Re: [asterisk-users] Screen pop based on incoming DID
On 10/3/06, Greg Delgado [EMAIL PROTECTED] wrote: I want to pop up a web page when a queue member phone rings but, instead of displaying the clid, I want to display the DID number the call came in. Any ideas how to best implement this? Checkout Asterisk Desktop Manager at http://adm.hamnett.org/ It might be able to do what you want with some customisation. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Picking up a call from queue?
Hi, Is it possible to pick up a call that's in queue and pass it to an agent directly. The use case is that some times some important calls land up in queue which I need to pickup immediatly and pass it on to an agent. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mediant 1000
Hi, I am looking for some docs to help configure a AudioCodes Mediant 1000 with asterisk, any tips or examples are appreciated. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call notification for queues?
Hi, Is there a way to do call notification to a desktop when a call is connected from a queue to an agent ? I have seen the call notification page in wiki, but they do not deal with queues. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue Messages now playing when caller is inside queue
Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using Playback with this stanza: exten = 900,1,Playback(queue-youarenext) exten = 900,2,Playback(queue-thereare) exten = 900,3,Playback(digits/three) exten = 900,4,Playback(queue-callswaiting) exten = 900,5,Playback(vm-ivr) The queue is invoked by: exten = s,1,Background(Welcome) exten = s,2,Queue(callcenter|tT|||300) extern = s,3,Hangup When I tried exten = s,2,Queue(callcenter|tTr|||300) It was ringing with out music on hold, but again with out any announcement. Queue.conf is: [general] [default] [callcenter] music=default leavewhenempty = yes monitor-format = wav strategy=rrmemory timeout=15 retry=5 servicelevel = 60 wrapuptime=5 eventwhencalled = yes eventmemberstatusoff = no maxlen = 0 announce-frequency = 120 announce-holdtime = yes queue-thankyou = vm-ivr queue-youarenext = queue-youarenext ; (You are now first in line.) queue-thereare = queue-thereare ; (You are Currently caller no) queue-callswaiting = queue-callswaiting ; (calls waiting.) queue-holdtime = queue-holdtime ; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) context=vm member = Agent/1000 member = Agent/1001 member = Agent/1002 member = Agent/1003 member = Agent/1004 member = Agent/1005 The funny part is that it's working perfectly in the old setup. Did I make some mistake some where? I am running on debian stable and asterisk was compiled with simple make;make install. raj ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Messages not playing when caller is inside queue
David Ankers wrote: Don't you need an exten = s,1,Answer The full sequence is: [ivr] ; Voice Menu exten = s, 1, wait(2) exten = s, 2, Answer exten = s, 3,Goto,MainMenu|s|1 [MainMenu] exten = s,1,Background(Welcome) exten = s,2,Queue(callcenter|tT|||600) extern = s,3,Hangup I am sorry that I missed this. The call is getting picked up and it goes to the agent in the queue. That part is fine. The only thing missing is that the messages (like queue-youarenext, queue-thankyou) are not played upon entering the queue. raj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rajkumar S Sent: Monday, 20 February 2006 3:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Queue Messages now playing when caller is insidequeue Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using Playback with this stanza: exten = 900,1,Playback(queue-youarenext) exten = 900,2,Playback(queue-thereare) exten = 900,3,Playback(digits/three) exten = 900,4,Playback(queue-callswaiting) exten = 900,5,Playback(vm-ivr) The queue is invoked by: exten = s,1,Background(Welcome) exten = s,2,Queue(callcenter|tT|||300) extern = s,3,Hangup When I tried exten = s,2,Queue(callcenter|tTr|||300) It was ringing with out music on hold, but again with out any announcement. Queue.conf is: [general] [default] [callcenter] music=default leavewhenempty = yes monitor-format = wav strategy=rrmemory timeout=15 retry=5 servicelevel = 60 wrapuptime=5 eventwhencalled = yes eventmemberstatusoff = no maxlen = 0 announce-frequency = 120 announce-holdtime = yes queue-thankyou = vm-ivr queue-youarenext = queue-youarenext ; (You are now first in line.) queue-thereare = queue-thereare ; (You are Currently caller no) queue-callswaiting = queue-callswaiting ; (calls waiting.) queue-holdtime = queue-holdtime ; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) context=vm member = Agent/1000 member = Agent/1001 member = Agent/1002 member = Agent/1003 member = Agent/1004 member = Agent/1005 The funny part is that it's working perfectly in the old setup. Did I make some mistake some where? I am running on debian stable and asterisk was compiled with simple make;make install. raj ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Messages not playing when caller is inside queue
