From helmut.kuper at ewetel.de Thu Jul 1 00:22:36 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 01 Jul 2010 09:22:36 +0200 Subject: [Freeswitch-users] ESL: Question about BACKGROUND_JOB events In-Reply-To: References: <4C22301C.3080405@ewetel.de> <4274374276864666200@unknownmsgid> <4C29BCF6.90701@ewetel.de> <21D9FC45-0FA9-4454-97AD-010E150470FF@avgs.ca> <4C2B16C7.6050200@ewetel.de> Message-ID: <4C2C423C.6020307@ewetel.de> Hi Michael, it's not a big thing to parse the events (in fact I did it already. I have only one Background_job per call in certain situations), but I simply tried to avoid needless traffic and the filter function is a great thing to achieve this. Don't get me wrong, I only want to understand and not annoying you. best regards Helmut On 30.06.2010 18:48, Michael Collins wrote: > 2010/6/30 Helmut Kuper > >> Hi, >> >> well, so I have to parse all BACKGROUND-JOB events in my app to find the >> event for me? >> >> How many processes out there doing "bgapi" do you have running at once? > Like bkw said, this isn't exactly a difficult task, even if you have many > such processes. > -MC From devel at thom.fr.eu.org Thu Jul 1 00:23:32 2010 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Thu, 01 Jul 2010 09:23:32 +0200 Subject: [Freeswitch-users] create a external FXO Trunk - A200 In-Reply-To: References: Message-ID: <14574f580e92f4591940e112751bc6a5@thom.fr.eu.org> I may be wrong, but it looks like it is not detecting the dial tone, and therefore not dailing the number. You should check that connection to the line is correct, and that the tones are correctly configured for the carrier you use. Fran?ois On Thu, 1 Jul 2010 16:16:36 +1000, Deya M wrote: Hi, Managed to ge I added the following to the dialplan: No ring. I get the following: 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:403 Set codec PCMA 20ms 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:1317 Connect outbound channel OpenZAP/1:4/ 2010-07-01 11:07:04.627394 [NOTICE] switch_channel.c:776 New Channel OpenZAP/1:4/ [3ca74e5b-7e26-4a0a-b5f7-72d7a2f33fbf] 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:1331 (OpenZAP/1:4/) State Change CS_NEW -> CS_INIT 2010-07-01 11:07:04.627394 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.627394 [DEBUG] ozmod_analog.c:59 Changing state on 1:4 from DOWN to DIALING 2010-07-01 11:07:04.627394 [DEBUG] ozmod_analog.c:293 ANALOG CHANNEL thread starting. 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_INIT 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:4/) State INIT 2010-07-01 11:07:04.628392 [DEBUG] mod_openzap.c:431 (OpenZAP/1:4/) State Change CS_INIT -> CS_ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:4/) State INIT going to sleep 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_channel.c:1471 (OpenZAP/1:4/) Callstate Change DOWN -> RINGING 2010-07-01 11:07:04.627394 [DEBUG] ozmod_wanpipe.c:608 Enabled DTMF events on chan 1:4 2010-07-01 11:07:04.628392 [DEBUG] ozmod_analog.c:464 Executing state handler on 1:4 for DIALING 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:4/) State ROUTING 2010-07-01 11:07:04.628392 [DEBUG] mod_openzap.c:454 OpenZAP/1:4/ CHANNEL ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_ivr_originate.c:64 (OpenZAP/1:4/) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:4/) State ROUTING going to sleep 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:4/) State CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:4/) State CONSUME_MEDIA going to sleep 2010-07-01 11:07:20.335200 [DEBUG] ozmod_wanpipe.c:1036 1:4 wanpipe returned event 5 2010-07-01 11:07:20.335200 [DEBUG] ozmod_wanpipe.c:1061 1:4 rxhook, state 2 2010-07-01 11:07:21.150195 [DEBUG] ozmod_wanpipe.c:1036 1:4 wanpipe returned event 5 2010-07-01 11:07:21.150195 [DEBUG] ozmod_wanpipe.c:1061 1:4 rxhook, state 1 2010-07-01 11:07:21.150195 [DEBUG] ozmod_analog.c:802 EVENT [ONHOOK][1:4] STATE [DIALING] 2010-07-01 11:07:21.150195 [DEBUG] ozmod_analog.c:838 Changing state on 1:4 from DIALING to DOWN 2010-07-01 11:07:21.166199 [DEBUG] ozmod_analog.c:464 Executing state handler on 1:4 for DOWN 2010-07-01 11:07:21.166199 [DEBUG] mod_openzap.c:1556 got FXO sig 1:4 [STOP] 2010-07-01 11:07:21.166199 [DEBUG] switch_channel.c:2261 (OpenZAP/1:4/) Callstate Change RINGING -> HANGUP 2010-07-01 11:07:21.166199 [NOTICE] mod_openzap.c:1577 Hangup OpenZAP/1:4/ [CS_CONSUME_MEDIA] [NONE] 2010-07-01 11:07:21.166199 [DEBUG] switch_channel.c:2277 Send signal OpenZAP/1:4/ [KILL] 2010-07-01 11:07:21.166199 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.166199 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_HANGUP 2010-07-01 11:07:21.166199 [DEBUG] zap_io.c:1388 channel done 1:4 2010-07-01 11:07:21.166199 [DEBUG] ozmod_analog.c:778 ANALOG CHANNEL 1:4 thread ended. 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:500 (OpenZAP/1:4/) State HANGUP 2010-07-01 11:07:21.168191 [WARNING] mod_openzap.c:517 VETO Changing state on 1:4 from DOWN to HANGUP 2010-07-01 11:07:21.168191 [DEBUG] mod_openzap.c:556 OpenZAP/1:4/ CHANNEL HANGUP 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:46 OpenZAP/1:4/ Standard HANGUP, cause: NONE 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:500 (OpenZAP/1:4/) State HANGUP going to sleep 2010-07-01 11:07:21.168191 [DEBUG] switch_ivr_originate.c:3369 Originate Resulted in Error Cause: 19 [NO_ANSWER] 2010-07-01 11:07:21.168191 [INFO] mod_dptools.c:2382 Originate Failed. Cause: NO_ANSWER 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:333 (OpenZAP/1:4/) State Change CS_HANGUP -> CS_REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:591 (OpenZAP/1:4/) State REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:53 OpenZAP/1:4/ Standard REPORTING, cause: NONE 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:591 (OpenZAP/1:4/) State REPORTING going to sleep 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:327 (OpenZAP/1:4/) State Change CS_REPORTING -> CS_DESTROY 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1175 Session 71 (OpenZAP/1:4/) Locked, Waiting on external entities 2010-07-01 11:07:21.168191 [NOTICE] switch_core_session.c:1193 Session 71 (OpenZAP/1:4/) Ended 2010-07-01 11:07:21.168191 [NOTICE] switch_core_session.c:1195 Close Channel OpenZAP/1:4/ [CS_DESTROY] But no ringing is heard. thanks, -:D On Thu, Jul 1, 2010 at 2:49 PM, wrote: Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org [2] To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [3] or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org [4] You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org [5] When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: Default Dial Plan: action application bridge (Deya M) ---------- Forwarded message ---------- From: Deya M To: freeswitch-users at lists.freeswitch.org [7] Date: Thu, 1 Jul 2010 14:49:26 +1000 Subject: Re: [Freeswitch-users] Default Dial Plan: action application bridge I generated samples, and started from scratch. I also made a copy of the conf after installation, rm it, and copied the original conf before testing! -:D On Thu, Jul 1, 2010 at 1:37 PM, wrote: Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org [9] To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [10] or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org [11] You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org [12] When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: Query !! freeswitch, dingaling, jingle, skypopen (Anthony Minessale) 2. continue_on_fail and hangup_after_bridge with transfer (Vladimir Elizarov) 3. Re: SIP header on only one fork of a bridge (Tim St. Pierre) 4. Default Dial Plan: action application bridge (Deya M) 5. Re: Default Dial Plan: action application bridge (Anthony Minessale) ---------- Forwarded message ---------- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org [14] Date: Wed, 30 Jun 2010 17:40:27 -0500 Subject: Re: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, skypopen If he didn't like to joke, he'd never survive in telephony. On Wed, Jun 30, 2010 at 3:39 PM, Sameer Khan wrote: no problem. but did't expect from the creator of mod_skypopen, the wonder i would say On Wed, Jun 30, 2010 at 9:19 PM, Giovanni Maruzzelli wrote: On Wed, Jun 30, 2010 at 6:02 PM, Sameer Khan wrote: > Mr Giovani what was that ? just jokin, nevermind -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [18] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [19] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [20] http://www.freeswitch.org [21] _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [22] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [23] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [24] http://www.freeswitch.org [25] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [26] ClueCon http://www.cluecon.com/ [27] Twitter: http://twitter.com/FreeSWITCH_wire [28] AIM: anthm MSN:anthony_minessale at hotmail.com [29] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [30] IRC: irc.freenode.net [31] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [32] googletalk:conf+888 at conference.freeswitch.org [33] pstn:+19193869900 ---------- Forwarded message ---------- From: Vladimir Elizarov To: freeswitch-users at lists.freeswitch.org [35] Date: Thu, 1 Jul 2010 03:29:09 +0400 Subject: [Freeswitch-users] continue_on_fail and hangup_after_bridge with transfer I'm trying to make a dialplan: expression="^(^([0-9]{10})$|^([0-9]{11})$|^([0-9]{12})$|^+([0-9]{11})$)$"> data="hangup_after_bridge=true"/> application="set"data="continue_on_fail=NO_ROUTE_DESTINATION"/> break="on-true"> data="dtmf=WWWWWWWWWWWWWWWWWWWWW111222#WWWWWW89$2#@100"/> data="sofia/gateway/gw1/89$2|sofia/gateway/gw2/89$2|sofia/gateway/gw3/89$2"/> data="{execute_on_answer=send_dtmfs${dtmf}}sofia/internal/${real_dialed_number}@192.168.50.53:5061 [36]"/> The logic of his work: if unavailable gateway to the next. if not available all the gateway to go to the extension transfer. A problem in If we get a code busy here, it is not satisfied hangup_after_bridge. The call goes to the transfer. Why is this so? trace: http://pastebin.freeswitch.org/13244 [37] -- Best regards, Vladimir Elizarov ---------- Forwarded message ---------- From: "Tim St. Pierre" To: freeswitch-users at lists.freeswitch.org [39] Date: Wed, 30 Jun 2010 22:45:19 -0400 Subject: Re: [Freeswitch-users] SIP header on only one fork of a bridge Thanks all, I read that bit, but didn't fully understand it's implications. I'll give that a go. -Tim Steven Ayre wrote: > > data="{this_is_global=true}[this_is_gw1_only=true]sofia/gateway/gw1/$1|[this_is_gw2_only=true]sofia/gateway/gw2/$1"/> > > > > On 28 June 2010 19:45, Tim St. Pierre fs-list at communicatefreely.net [41]>> wrote: > > Hello list, > > I would like to bridge a call to multiple SIP endpoints, but add > different headers to each. > > I'm not entirely sure how to do this. I have no problem exporting a > SIP header that does what I > want for one destination, but I'm not sure how to set it for two. > > My application is that I want two IP phones to ring - one with the > internal ring-ring splash, the > others with a group-answer (single ring, then lamp flash only), for > administrative assistants, etc. > > How do I export different variables to each branch? > > Thanks! > > -Tim > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org [42] > FreeSWITCH-users at lists.freeswitch.org [43]> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [44] > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [45] > http://www.freeswitch.org [46] > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org [47] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [48] > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [49] > http://www.freeswitch.org [50] ---------- Forwarded message ---------- From: Deya M To: freeswitch-users at lists.freeswitch.org [52] Date: Thu, 1 Jul 2010 06:19:51 +0300 Subject: [Freeswitch-users] Default Dial Plan: action application bridge Hi, In the default dial plan, with two extensions defined, 1000 and 1001, when I call from 1000 to 1001, I always get Voicemail, using the default config files: conf/dialplan/default.xml I changed the following from : TO and it did work! Not sure if the first / default one, should work, but something is missing ? New: Old: :-D ---------- Forwarded message ---------- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org [54] Date: Wed, 30 Jun 2010 22:37:15 -0500 Subject: Re: [Freeswitch-users] Default Dial Plan: action application bridge you must have changed more than you think. you might want to diff you configs against the in-tree ones. On Wed, Jun 30, 2010 at 10:19 PM, Deya M wrote: Hi, In the default dial plan, with two extensions defined, 1000 and 1001, when I call from 1000 to 1001, I always get Voicemail, using the default config files: conf/dialplan/default.xml I changed the following from : TO and it did work! Not sure if the first / default one, should work, but something is missing ? New: Old: :-D _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [56] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [57] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [58] http://www.freeswitch.org [59] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [60] ClueCon http://www.cluecon.com/ [61] Twitter: http://twitter.com/FreeSWITCH_wire [62] AIM: anthm MSN:anthony_minessale at hotmail.com [63] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [64] IRC: irc.freenode.net [65] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [66] googletalk:conf+888 at conference.freeswitch.org [67] pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [68] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [69] 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/bb90441f/attachment-0001.html From lakindia89 at gmail.com Thu Jul 1 02:10:13 2010 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Thu, 1 Jul 2010 14:40:13 +0530 Subject: [Freeswitch-users] mod_freetdm is not loading - Saying error Message-ID: Hi all, I've downloaded and compiled wanpipe and freeswitch as it was given in http://wiki.sangoma.com/wanpipe-api-freetdm-linux Here is my freetdm.conf: [span wanpipe wp1] number =>1 trunk_type =>e1 group=>grp1 b-channel => 1:1-15 b-channel => 1:17-31 I started freeswitch In the CLI, I gave "load mod_freetdm" and it gives me the following error. http://pastebin.freeswitch.org/13327 It says "Opening freetdm.conf failed". Can someone tell me how to solve this one. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/b3fcd891/attachment.html From irmatov at gmail.com Thu Jul 1 02:34:24 2010 From: irmatov at gmail.com (Timur Irmatov) Date: Thu, 1 Jul 2010 14:34:24 +0500 Subject: [Freeswitch-users] mod_erlang_event problem: REPRODUCIBLE, fix needed Message-ID: Hi, All and Andrew especially! ;-) I have managed to found a reason of my erlang and freeswitch mis-communication. I am receiving a call through 'external' sofia profile. Then I am sending it to my erlang application, *without* answering. The intent is for erlang application to test 'validity' of the call and then bridge it to external destination. Voice path needs to go directly between caller and callee, so I set inbound-bypass-media to true in external profile. When inbound-bypass-media is set to false, I am successfully managing a call through erlang application. When it is set to false, I am seeing 'got pid' and 'exit erlang_outbound_function' in log files, and I cannot manage calls in erlang. So, I guess there is a bug somewhere in erlang_event module or freeswitch itself. What should I do? Open a bug report in jira? -- Timur Irmatov, xmpp:irmatov at jabber.ru From irmatov at gmail.com Thu Jul 1 03:16:05 2010 From: irmatov at gmail.com (Timur Irmatov) Date: Thu, 1 Jul 2010 15:16:05 +0500 Subject: [Freeswitch-users] mod_erlang_event problem: REPRODUCIBLE, fix needed In-Reply-To: References: Message-ID: On Thu, Jul 1, 2010 at 2:34 PM, Timur Irmatov wrote: > When inbound-bypass-media is set to false, I am > successfully managing a call through erlang application. When it is > set to false, I am seeing 'got pid' and 'exit > erlang_outbound_function' in log files, and I cannot manage calls in > erlang. When it is set to *true* I am having a problem. -- Timur Irmatov, xmpp:irmatov at jabber.ru From mcampbellsmith at gmail.com Thu Jul 1 03:40:27 2010 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 1 Jul 2010 20:40:27 +1000 Subject: [Freeswitch-users] groups and g729 codec Message-ID: Hi! I have a group defined to call three extensions. The inbound route uses G729 and I only have one license. When a call comes into FS with a g729 codec chosen, the group call fails with encoding errors. Only one extension in the group supports the g729 codec. 2010-07-01 20:32:17.685087 [ERR] mod_com_g729.c:142 DECODER CREATE FAILED - 0xb6738248 (nil) 2010-07-01 20:32:17.686085 [ERR] switch_core_io.c:352 Codec G.729 decoder error! Is this because FS is trying to set up 3 channels on the b-leg with g729? I thought by activating late-negotiation this might work. No luck. I thought by creating a specific dialplan in the users setting it might work. I couldn't find the correct format. Would this work? Is there a way I can get this to work, or do I need to force the inbound call to use another codec? Thanks for your help From brian at freeswitch.org Thu Jul 1 06:08:05 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 1 Jul 2010 08:08:05 -0500 Subject: [Freeswitch-users] groups and g729 codec In-Reply-To: References: Message-ID: <3B251250-5AB7-4D3D-843D-018D7117C93D@freeswitch.org> Looks like your license server isn't running. email me the output of ifconfig -a and the contents of your license files off list please. /b On Jul 1, 2010, at 5:40 AM, Mark Campbell-Smith wrote: > Hi! > > I have a group defined to call three extensions. > > The inbound route uses G729 and I only have one license. When a call > comes into FS with a g729 codec chosen, the group call fails with > encoding errors. Only one extension in the group supports the g729 > codec. > > 2010-07-01 20:32:17.685087 [ERR] mod_com_g729.c:142 DECODER CREATE > FAILED - 0xb6738248 (nil) > 2010-07-01 20:32:17.686085 [ERR] switch_core_io.c:352 Codec G.729 decoder error! > > Is this because FS is trying to set up 3 channels on the b-leg with g729? > > I thought by activating late-negotiation this might work. No luck. > I thought by creating a specific dialplan in the users setting it > might work. I couldn't find the correct format. Would this work? > value="{absolute_codec_string=GSM}{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> > > Is there a way I can get this to work, or do I need to force the > inbound call to use another codec? > > Thanks for your help From jeff at jefflenk.com Thu Jul 1 06:34:55 2010 From: jeff at jefflenk.com (Jeff Lenk) Date: Thu, 1 Jul 2010 06:34:55 -0700 (PDT) Subject: [Freeswitch-users] mod_freetdm is not loading - Saying error In-Reply-To: References: Message-ID: <1277991295341-5243408.post@n2.nabble.com> Place the freetdm.conf.xml from /src/libs/freetdm/conf into your installed fs location conf/autoload_configs - where src is where you have FreeSWITCH source located you must configure the file as well! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-freetdm-is-not-loading-Saying-error-tp5242655p5243408.html Sent from the freeswitch-users mailing list archive at Nabble.com. From sanms.zhang at gmail.com Thu Jul 1 01:30:57 2010 From: sanms.zhang at gmail.com (chi zhang) Date: Thu, 1 Jul 2010 16:30:57 +0800 Subject: [Freeswitch-users] Problem of FreeSwitch and T.38 Test Message-ID: Hi,guys. This is sammy. Here i met some problem in T.38 fax test on new module: mod_spandsp(based on the latest src) When simulate one T.38 fax receive process with sipP as below: 1) 1000 calls to 666666, and call is established. 666666 is one extension in public.xml 2) 666666 send re-INVITE to 1000 with T.38 SDP. Here re-INVITE retrys 5 times, and FS return " Fax processing not successful - result (48) Disconnected after permitted retries. " i print some logs for libs/spandsp T.30 handle, found in function t30::repeat_last_command(), t30_state_t->state is T30_STATE_R(17) , so FS return " Disconnected after permitted retries ". Does anyone know the reason? Where is the span_log of spandsp lib ? i cannt find it... regards sammy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/0cb60d53/attachment.html From brian at freeswitch.org Thu Jul 1 06:44:09 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 1 Jul 2010 08:44:09 -0500 Subject: [Freeswitch-users] Problem of FreeSwitch and T.38 Test In-Reply-To: References: Message-ID: <5ECC3A8A-CDF9-4EF4-AD87-A3FAEDFF26AB@freeswitch.org> You'll need to provide the sip trace and full debug logs. /b On Jul 1, 2010, at 3:30 AM, chi zhang wrote: > Does anyone know the reason? > Where is the span_log of spandsp lib ? i cannt find it... > From dome at tel.co.th Thu Jul 1 10:03:36 2010 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 2 Jul 2010 00:03:36 +0700 Subject: [Freeswitch-users] How to config acl from web GUI ? Message-ID: I want to make wholesale service. so now i use acl to allow ip. so i want to use web gui. what's best way to do that ? 1. use xml_curl binding configuration to web application. i'm worry about peformance because FS request to web apps every call setup. 