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Duplicate UUID! message when using origination_uuid w/ originate
by Phillip Jones 23 Jul '10

23 Jul '10

Hi there, I have the following simple test script running in mod_managed: Session.Answer(); string Uuid = fsApi.ExecuteString("create_uuid"); string originateString = string.Format(@"{{ignore_early_media=true}}[origination_uuid={0}]user/3478309995(a)64.40.xx.xxx'&park'", Uuid); apiResult = fsApi.Execute("originate", originateString); Session.Hangup("NORMAL_CLEARING"); When I run this script and I anwser the call, the console shows the error message: [CRIT] switch_core_session.c:1287 Duplicate UUID! However 'show channels' reveals that the UUID is the expected value (that given in origination_uuid=), when the call is ringing and after it is answered. It is as if, FS is creating the call with the given UUID, and then on answer trying to set the UUID to the given UUID again, but failing when it see the existing call. I updated to the latest GIT HEAD just before sending this email. Any ideas? Is this something I should be worried about - or am I using origination_uuid incorrectly? Thanks 2010-07-22 21:49:37.843750 [DEBUG] sofia_glue.c:2442 Set Codec sofia/internal/sip:3478309995@71.225.60.199:5900 PCMU/8000 20 ms 160 samples 2010-07-22 21:49:37.843750 [DEBUG] sofia_glue.c:3935 Set 2833 dtmf send payload to 101 2010-07-22 21:49:37.843750 [DEBUG] sofia_glue.c:2682 AUDIO RTP [sofia/internal/sip:3478309995@71.225.60.199:5900] 64.40.12.122 port 26654 -> 71.225.60.199 port 32126 codec: 0 ms: 20 2010-07-22 21:49:37.843750 [DEBUG] switch_rtp.c:1413 Starting timer [soft] 160 bytes per 20ms 2010-07-22 21:49:37.843750 [DEBUG] sofia_glue.c:2892 Set 2833 dtmf send payload to 101 2010-07-22 21:49:37.843750 [DEBUG] sofia_glue.c:2897 Set 2833 dtmf receive payload to 101 2010-07-22 21:49:37.843750 [DEBUG] switch_channel.c:2539 (sofia/internal/ sip:3478309995@71.225.60.199:5900) Callstate Change RINGING -> ACTIVE 2010-07-22 21:49:37.843750 [NOTICE] sofia.c:4876 Channel [sofia/internal/ sip:3478309995@71.225.60.199:5900] has been answered 2010-07-22 21:49:37.843750 [DEBUG] switch_ivr_originate.c:3351 Originate Resulted in Success: [sofia/internal/sip:3478309995@71.225.60.199:5900] 2010-07-22 21:49:37.843750 [CRIT] switch_core_session.c:1287 Duplicate UUID! 2010-07-22 21:49:37.843750 [CRIT] switch_core_session.c:457 sofia/internal/ sip:3478309995@71.225.60.199:5900 set UUID=73e6a583-a88c-42be-8908-860a637a67cb FAILED 2010-07-22 21:49:37.843750 [DEBUG] switch_ivr_originate.c:3351 Originate Resulted in Success: [sofia/internal/sip:3478309995@71.225.60.199:5900] 2010-07-22 21:49:37.843750 [DEBUG] mod_commands.c:2936 (sofia/internal/ sip:3478309995@71.225.60.199:5900) State Change CS_CONSUME_MEDIA -> CS_EXECUTE 2010-07-22 21:49:37.843750 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/sip:3478309995@71.225.60.199:5900 [BREAK] 2010-07-22 21:49:37.843750 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:3478309995@71.225.60.199:5900) Running State Change CS_EXECUTE 2010-07-22 21:49:37.843750 [ALERT] switch_cpp.cpp:1177 +OK 73e6a583-a88c-42be-8908-860a637a67cb 2010-07-22 21:49:37.843750 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/sip:3478309995@71.225.60.199:5900) State EXECUTE 2010-07-22 21:49:37.843750 [DEBUG] mod_sofia.c:235 sofia/internal/ sip:3478309995@71.225.60.199:5900 SOFIA EXECUTE 2010-07-22 21:49:37.843750 [DEBUG] switch_core_state_machine.c:157 sofia/internal/sip:3478309995@71.225.60.199:5900 Standard EXECUTE EXECUTE sofia/internal/sip:3478309995@71.225.60.199:5900 park() 2010-07-22 21:49:37.843750 [DEBUG] switch_cpp.cpp:602 CoreSession::hangup