Peter Fern wrote: In queues.conf: [queuename] announce-frequency = XX ; where XX = number of seconds I had already given it. From my orig mail: [callcenter] music=default leavewhenempty = yes monitor-format = wav strategy=rrmemory timeout=15 retry=5 servicelevel = 60 wrapuptime=5 eventwhencalled = yes eventmemberstatusoff = no maxlen = 0 announce-frequency = 120 announce-holdtime = yes queue-thankyou = vm-ivr queue-youarenext = queue-youarenext ; (You are now first in line.) queue-thereare = queue-thereare ; (You are Currently caller no) queue-callswaiting = queue-callswaiting ; (calls waiting.) queue-holdtime = queue-holdtime ; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) context=vm member = Agent/1000 member = Agent/1001 member = Agent/1002 member = Agent/1003 member = Agent/1004 member = Agent/1005 Again this config is working perfectly in the 1.0.9-BRIstuffed-0.2.0-RC8h (Xorcom Rapid), but not in 1.2.4 raj ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitor Logged in Agent's conversation
Hi, Is it possible to monitor conversation of logged in Agents? Currently I am using ZapScan to monitor incoming calls, but I would like to monitor individual agents. raj ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 AgentCallbackLogin Questions
Hi, We have a small callcenter with about 5 agents, logging in via SIP (SJPhone) using AgentCallbackLogin and incoming calls via Zap. I am running Asterisk 1.0.9-BRIstuffed-0.2.0-RC8h (Rapid Distribution) Some times some agents forget to logout when they go and when the next person comes in, he logins with his username, while the old login persists. When an incoming call comes in while all agents are busy * assumes that the forgotten login is free and tries to send the call there. This is wrong behavior. In order to prevent this situation is it possible to deny logging in from a SIP device if some one else is also logged in ? Possibly telling that Agent no blah blah is logged in already. Second problem happens when some times the windows machines used by agents freezes. Once the machine freezes the agents reboot the machine. The problem is if they were on call while the machine froze the Zap channel does not gets hanged up and even otherwise the Agent will not be able to Log back again. When this happens I soft hangup any Zap or Local channel associated with the SIP device, and agents will be able to log back again. Is there any way to clear all channels associated with a sip channel with out using the soft hangup ? Basically I want agents to call a particular number if their machine hangs and that should clear all dead channels and agent also should be logged out. The agent can log back again after this. Thanks for reading my rant and any help will be much appreciated. raj ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallerID for BSNL (India) phones
Hi, What must be done to enable callerid and call progress monitoring (disconnect notification) for Zap lines connected to BSNL phones in India. I am willing to get documentation, test or write the necessary code to get it working. I have gone through the indications.conf, will that be sufficient? Any one to help me get there? raj ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID for BSNL (India) phones
Gurminder Arora wrote: Hi raj, Perhaps both of us are going through same tunnel... Along with all the Zap users in India :) Occasionally there will be a post in the list about Zap support for India, but even now there is no CallerID or Call Progress monitoring for India. I know a bit of coding and am willing to work on it. raj ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Callcenter and Softphone hanging
Hi, I run a small inbound callcenter with 3 agents doing techsupport. The agents are logged in via softphone, using agentcallback login. Some times the agents PC running softphone hangs, and they reboot the PC. But * is not aware of this and tries to send calls to the PC, which gets rejected. -- outgoing agentcall, to agent '1009', on 'Local/[EMAIL PROTECTED],1' -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1002) in new stack Oct 1 23:16:51 NOTICE[16907]: app_dial.c:777 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time -- Called Agent/1009 -- Timeout on Local/[EMAIL PROTECTED],2 == CDR updated on Local/[EMAIL PROTECTED],2 -- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack Is there any way to logoff an agent from the queue in such cases from the * prompt? Any better way to handle this issue? raj ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] didgium card in india
Capt MS wrote: thanks for the reply Is Digium card compatible with EPABX standards available in india , further how much does a card with three FXS and one FXO interface cost, Do u have any experience of implenting the same , I am in army what we lookin at is voice gateway to interface our PBX with the data network so that we have one underlying network to handle , any suggestions on how to implement in a cost effective manner. I am using Digium card in India (Trivandrum, Kerala) for a small call center application. What I did was to purchase the card in US, send it across to my friend in his US address and he brought it along when he came, but I guess this option is not applicable to you. 3 FXS and 1 FSO will cost some thing under Rs. 15,000, with out duty. See here for exact prices. http://store.yahoo.com/asteriskpbx/noname.html I tried it here with BSNL and a Siemens PBX, I am not receiving the callerid and it does not detect remote hangup. Pl mail me offline if you need further information. regards, raj ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call getting disconnected in queue