2. user web apps direct manage acl.conf.xml and use webapi reload acl Best Reguards. Dome C. From brian at freeswitch.org Thu Jul 1 10:14:44 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 1 Jul 2010 12:14:44 -0500 Subject: [Freeswitch-users] How to config acl from web GUI ? In-Reply-To: References: Message-ID: <9C90F33B-F419-4D01-A13D-CB1E8AA567F8@freeswitch.org> If you're using XML_CURL then you don't. You'll use your CGI thats responding to decide what to respond with. /b On Jul 1, 2010, at 12:03 PM, Dome Charoenyost wrote: > I want to make wholesale service. so now i use acl to allow ip. so i > want to use web gui. what's best way to do that ? > 1. use xml_curl binding configuration to web application. i'm worry > about peformance because FS request to web apps every call setup. > 2. user web apps direct manage acl.conf.xml and use webapi reload acl > > Best Reguards. > > Dome C. From dome at tel.co.th Thu Jul 1 10:43:44 2010 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 2 Jul 2010 00:43:44 +0700 Subject: [Freeswitch-users] How to config acl from web GUI ? In-Reply-To: <9C90F33B-F419-4D01-A13D-CB1E8AA567F8@freeswitch.org> References: <9C90F33B-F419-4D01-A13D-CB1E8AA567F8@freeswitch.org> Message-ID: 2010/7/2 Brian West : > If you're using XML_CURL then you don't. ?You'll use your CGI thats responding to decide what to respond with. XML_CURL good for dynamic like user. but acl update may be monthly or when got new customer :) so i'm thinking about http caching like a varnish also. Dome C. > > /b > > On Jul 1, 2010, at 12:03 PM, Dome Charoenyost wrote: > >> I want to make wholesale service. so now i use acl to allow ip. so i >> want to use web gui. what's best way to do that ? >> 1. use xml_curl binding configuration ?to web application. ?i'm worry >> about peformance because FS request to web apps every call setup. >> 2. user web apps direct ?manage acl.conf.xml and use webapi reload acl >> >> Best Reguards. >> >> Dome C. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Thu Jul 1 11:01:40 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 1 Jul 2010 11:01:40 -0700 Subject: [Freeswitch-users] How to config acl from web GUI ? In-Reply-To: References: <9C90F33B-F419-4D01-A13D-CB1E8AA567F8@freeswitch.org> Message-ID: On Thu, Jul 1, 2010 at 10:43 AM, Dome Charoenyost wrote: > 2010/7/2 Brian West : > > If you're using XML_CURL then you don't. You'll use your CGI thats > responding to decide what to respond with. > XML_CURL good for dynamic like user. but acl update may be monthly or > when got new customer :) > so i'm thinking about http caching like a varnish also. > > Dome C. > > If you have your ACL in acl.conf.xml you could have a GUI app that just re-writes that file. Then it could send "reloadacl reloadxml" whenever changes are made. Doing something like that once a month is probably not such a big deal... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/728571b4/attachment.html From infos at madovsky.org Thu Jul 1 11:07:00 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 1 Jul 2010 14:07:00 -0400 Subject: [Freeswitch-users] opensips + freeswitch Message-ID: <26A99B279622482ABEDB540C02BAE01D@MOBILEE1705> Is there any wiki or link to use Opensips as proxy with FS cluster ? Thanks F -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/f3d15bdc/attachment.html From msc at freeswitch.org Thu Jul 1 11:32:28 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 1 Jul 2010 11:32:28 -0700 Subject: [Freeswitch-users] opensips + freeswitch In-Reply-To: <26A99B279622482ABEDB540C02BAE01D@MOBILEE1705> References: <26A99B279622482ABEDB540C02BAE01D@MOBILEE1705> Message-ID: On Thu, Jul 1, 2010 at 11:07 AM, Madovsky wrote: > Is there any wiki or link to use > Opensips as proxy with FS cluster ? > > I can do better than that: dial 1-866-742-CLUE and register for ClueCon. Then sit in for these talks: Tuesday: 4:30 PM Daniel-Constantin Mierla Blending FreeSWITCH and Kamailio to Build Large Unified Communication Platforms Wednesday: 10:30 AM Bogdan-Andrei Iancu OpenSIPS Load-Balancing For A FreeSWITCH Server Cluster If you come to ClueCon you can actually *speak in person to the guy who created OpenSIPS*. That's just a smidge better than a wiki, no? ;) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/8f2356a5/attachment.html From infos at madovsky.org Thu Jul 1 11:48:53 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 1 Jul 2010 14:48:53 -0400 Subject: [Freeswitch-users] opensips + freeswitch References: <26A99B279622482ABEDB540C02BAE01D@MOBILEE1705> Message-ID: <448BCDB6C42A4951816FDE823EEB1900@MOBILEE1705> I have already talked to Bogdan months ago.. Just to say to you this month I even don't know how to pay my rent... so Cluecon is like dream for me today.... ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Thursday, July 01, 2010 2:32 PM Subject: Re: [Freeswitch-users] opensips + freeswitch On Thu, Jul 1, 2010 at 11:07 AM, Madovsky wrote: Is there any wiki or link to use Opensips as proxy with FS cluster ? I can do better than that: dial 1-866-742-CLUE and register for ClueCon. Then sit in for these talks: Tuesday: 4:30 PM Daniel-Constantin Mierla Blending FreeSWITCH and Kamailio to Build Large Unified Communication Platforms Wednesday: 10:30 AM Bogdan-Andrei Iancu OpenSIPS Load-Balancing For A FreeSWITCH Server Cluster If you come to ClueCon you can actually speak in person to the guy who created OpenSIPS. That's just a smidge better than a wiki, no? ;) -MC ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/f0a72e28/attachment.html From msc at freeswitch.org Thu Jul 1 11:58:26 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 1 Jul 2010 11:58:26 -0700 Subject: [Freeswitch-users] opensips + freeswitch In-Reply-To: <448BCDB6C42A4951816FDE823EEB1900@MOBILEE1705> References: <26A99B279622482ABEDB540C02BAE01D@MOBILEE1705> <448BCDB6C42A4951816FDE823EEB1900@MOBILEE1705> Message-ID: On Thu, Jul 1, 2010 at 11:48 AM, Madovsky wrote: > I have already talked to Bogdan months ago.. > Just to say to you this month I even don't know how to pay my rent... > so Cluecon is like dream for me today.... > Bummer. Well, I'll post a link to the viddler video when it's all done next month. :P -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/a6d9107c/attachment.html From andrew at hijacked.us Thu Jul 1 12:01:53 2010 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 1 Jul 2010 15:01:53 -0400 Subject: [Freeswitch-users] mod_erlang_event problem: REPRODUCIBLE, fix needed In-Reply-To: References: Message-ID: <20100701190152.GG17555@hijacked.us> On Thu, Jul 01, 2010 at 03:16:05PM +0500, Timur Irmatov wrote: > On Thu, Jul 1, 2010 at 2:34 PM, Timur Irmatov wrote: > > When inbound-bypass-media is set to false, I am > > successfully managing a call through erlang application. When it is > > set to false, I am seeing 'got pid' and 'exit > > erlang_outbound_function' in log files, and I cannot manage calls in > > erlang. > > When it is set to *true* I am having a problem. > So in the other thread you said it was working briefly until you restarted FreeSWITCH, is this the root cause of that problem, or is it another issue? I'm baffled as to why bypass media is having any effect here. Andrew From infos at madovsky.org Thu Jul 1 12:04:02 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 1 Jul 2010 15:04:02 -0400 Subject: [Freeswitch-users] opensips + freeswitch References: <26A99B279622482ABEDB540C02BAE01D@MOBILEE1705><448BCDB6C42A4951816FDE823EEB1900@MOBILEE1705> Message-ID: <07F18C441A4A4EA9B3253B83D8131851@MOBILEE1705> Thanks Michael, Hope to be more rich soon also... ;) ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Thursday, July 01, 2010 2:58 PM Subject: Re: [Freeswitch-users] opensips + freeswitch On Thu, Jul 1, 2010 at 11:48 AM, Madovsky wrote: I have already talked to Bogdan months ago.. Just to say to you this month I even don't know how to pay my rent... so Cluecon is like dream for me today.... Bummer. Well, I'll post a link to the viddler video when it's all done next month. :P -MC ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/11d27313/attachment.html From anthony.minessale at gmail.com Thu Jul 1 13:33:12 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 1 Jul 2010 15:33:12 -0500 Subject: [Freeswitch-users] mod_erlang_event problem: REPRODUCIBLE, fix needed In-Reply-To: <20100701190152.GG17555@hijacked.us> References: <20100701190152.GG17555@hijacked.us> Message-ID: pastebin a console log with debug and sip trace enabled http://pastebin.freeswitch.org sofia profile internal siptrace on console loglevel debug *if you use some other profile besides internal enable siptrace for that instead. do it for any profiles involved. On Thu, Jul 1, 2010 at 2:01 PM, Andrew Thompson wrote: > On Thu, Jul 01, 2010 at 03:16:05PM +0500, Timur Irmatov wrote: > > On Thu, Jul 1, 2010 at 2:34 PM, Timur Irmatov wrote: > > > When inbound-bypass-media is set to false, I am > > > successfully managing a call through erlang application. When it is > > > set to false, I am seeing 'got pid' and 'exit > > > erlang_outbound_function' in log files, and I cannot manage calls in > > > erlang. > > > > When it is set to *true* I am having a problem. > > > > So in the other thread you said it was working briefly until you > restarted FreeSWITCH, is this the root cause of that problem, or is it > another issue? I'm baffled as to why bypass media is having any effect > here. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/3e71fcf7/attachment-0001.html From msc at freeswitch.org Thu Jul 1 15:14:28 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 1 Jul 2010 15:14:28 -0700 Subject: [Freeswitch-users] Call For Assistance: Wiki updates for default dialplan changes Message-ID: Hey all, We have a need for assistance with updating the wiki. We recently changed some default dialplan extension numbers: 9878 through 9998 have been changed to 9178 to 9198, respectively 9999 has been changed to 9664 Any time you see a wiki page that references 99xx please edit it and put the new extension number in. I've already updated this page: http://wiki.freeswitch.org/wiki/Default_Dialplan_QRF ... so use it as a reference if you're not sure what the new extension number is. I've also done a search on "9999" and updated a handful of pages. Let's crowdsource this and bang it out real quick. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100701/a0bbfd5c/attachment.html From deya787 at gmail.com Thu Jul 1 15:53:27 2010 From: deya787 at gmail.com (Deya M) Date: Fri, 2 Jul 2010 08:53:27 +1000 Subject: [Freeswitch-users] Sangoma A200 FXO Problem Outgoing Message-ID: Hi, I posted this problem early today, but haven't got any reply. Problem is I cannot dial outside, I get no ring, and in the logs it ends up with a No Answer. I tried to dial, and configured an extension, and I was able to receive the call. Conf: openzap.conf: [span wanpipe FXO] name => OpenZAP fxo-channel => 1:1 fxo-channel => 1:2 fxo-channel => 1:3 fxo-channel => 1:4 openzap.conf.xml: #================================================ # WANPIPE1 Configuration File #================================================ # # Date: Mon Jul 31 17:10:23 EDT 2006 # # Note: This file was generated automatically # by /usr/local/sbin/setup-sangoma program. # # If you want to edit this file, it is # recommended that you use wancfg program # to do so. #================================================ # Sangoma Technologies Inc. #================================================ [devices] wanpipe1 = WAN_AFT_ANALOG, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 4 FE_MEDIA = FXO/FXS TDMV_LAW = ALAW TDMV_OPERMODE = FCC RM_BATTTHRESH = 3 RM_BATTDEBOUNCE = 16 FE_NETWORK_SYNC = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_HW_DTMF = YES TDMV_HW_FAX_DETECT = YES [w1g1] ACTIVE_CH = ALL MTU = 8 TDMV_HWEC = YES Logs: Dialplan: sofia/internal/1001 at 196.202.51.44 parsing [default->outgoing-sangoma] continue=false Dialplan: sofia/internal/1001 at 196.202.51.44 Regex (PASS) [outgoing-sangoma] destination_number(1234567890) =~ /^([0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9])$/ b reak=on-false Dialplan: sofia/internal/1001 at 196.202.51.44 Action set(dialed_ext=1234567890) Dialplan: sofia/internal/1001 at 196.202.51.44 Action bridge(openzap/1/1/${dialed_ext}) 2010-07-02 03:46:45.515582 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1001 at 196.202.51.44) State Change CS_ROUTING -> CS_EXECUTE 2010-07-02 03:46:45.515582 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1001 at 196.202.51.44 [BREAK] 2010-07-02 03:46:45.515582 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1001 at 196.202.51.44) State ROUTING going to sleep 2010-07-02 03:46:45.515582 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1001 at 196.202.51.44) Running State Change CS_EXECUTE 2010-07-02 03:46:45.515582 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1001 at 196.202.51.44) State EXECUTE 2010-07-02 03:46:45.515582 [DEBUG] mod_sofia.c:235 sofia/internal/ 1001 at 196.202.51.44 SOFIA EXECUTE 2010-07-02 03:46:45.515582 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1001 at 196.202.51.44 Standard EXECUTE EXECUTE sofia/internal/1001 at 196.202.51.44hash(insert/196.202.51.44-spymap/1001/4da2a91b-6c21-43e2-9591-ebba049e3488) EXECUTE sofia/internal/1001 at 196.202.51.44hash(insert/196.202.51.44-last_dial/1001/1234567890) EXECUTE sofia/internal/1001 at 196.202.51.44hash(insert/196.202.51.44-last_dial/global/4da2a91b-6c21-43e2-9591-ebba049e3488) EXECUTE sofia/internal/1001 at 196.202.51.44 set(dialed_ext=1234567890) 2010-07-02 03:46:45.516581 [DEBUG] mod_dptools.c:843 sofia/internal/ 1001 at 196.202.51.44 SET [dialed_ext]=[1234567890] EXECUTE sofia/internal/1001 at 196.202.51.44 bridge(openzap/1/1/1234567890) 2010-07-02 03:46:45.517583 [DEBUG] mod_openzap.c:403 Set codec PCMA 20ms 2010-07-02 03:46:45.517583 [DEBUG] mod_openzap.c:1317 Connect outbound channel OpenZAP/1:1/1234567890 2010-07-02 03:46:45.517583 [NOTICE] switch_channel.c:776 New Channel OpenZAP/1:1/1234567890 [6d5e43c8-39b9-45d4-9a52-4e17d4aa90d1] 2010-07-02 03:46:45.517583 [DEBUG] mod_openzap.c:1331 (OpenZAP/1:1/1234567890) State Change CS_NEW -> CS_INIT 2010-07-02 03:46:45.517583 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:1/1234567890 [BREAK] 2010-07-02 03:46:45.517583 [DEBUG] ozmod_analog.c:59 Changing state on 1:1 from DOWN to DIALING 2010-07-02 03:46:45.517583 [DEBUG] ozmod_analog.c:293 ANALOG CHANNEL thread starting. 2010-07-02 03:46:45.517583 [DEBUG] ozmod_wanpipe.c:608 Enabled DTMF events on chan 1:1 2010-07-02 03:46:45.517583 [DEBUG] ozmod_analog.c:464 Executing state handler on 1:1 for DIALING 2010-07-02 03:46:45.517583 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/1234567890) Running State Change CS_INIT 2010-07-02 03:46:45.517583 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:1/1234567890) State INIT 2010-07-02 03:46:45.517583 [DEBUG] mod_openzap.c:431 (OpenZAP/1:1/1234567890) State Change CS_INIT -> CS_ROUTING 2010-07-02 03:46:45.517583 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:1/1234567890 [BREAK] 2010-07-02 03:46:45.517583 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:1/1234567890) State INIT going to sleep 2010-07-02 03:46:45.517583 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/1234567890) Running State Change CS_ROUTING 2010-07-02 03:46:45.517583 [DEBUG] switch_channel.c:1471 (OpenZAP/1:1/1234567890) Callstate Change DOWN -> RINGING 2010-07-02 03:46:45.518582 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:1/1234567890) State ROUTING 2010-07-02 03:46:45.518582 [DEBUG] mod_openzap.c:454 OpenZAP/1:1/1234567890 CHANNEL ROUTING 2010-07-02 03:46:45.518582 [DEBUG] switch_ivr_originate.c:64 (OpenZAP/1:1/1234567890) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-07-02 03:46:45.518582 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:1/1234567890 [BREAK] 2010-07-02 03:46:45.518582 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:1/1234567890) State ROUTING going to sleep 2010-07-02 03:46:45.518582 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/1234567890) Running State Change CS_CONSUME_MEDIA 2010-07-02 03:46:45.518582 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:1/1234567890) State CONSUME_MEDIA 2010-07-02 03:46:45.518582 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:1/1234567890) State CONSUME_MEDIA going to sleep 2010-07-02 03:47:01.278384 [DEBUG] ozmod_wanpipe.c:1036 1:1 wanpipe returned event 5 2010-07-02 03:47:01.278384 [DEBUG] ozmod_wanpipe.c:1061 1:1 rxhook, state 2 2010-07-02 03:47:02.094372 [DEBUG] ozmod_wanpipe.c:1036 1:1 wanpipe returned event 5 2010-07-02 03:47:02.094372 [DEBUG] ozmod_wanpipe.c:1061 1:1 rxhook, state 1 2010-07-02 03:47:02.094372 [DEBUG] ozmod_analog.c:802 EVENT [ONHOOK][1:1] STATE [DIALING] 2010-07-02 03:47:02.094372 [DEBUG] ozmod_analog.c:838 Changing state on 1:1 from DIALING to DOWN 2010-07-02 03:47:02.096373 [DEBUG] ozmod_analog.c:464 Executing state handler on 1:1 for DOWN 2010-07-02 03:47:02.096373 [DEBUG] mod_openzap.c:1556 got FXO sig 1:1 [STOP] 2010-07-02 03:47:02.096373 [DEBUG] switch_channel.c:2261 (OpenZAP/1:1/1234567890) Callstate Change RINGING -> HANGUP 2010-07-02 03:47:02.096373 [NOTICE] mod_openzap.c:1577 Hangup OpenZAP/1:1/1234567890 [CS_CONSUME_MEDIA] [NONE] 2010-07-02 03:47:02.096373 [DEBUG] switch_channel.c:2277 Send signal OpenZAP/1:1/1234567890 [KILL] 2010-07-02 03:47:02.096373 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:1/1234567890 [BREAK] 2010-07-02 03:47:02.096373 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/1234567890) Running State Change CS_HANGUP 2010-07-02 03:47:02.096373 [DEBUG] zap_io.c:1388 channel done 1:1 2010-07-02 03:47:02.096373 [DEBUG] ozmod_analog.c:778 ANALOG CHANNEL 1:1 thread ended. 2010-07-02 03:47:02.096373 [DEBUG] switch_core_state_machine.c:500 (OpenZAP/1:1/1234567890) State HANGUP 2010-07-02 03:47:02.096373 [WARNING] mod_openzap.c:517 VETO Changing state on 1:1 from DOWN to HANGUP 2010-07-02 03:47:02.096373 [DEBUG] mod_openzap.c:556 OpenZAP/1:1/1234567890 CHANNEL HANGUP 2010-07-02 03:47:02.096373 [DEBUG] switch_core_state_machine.c:46 OpenZAP/1:1/1234567890 Standard HANGUP, cause: NONE 2010-07-02 03:47:02.096373 [DEBUG] switch_core_state_machine.c:500 (OpenZAP/1:1/1234567890) State HANGUP going to sleep 2010-07-02 03:47:02.098372 [DEBUG] switch_core_state_machine.c:333 (OpenZAP/1:1/1234567890) State Change CS_HANGUP -> CS_REPORTING 2010-07-02 03:47:02.098372 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:1/1234567890 [BREAK] 2010-07-02 03:47:02.098372 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/1234567890) Running State Change CS_REPORTING 2010-07-02 03:47:02.098372 [DEBUG] switch_core_state_machine.c:591 (OpenZAP/1:1/1234567890) State REPORTING 2010-07-02 03:47:02.098372 [DEBUG] switch_core_state_machine.c:53 OpenZAP/1:1/1234567890 Standard REPORTING, cause: NONE 2010-07-02 03:47:02.098372 [DEBUG] switch_core_state_machine.c:591 (OpenZAP/1:1/1234567890) State REPORTING going to sleep 2010-07-02 03:47:02.098372 [DEBUG] switch_core_state_machine.c:327 (OpenZAP/1:1/1234567890) State Change CS_REPORTING -> CS_DESTROY 2010-07-02 03:47:02.098372 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:1/1234567890 [BREAK] 2010-07-02 03:47:02.098372 [DEBUG] switch_core_session.c:1175 Session 26 (OpenZAP/1:1/1234567890) Locked, Waiting on external entities 2010-07-02 03:47:02.098372 [DEBUG] switch_ivr_originate.c:3369 Originate Resulted in Error Cause: 19 [NO_ANSWER] 2010-07-02 03:47:02.098372 [INFO] mod_dptools.c:2382 Originate Failed. Cause: NO_ANSWER 2010-07-02 03:47:02.098372 [NOTICE] switch_core_session.c:1193 Session 26 (OpenZAP/1:1/1234567890) Ended 2010-07-02 03:47:02.098372 [NOTICE] switch_core_session.c:1195 Close Channel OpenZAP/1:1/1234567890 [CS_DESTROY] 2010-07-02 03:47:02.098372 [DEBUG] switch_core_state_machine.c:427 (OpenZAP/1:1/1234567890) Callstate Change HANGUP -> DOWN 2010-07-02 03:47:02.098372 [DEBUG] switch_core_state_machine.c:430 (OpenZAP/1:1/1234567890) Running State Change CS_DESTROY 2010-07-02 03:47:02.098372 [DEBUG] switch_core_state_machine.c:440 (OpenZAP/1:1/1234567890) State DESTROY 2010-07-02 03:47:02.098372 [DEBUG] switch_core_state_machine.c:60 OpenZAP/1:1/1234567890 Standard DESTROY 2010-07-02 03:47:02.098372 [DEBUG] switch_core_state_machine.c:440 (OpenZAP/1:1/1234567890) State DESTROY going to sleep 2010-07-02 03:47:02.098372 [NOTICE] switch_core_state_machine.c:185 sofia/internal/1001 at 196.202.51.44 has executed the last dialplan instruction, hanging up. 2010-07-02 03:47:02.098372 [DEBUG] switch_channel.c:2261 (sofia/internal/ 1001 at 196.202.51.44) Callstate Change RINGING -> HANGUP 2010-07-02 03:47:02.098372 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/1001 at 196.202.51.44 [CS_EXECUTE] [NORMAL_CLEARING] 2010-07-02 03:47:02.098372 [DEBUG] switch_channel.c:2277 Send signal sofia/internal/1001 at 196.202.51.44 [KILL] 2010-07-02 03:47:02.098372 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1001 at 196.202.51.44 [BREAK] 2010-07-02 03:47:02.098372 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1001 at 196.202.51.44) State EXECUTE going to sleep 2010-07-02 03:47:02.098372 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1001 at 196.202.51.44) Running State Change CS_HANGUP 2010-07-02 03:47:02.098372 [DEBUG] switch_core_state_machine.c:500 (sofia/internal/1001 at 196.202.51.44) State HANGUP 2010-07-02 03:47:02.098372 [DEBUG] mod_sofia.c:453 Channel sofia/internal/ 1001 at 196.202.51.44 