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Re: [Freeswitch-users] Active Calls on a gateway
by David Ponzone 23 Jul '10

23 Jul '10

Sorry, I am tired. Of course, there is a way. You have to use mod_limit, setting variables in your all inbound dialplans where required. For instance, you will put this in all the required extensions in the context handling calls from the gateway: <action application="limit" data="hash gatewayX all" /> <action application="limit" data="hash gatewayX from" /> and you will put this in all the required extensions in the contexts handling calls to the gateway: <action application="limit" data="hash gatewayX all" /> <action application="limit" data="hash gatewayX to" /> This way, you will have a counter for all calls, one for calls coming from the gateway and one for calls going to the gateway. David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 23/07/2010 à 15:17, Ahmed Ajmal a écrit : > Show calls is too verbose plus it will show me calls on all the > gateways, I > am looking for something more specific. > > -ahmed > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of > Evgeniy > Zolotov > Sent: Friday, July 23, 2010 5:59 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Active Calls on a gateway > > Please try this command : > > show calls > > Ahmed Ajmal ?????: >> >> Hi >> >> Is there any way I can find the total number of active calls on a >> gateway. I have tried the following command on the CLI: >> >> sofia xmlstatus gateway gwname >> >> and it always returns 0 for Call In (assuming Call In represents the >> active calls). I am on freeswitch 1.0.6. >> >> Thanks >> >> Ahmed >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org

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Scalable System
by Samuel Macedo 23 Jul '10

23 Jul '10

Hi, I have a conference system that use ESL Outbound connection, users can access the conference by an IVR menu. Today my conference receives about 50 simultaneous calls and this number is increasing. I want to develop a kind of architecture to receive a large amount of calls, but I don't know how. This is the possibilities: 1. I have a Intel Xeon 3460 (2.8GHz) server. I can upgrade the machine and this will increase the number of calls. How many extra calls will the server support if I upgrade it? Is there any kind of statistics? 2. I can have a entrance server that will balance the calls in different machines, so I will transfer the calls to the correct server by the conference number. Conferences from 0 - 500 I send to *Server A* and conferences from 501 - 1000 send to *Server B*. The problem of this solution is the transfer operation - I suppose that it's an expensive action. Is there any other option? Thanks, -- Samuel Macedo

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Re: [Freeswitch-users] calls ending with MEDIA_TIMEOUT
by Dan 23 Jul '10