Bump! raj Rajkumar S wrote: Hi, I have a small call center with 4 Zap lines and 4 agents. Agents login using sip phones with AgentCallbackLogin. I occasionally gets a complaint that when customers call the call center, after the initial greeting is over the call gets cut after playing the thank you message. I started investigating and found that that happens when the call gets transferred to an agent who is making an outbound call (either calling customers or logging out). The debug logs of one such conversation is given below: As you can read below, the call gets fwd to agent 1005 at SIP/1004. But he is trying to log off at the same time, and call gets disconnected. Any help to fix this will be very much appreciated. regards, raj -- Executing Answer(Zap/2-1, ) in new stack -- Executing Goto(Zap/2-1, MainMenu|s|1) in new stack -- Goto (MainMenu,s,1) -- Executing BackGround(Zap/2-1, Welcome) in new stack -- Playing 'Welcome' (language 'en') -- Playing 'agent-incorrect' (language 'en') == Spawn extension (from-sip, 2002, 1) exited non-zero on 'SIP/1004-cc2f' -- Executing Queue(Zap/2-1, callcenter|tT|||300) in new stack -- Started music on hold, class 'default', on Zap/2-1 -- Stopped music on hold on Zap/2-1 -- Playing 'queue-youarenext' (language 'en') -- Executing AgentCallbackLogin(SIP/1004-e376, |l) in new stack -- Playing 'agent-user' (language 'en') -- Told Zap/2-1 in callcenter their queue position (which was 1) -- Playing 'queue-thankyou' (language 'en') -- Started music on hold, class 'default', on Zap/2-1 -- outgoing agentcall, to agent '1005', on 'Local/[EMAIL PROTECTED],1' -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1004) in new stack Sep 21 11:56:39 ERROR[12633]: chan_sip.c:1615 update_user_counter: Call from user '1004' rejected due to usage limit of 1 -- Couldn't call 1004 == Everyone is busy/congested at this time -- Called Agent/1005 -- Playing 'agent-incorrect' (language 'en') == Spawn extension (from-sip, 2002, 1) exited non-zero on 'SIP/1004-e376' -- Timeout on Local/[EMAIL PROTECTED],2 == CDR updated on Local/[EMAIL PROTECTED],2 -- Executing BackGround(Local/[EMAIL PROTECTED],2, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'en') -- Agent/1005 answered Zap/2-1 -- Stopped music on hold on Zap/2-1 -- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack == Spawn extension (from-sip, t, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-in.wav /var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-out.wav /var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010.wav rm -f /var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-* ) == Spawn extension (MainMenu, s, 2) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' sip.conf entry for the phone is [1004] host=dynamic type=friend dtmfmode=RFC2833 username=1004 secret=password context = from-sip disallow=all allow=speex allow=gsm incominglimit=1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call getting disconnected in queue
Hi, I have a small call center with 4 Zap lines and 4 agents. Agents login using sip phones with AgentCallbackLogin. I occasionally gets a complaint that when customers call the call center, after the initial greeting is over the call gets cut after playing the thank you message. I started investigating and found that that happens when the call gets transferred to an agent who is making an outbound call (either calling customers or logging out). The debug logs of one such conversation is given below: As you can read below, the call gets fwd to agent 1005 at SIP/1004. But he is trying to log off at the same time, and call gets disconnected. Any help to fix this will be very much appreciated. regards, raj -- Executing Answer(Zap/2-1, ) in new stack -- Executing Goto(Zap/2-1, MainMenu|s|1) in new stack -- Goto (MainMenu,s,1) -- Executing BackGround(Zap/2-1, Welcome) in new stack -- Playing 'Welcome' (language 'en') -- Playing 'agent-incorrect' (language 'en') == Spawn extension (from-sip, 2002, 1) exited non-zero on 'SIP/1004-cc2f' -- Executing Queue(Zap/2-1, callcenter|tT|||300) in new stack -- Started music on hold, class 'default', on Zap/2-1 -- Stopped music on hold on Zap/2-1 -- Playing 'queue-youarenext' (language 'en') -- Executing AgentCallbackLogin(SIP/1004-e376, |l) in new stack -- Playing 'agent-user' (language 'en') -- Told Zap/2-1 in callcenter their queue position (which was 1) -- Playing 'queue-thankyou' (language 'en') -- Started music on hold, class 'default', on Zap/2-1 -- outgoing agentcall, to agent '1005', on 'Local/[EMAIL PROTECTED],1' -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1004) in new stack Sep 21 11:56:39 ERROR[12633]: chan_sip.c:1615 update_user_counter: Call from user '1004' rejected due to usage limit of 1 -- Couldn't call 1004 == Everyone is busy/congested at this time -- Called Agent/1005 -- Playing 'agent-incorrect' (language 'en') == Spawn