hanging up, cause: NORMAL_CLEARING 2010-07-02 03:47:02.104376 [DEBUG] mod_sofia.c:515 Responding to INVITE with: 480 2010-07-02 03:47:02.104376 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1001 at 196.202.51.44 Standard HANGUP, cause: NORMAL_CLEARING 2010-07-02 03:47:02.104376 [DEBUG] switch_core_state_machine.c:500 (sofia/internal/1001 at 196.202.51.44) State HANGUP going to sleep 2010-07-02 03:47:02.106398 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1001 at 196.202.51.44) State Change CS_HANGUP -> CS_REPORTING 2010-07-02 03:47:02.106398 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1001 at 196.202.51.44 [BREAK] 2010-07-02 03:47:02.106398 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1001 at 196.202.51.44) Running State Change CS_REPORTING 2010-07-02 03:47:02.106398 [DEBUG] switch_core_state_machine.c:591 (sofia/internal/1001 at 196.202.51.44) State REPORTING 2010-07-02 03:47:02.106398 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1001 at 196.202.51.44 Standard REPORTING, cause: NORMAL_CLEARING 2010-07-02 03:47:02.106398 [DEBUG] switch_core_state_machine.c:591 (sofia/internal/1001 at 196.202.51.44) State REPORTING going to sleep 2010-07-02 03:47:02.106398 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1001 at 196.202.51.44) State Change CS_REPORTING -> CS_DESTROY 2010-07-02 03:47:02.106398 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1001 at 196.202.51.44 [BREAK] 2010-07-02 03:47:02.106398 [DEBUG] switch_core_session.c:1175 Session 25 (sofia/internal/1001 at 196.202.51.44) Locked, Waiting on external entities 2010-07-02 03:47:02.106398 [NOTICE] switch_core_session.c:1193 Session 25 (sofia/internal/1001 at 196.202.51.44) Ended 2010-07-02 03:47:02.106398 [NOTICE] switch_core_session.c:1195 Close Channel sofia/internal/1001 at 196.202.51.44 [CS_DESTROY] 2010-07-02 03:47:02.106398 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/1001 at 196.202.51.44) Callstate Change HANGUP -> DOWN 2010-07-02 03:47:02.106398 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/1001 at 196.202.51.44) Running State Change CS_DESTROY 2010-07-02 03:47:02.106398 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1001 at 196.202.51.44) State DESTROY 2010-07-02 03:47:02.106398 [DEBUG] mod_sofia.c:358 sofia/internal/ 1001 at 196.202.51.44 SOFIA DESTROY 2010-07-02 03:47:02.106398 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1001 at 196.202.51.44 Standard DESTROY 2010-07-02 03:47:02.106398 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1001 at 196.202.51.44) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100702/2bacec89/attachment-0001.html From sanms.zhang at gmail.com Thu Jul 1 18:23:30 2010 From: sanms.zhang at gmail.com (chi zhang) Date: Fri, 2 Jul 2010 09:23:30 +0800 Subject: [Freeswitch-users] Problem of FreeSwitch and T.38 Test In-Reply-To: <5ECC3A8A-CDF9-4EF4-AD87-A3FAEDFF26AB@freeswitch.org> References: <5ECC3A8A-CDF9-4EF4-AD87-A3FAEDFF26AB@freeswitch.org> Message-ID: hi,brian log as below: ------------------------ start --------------------------- [ 2010-07-01 14:15:16.287030 [DEBUG] sofia.c:5928 IP 192.168.26.39 Approved by acl "192.168.26.0/24[]". Access Granted. [36m2010-07-01 14:15:16.289031 [NOTICE] switch_channel.c:776 New Channel sofia/internal/1000 at 192.168.26.39:15060[d8dc7307-efd6-48ed-8d85-d5f2d52deb10] [ 2010-07-01 14:15:16.289031 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_NEW [ 2010-07-01 14:15:16.289031 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1000 at 192.168.26.39:15060) State NEW [ 2010-07-01 14:15:16.331035 [DEBUG] sofia.c:4297 Channel sofia/internal/ 1000 at 192.168.26.39:15060 entering state [received][100] [ 2010-07-01 14:15:16.331035 [DEBUG] sofia.c:4308 Remote SDP: v=0 o=user1 3748 3748 IN IP4 192.168.26.39 s=- c=IN IP4 192.168.26.39 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 [ 2010-07-01 14:15:16.331035 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [PCMA:8:8000:20]/[G7221:115:32000:20] [ 2010-07-01 14:15:16.331035 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [PCMA:8:8000:20]/[G7221:107:16000:20] [ 2010-07-01 14:15:16.331035 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [PCMA:8:8000:20]/[G722:9:8000:20] [ 2010-07-01 14:15:16.331035 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] [ 2010-07-01 14:15:16.331035 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] [ 2010-07-01 14:15:16.331035 [DEBUG] sofia_glue.c:2442 Set Codec sofia/internal/1000 at 192.168.26.39:15060 PCMA/8000 20 ms 160 samples [ 2010-07-01 14:15:16.331035 [DEBUG] sofia_glue.c:3798 Set 2833 dtmf send/recv payload to 101 [ 2010-07-01 14:15:16.331035 [DEBUG] sofia.c:4455 (sofia/internal/ 1000 at 192.168.26.39:15060) State Change CS_NEW -> CS_INIT [ 2010-07-01 14:15:16.331035 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] [ 2010-07-01 14:15:16.333036 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_INIT [ 2010-07-01 14:15:16.333036 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.26.39:15060) State INIT [ 2010-07-01 14:15:16.333036 [DEBUG] mod_sofia.c:83 sofia/internal/ 1000 at 192.168.26.39:15060 SOFIA INIT [ 2010-07-01 14:15:16.333036 [DEBUG] mod_sofia.c:119 (sofia/internal/ 1000 at 192.168.26.39:15060) State Change CS_INIT -> CS_ROUTING [ 2010-07-01 14:15:16.333036 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] [ 2010-07-01 14:15:16.333036 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.26.39:15060) State INIT going to sleep [ 2010-07-01 14:15:16.333036 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_ROUTING [ 2010-07-01 14:15:16.333036 [DEBUG] switch_channel.c:1471 (sofia/internal/ 1000 at 192.168.26.39:15060) Callstate Change DOWN -> RINGING [ 2010-07-01 14:15:16.333036 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING [ 2010-07-01 14:15:16.333036 [DEBUG] mod_sofia.c:142 sofia/internal/ 1000 at 192.168.26.39:15060 SOFIA ROUTING [ 2010-07-01 14:15:16.333036 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1000 at 192.168.26.39:15060 Standard ROUTING [32m2010-07-01 14:15:16.333036 [INFO] mod_dialplan_xml.c:331 Processing 1000->666666 in context public [ Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->fax] continue=false [ Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] destination_number(666666) =~ /^fax$/ break=on-false [ Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->4444] continue=false [ Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [4444] destination_number(666666) =~ /^(4444)$/ break=on-false [ Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->fax] continue=false [ Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] destination_number(666666) =~ /^fax$/ break=on-false [ Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->666666] continue=false [ Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (PASS) [666666] destination_number(666666) =~ /^(666666)$/ break=on-false [ Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action answer() [ Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action playback(silence_stream://2000) [ Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action rxfax(/tmp/999.tiff) [ Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action hangup() [ 2010-07-01 14:15:16.333036 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1000 at 192.168.26.39:15060) State Change CS_ROUTING -> CS_EXECUTE [ 2010-07-01 14:15:16.333036 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] [ 2010-07-01 14:15:16.333036 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING going to sleep [ 2010-07-01 14:15:16.333036 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_EXECUTE [ 2010-07-01 14:15:16.333036 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1000 at 192.168.26.39:15060) State EXECUTE [ 2010-07-01 14:15:16.333036 [DEBUG] mod_sofia.c:235 sofia/internal/ 1000 at 192.168.26.39:15060 SOFIA EXECUTE [ 2010-07-01 14:15:16.333036 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1000 at 192.168.26.39:15060 Standard EXECUTE [ EXECUTE sofia/internal/1000 at 192.168.26.39:15060 answer() [ 2010-07-01 14:15:16.367038 [DEBUG] sofia_glue.c:2682 AUDIO RTP [sofia/internal/1000 at 192.168.26.39:15060] 192.168.26.39 port 30734 -> 192.168.26.39 port 6000 codec: 8 ms: 20 [ 2010-07-01 14:15:16.369048 [DEBUG] switch_rtp.c:1413 Starting timer [soft] 160 bytes per 20ms [ 2010-07-01 14:15:16.371038 [DEBUG] sofia_glue.c:2892 Set 2833 dtmf send payload to 101 [ 2010-07-01 14:15:16.371038 [DEBUG] sofia_glue.c:2897 Set 2833 dtmf receive payload to 101 [ 2010-07-01 14:15:16.371038 [DEBUG] mod_sofia.c:669 Local SDP sofia/internal/1000 at 192.168.26.39:15060: v=0 o=FreeSWITCH 1277934182 1277934183 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=audio 30734 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv [ 2010-07-01 14:15:16.371038 [DEBUG] switch_core_session.c:647 Send signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] [ 2010-07-01 14:15:16.371038 [DEBUG] switch_channel.c:2494 (sofia/internal/ 1000 at 192.168.26.39:15060) Callstate Change RINGING -> ACTIVE [36m2010-07-01 14:15:16.371038 [NOTICE] mod_dptools.c:746 Channel [sofia/internal/1000 at 192.168.26.39:15060] has been answered [ 2010-07-01 14:15:16.371038 [DEBUG] sofia.c:4297 Channel sofia/internal/ 1000 at 192.168.26.39:15060 entering state [completed][200] [ 2010-07-01 14:15:16.371038 [DEBUG] sofia.c:4297 Channel sofia/internal/ 1000 at 192.168.26.39:15060 entering state [ready][200] [ EXECUTE sofia/internal/1000 at 192.168.26.39:15060playback(silence_stream://2000) [ 2010-07-01 14:15:16.371038 [DEBUG] switch_ivr_play_say.c:1161 Codec Activated L16 at 8000hz 1 channels 20ms [ 2010-07-01 14:15:18.364218 [DEBUG] switch_ivr_play_say.c:1468 done playing file [ EXECUTE sofia/internal/1000 at 192.168.26.39:15060 rxfax(/tmp/999.tiff) [ 2010-07-01 14:15:18.364218 [ERR] mod_spandsp.c:64 This is for fax test: receive fax [ 2010-07-01 14:15:18.364218 [ERR] mod_spandsp_fax.c:445 trans mode = 1 [ 2010-07-01 14:15:18.364218 [ERR] mod_spandsp_fax.c:591 This is for fax test: prag go to here!!! [ 2010-07-01 14:15:18.364218 [DEBUG] mod_spandsp_fax.c:1064 Raw read codec activation Success L16 20000 [ 2010-07-01 14:15:18.364218 [DEBUG] switch_core_codec.c:122 sofia/internal/1000 at 192.168.26.39:15060 Push codec L16:10 [ 2010-07-01 14:15:18.364218 [DEBUG] mod_spandsp_fax.c:1080 Raw write codec activation Success L16 [ 2010-07-01 14:15:18.645238 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 [ 2010-07-01 14:15:26.374918 [DEBUG] sofia.c:4297 Channel sofia/internal/ 1000 at 192.168.26.39:15060 entering state [received][100] [ 2010-07-01 14:15:26.374918 [DEBUG] sofia.c:4308 Remote SDP: v=0 o=root 0 0 IN IP4 192.168.26.39 s=Session SDP c=IN IP4 192.168.26.39 t=0 0 m=image 49172 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy [ 2010-07-01 14:15:26.382925 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 [ 2010-07-01 14:15:26.382925 [DEBUG] mod_spandsp_fax.c:744 T38FaxVersion = 0 [ 2010-07-01 14:15:26.382925 [DEBUG] mod_spandsp_fax.c:745 T38MaxBitRate = 9600 [ 2010-07-01 14:15:26.382925 [DEBUG] mod_spandsp_fax.c:746 T38FaxFillBitRemoval = 1 [ 2010-07-01 14:15:26.382925 [DEBUG] mod_spandsp_fax.c:747 T38FaxTranscodingMMR = 1 [ 2010-07-01 14:15:26.382925 [DEBUG] mod_spandsp_fax.c:748 T38FaxTranscodingJBIG = 1 [ 2010-07-01 14:15:26.382925 [DEBUG] mod_spandsp_fax.c:749 T38FaxRateManagement = 'transferredTCF' [ 2010-07-01 14:15:26.382925 [DEBUG] mod_spandsp_fax.c:750 T38FaxMaxBuffer = 200 [ 2010-07-01 14:15:26.382925 [DEBUG] mod_spandsp_fax.c:751 T38FaxMaxDatagram = 72 [ 2010-07-01 14:15:26.382925 [DEBUG] mod_spandsp_fax.c:752 T38FaxUdpEC = 't38UDPRedundancy' [ 2010-07-01 14:15:26.382925 [DEBUG] mod_spandsp_fax.c:753 T38VendorInfo = '' [ 2010-07-01 14:15:26.382925 [DEBUG] mod_spandsp_fax.c:754 ip = '192.168.26.39' [ 2010-07-01 14:15:26.382925 [DEBUG] mod_spandsp_fax.c:756 port = 49172 [ 2010-07-01 14:15:26.382925 [DEBUG] mod_sofia.c:1232 IMAGE UDPTL CHANGING DEST TO: [192.168.26.39:49172] [ 2010-07-01 14:15:26.382925 [DEBUG] sofia_glue.c:122 sofia/internal/ 1000 at 192.168.26.39:15060 image media sdp: v=0 o=FreeSWITCH 1277934182 1277934184 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 30734 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 [ 2010-07-01 14:15:26.382925 [ERR] mod_spandsp_fax.c:445 trans mode = 0 [ 2010-07-01 14:15:26.382925 [DEBUG] sofia.c:4297 Channel sofia/internal/ 1000 at 192.168.26.39:15060 entering state [completed][200] [ 2010-07-01 14:15:46.390709 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 This is for fax test : s->state= 17 This is for fax test: command exceed max [ 2010-07-01 14:15:49.052963 [DEBUG] mod_spandsp_fax.c:302 result = 48 [ 2010-07-01 14:15:49.052963 [DEBUG] mod_spandsp_fax.c:304 ============================================================================== [ 2010-07-01 14:15:49.052963 [DEBUG] mod_spandsp_fax.c:316 Fax processing not successful - result (48) Disconnected after permitted retries. [ 2010-07-01 14:15:49.052963 [DEBUG] mod_spandsp_fax.c:321 Remote station id: [ 2010-07-01 14:15:49.052963 [DEBUG] mod_spandsp_fax.c:322 Local station id: SpanDSP Fax Ident [ 2010-07-01 14:15:49.052963 [DEBUG] mod_spandsp_fax.c:323 Pages transferred: 0 [ 2010-07-01 14:15:49.052963 [DEBUG] mod_spandsp_fax.c:325 Total fax pages: 0 [ 2010-07-01 14:15:49.052963 [DEBUG] mod_spandsp_fax.c:326 Image resolution: 0x0 [ 2010-07-01 14:15:49.052963 [DEBUG] mod_spandsp_fax.c:327 Transfer Rate: 14400 [ 2010-07-01 14:15:49.052963 [DEBUG] mod_spandsp_fax.c:329 ECM status off [ 2010-07-01 14:15:49.052963 [DEBUG] mod_spandsp_fax.c:330 remote country: [ 2010-07-01 14:15:49.052963 [DEBUG] mod_spandsp_fax.c:331 remote vendor: [ 2010-07-01 14:15:49.052963 [DEBUG] mod_spandsp_fax.c:332 remote model: [ 2010-07-01 14:15:49.052963 [DEBUG] mod_spandsp_fax.c:334 ============================================================================== [ 2010-07-01 14:15:51.388172 [DEBUG] switch_core_codec.c:146 sofia/internal/1000 at 192.168.26.39:15060 Restore previous codec PCMA:8. [ EXECUTE sofia/internal/1000 at 192.168.26.39:15060 hangup() [ 2010-07-01 14:15:51.388172 [DEBUG] switch_channel.c:2261 (sofia/internal/ 1000 at 192.168.26.39:15060) Callstate Change ACTIVE -> HANGUP [36m2010-07-01 14:15:51.388172 [NOTICE] mod_dptools.c:732 Hangup sofia/internal/1000 at 192.168.26.39:15060 [CS_EXECUTE] [NORMAL_CLEARING] [ 2010-07-01 14:15:51.388172 [DEBUG] switch_channel.c:2277 Send signal sofia/internal/1000 at 192.168.26.39:15060 [KILL] [ 2010-07-01 14:15:51.388172 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] [ 2010-07-01 14:15:51.388172 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1000 at 192.168.26.39:15060) State EXECUTE going to sleep [ 2010-07-01 14:15:51.391171 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_HANGUP [ 2010-07-01 14:15:51.391171 [DEBUG] switch_core_state_machine.c:500 (sofia/internal/1000 at 192.168.26.39:15060) State HANGUP [ 2010-07-01 14:15:51.391171 [DEBUG] mod_sofia.c:447 Channel sofia/internal/1000 at 192.168.26.39:15060 hanging up, cause: NORMAL_CLEARING [ 2010-07-01 14:15:51.433177 [DEBUG] mod_sofia.c:490 Sending BYE to sofia/internal/1000 at 192.168.26.39:15060 [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1000 at 192.168.26.39:15060 Standard HANGUP, cause: NORMAL_CLEARING [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_state_machine.c:500 (sofia/internal/1000 at 192.168.26.39:15060) State HANGUP going to sleep [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1000 at 192.168.26.39:15060) State Change CS_HANGUP -> CS_REPORTING [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_REPORTING [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_state_machine.c:591 (sofia/internal/1000 at 192.168.26.39:15060) State REPORTING [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1000 at 192.168.26.39:15060 Standard REPORTING, cause: NORMAL_CLEARING [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_state_machine.c:591 (sofia/internal/1000 at 192.168.26.39:15060) State REPORTING going to sleep [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1000 at 192.168.26.39:15060) State Change CS_REPORTING -> CS_DESTROY [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_session.c:1175 Session 1 (sofia/internal/1000 at 192.168.26.39:15060) Locked, Waiting on external entities [36m2010-07-01 14:15:51.433177 [NOTICE] switch_core_session.c:1193 Session 1 (sofia/internal/1000 at 192.168.26.39:15060) Ended [36m2010-07-01 14:15:51.433177 [NOTICE] switch_core_session.c:1195 Close Channel sofia/internal/1000 at 192.168.26.39:15060 [CS_DESTROY] [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/1000 at 192.168.26.39:15060) Callstate Change HANGUP -> DOWN [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_DESTROY [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1000 at 192.168.26.39:15060) State DESTROY [ 2010-07-01 14:15:51.433177 [DEBUG] mod_sofia.c:352 sofia/internal/ 1000 at 192.168.26.39:15060 SOFIA DESTROY [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1000 at 192.168.26.39:15060 Standard DESTROY [ 2010-07-01 14:15:51.433177 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1000 at 192.168.26.39:15060) State DESTROY going to sleep ------------------------ end ------------------------- regards sammy 2010/7/1 Brian West > You'll need to provide the sip trace and full debug logs. > > /b > > On Jul 1, 2010, at 3:30 AM, chi zhang wrote: > > > Does anyone know the reason? > > Where is the span_log of spandsp lib ? i cannt find it... > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100702/dd90556c/attachment-0001.html From brian at freeswitch.org Thu Jul 1 18:29:03 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 1 Jul 2010 20:29:03 -0500 Subject: [Freeswitch-users] Problem of FreeSwitch and T.38 Test In-Reply-To: References: <5ECC3A8A-CDF9-4EF4-AD87-A3FAEDFF26AB@freeswitch.org> Message-ID: turn sip on sofia profile xxx siptrace on /b On Jul 1, 2010, at 8:23 PM, chi zhang wrote: > hi,brian > From sanms.zhang at gmail.com Thu Jul 1 18:51:22 2010 From: sanms.zhang at gmail.com (chi zhang) Date: Fri, 2 Jul 2010 09:51:22 +0800 Subject: [Freeswitch-users] Problem of FreeSwitch and T.38 Test In-Reply-To: References: <5ECC3A8A-CDF9-4EF4-AD87-A3FAEDFF26AB@freeswitch.org> Message-ID: I got it. -------------------log start-------------------------- recv 576 bytes from udp/[192.168.26.39]:15060 at 01:41:26.500127: ------------------------------------------------------------------------ INVITE sip:666666 at 192.168.26.39:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 From: 1000 ;tag=1 To: 666666 Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: sip:1000 at 192.168.26.39:15060 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 184 v=0 o=user1 3748 3748 IN IP4 192.168.26.39 s=- c=IN IP4 192.168.26.39 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 ------------------------------------------------------------------------ send 294 bytes to udp/[192.168.26.39]:15060 at 01:41:26.500477: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 From: 1000 ;tag=1 To: 666666 Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ 2010-07-02 09:41:26.498946 [DEBUG] sofia.c:5928 IP 192.168.26.39 Approved by acl "192.168.26.0/24[]". Access Granted. 