23 Jul '10

I appologize if I'm not making sense or barking up the wrong tree, I'm hoping that this will be obvious to someone else. I'm comparing a call that succeeds and one that times out. With both calls I see that after the initial invite from the far end device, freeswitch sends a 180 and attempts to reach the device's advertised RTP port before sending a 200 OK. With the call that times-out its interesting that it sends these two packets ( src port 27176 to dest port 27176) but then send 2 more with the same source port but to a different destination port that I don't see negotiated anywhere (src port 27176 to dest port 19488). The call that succeedes (see below below) does not have this behavior. The boxes have full visibility to each other, there is no nat-ing or firewalls between the two internal networks (just a router). Any thoughts? Media timeout call No. Time Source Source Port Destination Dest Port Protocol Info 1 0.000000 192.168.21.4 56397 192.168.24.22 onscreen SIP/SDP Request: INVITE sip:3103568788@192.168.24.22:5080, with session description 2 0.000367 192.168.24.22 onscreen 192.168.21.4 sip SIP Status: 100 Trying 3 0.000369 192.168.24.22 onscreen 192.168.21.4 sip SIP Status: 100 Trying 4 0.007171 192.168.24.22 27176 192.168.24.22 27176 UDP Source port: 27176 Destination port: 27176 5 0.017542 192.168.24.22 27176 192.168.24.22 27176 UDP Source port: 27176 Destination port: 27176 6 0.017613 192.168.24.22 27176 192.168.21.4 19488 UDP Source port: 27176 Destination port: 19488 7 0.017616 192.168.24.22 27176 192.168.21.4 19488 UDP Source port: 27176 Destination port: 19488 8 0.017809 192.168.24.22 onscreen 192.168.21.4 sip SIP/SDP Status: 200 OK, with session description 9 0.017810 192.168.24.22 onscreen 192.168.21.4 sip SIP/SDP Status: 200 OK, with session description 10 0.020167 192.168.21.4 56397 192.168.24.22 onscreen SIP Request: ACK sip:3103568788@192.168.24.22:5080;transport=udp 11 0.186076 192.168.21.4 16752 192.168.24.22 27176 RTP PT=ITU-T G.711 PCMU, SSRC=0x21B51504, Seq=10837, Time=1575317876, Mark 12 0.186121 192.168.24.22 27176 192.168.21.4 16752 RTP PT=ITU-T G.711 PCMU, SSRC=0x33CF6C8, Seq=5202, Time=240, Mark 13 0.186126 192.168.24.22 27176 192.168.21.4 16752 RTP PT=ITU-T G.711 PCMU, SSRC=0x33CF6C8, Seq=5202, Time=240, Mark 14 0.216085 192.168.21.4 16752 192.168.24.22 27176 RTP PT=ITU-T G.711 PCMU, SSRC=0x21B51504, Seq=10838, Time=1575318116 15 0.246105 192.168.21.4 16752 192.168.24.22 27176 RTP PT=ITU-T G.711 PCMU, SSRC=0x21B51504, Seq=10839, Time=1575318356 16 0.276137 192.168.21.4 16752 192.168.24.22 27176 RTP PT=ITU-T G.711 PCMU, SSRC=0x21B51504, Seq=10840, Time=1575318596 17 0.306164 192.168.21.4 16752 192.168.24.22 27176 RTP PT=ITU-T G.711 PCMU, SSRC=0x21B51504, Seq=10841, Time=1575318836 18 0.336178 192.168.21.4 16752 192.168.24.22 27176 RTP PT=ITU-T G.711 PCMU, SSRC=0x21B51504, Seq=10842, Time=1575319076 19 0.366208 192.168.21.4 16752 192.168.24.22 27176 RTP PT=ITU-T G.711 PCMU, SSRC=0x21B51504, Seq=10843, Time=1575319316 20 0.396232 192.168.21.4 16752 192.168.24.22 27176 RTP PT=ITU-T G.711 PCMU, SSRC=0x21B51504, Seq=10844, Time=1575319556 21 0.426256 192.168.21.4 16752 192.168.24.22 27176 RTP PT=ITU-T G.711 PCMU, SSRC=0x21B51504, Seq=10845, Time=1575319796 22 0.456276 192.168.21.4 16752 192.168.24.22 27176 RTP PT=ITU-T G.711 PCMU, SSRC=0x21B51504, Seq=10846, Time=1575320036 23 0.486311 192.168.21.4 16752 192.168.24.22 27176 RTP PT=ITU-T G.711 PCMU, SSRC=0x21B51504, Seq=10847, Time=1575320276 .....many more of 192.168.21.4 to 192.168.24.22 57 1.477101 192.168.24.22 16752 192.168.21.4 27176 ICMP Destination unreachable (Port unreachable) 58 1.477101 192.168.24.22 16752 192.168.21.4 27176 ICMP Destination unreachable (Port unreachable) .... RTP pattern repeats Good Call No. Time Source Src Port Destination DesT Port Protocol Info 7 3.170800 192.168.21.4 57394 