extension (from-sip, 2002, 1) exited non-zero on 'SIP/1004-e376' -- Timeout on Local/[EMAIL PROTECTED],2 == CDR updated on Local/[EMAIL PROTECTED],2 -- Executing BackGround(Local/[EMAIL PROTECTED],2, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'en') -- Agent/1005 answered Zap/2-1 -- Stopped music on hold on Zap/2-1 -- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack == Spawn extension (from-sip, t, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-in.wav /var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-out.wav /var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010.wav rm -f /var/spool/asterisk/monitor/agent-1005-asterisk-12632-1127283999-1010-* ) == Spawn extension (MainMenu, s, 2) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' sip.conf entry for the phone is [1004] host=dynamic type=friend dtmfmode=RFC2833 username=1004 secret=password context = from-sip disallow=all allow=speex allow=gsm incominglimit=1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AgentCallbackLogin and calling outside
Hi, I have a small callcenter with 3 agents who login using AgentCallbackLogin. They normally receive calls, but needs to call outside also. When they call outside, though they are busy the show agents shows them as available, and calls gets routed to them. How can I make them busy when they call outside. Also they also need to move out for couple of minutes or to send a mails etc in between calling. Right now only way to do this is to logoff and then login back. This is bit tedious esp for short breaks. If I put DND in the client (sjphone) the call will land in the agent and gets disconnected. Is there any way to achieve this? with warm regards, raj ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AgentCallbackLogin and calling outside
BJ Weschke wrote: For your outbound calling problem, if you're operating with CVS-HEAD you can PauseQueueMember and then UnpauseQueueMember as part of the dial-plan for your outbound calls for those agents. Thanks, I think this will do the trick. For short breaks, I can wrap this around an echo test and let the agents call that number to take a break. I am using the Xorcom Rapid 1.1, I will have to check if PauseQueueMember is available in that. raj ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] show queue callcenter output?
Hi, Can some one tell me what is the meaning of all the fields of show queue callcenter? for example in my system it gives: callcenter has 0 calls (max unlimited) in 'roundrobin' strategy (33s holdtime), C:429, A:12, SL:0.0% within 0s How is the holdtime calculated? what is A and SL? Also how can I see which of my zap interfaces are busy currently? I did a zap show channels I get this output, but no indication as to which is busy and which is free. rapid*CLI zap show channels Chan Extension Context Language MusicOnHold pseudoivr 1ivr 2ivr 3ivr 4ivr Thanks and regards, raj ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitoring RTP protocol
Bohuslav Coufal wrote: Hi all, is it possible to monitor RTP protocol (latency, errors, ...) by Asterisk or other software. Try http://tstat.tlc.polito.it/ quote Tstat, a passive sniffer able to provide several insight on the traffic patterns at both the the network and transport levels. /quote I have not tried it myself, just have it in my bookmarks. raj ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (cause 66 - Channel not implemented) -- IAX?
Joseph wrote: [EMAIL PROTECTED] wrote: I am using firefly as my iax client, and it does not seems to work when I use 1001,1,Dial(IAX2/1001) instead of 1001,1,Dial(IAX/1001) Change the lines below from IAX to IAX2 Thanks a lot Joseph for your reply. As you can see from my mail, I had done that. But firefly is not working with (nor with sip). If Dial(IAX/ will not work, any sugg for a good windows IAX2 client? raj ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap to zap bridging not hanging up
Paradise Dove wrote: i have the same problem. it seems to be a bug. Is this related to the problem i posted yesterday (in a mail with subject Zap channel not hangingup raj On 6/5/05, Master Abi [EMAIL PROTECTED] wrote: Hi I am trying to develop a night divert. Caller dials in after hours on Zap and it gets divert to a mobile number via a second Zap. The call bridges but will not hangup the channels when the parties finish. Is there something I am missing or an dial option that I should be using. I am using latest CVS. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap to zap bridging not hanging up
Rich Adamson wrote: The *proper* way to see it is with a voltmeter. your off-hook voltage should be between roughly -5 and -15 Volts DC. CPD should either disconnect the battery (0V) or reverse the battery (-5-15VDC) briefly upon remote party hangup. Just to add to Andrew's comment above, the majority of US analog pstn lines will disconnect battery for about 400 milliseconds. However, a fair number of non-US countries either provide no disconnect supervision, or use some other approach (eg, tones). Thanks for your suggestions. Let me check with my provider if the network supports CPD. raj ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