2010-07-02 09:41:26.498946 [NOTICE] switch_channel.c:776 New Channel sofia/internal/1000 at 192.168.26.39:15060[f588c66c-48c8-4220-a944-8de287adb3ab] 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_NEW 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1000 at 192.168.26.39:15060) State NEW 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4297 Channel sofia/internal/ 1000 at 192.168.26.39:15060 entering state [received][100] 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4308 Remote SDP: v=0 o=user1 3748 3748 IN IP4 192.168.26.39 s=- c=IN IP4 192.168.26.39 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [PCMA:8:8000:20]/[G7221:115:32000:20] 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [PCMA:8:8000:20]/[G7221:107:16000:20] 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [PCMA:8:8000:20]/[G722:9:8000:20] 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:2442 Set Codec sofia/internal/1000 at 192.168.26.39:15060 PCMA/8000 20 ms 160 samples 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3798 Set 2833 dtmf send/recv payload to 101 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4455 (sofia/internal/ 1000 at 192.168.26.39:15060) State Change CS_NEW -> CS_INIT 2010-07-02 09:41:26.542951 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_INIT 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.26.39:15060) State INIT 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:83 sofia/internal/ 1000 at 192.168.26.39:15060 SOFIA INIT 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:119 (sofia/internal/ 1000 at 192.168.26.39:15060) State Change CS_INIT -> CS_ROUTING 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.26.39:15060) State INIT going to sleep 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_ROUTING 2010-07-02 09:41:26.544953 [DEBUG] switch_channel.c:1471 (sofia/internal/ 1000 at 192.168.26.39:15060) Callstate Change DOWN -> RINGING 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:142 sofia/internal/ 1000 at 192.168.26.39:15060 SOFIA ROUTING 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1000 at 192.168.26.39:15060 Standard ROUTING 2010-07-02 09:41:26.544953 [INFO] mod_dialplan_xml.c:331 Processing 1000->666666 in context public Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->fax] continue=false Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] destination_number(666666) =~ /^fax$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->4444] continue=false Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [4444] destination_number(666666) =~ /^(4444)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->fax] continue=false Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] destination_number(666666) =~ /^fax$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->666666] continue=false Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (PASS) [666666] destination_number(666666) =~ /^(666666)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action answer() Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action playback(silence_stream://2000) Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action rxfax(/tmp/999.tiff) Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action hangup() 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1000 at 192.168.26.39:15060) State Change CS_ROUTING -> CS_EXECUTE 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING going to sleep 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_EXECUTE 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1000 at 192.168.26.39:15060) State EXECUTE 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:235 sofia/internal/ 1000 at 192.168.26.39:15060 SOFIA EXECUTE 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1000 at 192.168.26.39:15060 Standard EXECUTE EXECUTE sofia/internal/1000 at 192.168.26.39:15060 answer() 2010-07-02 09:41:26.580967 [DEBUG] sofia_glue.c:2682 AUDIO RTP [sofia/internal/1000 at 192.168.26.39:15060] 192.168.26.39 port 22464 -> 192.168.26.39 port 6000 codec: 8 ms: 20 2010-07-02 09:41:26.580967 [DEBUG] switch_rtp.c:1413 Starting timer [soft] 160 bytes per 20ms 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2892 Set 2833 dtmf send payload to 101 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2897 Set 2833 dtmf receive payload to 101 2010-07-02 09:41:26.582952 [DEBUG] mod_sofia.c:669 Local SDP sofia/internal/ 1000 at 192.168.26.39:15060: v=0 o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=audio 22464 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv send 1091 bytes to udp/[192.168.26.39]:15060 at 01:41:26.584093: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 From: 1000 ;tag=1 To: 666666 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "666666" >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=audio 22464 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ recv 369 bytes from udp/[192.168.26.39]:15060 at 01:41:26.584205: ------------------------------------------------------------------------ ACK sip:666666 at 192.168.26.39:25060 SIP/2.0 Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-4 From: 1000 ;tag=1 To: 666666 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 ACK Contact: sip:1000 at 192.168.26.39:15060 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ------------------------------------------------------------------------ 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/ 1000 at 192.168.26.39:15060 entering state [completed][200] 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/ 1000 at 192.168.26.39:15060 entering state [ready][200] 2010-07-02 09:41:26.582952 [DEBUG] switch_core_session.c:647 Send signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] 2010-07-02 09:41:26.582952 [DEBUG] switch_channel.c:2494 (sofia/internal/ 1000 at 192.168.26.39:15060) Callstate Change RINGING -> ACTIVE 2010-07-02 09:41:26.582952 [NOTICE] mod_dptools.c:746 Channel [sofia/internal/1000 at 192.168.26.39:15060] has been answered EXECUTE sofia/internal/1000 at 192.168.26.39:15060playback(silence_stream://2000) 2010-07-02 09:41:26.584953 [DEBUG] switch_ivr_play_say.c:1161 Codec Activated L16 at 8000hz 1 channels 20ms 2010-07-02 09:41:28.578954 [DEBUG] switch_ivr_play_say.c:1468 done playing file EXECUTE sofia/internal/1000 at 192.168.26.39:15060 rxfax(/tmp/999.tiff) 2010-07-02 09:41:28.578954 [ERR] mod_spandsp.c:64 This is for fax test: receive fax 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:445 trans mode = 1 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:591 This is for fax test: prag go to here!!! 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1064 Raw read codec activation Success L16 20000 2010-07-02 09:41:28.578954 [DEBUG] switch_core_codec.c:122 sofia/internal/ 1000 at 192.168.26.39:15060 Push codec L16:10 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1080 Raw write codec activation Success L16 2010-07-02 09:41:28.857958 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 send 956 bytes to udp/[192.168.21.76]:5060 at 01:41:29.754477: ------------------------------------------------------------------------ NOTIFY sip:1001 at 192.168.21.76:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.26.39:25060;rport;branch=z9hG4bKS989SFgmXvmNF Max-Forwards: 70 From: "1001" ;tag=vr6p3K1XK247r To: "1001" ;tag=26647676 Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. CSeq: 132897310 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=timeout Content-Type: application/simple-message-summary Content-Length: 65 Messages-Waiting: no Message-Account: sip:1001 at 192.168.26.39 ------------------------------------------------------------------------ 2010-07-02 09:41:29.757979 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 recv 389 bytes from udp/[192.168.21.76]:5060 at 01:41:29.758362: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.26.39:25060 ;rport=25060;branch=z9hG4bKS989SFgmXvmNF To: "1001";tag=26647676 From: "1001";tag=vr6p3K1XK247r Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. CSeq: 132897310 NOTIFY Accept-Language: en Content-Length: 0 ------------------------------------------------------------------------ 2010-07-02 09:41:36.557090 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:36.587253: ------------------------------------------------------------------------ INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: sip:666666 at 192.168.26.39:25060 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 345 v=0 o=root 0 0 IN IP4 192.168.26.39 s=Session SDP c=IN IP4 192.168.26.39 t=0 0 m=image 49172 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy ------------------------------------------------------------------------ send 312 bytes to udp/[192.168.26.39]:25060 at 01:41:36.587546: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ recv 312 bytes from udp/[192.168.26.39]:25060 at 01:41:36.587641: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4297 Channel sofia/internal/1000 at 192.168.26.39:15060 entering state [received][100] User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ 2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4308 Remote SDP: v=0 o=root 0 0 IN IP4 192.168.26.39 s=Session SDP c=IN IP4 192.168.26.39 t=0 0 m=image 49172 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:744 T38FaxVersion = 0 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:745 T38MaxBitRate = 9600 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:746 T38FaxFillBitRemoval = 1 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:747 T38FaxTranscodingMMR = 1 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:748 T38FaxTranscodingJBIG = 1 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:749 T38FaxRateManagement = 'transferredTCF' 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:750 T38FaxMaxBuffer = 200 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:751 T38FaxMaxDatagram = 72 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:752 T38FaxUdpEC = 't38UDPRedundancy' 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:753 T38VendorInfo = '' 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:754 ip = '192.168.26.39' 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:756 port = 49172 2010-07-02 09:41:36.597094 [DEBUG] mod_sofia.c:1232 IMAGE UDPTL CHANGING DEST TO: [192.168.26.39:49172] 2010-07-02 09:41:36.597094 [DEBUG] sofia_glue.c:122 sofia/internal/ 1000 at 192.168.26.39:15060 image media sdp: v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:445 trans mode = 0 send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:36.597525: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ 2010-07-02 09:41:36.597094 [DEBUG] sofia.c:4297 Channel sofia/internal/ 1000 at 192.168.26.39:15060 entering state [completed][200] recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:36.597657: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ 2010-07-02 09:41:36.607095 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:37.089156: ------------------------------------------------------------------------ INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: sip:666666 at 192.168.26.39:25060 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 345 v=0 o=root 0 0 IN IP4 192.168.26.39 s=Session SDP c=IN IP4 192.168.26.39 t=0 0 m=image 49172 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy ------------------------------------------------------------------------ send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.089302: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.089395: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.098259: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.098339: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:38.091157: ------------------------------------------------------------------------ INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: sip:666666 at 192.168.26.39:25060 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 345 v=0 o=root 0 0 IN IP4 192.168.26.39 s=Session SDP c=IN IP4 192.168.26.39 t=0 0 m=image 49172 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy ------------------------------------------------------------------------ send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.091305: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.091406: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.098260: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.098344: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ This is for fax test: dis 5This is for fax test: cause disconnect 4 recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:40.093267: ------------------------------------------------------------------------ INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: sip:666666 at 192.168.26.39:25060 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 345 v=0 o=root 0 0 IN IP4 192.168.26.39 s=Session SDP c=IN IP4 192.168.26.39 t=0 0 m=image 49172 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy ------------------------------------------------------------------------ send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.093421: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.093541: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.098263: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.098418: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ 2010-07-02 09:41:41.606147 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:44.095284: ------------------------------------------------------------------------ INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: sip:666666 at 192.168.26.39:25060 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 345 v=0 o=root 0 0 IN IP4 192.168.26.39 s=Session SDP c=IN IP4 192.168.26.39 t=0 0 m=image 49172 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy ------------------------------------------------------------------------ send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.095457: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.095556: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.098263: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.098399: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 ------------------------------------------------------------------------ 2010-07-02 09:41:46.607257 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:48.098287: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 From: 666666 ;tag=1 To: 1000 ;tag=85ySHt6rmaSyp Call-ID: 1-16005 at 192.168.26.39 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 341 v=0 o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=image 22464 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 -----------------------------log end---------------------------------- 2010/7/2 Brian West > turn sip on > > sofia profile xxx siptrace on > > /b > > On Jul 1, 2010, at 8:23 PM, chi zhang wrote: > > > hi,brian > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100702/4ef443fc/attachment-0001.html From nagalenoj at gmail.com Thu Jul 1 22:25:36 2010 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Fri, 2 Jul 2010 10:55:36 +0530 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: I've pasted the console log here, http://pastebin.freeswitch.org/13333 On Wed, Jun 30, 2010 at 11:15 PM, Michael Collins wrote: > Can you supply a console log of these calls? > -MC > > > On Wed, Jun 30, 2010 at 7:37 AM, Nagalenoj H. wrote: > >> Dear Anthony, >> I've tried using the group_confirm_cancel_timeout as per the >> discussion we had in IRC. You wanted to used it as part of dial string and >> not as a channel variable. >> But, It doesn't work for me. >> >> Here is how I've given the commands and the script I've executed. Even >> when I give group_confirm_cancel_timeout, the callee's leg is getting >> disconnected after legtimeout. >> >> >> connect >> >> sendmsg >> call-command: execute >> execute-app-name:answer >> >> sendmsg >> call-command: execute >> execute-app-name: set >> execute-app-arg: group_confirm_key=exec >> >> sendmsg >> call-command: execute >> execute-app-name: set >> execute-app-arg: group_confirm_file=perl /root/bridge.pl >> >> sendmsg >> call-command: execute >> execute-app-name: bridge >> execute-app-arg: {group_confirm_cancel_timeout=1}[leg_timeout=10]user/1005 >> >> >> >> bridge.pl: >> #!/usr/bin/perl >> use freeswitch; >> >> our $session; >> freeswitch::consoleLog("info","Goint to get the digits"); >> # To simulate the scenario I used sleep here. >> sleep(30); >> 1; >> >> Kindly tell me whats wrong in the above. >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100702/bf1d7cb0/attachment.html From irmatov at gmail.com Fri Jul 2 00:09:47 2010 From: irmatov at gmail.com (Timur Irmatov) Date: Fri, 2 Jul 2010 12:09:47 +0500 Subject: [Freeswitch-users] mod_erlang_event problem: REPRODUCIBLE, fix needed In-Reply-To: <20100701190152.GG17555@hijacked.us> References: <20100701190152.GG17555@hijacked.us> Message-ID: On Fri, Jul 2, 2010 at 12:01 AM, Andrew Thompson wrote: > On Thu, Jul 01, 2010 at 03:16:05PM +0500, Timur Irmatov wrote: >> On Thu, Jul 1, 2010 at 2:34 PM, Timur Irmatov wrote: >> > When inbound-bypass-media is set to false, I am >> > successfully managing a call through erlang application. When it is >> > set to false, I am seeing 'got pid' and 'exit >> > erlang_outbound_function' in log files, and I cannot manage calls in >> > erlang. >> >> When it is set to *true* I am having a problem. >> > > So in the other thread you said it was working briefly until you > restarted FreeSWITCH, is this the root cause of that problem, or is it > another issue? I'm baffled as to why bypass media is having any effect > here. This is same issue, and bypass media is the root cause of the problem. I'm baffled too. Actually, changing bypass media was the reason of the restart. But, same as you, I failed to realize that I *changed* configuration of FreeSWITCH and after that mod_erlang_event failed to deliver events to erlang application because I couldn't imagine that bypass media would affect mod_erlang_event. Problem is stable. With bypass media set to false, mod_erlang_event works as expected. I change bypass media to true, reloadxml, sofia profile external restart, my problem reappears. Change bypass media back to false, reloadxml, sofia profile external restart, works again. -- Timur Irmatov, xmpp:irmatov at jabber.ru From irmatov at gmail.com Fri Jul 2 00:23:43 2010 From: irmatov at gmail.com (Timur Irmatov) Date: Fri, 2 Jul 2010 12:23:43 +0500 Subject: [Freeswitch-users] mod_erlang_event problem: REPRODUCIBLE, fix needed In-Reply-To: References: <20100701190152.GG17555@hijacked.us> Message-ID: On Fri, Jul 2, 2010 at 1:33 AM, Anthony Minessale wrote: > pastebin a console log with debug and sip trace enabled > http://pastebin.freeswitch.org > sofia profile internal siptrace on > console loglevel debug > *if you use some other profile besides internal enable siptrace for that > instead. ?do it for any profiles involved. I use only external profile. Here is the trace of failed call with inbound-bypass-media=true: http://pastebin.freeswitch.org/13334 Here is successfull call with inbound-bypass-media=false: http://pastebin.freeswitch.org/13335 -- Timur Irmatov, xmpp:irmatov at jabber.ru From irmatov at gmail.com Fri Jul 2 00:27:48 2010 From: irmatov at gmail.com (Timur Irmatov) Date: Fri, 2 Jul 2010 12:27:48 +0500 Subject: [Freeswitch-users] mod_erlang_event problem: REPRODUCIBLE, fix needed In-Reply-To: References: <20100701190152.GG17555@hijacked.us> Message-ID: Issue is descirbed in http://jira.freeswitch.org/browse/MODEVENT-65 with same info and logs. On Fri, Jul 2, 2010 at 12:23 PM, Timur Irmatov wrote: > On Fri, Jul 2, 2010 at 1:33 AM, Anthony Minessale > wrote: >> pastebin a console log with debug and sip trace enabled >> http://pastebin.freeswitch.org >> sofia profile internal siptrace on >> console loglevel debug >> *if you use some other profile besides internal enable siptrace for that >> instead. ?do it for any profiles involved. > > I use only external profile. Here is the trace of failed call with > inbound-bypass-media=true: > > http://pastebin.freeswitch.org/13334 > > Here is successfull call with inbound-bypass-media=false: > > http://pastebin.freeswitch.org/13335 > > > -- > Timur Irmatov, xmpp:irmatov at jabber.ru > -- Timur Irmatov, xmpp:irmatov at jabber.ru From svetikvoip at gmail.com Fri Jul 2 07:17:21 2010 From: svetikvoip at gmail.com (Svetik) Date: Fri, 02 Jul 2010 10:17:21 -0400 Subject: [Freeswitch-users] Upgraded from 1.0.4 pre8 to the latest Git tree. Skype does not work anymore. References: 4C2A0CF5.4090606@gmail.com Message-ID: <4C2DF4F1.3070308@gmail.com> Giovanni, Thank you for your reply. I did not use my original configuration file, I have migrated changes into new configuration file. That is how it looks like: I have tried to use old configuration file from skypiax module, but it gives me the same problem. Any thoughts? Can I try something that helps to diagnose this problem? Igor >>On Tue, Jun 29, 2010 at 5:10 PM, Svetik> wrote: >>/ Hi, />>/ />>/ On weekend I have upgraded to the latest Git tree from 1.0.4 pre8 which />>/ I was running for a long time, year may be. Everything went smooth, />>/ except Skype does not work anymore. Basically I followed Download& / >Have you updated the configuration file? Maybe the configuration file >format has changed from the one you are using... >-giovanni From a.afzali2003 at gmail.com Fri Jul 2 09:17:15 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Fri, 2 Jul 2010 20:47:15 +0430 Subject: [Freeswitch-users] Using transfer_after_bridge To Handle Post Agent Visit Services Message-ID: Hi FreeSWITCH, I'm using the transfer_after_bridge variable to capture calls which have been serviced by my agents (via a single FIFO) to do post services (such as say goodbye) successfully. Mean while I've enabled transfer feature ( extension dx ) on my agents legs to be enable to transfer their callers to desire extensions. Here is my problem. When agent transfer his caller to an extension as I expected, caller routes to the specified extension ( post_agent_visit ) but I don't know how I could retrieve that extension which agent want to transfer to. BEST -- afshin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100702/d2d60d7a/attachment.html From anthony.minessale at gmail.com Fri Jul 2 09:39:35 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 2 Jul 2010 11:39:35 -0500 Subject: [Freeswitch-users] Using transfer_after_bridge To Handle Post Agent Visit Services In-Reply-To: References: Message-ID: check ${DNIS} On Fri, Jul 2, 2010 at 11:17 AM, afshin afzali wrote: > Hi FreeSWITCH, > > I'm using the transfer_after_bridge variable to capture calls which have > been serviced by my agents (via a single FIFO) to do post services (such as > say goodbye) successfully. > > data="transfer_after_bridge=post_agent_visit:XML:public"/> > > Mean while I've enabled transfer feature ( extension dx ) on my agents legs > to be enable to transfer their callers to desire extensions. Here is my > problem. When agent transfer his caller to an extension as I expected, > caller routes to the specified extension ( post_agent_visit ) but I don't > know how I could retrieve that extension which agent want to transfer to. > > BEST > -- afshin > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100702/2633c608/attachment.html From anthony.minessale at gmail.com Fri Jul 2 09:45:06 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 2 Jul 2010 11:45:06 -0500 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: update and reproduce that same log with latest GIT the version you are using has an issue. On Fri, Jul 2, 2010 at 12:25 AM, Nagalenoj H. wrote: > I've pasted the console log here, > > http://pastebin.freeswitch.org/13333 > > > On Wed, Jun 30, 2010 at 11:15 PM, Michael Collins wrote: > >> Can you supply a console log of these calls? >> -MC >> >> >> On Wed, Jun 30, 2010 at 7:37 AM, Nagalenoj H. wrote: >> >>> Dear Anthony, >>> I've tried using the group_confirm_cancel_timeout as per the >>> discussion we had in IRC. You wanted to used it as part of dial string and >>> not as a channel variable. >>> But, It doesn't work for me. >>> >>> Here is how I've given the commands and the script I've executed. Even >>> when I give group_confirm_cancel_timeout, the callee's leg is getting >>> disconnected after legtimeout. >>> >>> >>> connect >>> >>> sendmsg >>> call-command: execute >>> execute-app-name:answer >>> >>> sendmsg >>> call-command: execute >>> execute-app-name: set >>> execute-app-arg: group_confirm_key=exec >>> >>> sendmsg >>> call-command: execute >>> execute-app-name: set >>> execute-app-arg: group_confirm_file=perl /root/bridge.pl >>> >>> sendmsg >>> call-command: execute >>> execute-app-name: bridge >>> execute-app-arg: >>> {group_confirm_cancel_timeout=1}[leg_timeout=10]user/1005 >>> >>> >>> >>> bridge.pl: >>> #!