192.168.24.22 onscreen SIP/SDP Request: INVITE sip:3103568788@192.168.24.22:5080, with session description 8 3.171165 192.168.24.22 onscreen 192.168.21.4 sip SIP Status: 100 Trying 9 3.171171 192.168.24.22 onscreen 192.168.21.4 sip SIP Status: 100 Trying 36 3.185636 192.168.24.22 19472 192.168.24.22 19472 UDP Source port: 19472 Destination port: 19472 40 3.194949 192.168.24.22 19472 192.168.24.22 19472 UDP Source port: 19472 Destination port: 19472 44 3.195565 192.168.24.22 onscreen 192.168.21.4 sip SIP/SDP Status: 200 OK, with session description 45 3.195568 192.168.24.22 onscreen 192.168.21.4 sip SIP/SDP Status: 200 OK, with session description 46 3.197975 192.168.21.4 57394 192.168.24.22 onscreen SIP Request: ACK sip:3103568788@192.168.24.22:5080;transport=udp 140 3.320438 192.168.24.22 19472 192.168.21.4 17148 RTP PT=ITU-T G.711 PCMU, SSRC=0x287D4E7, Seq=11773, Time=240, Mark 141 3.320441 192.168.24.22 19472 192.168.21.4 17148 RTP PT=ITU-T G.711 PCMU, SSRC=0x287D4E7, Seq=11773, Time=240, Mark 192 3.450347 192.168.21.4 17148 192.168.24.22 19472 RTP PT=ITU-T G.711 PCMU, SSRC=0x6B1504, Seq=36119, Time=96783955, Mark 199 3.480344 192.168.21.4 17148 192.168.24.22 19472 RTP PT=ITU-T G.711 PCMU, SSRC=0x6B1504, Seq=36120, Time=96784195 200 3.480393 192.168.24.22 19472 192.168.21.4 17148 RTP PT=ITU-T G.711 PCMU, SSRC=0x287D4E7, Seq=11774, Time=480 201 3.480395 192.168.24.22 19472 192.168.21.4 17148 RTP PT=ITU-T G.711 PCMU, SSRC=0x287D4E7, Seq=11774, Time=480 213 3.510382 192.168.21.4 17148 192.168.24.22 19472 RTP PT=ITU-T G.711 PCMU, SSRC=0x6B1504, Seq=36121, Time=96784435 ..... RTP pattern repeats for duration of call From: "Dan" <freeswitch-users(a)digitaldan.com> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Wednesday, July 21, 2010 9:14:01 AM Subject: Re: [Freeswitch-users] calls ending with MEDIA_TIMEOUT Gotcha, if I sort wireshark by source freeswitch sends 2 RTP packets and then a handful if ICMP errors, this pattern repeats for the entire call (2 voice packets and 5+ icmp errors). 192.168.24.22 = freeswitch 192.168.21.4 = endpoint ID Time Source Src Port Destination Dest Port 12 0.186121 192.168.24.22 27176 192.168.21.4 16752 RTP PT=ITU-T G.711 PCMU, SSRC=0x33CF6C8, Seq=5202, Time=240, Mark 13 0.186126 192.168.24.22 27176 192.168.21.4 16752 RTP PT=ITU-T G.711 PCMU, SSRC=0x33CF6C8, Seq=5202, Time=240, Mark 57 1.477101 192.168.24.22 16752 192.168.21.4 27176 ICMP Destination unreachable (Port unreachable) 92 2.467948 192.168.24.22 16752 192.168.21.4 27176 ICMP Destination unreachable (Port unreachable) 93 2.467949 192.168.24.22 16752 192.168.21.4 27176 ICMP Destination unreachable (Port unreachable) 194 5.465388 192.168.24.22 16752 192.168.21.4 27176 ICMP Destination unreachable (Port unreachable) The ICMP erros have the ports reversed, is this normal ? Thanks, Dan- From: "Brian West" <brian(a)freeswitch.org> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Wednesday, July 21, 2010 8:30:40 AM Subject: Re: [Freeswitch-users] calls ending with MEDIA_TIMEOUT NO that would mean that it was sending RTP to that IP and the firewall returned and ICMP port unreachable. /b On Jul 21, 2010, at 8:43 AM, Dan wrote: > I was able to pull the box out yesterday and try some load testing under tcpdump. It looks like under load freeswitch was trying to ping the far end RTP port which was failing: > > 57 1.477101 192.168.24.22 192.168.21.4 ICMP Destination unreachable (Port unreachable) > > I'm guessing this is part of auto nat detection? Is this a tunable option? I don't need stun/nat functionality at this point so I'm running freeswitch with the -nonat flag right now to see if its playing a part here. > > I have the tcpdump of a call if anyone is interested. > > Thanks > Dan- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Re: [Freeswitch-users] Unstable hangup hook in javascript
by Dan 23 Jul '10