/usr/bin/perl >>> use freeswitch; >>> >>> our $session; >>> freeswitch::consoleLog("info","Goint to get the digits"); >>> # To simulate the scenario I used sleep here. >>> sleep(30); >>> 1; >>> >>> Kindly tell me whats wrong in the above. >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > Regards, > Nagalenoj H. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100702/3088895c/attachment-0001.html From anthony.minessale at gmail.com Fri Jul 2 09:56:28 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 2 Jul 2010 11:56:28 -0500 Subject: [Freeswitch-users] mod_erlang_event problem: REPRODUCIBLE, fix needed In-Reply-To: References: <20100701190152.GG17555@hijacked.us> Message-ID: The problem is that you cannot park channels in the bypass media state. I did a patch to the code that will probably allow this so you can update and test it. The better approach is probably to set the bypass_media=true channel variable on an as-needed basis rather than setting that explicit flag or bypass_media_after_bridge=true channel variable so the call is using media during the park and takes it away when the call is bridged. On Fri, Jul 2, 2010 at 2:27 AM, Timur Irmatov wrote: > Issue is descirbed in http://jira.freeswitch.org/browse/MODEVENT-65 > with same info and logs. > > On Fri, Jul 2, 2010 at 12:23 PM, Timur Irmatov wrote: > > On Fri, Jul 2, 2010 at 1:33 AM, Anthony Minessale > > wrote: > >> pastebin a console log with debug and sip trace enabled > >> http://pastebin.freeswitch.org > >> sofia profile internal siptrace on > >> console loglevel debug > >> *if you use some other profile besides internal enable siptrace for that > >> instead. do it for any profiles involved. > > > > I use only external profile. Here is the trace of failed call with > > inbound-bypass-media=true: > > > > http://pastebin.freeswitch.org/13334 > > > > Here is successfull call with inbound-bypass-media=false: > > > > http://pastebin.freeswitch.org/13335 > > > > > > -- > > Timur Irmatov, xmpp:irmatov at jabber.ru > > > > > > -- > Timur Irmatov, xmpp:irmatov at jabber.ru > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100702/d9b81b45/attachment.html From saeedahmad1981 at gmail.com Fri Jul 2 13:07:08 2010 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Fri, 2 Jul 2010 22:07:08 +0200 Subject: [Freeswitch-users] ooH323 vs mod_h323 Message-ID: Dear FSers, which one is better to use when there are higher number of concurrent calls? *ooH323 *or *mod_h323*. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100702/5184620e/attachment.html From brian at freeswitch.org Fri Jul 2 13:14:21 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 2 Jul 2010 15:14:21 -0500 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: Message-ID: We don't have an ooh323 driver in FreeSWITCH. /b On Jul 2, 2010, at 3:07 PM, Saeed Ahmed wrote: > Dear FSers, > > which one is better to use when there are higher number of concurrent calls? ooH323 or mod_h323. > > Thanks From anthony.minessale at gmail.com Fri Jul 2 13:15:55 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 2 Jul 2010 15:15:55 -0500 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: Message-ID: The one you tell us about after you test them both =D I honestly don't know. On Fri, Jul 2, 2010 at 3:07 PM, Saeed Ahmed wrote: > Dear FSers, > > which one is better to use when there are higher number of concurrent > calls? *ooH323 *or *mod_h323*. > > Thanks > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100702/6a35d306/attachment.html From anthony.minessale at gmail.com Fri Jul 2 13:24:19 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 2 Jul 2010 15:24:19 -0500 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: Message-ID: i think he meant opal On Fri, Jul 2, 2010 at 3:14 PM, Brian West wrote: > We don't have an ooh323 driver in FreeSWITCH. > > /b > > On Jul 2, 2010, at 3:07 PM, Saeed Ahmed wrote: > > > Dear FSers, > > > > which one is better to use when there are higher number of concurrent > calls? ooH323 or mod_h323. > > > > Thanks > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100702/1e9029ae/attachment.html From brian at freeswitch.org Fri Jul 2 13:28:14 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 2 Jul 2010 15:28:14 -0500 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: Message-ID: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> DOH... btw ooh323 is now license compatible with FreeSWITCH if anyone wants to write one. /b On Jul 2, 2010, at 3:24 PM, Anthony Minessale wrote: > i think he meant opal > > > On Fri, Jul 2, 2010 at 3:14 PM, Brian West wrote: > We don't have an ooh323 driver in FreeSWITCH. > > /b From sos at sokhapkin.dyndns.org Fri Jul 2 13:28:36 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 2 Jul 2010 16:28:36 -0400 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: Message-ID: <201007021628.36733.sos@sokhapkin.dyndns.org> Use SIP :-) On Friday 02 July 2010, Saeed Ahmed wrote: > Dear FSers, > > which one is better to use when there are higher number of concurrent > calls? *ooH323 *or *mod_h323*. > > Thanks > From saeedahmad1981 at gmail.com Fri Jul 2 13:45:35 2010 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Fri, 2 Jul 2010 22:45:35 +0200 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: References: Message-ID: Ok i'll test mod_h323 first. Will submit my results :) On Fri, Jul 2, 2010 at 10:15 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > The one you tell us about after you test them both =D > I honestly don't know. > > > On Fri, Jul 2, 2010 at 3:07 PM, Saeed Ahmed wrote: > >> Dear FSers, >> >> which one is better to use when there are higher number of concurrent >> calls? *ooH323 *or *mod_h323*. >> >> Thanks >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100702/5a6e0a18/attachment-0001.html From gmaruzz at celliax.org Fri Jul 2 15:56:56 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 3 Jul 2010 00:56:56 +0200 Subject: [Freeswitch-users] Upgraded from 1.0.4 pre8 to the latest Git tree. Skype does not work anymore. In-Reply-To: <4C2DF4F1.3070308@gmail.com> References: <4C2DF4F1.3070308@gmail.com> Message-ID: On Fri, Jul 2, 2010 at 4:17 PM, Svetik wrote: > I did not use my original configuration file, I have migrated changes into > new configuration file. That is how it looks like: Thanks a lot for the infos, Svetik There has been other reports on a possible malfunctioning of mod_skypopen with recent FS code. I'm traveling right now, but I'll try to sort it out beginning next week. Thanks again for reporting, and please open a Jira adding all the information you can. -giovanni > > > ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? > ? > ? > ? ? > ? ? ? ? > ? ? ? ? > ? ? > ? > ? > > > I have tried to use old configuration file from skypiax module, but it gives me the same problem. > > Any thoughts? > Can I try something that helps to diagnose this problem? > > > Igor > > >>>On Tue, Jun 29, 2010 at 5:10 PM, Svetik> ?wrote: >>>/ ?Hi, > />>/ > />>/ ?On weekend I have upgraded to the latest Git tree from 1.0.4 pre8 which > />>/ ?I was running for a long time, year may be. Everything went smooth, > />>/ ?except Skype does not work anymore. Basically I followed Download& > / > >>Have you updated the configuration file? Maybe the configuration file >>format has changed from the one you are using... > >>-giovanni > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From steveu at coppice.org Fri Jul 2 19:36:57 2010 From: steveu at coppice.org (Steve Underwood) Date: Sat, 03 Jul 2010 10:36:57 +0800 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> Message-ID: <4C2EA249.4070704@coppice.org> On 07/03/2010 04:28 AM, Brian West wrote: > DOH... btw ooh323 is now license compatible with FreeSWITCH if anyone wants to write one. > > /b > Where did you see that? The current ooh323c tarball still says GPL 2. Steve From a.afzali2003 at gmail.com Sat Jul 3 01:59:55 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sat, 3 Jul 2010 12:29:55 +0330 Subject: [Freeswitch-users] Using transfer_after_bridge To Handle Post Agent Visit Services In-Reply-To: References: Message-ID: Hi Anthony, Unfortunately my log shows the DNIS is undefined / empty. BEST -- afshin my console log : http://pastebin.freeswitch.org/13345 my public dialplan : http://pastebin.freeswitch.org/13346 On 7/2/10, Anthony Minessale wrote: > check ${DNIS} > > On Fri, Jul 2, 2010 at 11:17 AM, afshin afzali > wrote: > >> Hi FreeSWITCH, >> >> I'm using the transfer_after_bridge variable to capture calls which have >> been serviced by my agents (via a single FIFO) to do post services (such >> as >> say goodbye) successfully. >> >> > data="transfer_after_bridge=post_agent_visit:XML:public"/> >> >> Mean while I've enabled transfer feature ( extension dx ) on my agents >> legs >> to be enable to transfer their callers to desire extensions. Here is my >> problem. When agent transfer his caller to an extension as I expected, >> caller routes to the specified extension ( post_agent_visit ) but I don't >> know how I could retrieve that extension which agent want to transfer to. >> >> BEST >> -- afshin >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > From sameer2k3t at gmail.com Sat Jul 3 03:29:42 2010 From: sameer2k3t at gmail.com (Sameer Khan) Date: Sat, 3 Jul 2010 15:29:42 +0500 Subject: [Freeswitch-users] Skype calls Message-ID: Hello everyone, When i call skype user by my skype interface from my softphone It works fine but there is only a little issue which i cannot find why it is. when I end call from soft phone skype doesn't hangs up and remains busy until i cancelled the call from remote skype id. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100703/baf38813/attachment.html From john_re at fastmail.us Sat Jul 3 03:53:15 2010 From: john_re at fastmail.us (giovanni_re) Date: Sat, 03 Jul 2010 03:53:15 -0700 Subject: [Freeswitch-users] Join July Global via VOIP Free SW HW Culture Mtgs - BerkeleyTIP Message-ID: <1278154395.20536.1383107391@webmail.messagingengine.com> Watch some videos. Mark your calendar. Invite your friends. Join in on IRC or Voice. Join the mailing list, say "Hi. :)" ===== 1) 2010.7 Videos: How to write VOIP client in less then 2 minutes, rpdammu Schizophrenic Firewalls - virtualized net stack OpenBSD, Claudio Jeker Motorola Droid Metro PCS Apps, makeitcricket.com Open Wonderland virtual worlds platform, Nicole Yankelovich, iED How to Succeed in Mobile, Girl Geek Dinner, Kris Corzine Using KDE Marble to research your next vacation, Justin Kirby Meet Google Founder Larry Page, Google Faculty Summit 2009 Building the Python Community, Steve Holden, PyCon 2010 How Python, TurboGears, and MongoDB are Transforming SourceForge.net, Rick Copeland, PyCon Introducing Numpy Arrays, unpingco http://sites.google.com/site/berkeleytip/talk-videos/2010-7-videos == July Meetings - Mark your calendar: 3 Sat 12N-3P PST = 3-6P EST = 19-22 UTC 12 Mon 5 -6P PST = 8-9P EST = 0- 1 UTC Tues 13 18 Sun 12N-3P PST = 3-6P EST = 19-22 UTC 27 Tue 5 -6P PST = 8-9P EST = 0- 1 UTC Wed 28 ===== You're invited to join in with the friendly people at the BerkeleyTIP global meeting - newbie to Ph.D. - everyone is invited. Get a headset & join using VOIP online, or come to Berkeley. 1st step: Join the mailing list: http://groups.google.com/group/BerkTIPGlobal Watch the videos. Discuss them on VOIP. 10 great videos/talks this month. Join with us at the Golden Bear Cafe, Upper Sproul Plaza, UCB at the University of California at Berkeley, or join from your home via VOIP, or send this email locally, create a local meeting, & join via VOIP: Tip: a wifi cafe is a great place to meet. :) PLEASE VIEW THE BTIP WEBSITE & MAILING LIST FOR LATEST DETIALS. http://sites.google.com/site/berkeleytip BerkeleyTIP - Educational, Productive, Social For Learning about, Sharing, & Producing, All Free SW HW & Culture. TIP == Talks, Installfest, Project & Programming Party ===== CONTENTS: 1) 2010 JULY VIDEOS; 2) 2010 JULY MEETING DAYS, TIMES, LOCATIONS; 3) LOCAL MEETING AT U. C. Berkeley; 4) HOT TOPICS; 5) PLEASE RSVP PROBABILISTICALLY, THANKS :) ; 6) INSTALLFEST; 7) ARRIVING FIRST AT THE MEETING: MAKE A "BerkeleyTIP" SIGN; 8) IRC: #berkeleytip on irc.freenode.net; 9) VOIP FOR GLOBAL MEETING; 10) VOLUNTEERING, TO DOs; 11) MAILING LISTS: BerkeleyTIP-Global, LocalBerkeley, Announce; 12) ANYTHING I FORGOT TO MENTION?; 13) FOR FORWARDING ======================================================================= ===== 1) - See videos list at top of this email. Thanks to all the speakers, organizations, & videographers. :) [Please alert the speakers that their talks are scheduled for BTIP (if you are with the group that recorded their talk), because I may not have time to do that. Thanks. :) ] Download & watch these talks before the BTIP meetings. Discuss at the meeting. Email the mailing list, tell us what videos you'll watch & want to discuss. Know any other video sources? - please email me. _Your_ group should video record & post online your meeting's talks! ===== 2) 2010 JULY MEETING DAYS, TIMES, LOCATIONS http://sites.google.com/site/berkeleytip/schedule http://sites.google.com/site/berkeleytip/directions In person meetings on 1st Saturday & 3rd Sunday, every month. July 3 & 18, 12N-3P USA-Pacific time, Saturday, Sunday July 3 = Golden Bear Cafe, Upper Sproul Plaza, UCB July 18 = Free Speech Cafe, Moffitt Library, UCB Online only meeting using VOIP - 9 days after weekend meetings: July 12 & 27, 5-6P USA-Pacific time, Monday, Tuesday Mark your calendars. ===== 3) LOCAL MEETING AT U. C. BERKELEY http://sites.google.com/site/berkeleytip/directions RSVP please. See below. It greatly helps my planning. But, _do_ come if you forgot to RSVP. ALWAYS BE SURE TO CHECK THE BTIP WEBSITE _&_ MAILING LIST FOR THE LATEST LAST MINUTE DETAILS & CHANGES, BEFORE COMING TO THE MEETING! :) DO BRING A VOIP HEADSET, available for $10-30 at most electronics retail stores, & a laptop computer, so you are able to communicate with the global BTIP community via VOIP. It is highly recommended that you have a voip headset, & not rely on a laptop's built in microphone & speakers, because the headphones keep the noise level down. Bringing a headset is not required, but is a great part of the being able to communicate with the global community. :) Clothing: Typically 55-80 degrees F. Weather: http://www.wunderground.com/auto/sfgate/CA/Berkeley.html Other location local meeting possibilities: http://sites.google.com/site/berkeleytip/local-meetings Create a local meeting in your town. Invite your friends. :) ===== 4) HOT TOPICS Android phones - Besting iPhone? worthwhile? How knowable is the hw? iPad, iPhone4 & iPod- rooting & running GNU(Linux) Skype for group video conferencing? Open Wonderland virtual worlds platform ===== 5) PLEASE RSVP PROBABILISTICALLY, THANKS :) If you think there is a >70% chance ("likely") you'll come to the in person meeting in Berkeley, please RSVP to me. Thanks. It helps my planning. Please _do_ come even if you haven't RSVP'd, it's not required. Better yet, join the BerkeleyTIP-Global mailing list, send the RSVP there, & tell us what things you're interested in, or what videos you'll try to watch - so we can know what videos are popular, & we might watch them too. :) http://groups.google.com/group/BerkTIPGlobal ===== 6) INSTALLFEST Get help installing & using Free Software, Hardware & Culture. Laptops only, typically. There isn't easy access for physically bringing desktop boxes here. RSVP _HIGHLY RECOMMENDED_ if you want installfest help. Please RSVP to me, Giovanni, at the from address for this announcement, or better, join & send email to the BTIP-Global mailing list telling us what you'd like help with. This way we can be better prepared to help you, & you might get valuable advice from the mailing list members. If you are new to using free software, an excellent system would be the KUbuntu GNU(Linux) software. It is very comprehensive, fairly easy to use (similar to Windows or Mac), & suitable for personal, home, university, or business use. We are also glad to try to help people with software who join via VOIP. Please email the mailing list with requests that you want help with, so we can try to be prepared better to help you. Installfest volunteers/helpers always welcome, in person, or via VOIP. :) ===== 7) ARRIVING FIRST AT THE MEETING: MAKE A "BerkeleyTIP" SIGN If you get to the meeting & don't see a "BerkeleyTIP" sign up yet, please: 1) Make a BTIP sign on an 8x11 paper & put it at your table, 2) Email the mailing list, or join on IRC, & let us know you are there. Ask someone if you could use their computer for a minute to look something up, or send an email. People are usually very friendly & willing to help. We can also email you a temporary guest AirBears account login. We will have wifi guest accounts available for BTIP attendees. Be sure you have wifi capable equipment. Be Prepared: Bring a multi-outlet extension power cord. ===== 8) IRC: #berkeleytip on irc.freenode.net For help with anything, especially how to get VOIP working, & text communication. ===== 9) VOIP FOR GLOBAL MEETING Speak & listen to everyone globally using VOIP. Get a headset! See some VOIP instructions here: http://sites.google.com/site/berkeleytip/voice-voip-conferencing ===== 10) VOLUNTEERING, TO DOs Enjoy doing or learning something(s)? Help out BTIP in that area. Website development, mailing list management, video locating, VOIP server (FreeSwitch, Asterisk) or client (Ekiga, SFLPhone,...), creating a local meeting. Join the mailing list & let us know. Your offers of free help are always welcome here. :) ===== 11) MAILING LISTS: BerkeleyTIP-Global, LocalBerkeley, Announce Everyone should join the BerkeleyTIP-Global list: http://groups.google.com/group/BerkTIPGlobal Say "hi", tell us your interests, & what videos you'll like to watch. Info on all lists here: http://sites.google.com/site/berkeleytip/mailing-lists ===== 12) ANYTHING I FORGOT TO MENTION? Please join & email the BerkeleyTIP-Global mailing list. ===== 13) FOR FORWARDING You are invited to forward this message anywhere it would be appreciated. Better yet, use it to create a local meeting. Invite & get together with your friends locally, & join in with us all globally. :) Looking forward to meeting with you in person, or online. :) Giovanni == Join in the Global weekly meetings, via VOIP, about all Free SW HW & Culture http://sites.google.com/site/berkeleytip/ From nagalenoj at gmail.com Sat Jul 3 06:22:14 2010 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Sat, 3 Jul 2010 18:52:14 +0530 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: Here is the log got from the latest GIT source. http://pastebin.freeswitch.org/13347 On Fri, Jul 2, 2010 at 10:15 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > update and reproduce that same log with latest GIT the version you are > using has an issue. > > > On Fri, Jul 2, 2010 at 12:25 AM, Nagalenoj H. wrote: > >> I've pasted the console log here, >> >> http://pastebin.freeswitch.org/13333 >> >> >> On Wed, Jun 30, 2010 at 11:15 PM, Michael Collins wrote: >> >>> Can you supply a console log of these calls? >>> -MC >>> >>> >>> On Wed, Jun 30, 2010 at 7:37 AM, Nagalenoj H. wrote: >>> >>>> Dear Anthony, >>>> I've tried using the group_confirm_cancel_timeout as per the >>>> discussion we had in IRC. You wanted to used it as part of dial string and >>>> not as a channel variable. >>>> But, It doesn't work for me. >>>> >>>> Here is how I've given the commands and the script I've executed. Even >>>> when I give group_confirm_cancel_timeout, the callee's leg is getting >>>> disconnected after legtimeout. >>>> >>>> >>>> connect >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name:answer >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: set >>>> execute-app-arg: group_confirm_key=exec >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: set >>>> execute-app-arg: group_confirm_file=perl /root/bridge.pl >>>> >>>> sendmsg >>>> call-command: execute >>>> execute-app-name: bridge >>>> execute-app-arg: >>>> {group_confirm_cancel_timeout=1}[leg_timeout=10]user/1005 >>>> >>>> >>>> >>>> bridge.pl: >>>> #!/usr/bin/perl >>>> use freeswitch; >>>> >>>> our $session; >>>> freeswitch::consoleLog("info","Goint to get the digits"); >>>> # To simulate the scenario I used sleep here. >>>> sleep(30); >>>> 1; >>>> >>>> Kindly tell me whats wrong in the above. >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> Regards, >> Nagalenoj H. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100703/de8630a8/attachment-0001.html From msc at freeswitch.org Sat Jul 3 10:32:27 2010 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 3 Jul 2010 10:32:27 -0700 Subject: [Freeswitch-users] Using transfer_after_bridge To Handle Post Agent Visit Services In-Reply-To: References: Message-ID: <075A3ED6-EB04-4733-9CED-B79070C1F54B@freeswitch.org> What about ${rdnis} ? -MC Sent from my iPhone On Jul 3, 2010, at 1:59 AM, afshin afzali wrote: > Hi Anthony, > > Unfortunately my log shows the DNIS is undefined / empty. > > BEST > -- afshin > > my console log : > http://pastebin.freeswitch.org/13345 > > my public dialplan : > http://pastebin.freeswitch.org/13346 > > On 7/2/10, Anthony Minessale wrote: >> check ${DNIS} >> >> On Fri, Jul 2, 2010 at 11:17 AM, afshin afzali >> wrote: >> >>> Hi FreeSWITCH, >>> >>> I'm using the transfer_after_bridge variable to capture calls >>> which have >>> been serviced by my agents (via a single FIFO) to do post services >>> (such >>> as >>> say goodbye) successfully. >>> >>> >> data="transfer_after_bridge=post_agent_visit:XML:public"/> >>> >>> Mean while I've enabled transfer feature ( extension dx ) on my >>> agents >>> legs >>> to be enable to transfer their callers to desire extensions. Here >>> is my >>> problem. When agent transfer his caller to an extension as I >>> expected, >>> caller routes to the specified extension ( post_agent_visit ) but >>> I don't >>> know how I could retrieve that extension which agent want to >>> transfer to. >>> >>> BEST >>> -- afshin >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at hotmail.com > %3Aanthony_minessale at hotmail.com> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com> %3Aanthony.minessale at gmail.com> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org > %3A888 at conference.freeswitch.org> >> googletalk:conf+888 at conference.freeswitch.org> %2B888 at conference.freeswitch.org> >> pstn:+19193869900 >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From a.afzali2003 at gmail.com Sat Jul 3 10:48:53 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sat, 3 Jul 2010 22:18:53 +0430 Subject: [Freeswitch-users] Using transfer_after_bridge To Handle Post Agent Visit Services In-Reply-To: <075A3ED6-EB04-4733-9CED-B79070C1F54B@freeswitch.org> References: <075A3ED6-EB04-4733-9CED-B79070C1F54B@freeswitch.org> Message-ID: I'll check, BEST -- afshin On Sat, Jul 3, 2010 at 10:02 PM, Michael S Collins wrote: > What about ${rdnis} ? > -MC > > Sent from my iPhone > > On Jul 3, 2010, at 1:59 AM, afshin afzali > wrote: > > > Hi Anthony, > > > > Unfortunately my log shows the DNIS is undefined / empty. > > > > BEST > > -- afshin > > > > my console log : > > http://pastebin.freeswitch.org/13345 > > > > my public dialplan : > > http://pastebin.freeswitch.org/13346 > > > > On 7/2/10, Anthony Minessale wrote: > >> check ${DNIS} > >> > >> On Fri, Jul 2, 2010 at 11:17 AM, afshin afzali > >> wrote: > >> > >>> Hi FreeSWITCH, > >>> > >>> I'm using the transfer_after_bridge variable to capture calls > >>> which have > >>> been serviced by my agents (via a single FIFO) to do post services > >>> (such > >>> as > >>> say goodbye) successfully. > >>> > >>> >>> data="transfer_after_bridge=post_agent_visit:XML:public"/> > >>> > >>> Mean while I've enabled transfer feature ( extension dx ) on my > >>> agents > >>> legs > >>> to be enable to transfer their callers to desire extensions. Here > >>> is my > >>> problem. When agent transfer his caller to an extension as I > >>> expected, > >>> caller routes to the specified extension ( post_agent_visit ) but > >>> I don't > >>> know how I could retrieve that extension which agent want to > >>> transfer to. > >>> > >>> BEST > >>> -- afshin > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at 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 at hotmail.com >> %3Aanthony_minessale at hotmail.com> > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >> %3Aanthony.minessale at gmail.com> > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888 at conference.freeswitch.org >> %3A888 at conference.freeswitch.org> > >> googletalk:conf+888 at conference.freeswitch.org > >> %2B888 at conference.freeswitch.org> > >> pstn:+19193869900 > >> > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100703/d3e1337b/attachment.html From gmaruzz at celliax.org Sat Jul 3 12:39:05 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 3 Jul 2010 21:39:05 +0200 Subject: [Freeswitch-users] Skype calls In-Reply-To: References: Message-ID: On Sat, Jul 3, 2010 at 12:29 PM, Sameer Khan wrote: > When i call skype user by my skype interface from my softphone It works fine > but there is only a little issue which i cannot find why it is. when I end > call from soft phone skype doesn't hangs up and remains busy until i > cancelled the call from remote skype id. Dear Samir, thanks for reporting. Please consult the wiki page on skypopen at wiki.freeswitch.org and follow the guidelines for reporting the problem on Jira. Please put in the jira report all the information you can. -giovanni > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From fraserredmond at gmail.com Sat Jul 3 13:21:14 2010 From: fraserredmond at gmail.com (Fraser Redmond) Date: Sat, 3 Jul 2010 21:21:14 +0100 Subject: [Freeswitch-users] Installation: make is failing at mod_conference.c:3804 Message-ID: I'm trying to do a fairly normal download, configure, make, make install of last night's tarball, but the make is failing at mod_conference: http://pastebin.freeswitch.org/13350 This is on an Amazon EC2 install of Ubuntu Lucid (10.04). I followed the exact same procedures about 3-4 days ago and it installed fine then, so it may be something that has changed this week. Apart from the default configs, I'm enabling mod_silk, mod_flite, mod_shout, mod_xml_curl. Any ideas? Cheers, Fraser -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100703/1413c533/attachment.html From mrene_lists at avgs.ca Sat Jul 3 13:49:59 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sat, 3 Jul 2010 16:49:59 -0400 Subject: [Freeswitch-users] Installation: make is failing at mod_conference.c:3804 In-Reply-To: References: Message-ID: Was fixed in git last tuesday, update. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-07-03, at 4:21 PM, Fraser Redmond wrote: > I'm trying to do a fairly normal download, configure, make, make install of last night's tarball, but the make is failing at mod_conference: > http://pastebin.freeswitch.org/13350 > > This is on an Amazon EC2 install of Ubuntu Lucid (10.04). I followed the exact same procedures about 3-4 days ago and it installed fine then, so it may be something that has changed this week. > > Apart from the default configs, I'm enabling mod_silk, mod_flite, mod_shout, mod_xml_curl. > > Any ideas? > > Cheers, > Fraser > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100703/268cf040/attachment.html From uzairsh at yahoo.com Sat Jul 3 18:35:17 2010 From: uzairsh at yahoo.com (Syed Hussain) Date: Sat, 3 Jul 2010 18:35:17 -0700 (PDT) Subject: [Freeswitch-users] Call-ID format Message-ID: <98731.23464.qm@web30508.mail.mud.yahoo.com> Hi, Is it possible to change the Call-ID: on the outbound leg while transferring the call to external service provider ? example froms snippet below -------------------------------------------- 02:23:10.513759 IP (tos 0x0, ttl 239, id 38921, offset 0, flags [DF], proto: UDP (17), length: 311) 64.122.22.22.sip > 66.89.245.66.5080: [udp sum ok] UDP, length 283 E..7. @....]@.A.Y.._.....#8.SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.89.245.66:5080;rport;branch=z9hG4bK9a91UZ2te93Fc From: "ecsvext00" ;tag=9rev06ra96D8K To: Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 CSeq: 132984959 INVITE Content-Length: 0 ------------------------------------------ >From above This Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 to s to Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 at 66.89.245.66 I have tried please advise , thank you regards S From gerrit308 at gmail.com Sat Jul 3 19:15:14 2010 From: gerrit308 at gmail.com (humbr) Date: Sat, 3 Jul 2010 19:15:14 -0700 (PDT) Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <1278204209127-5252219.post@n2.nabble.com> References: <1278204209127-5252219.post@n2.nabble.com> Message-ID: <1278209714063-5252353.post@n2.nabble.com> Hi The answer you will get from here is to downgrade to an older compiler. This is thankfully easy enough to do with OpenWRT. A workaround, not a fix but it gets you going. Gerrit (who also posted this quite a while ago on the OpenWRT forum :-) mazilo wrote: > > I tried to compile freeswitch-1.0.6 on OpenWRT for a Marvell Kirkwood > (ARM) platform to no avail. The same source codes compiled just fine for a > Broadcomm (MIPS/MIPS32) platform. What fails to compile is the source code > of sofia.c at line 1574 with the following error messages: > > ++++++++++++++++++++++++++ > > -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5252353.html Sent from the freeswitch-users mailing list archive at Nabble.com. From elihay at savion.huji.ac.il Sat Jul 3 21:52:54 2010 From: elihay at savion.huji.ac.il (Eli Hayun) Date: Sun, 04 Jul 2010 07:52:54 +0300 Subject: [Freeswitch-users] Getting DTMF during Fifo Message-ID: <1278219174.2180.10.camel@localhost.localdomain> Hi I created a call senter using fifo. I used a LUA script to detect the caller position in the queue, as explained in the wiki. I want to play a file to tell the caller to enter his pin number (or SSN) and I cannot find any way to do that. I have his UUID but all the samples to get DTMF digits are from the session. How do I get DTMF digits if all I have is UUID? Thanks Eli From a.afzali2003 at gmail.com Sun Jul 4 03:04:50 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sun, 4 Jul 2010 13:34:50 +0330 Subject: [Freeswitch-users] Using transfer_after_bridge To Handle Post Agent Visit Services In-Reply-To: References: <075A3ED6-EB04-4733-9CED-B79070C1F54B@freeswitch.org> Message-ID: It's there :) Thank you Michael, Thank you Anthony. appreciate, -- afshin On 7/3/10, afshin afzali wrote: > I'll check, > > BEST > -- afshin > > On Sat, Jul 3, 2010 at 10:02 PM, Michael S Collins > wrote: > >> What about ${rdnis} ? >> -MC >> >> Sent from my iPhone >> >> On Jul 3, 2010, at 1:59 AM, afshin afzali >> wrote: >> >> > Hi Anthony, >> > >> > Unfortunately my log shows the DNIS is undefined / empty. >> > >> > BEST >> > -- afshin >> > >> > my console log : >> > http://pastebin.freeswitch.org/13345 >> > >> > my public dialplan : >> > http://pastebin.freeswitch.org/13346 >> > >> > On 7/2/10, Anthony Minessale wrote: >> >> check ${DNIS} >> >> >> >> On Fri, Jul 2, 2010 at 11:17 AM, afshin afzali >> >> wrote: >> >> >> >>> Hi FreeSWITCH, >> >>> >> >>> I'm using the transfer_after_bridge variable to capture calls >> >>> which have >> >>> been serviced by my agents (via a single FIFO) to do post services >> >>> (such >> >>> as >> >>> say goodbye) successfully. >> >>> >> >>> > >>> data="transfer_after_bridge=post_agent_visit:XML:public"/> >> >>> >> >>> Mean while I've enabled transfer feature ( extension dx ) on my >> >>> agents >> >>> legs >> >>> to be enable to transfer their callers to desire extensions. Here >> >>> is my >> >>> problem. When agent transfer his caller to an extension as I >> >>> expected, >> >>> caller routes to the specified extension ( post_agent_visit ) but >> >>> I don't >> >>> know how I could retrieve that extension which agent want to >> >>> transfer to. >> >>> >> >>> BEST >> >>> -- afshin >> >>> >> >>> _______________________________________________ >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at 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 at hotmail.com >> >> > >> %3Aanthony_minessale at hotmail.com> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > >> %3Aanthony.minessale at gmail.com> >> >> IRC: irc.freenode.net #freeswitch >> >> >> >> FreeSWITCH Developer Conference >> >> sip:888 at conference.freeswitch.org >> >> > >> %3A888 at conference.freeswitch.org> >> >> googletalk:conf+888 at conference.freeswitch.org >> > >> %2B888 at conference.freeswitch.org> >> >> pstn:+19193869900 >> >> >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From yehavi.bourvine at gmail.com Sun Jul 4 05:08:12 2010 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 4 Jul 2010 15:08:12 +0300 Subject: [Freeswitch-users] Voicemail: can it advance to the next message when no user's input? Message-ID: Hello, Our users are used to a voicemail system in which when you hear a voicemail message and do not touch any DTMF for 2 seconds it proceeds to the next message. Can this be done with Freeswitch as well? Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100704/8adec82f/attachment.html From a.afzali2003 at gmail.com Sun Jul 4 05:17:36 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sun, 4 Jul 2010 15:47:36 +0330 Subject: [Freeswitch-users] Getting DTMF during Fifo In-Reply-To: <1278219174.2180.10.camel@localhost.localdomain> References: <1278219174.2180.10.camel@localhost.localdomain> Message-ID: Hi Eli, You could use bind_meta_app application to detect DTMF digits on a / b leg of a bridge and do what you want. You will find a great usage of this application in Local_Extension extension in default.xml . -- afshin On 7/4/10, Eli Hayun wrote: > Hi > I created a call senter using fifo. I used a LUA script to detect the > caller position in the queue, as explained in the wiki. I want to play a > file to tell the caller to enter his pin number (or SSN) and I cannot > find any way to do that. I have his UUID but all the samples to get DTMF > digits are from the session. How do I get DTMF digits if all I have is > UUID? > > Thanks > > Eli > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From vetali100 at gmail.com Sun Jul 4 05:29:02 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Sun, 4 Jul 2010 15:29:02 +0300 Subject: [Freeswitch-users] Getting WTF1..N messages in the fs log Message-ID: Hi, I am testing FreeSWITCH on Xen Linux VPS sever. FreeSWITCH Version 1.0.head (git-7566575 2010-05-26 10-45-52 +0200) Linux: Ubuntu 9.10. When I view the log using ./fs_cli I can see periodic messages in the log: WTF1 WTF2 WTF3 WTF4 WTF5 WTF1 WTF2 WTF3 WTF4 WTF2 ... Looks like FreeSWITCH works properly, however these messages make me warn. I never seen these messages on the other servers - could you please advise what can I check? Thank you, Vitalie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100704/f27c4ac6/attachment.html From rupa at rupa.com Sun Jul 4 05:48:41 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Sun, 4 Jul 2010 07:48:41 -0500 Subject: [Freeswitch-users] Getting WTF1..N messages in the fs log In-Reply-To: References: Message-ID: You must be using freetdm or openzap? Loks like some debug code left in being dumped to the console. Safe to ignore. libs/freetdm/src/priserver.c 293: printf("WTF %d\n", debug); libs/openzap/src/priserver.c 293: printf("WTF %d\n", debug); On Sun, Jul 4, 2010 at 7:29 AM, Vitalii Colosov wrote: > Hi, > > I am testing FreeSWITCH on Xen Linux VPS sever. > FreeSWITCH Version 1.0.head (git-7566575 2010-05-26 10-45-52 +0200) > Linux: Ubuntu 9.10. > > When I view the log using ./fs_cli I can see periodic messages in the log: > > WTF1 > WTF2 > WTF3 > WTF4 > WTF5 > WTF1 > WTF2 > WTF3 > WTF4 > WTF2 > ... > > > Looks like FreeSWITCH works properly, however these messages make me warn. > > I never seen these messages on the other servers - could you please advise > what can I check? > > Thank you, > Vitalie > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100704/4fb2c17c/attachment.html From vetali100 at gmail.com Sun Jul 4 06:04:12 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Sun, 4 Jul 2010 16:04:12 +0300 Subject: [Freeswitch-users] Getting WTF1..N messages in the fs log In-Reply-To: References: Message-ID: Thanks Rupa, Neither freetdm nor openzap is being used. Maybe it is similar debug message from some other part of the code... Thank you, Vitalie 2010/7/4 Rupa Schomaker > You must be using freetdm or openzap? Loks like some debug code left in > being dumped to the console. Safe to ignore. > > libs/freetdm/src/priserver.c > 293: printf("WTF %d\n", debug); > > libs/openzap/src/priserver.c > 293: printf("WTF %d\n", debug); > > > On Sun, Jul 4, 2010 at 7:29 AM, Vitalii Colosov wrote: > >> Hi, >> >> I am testing FreeSWITCH on Xen Linux VPS sever. >> FreeSWITCH Version 1.0.head (git-7566575 2010-05-26 10-45-52 +0200) >> Linux: Ubuntu 9.10. >> >> When I view the log using ./fs_cli I can see periodic messages in the log: >> >> WTF1 >> WTF2 >> WTF3 >> WTF4 >> WTF5 >> WTF1 >> WTF2 >> WTF3 >> WTF4 >> WTF2 >> ... >> >> >> Looks like FreeSWITCH works properly, however these messages make me >> warn. >> >> I never seen these messages on the other servers - could you please advise >> what can I check? >> >> Thank you, >> Vitalie >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100704/9e471ba5/attachment-0001.html From jan.berger at video24.no Sun Jul 4 07:33:17 2010 From: jan.berger at video24.no (Jan Berger) Date: Sun, 4 Jul 2010 16:33:17 +0200 Subject: [Freeswitch-users] [Freeswitch-dev] Unable to register with Freeswitch In-Reply-To: <838995.92066.qm@web50301.mail.re2.yahoo.com> References: <838995.92066.qm@web50301.mail.re2.yahoo.com> Message-ID: <0CEBAB809EB44A06957637B4EFC03B0C@dell9400> Thirupathy, You will get more responce on these questions in the user list than in the dev list. Firstly can you actually see in the log that FreeSWITCH is registered on SIP? Secondly - what IP address do you connect to from your SIP phone? FreeSWITCH will be silence unless you hit the correct address. Jan _____ From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Thirupathi Sent: 4. juli 2010 16:12 To: freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-dev] Unable to register with Freeswitch Hi, I have installed and running freeswitch on ubuntu. But when i am trying to register IP phone (localhost) with freeswitch (1001 and 1234) it is not happening. Also i don't receive any log in the freeswitch when i am trying to register the IP phone. How to fix this issue? Any one help me thanks in advance. Thanks and Regards, Thirupathi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100704/6c905bb5/attachment.html From gerrit308 at gmail.com Sun Jul 4 07:49:55 2010 From: gerrit308 at gmail.com (humbr) Date: Sun, 4 Jul 2010 07:49:55 -0700 (PDT) Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <1278210527955-5252372.post@n2.nabble.com> References: <1278204209127-5252219.post@n2.nabble.com> <1278209714063-5252353.post@n2.nabble.com> <1278210527955-5252372.post@n2.nabble.com> Message-ID: <1278254995255-5253368.post@n2.nabble.com> Hi I used 4.3.3 (without codesourcery enhancements), all the libs were whatever 'make menuconfig' preset. Some more details are in jira: http://jira.freeswitch.org/browse/MODSOFIA-68 Gerrit mazilo wrote: > > Thank you for your quick response and suggestion. Do you know which > version of GCC should I downgrade to? I configured my OpenWRT SVN trunk > with the versions of binutils-2.18, uClibc-0.9.30.1, and gcc-4.3.3 with > CodeSourcery enhancements to compile freeswitch-1.0.6. > -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5253368.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ali.stgt at gmail.com Fri Jul 2 23:29:51 2010 From: ali.stgt at gmail.com (=?UTF-8?B?RHVybXXFnyBBbGkgw5Z6dMO8cms=?