23 Jul '10

So If I call session.recordFile which blocks in code until the user hangs up, and can't make periodic calls to session.ready, is the hangup hook not realiable in this scenario? From: "Anthony Minessale" <anthony.minessale(a)gmail.com> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Thursday, July 22, 2010 10:30:48 AM Subject: Re: [Freeswitch-users] Unstable hangup hook in javascript You must make periodic calls to session.ready() or some other freeswitch api functions to trigger the hangup hook. The hangup hook in scripts must be executed by your call's thread, if another thread does the hangup on your channel, it queues in the internal object that it needs to call the hanguphook method. This is triggered by doing anything new in your script. So if your channel is blocking on something while its being hungup, it will eventually pick up on it when it exits the blocking call and runs one of many methods that have a check for if it has to run the hangup hook. Manually calling hangup is a bad example, because you know you are hanging up so there is no real need to trigger the hanguphook but if you want to ensure it does you have to call session.ready() one last time. If you are writing a critical application, consider C. It's much more robust when it comes to this type of problem because it supports threading. Or consider ESL external scripts where you can run async operations and get events etc. On Thu, Jul 22, 2010 at 10:55 AM, Chris Fowler <chris(a)fowler.cc> wrote: Did this topic get resolved? We’re seeing the same occasional problem using LUA scripts where on occasion the hangup hook doesn’t get called. Our app for this is the 24x7 emergency paging system for our customers – so we’d like to resolve! This in on our production system (which is very stable) – hence the out of date build: FreeSWITCH Version 1.0.trunk (16535). Thanks, Chris. From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto: freeswitch-users-bounces(a)lists.freeswitch.org ] On Behalf Of xuyan yang Sent: Sunday, July 11, 2010 08:08 To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Unstable hangup hook in javascript Hi, I have tested the hangup hook function using the following javascript. During the speech in the call I hangup the call to see the result. Sometimes hangup function works, but sometimes it did not. The chance of fail is about 1 out of 10. I am running windows XP, FS git version cloned and compiled at end of Jun. Does anyone meet the same problem? Or do I make some mistake in the script? Thanks. ////test.js function on_hangup(session, how){ console_log("notice","hangup!!!!!!!!!!!!!\n"); } session = new Session("sofia/gateway/12voip-out/*********"); session.waitForAnswer(); session.execute("set","continue_on_fail=true"); var res=session.setHangupHook(on_hangup); console_log("notice","res:"+res+"\n"); session.execute("speak", "This is an example of flite text to speech engine. We can test the work of hanguphook."); session.execute("speak","you are calling from " + session.caller_id_num + " You are calling " + session.destination ); session.hangup(); _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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autoanswer or paging
by babak yakhchali 23 Jul '10

23 Jul '10

Hi There is a feature in Panasonic phones used with Panasonic centrals so that when calling a number, after some rings u can press * and phone is answered automatically and u can page someone. is it possible to implement this with FreeSWITCH? thanx

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Re: [Freeswitch-users] Active Calls on a gateway
by David Ponzone 23 Jul '10

23 Jul '10

no, calls-in is the total number of incoming calls from this gateway since the last reset of the counter. There is no way, AFAIK, to get the number of active calls to/from a specific gateway. With show calls, you will see the list of calls, and the gateway of the inbound leg, but the gateway of the outbound leg is not shown, so I don't think you can do something with that. David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 23/07/2010 à 14:47, Ahmed Ajmal a écrit : > Hi > > Is there any way I can find the total number of active calls on a > gateway. I have tried the following command on the CLI: > > sofia xmlstatus gateway gwname > > and it always returns 0 for Call In (assuming Call In represents the > active calls). I am on freeswitch 1.0.6. > > Thanks > Ahmed > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org

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Re: [Freeswitch-users] Active Calls on a gateway
by Evgeniy Zolotov 23 Jul '10

23 Jul '10

Please try this command : show calls Ahmed Ajmal ?????: > > Hi > > Is there any way I can find the total number of active calls on a > gateway. I have tried the following command on the CLI: > > sofia xmlstatus gateway gwname > > and it always returns 0 for Call In (assuming Call In represents the > active calls). I am on freeswitch 1.0.6. > > Thanks > > Ahmed > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

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Active Calls on a gateway
by Ahmed Ajmal 23 Jul '10

23 Jul '10

Hi Is there any way I can find the total number of active calls on a gateway. I have tried the following command on the CLI: sofia xmlstatus gateway gwname and it always returns 0 for Call In (assuming Call In represents the active calls). I am on freeswitch 1.0.6. Thanks Ahmed

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xml_cdr and contact header question
by Yuriy Ivzhenko 23 Jul '10

23 Jul '10

Hello. I develop report system on database from mod_xml_cdr I wish to select all calls to the user XX For this purpose I do select on callflow -> profile_index = "1" caller_profile -> destination_number = XX (exact select: http://pastebin.freeswitch.org/13519 ) And all is good but FS get destination_number from a field <contact> which the user defines at registration. So there is no useful. Questions. 1. As it is possible to forbid registration of the user with number in the field <contact> distinct from username? 2. Whether there are other variants to select all calls to the user XX?

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