=) Date: Sat, 3 Jul 2010 09:29:51 +0300 Subject: [Freeswitch-users] How to integrate FreeSWITCH in our application Message-ID: Hello, I have downloaded and compiled the sources without any problems. Now we want to use the library to make a call controlled by our own business logic (We dont want to use a dialplan) We didnt find any documents or samples how we can 'manually' establish a call and play a media file. Can you give us some coding tips for this? Thanks in advance!! Ali -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100703/b155ef78/attachment-0001.html From fabiodive at gmail.com Sat Jul 3 07:37:42 2010 From: fabiodive at gmail.com (Fabio Dive) Date: Sat, 03 Jul 2010 16:37:42 +0200 Subject: [Freeswitch-users] radius SSL accounting Message-ID: <4C2F4B36.7080001@gmail.com> hello, I am looking for a way to SSL encrypt accounting messages between Freeswitch and remote Freeradius, actually I can do only clear text accounting with simple shared key auth. many thanks, cheers, Fabio From Nabble at slickdeals.endjunk.com Sat Jul 3 17:43:29 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Sat, 3 Jul 2010 17:43:29 -0700 (PDT) Subject: [Freeswitch-users] Failed to compile sofia.c for ARM Message-ID: <1278204209127-5252219.post@n2.nabble.com> I tried to compile freeswitch-1.0.6 on OpenWRT for a Marvell Kirkwood (ARM) platform to no avail. The same source codes compiled just fine for a Broadcomm (MIPS/MIPS32) platform. What fails to compile is the source code of sofia.c at line 1574 with the following error messages: ++++++++++++++++++++++++++ make[8]: Entering directory `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.2_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' Creating mod_sofia_la-mod_sofia.lo mkdir .libs Compiling mod_sofia.c ... Creating mod_sofia_la-sofia.lo Compiling sofia.c ... sofia.c: In function 'logger': sofia.c:1574: error: used struct type value where scalar is required make[8]: *** [mod_sofia_la-sofia.lo] Error 1 make[8]: Leaving directory `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.2_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' make[7]: *** [all] Error 2 make[7]: Leaving directory `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.2_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' make[6]: *** [mod_sofia-all] Error 1 make[6]: Leaving directory `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.2_eabi/freeswitch-1.0.6/src/mod' make[5]: *** [all-recursive] Error 1 make[5]: Leaving directory `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.2_eabi/freeswitch-1.0.6/src' ++++++++++++++++++++++++++ The pertaining sofia.c source code at line 1574 is shown below: ++++++++++++++++++++++++++ 1569 static void logger(void *logarg, char const *fmt, va_list ap) 1570 { 1571 if (!fmt) 1572 return; 1573 1574 if (ap) { 1575 switch_log_vprintf(SWITCH_CHANNEL_LOG_CLEAN, mod_sofia_glob als.tracelevel, fmt, ap); 1576 } else { 1577 switch_log_printf(SWITCH_CHANNEL_LOG_CLEAN, mod_sofia_globa ls.tracelevel, "%s", fmt); 1578 } 1579 } ++++++++++++++++++++++++++ I am no programmer; however, isn't va_list ap indicates ap is a struct type value while the if(ap) expects ap to be a scalar value? If so, why sofia.c doesn't fail to compile for a MIPS/MIPS32 platform but fails to compile for an ARM platform? That said, how does one go about to fix this so that sofia.c source will compile cleanly on both platforms? Thanks. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5252219.html Sent from the freeswitch-users mailing list archive at Nabble.com. From Nabble at slickdeals.endjunk.com Sat Jul 3 19:28:47 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Sat, 3 Jul 2010 19:28:47 -0700 (PDT) Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <1278209714063-5252353.post@n2.nabble.com> References: <1278204209127-5252219.post@n2.nabble.com> <1278209714063-5252353.post@n2.nabble.com> Message-ID: <1278210527955-5252372.post@n2.nabble.com> Thank you for your quick response and suggestion. Do you know which version of GCC should I downgrade to? I configured my OpenWRT SVN trunk with the versions of binutils-2.18, uClibc-0.9.30.1, and gcc-4.3.3 with CodeSourcery enhancements to compile freeswitch-1.0.6. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5252372.html Sent from the freeswitch-users mailing list archive at Nabble.com. From motosota at gmail.com Sun Jul 4 00:35:25 2010 From: motosota at gmail.com (Mike) Date: Sun, 4 Jul 2010 08:35:25 +0100 Subject: [Freeswitch-users] Problems with git installation Message-ID: Hello, I'm relatively new to Freeswitch - having only really been playing around with it for the last few weeks. Apologies in advance if the following questions are as a result of something I'm doing wrong (more than likely) - I have the book pre-ordered from Amazon - hopefully that will help me a lot ;) A little background context - I already have Freeswitch up and running well on an Amazon EC2 (Fedora) instance. I had to get this up, running and in service very quickly for various reasons, being completely new to Freeswitch I installed it by downloading the file: wget http://files.freeswitch.org/freeswitch-1.0.6.tar.gz And followed the installation instructions. So far that installation has performed exactly as expected. Now I want to install a test and development Freeswitch on another EC2 instance. For this one I thought I'd follow the recommendations to install from the latest git tree. Bear in mind that my main objective here is to have a basic / default / vanilla installation to use for testing / exploring / playing. At this stage I don't want / need anything fancy. The exact commands I ran were: yum install git yum install gcc yum install gcc-c++ yum install autoconf yum install automake yum install libtool yum install ncurses-devel cd /usr/local/src git clone git://git.freeswitch.org/freeswitch.git cd /usr/local/src/freeswitch ./bootstrap.sh ./configure make all install sounds-install moh-install got a coffee... or two... I edited the vars.xml file to have the correct external (elastic IP address) for the RTP and SIP signalling and opened up the right ports in the Security Group (firewall). On starting Freeswitch I see the following errors, but am able to register my two test phones fine. 2010-07-04 02:32:38.211909 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_voicemail.so **/usr/local/freeswitch/mod/mod_voicemail.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.212462 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_valet_parking.so **/usr/local/freeswitch/mod/mod_valet_parking.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.233480 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_spandsp.so **/usr/local/freeswitch/mod/mod_spandsp.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.234111 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_speex.so **/usr/local/freeswitch/mod/mod_speex.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.236968 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_tone_stream.so **/usr/local/freeswitch/mod/mod_tone_stream.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.237129 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/mod/mod_spidermonkey.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.239198 [CONSOLE] mod_local_stream.c:161 Can't open directory: /usr/local/freeswitch/sounds/music/16000 2010-07-04 02:32:38.239337 [CONSOLE] mod_local_stream.c:161 Can't open directory: /usr/local/freeswitch/sounds/music/32000 When I try to dial the Music On Hold from one of the phones I get the following error: 2010-07-04 02:57:24.513520 [ERR] switch_core_file.c:122 Invalid file format [silence_stream] for [2000]! and unsurprisingly given the above errors, trying to dial voicemail gives: 2010-07-04 02:58:55.846720 [ERR] switch_core_session.c:1752 Invalid Application voicemail So guys - please advise - what am I doing wrong? Mike From xyangni at gmail.com Sun Jul 4 06:49:49 2010 From: xyangni at gmail.com (xuyan yang) Date: Sun, 4 Jul 2010 21:49:49 +0800 Subject: [Freeswitch-users] Only Eutelia not working on NAT Message-ID: My freeswitch is working fine behind a NAT with 12voip, sipgate, voipcheap, VoXalot as my router support UPnP. Recently, I have applied some DID numbers from Eutelia.it and tested the account with x-lite. But when configured to FS, it can only make out-going call and fail to receive any form this gateway. In the meanwhile 12voip,sipgate are still working well. I tried many parameters but none of them works unless the router was removed and FS connected directly to public IP. It is quite strange for only a single gateway fail behind NAT, while all others is fine. From gcd at i.ph Sun Jul 4 08:28:05 2010 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 4 Jul 2010 23:28:05 +0800 Subject: [Freeswitch-users] How to integrate FreeSWITCH in our application In-Reply-To: References: Message-ID: did u execute "make samples" after "make install"? this will install the sample configuration w/c you can modify to suit your needs. On Sat, Jul 3, 2010 at 2:29 PM, Durmu? Ali ?zt?rk wrote: > Hello, > > I have downloaded and compiled the sources without any problems. Now we > want to use the library to make a call controlled by our own business logic > (We dont want to use a dialplan) > > We didnt find any documents or samples how we can 'manually' establish a > call and play a media file. > > Can you give us some coding tips for this? > > Thanks in advance!! > Ali > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100704/5c5eda3b/attachment.html From peder at networkoblivion.com Sun Jul 4 08:29:46 2010 From: peder at networkoblivion.com (Peder) Date: Sun, 4 Jul 2010 10:29:46 -0500 Subject: [Freeswitch-users] Problems with git installation In-Reply-To: References: Message-ID: <045d01cb1b8d$bbf7f7c0$33e7e740$@com> You might want to do the last 3 items again and watch for errors. Your make is probably erroring out during the compile phase and so some of the modules are not actually being created. Try: cd /usr/local/freeswitch/mod ls Is mod_voicemail.so or mod_valet_parking.so in there? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike Sent: Sunday, July 04, 2010 2:35 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Problems with git installation Hello, I'm relatively new to Freeswitch - having only really been playing around with it for the last few weeks. Apologies in advance if the following questions are as a result of something I'm doing wrong (more than likely) - I have the book pre-ordered from Amazon - hopefully that will help me a lot ;) A little background context - I already have Freeswitch up and running well on an Amazon EC2 (Fedora) instance. I had to get this up, running and in service very quickly for various reasons, being completely new to Freeswitch I installed it by downloading the file: wget http://files.freeswitch.org/freeswitch-1.0.6.tar.gz And followed the installation instructions. So far that installation has performed exactly as expected. Now I want to install a test and development Freeswitch on another EC2 instance. For this one I thought I'd follow the recommendations to install from the latest git tree. Bear in mind that my main objective here is to have a basic / default / vanilla installation to use for testing / exploring / playing. At this stage I don't want / need anything fancy. The exact commands I ran were: yum install git yum install gcc yum install gcc-c++ yum install autoconf yum install automake yum install libtool yum install ncurses-devel cd /usr/local/src git clone git://git.freeswitch.org/freeswitch.git cd /usr/local/src/freeswitch ./bootstrap.sh ./configure make all install sounds-install moh-install got a coffee... or two... I edited the vars.xml file to have the correct external (elastic IP address) for the RTP and SIP signalling and opened up the right ports in the Security Group (firewall). On starting Freeswitch I see the following errors, but am able to register my two test phones fine. 2010-07-04 02:32:38.211909 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_voicemail.so **/usr/local/freeswitch/mod/mod_voicemail.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.212462 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_valet_parking.so **/usr/local/freeswitch/mod/mod_valet_parking.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.233480 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_spandsp.so **/usr/local/freeswitch/mod/mod_spandsp.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.234111 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_speex.so **/usr/local/freeswitch/mod/mod_speex.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.236968 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_tone_stream.so **/usr/local/freeswitch/mod/mod_tone_stream.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.237129 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/mod/mod_spidermonkey.so: cannot open shared object file: No such file or directory** 2010-07-04 02:32:38.239198 [CONSOLE] mod_local_stream.c:161 Can't open directory: /usr/local/freeswitch/sounds/music/16000 2010-07-04 02:32:38.239337 [CONSOLE] mod_local_stream.c:161 Can't open directory: /usr/local/freeswitch/sounds/music/32000 When I try to dial the Music On Hold from one of the phones I get the following error: 2010-07-04 02:57:24.513520 [ERR] switch_core_file.c:122 Invalid file format [silence_stream] for [2000]! and unsurprisingly given the above errors, trying to dial voicemail gives: 2010-07-04 02:58:55.846720 [ERR] switch_core_session.c:1752 Invalid Application voicemail So guys - please advise - what am I doing wrong? Mike _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From stephen at mymessage.us Sun Jul 4 08:48:53 2010 From: stephen at mymessage.us (Stephen Cattaneo) Date: Sun, 4 Jul 2010 11:48:53 -0400 Subject: [Freeswitch-users] Problems with git installation In-Reply-To: <045d01cb1b8d$bbf7f7c0$33e7e740$@com> References: <045d01cb1b8d$bbf7f7c0$33e7e740$@com> Message-ID: i found it easier to go into those modules directories and compile them individually to look for errors. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print On Sun, Jul 4, 2010 at 11:29 AM, Peder wrote: > You might want to do the last 3 items again and watch for errors. Your > make > is probably erroring out during the compile phase and so some of the > modules > are not actually being created. > > Try: > cd /usr/local/freeswitch/mod > ls > > Is mod_voicemail.so or mod_valet_parking.so in there? > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike > Sent: Sunday, July 04, 2010 2:35 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Problems with git installation > > Hello, > > I'm relatively new to Freeswitch - having only really been playing > around with it for the last few weeks. Apologies in advance if the > following questions are as a result of something I'm doing wrong (more > than likely) - I have the book pre-ordered from Amazon - hopefully > that will help me a lot ;) > > A little background context - I already have Freeswitch up and running > well on an Amazon EC2 (Fedora) instance. I had to get this up, running > and in service very quickly for various reasons, being completely new > to Freeswitch I installed it by downloading the file: > > wget http://files.freeswitch.org/freeswitch-1.0.6.tar.gz > > And followed the installation instructions. So far that installation > has performed exactly as expected. > > Now I want to install a test and development Freeswitch on another EC2 > instance. For this one I thought I'd follow the recommendations to > install from the latest git tree. Bear in mind that my main objective > here is to have a basic / default / vanilla installation to use for > testing / exploring / playing. At this stage I don't want / need > anything fancy. > > The exact commands I ran were: > yum install git > yum install gcc > yum install gcc-c++ > yum install autoconf > yum install automake > yum install libtool > yum install ncurses-devel > cd /usr/local/src > git clone git://git.freeswitch.org/freeswitch.git > cd /usr/local/src/freeswitch > ./bootstrap.sh > ./configure > make all install sounds-install moh-install > got a coffee... or two... > > I edited the vars.xml file to have the correct external (elastic IP > address) for the RTP and SIP signalling and opened up the right ports > in the Security Group (firewall). On starting Freeswitch I see the > following errors, but am able to register my two test phones fine. > > 2010-07-04 02:32:38.211909 [CRIT] switch_loadable_module.c:926 Error > Loading module /usr/local/freeswitch/mod/mod_voicemail.so > **/usr/local/freeswitch/mod/mod_voicemail.so: cannot open shared > object file: No such file or directory** > 2010-07-04 02:32:38.212462 [CRIT] switch_loadable_module.c:926 Error > Loading module /usr/local/freeswitch/mod/mod_valet_parking.so > **/usr/local/freeswitch/mod/mod_valet_parking.so: cannot open shared > object file: No such file or directory** > 2010-07-04 02:32:38.233480 [CRIT] switch_loadable_module.c:926 Error > Loading module /usr/local/freeswitch/mod/mod_spandsp.so > **/usr/local/freeswitch/mod/mod_spandsp.so: cannot open shared object > file: No such file or directory** > 2010-07-04 02:32:38.234111 [CRIT] switch_loadable_module.c:926 Error > Loading module /usr/local/freeswitch/mod/mod_speex.so > **/usr/local/freeswitch/mod/mod_speex.so: cannot open shared object > file: No such file or directory** > 2010-07-04 02:32:38.236968 [CRIT] switch_loadable_module.c:926 Error > Loading module /usr/local/freeswitch/mod/mod_tone_stream.so > **/usr/local/freeswitch/mod/mod_tone_stream.so: cannot open shared > object file: No such file or directory** > 2010-07-04 02:32:38.237129 [CRIT] switch_loadable_module.c:926 Error > Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so > **/usr/local/freeswitch/mod/mod_spidermonkey.so: cannot open shared > object file: No such file or directory** > 2010-07-04 02:32:38.239198 [CONSOLE] mod_local_stream.c:161 Can't open > directory: /usr/local/freeswitch/sounds/music/16000 > 2010-07-04 02:32:38.239337 [CONSOLE] mod_local_stream.c:161 Can't open > directory: /usr/local/freeswitch/sounds/music/32000 > > When I try to dial the Music On Hold from one of the phones I get the > following error: > 2010-07-04 02:57:24.513520 [ERR] switch_core_file.c:122 Invalid file > format [silence_stream] for [2000]! > and unsurprisingly given the above errors, trying to dial voicemail gives: > 2010-07-04 02:58:55.846720 [ERR] switch_core_session.c:1752 Invalid > Application voicemail > > So guys - please advise - what am I doing wrong? > > Mike > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100704/1a3c55c0/attachment.html From steveayre at gmail.com Sun Jul 4 12:05:27 2010 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 4 Jul 2010 20:05:27 +0100 Subject: [Freeswitch-users] radius SSL accounting In-Reply-To: <4C2F4B36.7080001@gmail.com> References: <4C2F4B36.7080001@gmail.com> Message-ID: Radius only works unencrypted... A host-host ipsec tunnel might work though? -Steve On 3 July 2010 15:37, Fabio Dive wrote: > hello, > > I am looking for a way to SSL encrypt accounting messages between > Freeswitch and remote Freeradius, > actually I can do only clear text accounting with simple shared key auth. > > many thanks, > cheers, > > Fabio > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100704/2d2db67e/attachment-0001.html From jmesquita at freeswitch.org Sun Jul 4 12:37:14 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 4 Jul 2010 16:37:14 -0300 Subject: [Freeswitch-users] Voicemail: can it advance to the next message when no user's input? In-Reply-To: References: Message-ID: Yes, but requires a patch on mod_voicemail code. Maybe suitable for a bounty? Regards, Jo?o Mesquita On Sun, Jul 4, 2010 at 9:08 AM, Yehavi Bourvine wrote: > Hello, > > Our users are used to a voicemail system in which when you hear a > voicemail message and do not touch any DTMF for 2 seconds it proceeds to the > next message. Can this be done with Freeswitch as well? > > Thanks, __Yehavi: > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100704/f6a08ed8/attachment.html From msc at freeswitch.org Sun Jul 4 12:46:58 2010 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 4 Jul 2010 12:46:58 -0700 Subject: [Freeswitch-users] How to integrate FreeSWITCH in our application In-Reply-To: References: Message-ID: <7DA97194-8283-43D4-92FC-F90ABBB666BD@freeswitch.org> There are several ways to accomplish this. I would start by reading up on the "originate" command, which is used from the CLI or event socket like this: originate sofia/gateway/mygw/12223334444 &bridge(user/1001) Of course, it's a lot more powerful than just this example. You can originate calls from the event socket and control what happens, including playing files and getting digits. There are simple ESL examples in libs/esl. Do a wiki search on esl or event socket library and you'll see more. My last piece of advice is this: take a deep breath! This telephony stuff can be overwhelming at first. Keep plugging away at it. Also, join the IRC channel and listen to the weekly conf calls on Wednesdays. Welcome to FreeSWITCH! -MC Sent from my iPhone On Jul 2, 2010, at 11:29 PM, Durmu? Ali ?zt?rk wrote: > Hello, > > I have downloaded and compiled the sources without any problems. Now > we want to use the library to make a call controlled by our own > business logic (We dont want to use a dialplan) > > We didnt find any documents or samples how we can 'manually' > establish a call and play a media file. > > Can you give us some coding tips for this? > > Thanks in advance!! > Ali > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From Nabble at slickdeals.endjunk.com Sun Jul 4 14:06:16 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Sun, 4 Jul 2010 14:06:16 -0700 (PDT) Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <1278254995255-5253368.post@n2.nabble.com> References: <1278204209127-5252219.post@n2.nabble.com> <1278209714063-5252353.post@n2.nabble.com> <1278210527955-5252372.post@n2.nabble.com> <1278254995255-5253368.post@n2.nabble.com> Message-ID: <1278277576027-5254201.post@n2.nabble.com> humbr wrote: > I used 4.3.3 (without codesourcery enhancements), all the libs were > whatever 'make menuconfig' preset. Thanks for the suggestion and that makes a lot of difference. Now, I got FreeSwitch-1.0.6 built and installed on my Seagate DockStar. It is time to start playing with it. BTW, if anyone is interested in get a Seagate DockStar, currently the only store Buy is selling a Seagate DockStar for $24.99 + Free S/H (to US). ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5254201.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mashudi72 at gmail.com Sun Jul 4 18:14:16 2010 From: mashudi72 at gmail.com (mashudi72 -) Date: Mon, 5 Jul 2010 08:14:16 +0700 Subject: [Freeswitch-users] Call-ID format In-Reply-To: <98731.23464.qm@web30508.mail.mud.yahoo.com> References: <98731.23464.qm@web30508.mail.mud.yahoo.com> Message-ID: Hi Syed Hussain, You could define it on effective_caller_id_number 2010/7/4 Syed Hussain > Hi, > > Is it possible to change the Call-ID: on the outbound leg while > transferring the call to external service provider ? > > example froms snippet below > -------------------------------------------- > 02:23:10.513759 IP (tos 0x0, ttl 239, id 38921, offset 0, flags [DF], > proto: UDP (17), length: 311) 64.122.22.22.sip > 66.89.245.66.5080: [udp sum > ok] UDP, length 283 > E..7. @....]@.A.Y.._.....#8.SIP/2.0 100 Trying > Via: SIP/2.0/UDP 66.89.245.66:5080;rport;branch=z9hG4bK9a91UZ2te93Fc > From: "ecsvext00" > >;tag=9rev06ra96D8K > To: > > Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 > CSeq: 132984959 INVITE > Content-Length: 0 > > ------------------------------------------ > >From above > This > > Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 to s > to > Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 at 66.89.245.66 > > I have tried > > > please advise , thank you > > regards > > S > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/8e12df63/attachment.html From sos at sokhapkin.dyndns.org Sun Jul 4 18:33:21 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 4 Jul 2010 21:33:21 -0400 Subject: [Freeswitch-users] Call-ID format In-Reply-To: References: <98731.23464.qm@web30508.mail.mud.yahoo.com> Message-ID: <201007042133.21253.sos@sokhapkin.dyndns.org> The question was about SIP call id, but not caller id number. I don't think FS allows to specify custom SIP call id. However, I wonder why the original poster needs it... On Sunday 04 July 2010, mashudi72 - wrote: > Hi Syed Hussain, > > You could define it on effective_caller_id_number > > 2010/7/4 Syed Hussain > > > Hi, > > > > Is it possible to change the Call-ID: on the outbound leg while > > transferring the call to external service provider ? > > > > example froms snippet below > > -------------------------------------------- > > 02:23:10.513759 IP (tos 0x0, ttl 239, id 38921, offset 0, flags [DF], > > proto: UDP (17), length: 311) 64.122.22.22.sip > 66.89.245.66.5080: [udp > > sum ok] UDP, length 283 > > E..7. @....]@.A.Y.._.....#8.SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 66.89.245.66:5080;rport;branch=z9hG4bK9a91UZ2te93Fc > > From: "ecsvext00" > > > > >;tag=9rev06ra96D8K > > > > To: > > > Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 > > CSeq: 132984959 INVITE > > Content-Length: 0 > > > > ------------------------------------------ > > > > >From above > > > > This > > > > Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 to s > > to > > Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 at 66.89.245.66 > > > > I have tried > > > > > > please advise , thank you > > > > regards > > > > S > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > From jmesquita at freeswitch.org Sun Jul 4 18:37:56 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 4 Jul 2010 22:37:56 -0300 Subject: [Freeswitch-users] Call-ID format In-Reply-To: <201007042133.21253.sos@sokhapkin.dyndns.org> References: <98731.23464.qm@web30508.mail.mud.yahoo.com> <201007042133.21253.sos@sokhapkin.dyndns.org> Message-ID: As far as I know, you can only change the Call-Id to the channel's Unique-ID. There's a profile config for that, I suppose. JM On Sun, Jul 4, 2010 at 10:33 PM, Sergey Okhapkin wrote: > The question was about SIP call id, but not caller id number. I don't think > FS > allows to specify custom SIP call id. However, I wonder why the original > poster needs it... > > On Sunday 04 July 2010, mashudi72 - wrote: > > Hi Syed Hussain, > > > > You could define it on effective_caller_id_number > > > > 2010/7/4 Syed Hussain > > > > > Hi, > > > > > > Is it possible to change the Call-ID: on the outbound leg while > > > transferring the call to external service provider ? > > > > > > example froms snippet below > > > -------------------------------------------- > > > 02:23:10.513759 IP (tos 0x0, ttl 239, id 38921, offset 0, flags [DF], > > > proto: UDP (17), length: 311) 64.122.22.22.sip > 66.89.245.66.5080: > [udp > > > sum ok] UDP, length 283 > > > E..7. @....]@.A.Y.._.....#8.SIP/2.0 100 Trying > > > Via: SIP/2.0/UDP 66.89.245.66:5080;rport;branch=z9hG4bK9a91UZ2te93Fc > > > From: "ecsvext00" < > sip%3A6003 at 89.200.141.95 > > > > > > > >;tag=9rev06ra96D8K > > > > > > To: < > sip%3A0044202929208 at 64.122.22.22 >> > > > Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 > > > CSeq: 132984959 INVITE > > > Content-Length: 0 > > > > > > ------------------------------------------ > > > > > > >From above > > > > > > This > > > > > > Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 to s > > > to > > > Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 at 66.89.245.66 > > > > > > I have tried > > > > > > > > > please advise , thank you > > > > > > regards > > > > > > S > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100704/a3d7445b/attachment.html From sos at sokhapkin.dyndns.org Sun Jul 4 18:47:05 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 4 Jul 2010 21:47:05 -0400 Subject: [Freeswitch-users] Call-ID format In-Reply-To: References: <98731.23464.qm@web30508.mail.mud.yahoo.com> <201007042133.21253.sos@sokhapkin.dyndns.org> Message-ID: <201007042147.05294.sos@sokhapkin.dyndns.org> Hmm, both b-leg Call-id and channel's Unique-ID are the same - channel's uuid. Correct me if I'm wrong. On Sunday 04 July 2010, Jo?o Mesquita wrote: > As far as I know, you can only change the Call-Id to the channel's > Unique-ID. There's a profile config for that, I suppose. > > JM > > > On Sun, Jul 4, 2010 at 10:33 PM, Sergey Okhapkin > > wrote: > > The question was about SIP call id, but not caller id number. I don't > > think FS > > allows to specify custom SIP call id. However, I wonder why the original > > poster needs it... > > > > On Sunday 04 July 2010, mashudi72 - wrote: > > > Hi Syed Hussain, > > > > > > You could define it on effective_caller_id_number > > > > > > 2010/7/4 Syed Hussain > > > > > > > Hi, > > > > > > > > Is it possible to change the Call-ID: on the outbound leg while > > > > transferring the call to external service provider ? > > > > > > > > example froms snippet below > > > > -------------------------------------------- > > > > 02:23:10.513759 IP (tos 0x0, ttl 239, id 38921, offset 0, flags [DF], > > > > proto: UDP (17), length: 311) 64.122.22.22.sip > 66.89.245.66.5080: > > > > [udp > > > > > > sum ok] UDP, length 283 > > > > E..7. @....]@.A.Y.._.....#8.SIP/2.0 100 Trying > > > > Via: SIP/2.0/UDP 66.89.245.66:5080;rport;branch=z9hG4bK9a91UZ2te93Fc > > > > From: "ecsvext00" > > > > < > > > > sip%3A6003 at 89.200.141.95 > > > > > > > >;tag=9rev06ra96D8K > > > > > > > > To: > > > < > > > > sip%3A0044202929208 at 64.122.22.22 >> > > > > > > Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 > > > > CSeq: 132984959 INVITE > > > > Content-Length: 0 > > > > > > > > ------------------------------------------ > > > > > > > > >From above > > > > > > > > This > > > > > > > > Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 to s > > > > to > > > > Call-ID: 8c9889b0-01ad-122e-1aa9-00163e000001 at 66.89.245.66 > > > > > > > > I have tried > > > > > > > > > > > > please advise , thank you > > > > > > > > regards > > > > > > > > S > > > > > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > From jcasale at activenetwerx.com Sun Jul 4 23:13:28 2010 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 5 Jul 2010 06:13:28 +0000 Subject: [Freeswitch-users] FreeTDM and Digium analog cards Message-ID: Does FreeTDM support TDM cards, or just the digital variants? Reading through Sangoma's docs on FreeTDM, it's not clear? Thanks! From tony.tin at noahmedia.com.hk Sun Jul 4 23:48:10 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Mon, 5 Jul 2010 14:48:10 +0800 Subject: [Freeswitch-users] how to check whether openzap channel is idle Message-ID: Hi, Could anyone please tell me that how to check whether a openzap channel is in idle, so I can use it to originate a call. Thanks. Regards, Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/bb21a9fd/attachment.html From irmatov at gmail.com Mon Jul 5 01:00:35 2010 From: irmatov at gmail.com (Timur Irmatov) Date: Mon, 5 Jul 2010 13:00:35 +0500 Subject: [Freeswitch-users] mod_erlang_event problem: REPRODUCIBLE, fix needed In-Reply-To: References: <20100701190152.GG17555@hijacked.us> Message-ID: On Fri, Jul 2, 2010 at 9:56 PM, Anthony Minessale wrote: > The problem is that you cannot park channels in the bypass media state. > I did a patch to the code that will probably allow this so you can update > and test it. It works! Thank you very much! -- Timur Irmatov, xmpp:irmatov at jabber.ru From lakindia89 at gmail.com Mon Jul 5 03:55:28 2010 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Mon, 5 Jul 2010 16:25:28 +0530 Subject: [Freeswitch-users] how to check whether openzap channel is idle In-Reply-To: References: Message-ID: Hi, If you say openzap/1/a/, then freeswitch will automatically use an idle channel. On Mon, Jul 5, 2010 at 12:18 PM, Tony Tin wrote: > Hi, > > Could anyone please tell me that how to check whether a openzap channel is > in idle, so I can use it to originate a call. Thanks. > > Regards, > Tony > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/770daa74/attachment.html From luixsansan at hotmail.com Mon Jul 5 05:02:30 2010 From: luixsansan at hotmail.com (luixsansan at hotmail.com) Date: Mon, 5 Jul 2010 14:02:30 +0200 Subject: [Freeswitch-users] voice mail password Message-ID: Hello, I have the following user as 1001.xml I can call to this extension and if it does not answer I can leave a message in its mailbox, I use the default configuration and voicemail is accessed in extension 4000. If I want to listen to the messages the password I have to dial "1001" not "0288", why is this so?, where is 1001 set as the password for the mailbox?, how do I change the XML configuration files to use 0288 or any other number I may choose? Thank you for your help. Luis. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/17ac2e20/attachment.html From anthony.minessale at gmail.com Mon Jul 5 06:37:09 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 5 Jul 2010 08:37:09 -0500 Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <1278204209127-5252219.post@n2.nabble.com> References: <1278204209127-5252219.post@n2.nabble.com> Message-ID: usually va_list is expressed externally as void * Is this the only place with an error? try altering the code so the if and else is replaced by just the contents of the if. ie the switch va_sprintf line On Sat, Jul 3, 2010 at 7:43 PM, mazilo wrote: > > I tried to compile freeswitch-1.0.6 on OpenWRT for a Marvell Kirkwood (ARM) > platform to no avail. The same source codes compiled just fine for a > Broadcomm (MIPS/MIPS32) platform. What fails to compile is the source code > of sofia.c at line 1574 with the following error messages: > > ++++++++++++++++++++++++++ > make[8]: Entering directory > > `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.2_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' > Creating mod_sofia_la-mod_sofia.lo > mkdir .libs > Compiling mod_sofia.c ... > Creating mod_sofia_la-sofia.lo > Compiling sofia.c ... > sofia.c: In function 'logger': > sofia.c:1574: error: used struct type value where scalar is required > make[8]: *** [mod_sofia_la-sofia.lo] Error 1 > make[8]: Leaving directory > > `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.2_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' > make[7]: *** [all] Error 2 > make[7]: Leaving directory > > `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.2_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' > make[6]: *** [mod_sofia-all] Error 1 > make[6]: Leaving directory > > `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.2_eabi/freeswitch-1.0.6/src/mod' > make[5]: *** [all-recursive] Error 1 > make[5]: Leaving directory > > `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.2_eabi/freeswitch-1.0.6/src' > ++++++++++++++++++++++++++ > > The pertaining sofia.c source code at line 1574 is shown below: > > ++++++++++++++++++++++++++ > 1569 static void logger(void *logarg, char const *fmt, va_list ap) > 1570 { > 1571 if (!fmt) > 1572 return; > 1573 > 1574 if (ap) { > 1575 switch_log_vprintf(SWITCH_CHANNEL_LOG_CLEAN, > mod_sofia_glob als.tracelevel, fmt, ap); > 1576 } else { > 1577 switch_log_printf(SWITCH_CHANNEL_LOG_CLEAN, > mod_sofia_globa ls.tracelevel, "%s", fmt); > 1578 } > 1579 } > ++++++++++++++++++++++++++ > > I am no programmer; however, isn't va_list ap indicates ap is a struct type > value while the if(ap) expects ap to be a scalar value? If so, why sofia.c > doesn't fail to compile for a MIPS/MIPS32 platform but fails to compile for > an ARM platform? That said, how does one go about to fix this so that > sofia.c source will compile cleanly on both platforms? > > Thanks. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5252219.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/9983674d/attachment.html From motosota at gmail.com Mon Jul 5 06:41:22 2010 From: motosota at gmail.com (Mike) Date: Mon, 5 Jul 2010 14:41:22 +0100 Subject: [Freeswitch-users] Problems with git installation In-Reply-To: <045d01cb1b8d$bbf7f7c0$33e7e740$@com> References: <045d01cb1b8d$bbf7f7c0$33e7e740$@com> Message-ID: Peder - thanks - you were right. /usr/bin/ld: cannot find -ljpeg was causing errors in the compliation. make[4]: *** [mod_spandsp.la] Error 1 I guess I was working under the lazy premise that the configure process would crash out if something vital was needed (like it did for the other components that I installed). Lesson learned - I'll pay more attention to the output messages in the future. Did a yum install libjpeg-devel and everything was cool after that. Cheers Mike On Sun, Jul 4, 2010 at 4:29 PM, Peder wrote: > You might want to do the last 3 items again and watch for errors. Your > make > is probably erroring out during the compile phase and so some of the > modules > are not actually being created. > > Try: > cd /usr/local/freeswitch/mod > ls > > Is mod_voicemail.so or mod_valet_parking.so in there? > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike > Sent: Sunday, July 04, 2010 2:35 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Problems with git installation > > Hello, > > I'm relatively new to Freeswitch - having only really been playing > around with it for the last few weeks. Apologies in advance if the > following questions are as a result of something I'm doing wrong (more > than likely) - I have the book pre-ordered from Amazon - hopefully > that will help me a lot ;) > > A little background context - I already have Freeswitch up and running > well on an Amazon EC2 (Fedora) instance. I had to get this up, running > and in service very quickly for various reasons, being completely new > to Freeswitch I installed it by downloading the file: > > wget http://files.freeswitch.org/freeswitch-1.0.6.tar.gz > > And followed the installation instructions. So far that installation > has performed exactly as expected. > > Now I want to install a test and development Freeswitch on another EC2 > instance. For this one I thought I'd follow the recommendations to > install from the latest git tree. Bear in mind that my main objective > here is to have a basic / default / vanilla installation to use for > testing / exploring / playing. At this stage I don't want / need > anything fancy. > > The exact commands I ran were: > yum install git > yum install gcc > yum install gcc-c++ > yum install autoconf > yum install automake > yum install libtool > yum install ncurses-devel > cd /usr/local/src > git clone git://git.freeswitch.org/freeswitch.git > cd /usr/local/src/freeswitch > ./bootstrap.sh > ./configure > make all install sounds-install moh-install > got a coffee... or two... > > I edited the vars.xml file to have the correct external (elastic IP > address) for the RTP and SIP signalling and opened up the right ports > in the Security Group (firewall). On starting Freeswitch I see the > following errors, but am able to register my two test phones fine. > > 2010-07-04 02:32:38.211909 [CRIT] switch_loadable_module.c:926 Error > Loading module /usr/local/freeswitch/mod/mod_voicemail.so > **/usr/local/freeswitch/mod/mod_voicemail.so: cannot open shared > object file: No such file or directory** > 2010-07-04 02:32:38.212462 [CRIT] switch_loadable_module.c:926 Error > Loading module /usr/local/freeswitch/mod/mod_valet_parking.so > **/usr/local/freeswitch/mod/mod_valet_parking.so: cannot open shared > object file: No such file or directory** > 2010-07-04 02:32:38.233480 [CRIT] switch_loadable_module.c:926 Error > Loading module /usr/local/freeswitch/mod/mod_spandsp.so > **/usr/local/freeswitch/mod/mod_spandsp.so: cannot open shared object > file: No such file or directory** > 2010-07-04 02:32:38.234111 [CRIT] switch_loadable_module.c:926 Error > Loading module /usr/local/freeswitch/mod/mod_speex.so > **/usr/local/freeswitch/mod/mod_speex.so: cannot open shared object > file: No such file or directory** > 2010-07-04 02:32:38.236968 [CRIT] switch_loadable_module.c:926 Error > Loading module /usr/local/freeswitch/mod/mod_tone_stream.so > **/usr/local/freeswitch/mod/mod_tone_stream.so: cannot open shared > object file: No such file or directory** > 2010-07-04 02:32:38.237129 [CRIT] switch_loadable_module.c:926 Error > Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so > **/usr/local/freeswitch/mod/mod_spidermonkey.so: cannot open shared > object file: No such file or directory** > 2010-07-04 02:32:38.239198 [CONSOLE] mod_local_stream.c:161 Can't open > directory: /usr/local/freeswitch/sounds/music/16000 > 2010-07-04 02:32:38.239337 [CONSOLE] mod_local_stream.c:161 Can't open > directory: /usr/local/freeswitch/sounds/music/32000 > > When I try to dial the Music On Hold from one of the phones I get the > following error: > 2010-07-04 02:57:24.513520 [ERR] switch_core_file.c:122 Invalid file > format [silence_stream] for [2000]! > and unsurprisingly given the above errors, trying to dial voicemail gives: > 2010-07-04 02:58:55.846720 [ERR] switch_core_session.c:1752 Invalid > Application voicemail > > So guys - please advise - what am I doing wrong? > > Mike > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/3af29f9c/attachment-0001.html From yehavi.bourvine at gmail.com Mon Jul 5 07:11:26 2010 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Mon, 5 Jul 2010 17:11:26 +0300 Subject: [Freeswitch-users] Voicemail: can it advance to the next message when no user's input? In-Reply-To: References: Message-ID: We thought of doing it ourselves... How much bounty you think this deserves? Thanks, Yehavi On 04/07/2010 22:42, "Jo?o Mesquita" wrote: Yes, but requires a patch on mod_voicemail code. Maybe suitable for a bounty? Regards, Jo?o Mesquita On Sun, Jul 4, 2010 at 9:08 AM, Yehavi Bourvine wrote: > > > > Hello, > > > > Our users are used to a voicemail system in which when you hear a > voicemail messa... > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/c4b123c6/attachment.html From brian at freeswitch.org Mon Jul 5 07:28:32 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 5 Jul 2010 09:28:32 -0500 Subject: [Freeswitch-users] ooH323 vs mod_h323 In-Reply-To: <4C2EA249.4070704@coppice.org> References: <2BD0DC60-D428-4323-A020-D9AD20E76553@freeswitch.org> <4C2EA249.4070704@coppice.org> Message-ID: <267C8398-6CA3-442E-B973-BE096D728B28@freeswitch.org> It has a FLOSS exception in the GPL license. /b On Jul 2, 2010, at 9:36 PM, Steve Underwood wrote: > On 07/03/2010 04:28 AM, Brian West wrote: >> DOH... btw ooh323 is now license compatible with FreeSWITCH if anyone wants to write one. >> >> /b >> > Where did you see that? The current ooh323c tarball still says GPL 2. > > Steve From brian at freeswitch.org Mon Jul 5 07:31:06 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 5 Jul 2010 09:31:06 -0500 Subject: [Freeswitch-users] Call-ID format In-Reply-To: <201007042147.05294.sos@sokhapkin.dyndns.org> References: <98731.23464.qm@web30508.mail.mud.yahoo.com> <201007042133.21253.sos@sokhapkin.dyndns.org> <201007042147.05294.sos@sokhapkin.dyndns.org> Message-ID: Does nobody ever bother reading the source code? Setting sip_call_id would do the trick. /b On Jul 4, 2010, at 8:47 PM, Sergey Okhapkin wrote: > Hmm, both b-leg Call-id and channel's Unique-ID are the same - channel's uuid. > Correct me if I'm wrong. > > On Sunday 04 July 2010, Jo?o Mesquita wrote: >> As far as I know, you can only change the Call-Id to the channel's >> Unique-ID. There's a profile config for that, I suppose. >> >> JM > From moises.silva at gmail.com Mon Jul 5 07:33:43 2010 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 5 Jul 2010 10:33:43 -0400 Subject: [Freeswitch-users] FreeTDM and Digium analog cards In-Reply-To: References: Message-ID: I don't understand your question. TDM refers to http://en.wikipedia.org/wiki/Time-division_multiplexing T1/E1 use TDM. So what do you mean by digital variants? You make it sound like TDM cards and the digital variants you talk about are different things. FreeTDM support analog and digital cards for both Sangoma and Digium. There is some PIKA support but it may be broken by now since few or no people seems to use it. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com On Mon, Jul 5, 2010 at 2:13 AM, Joseph L. Casale wrote: > Does FreeTDM support TDM cards, or just the digital variants? > Reading through Sangoma's docs on FreeTDM, it's not clear? > > Thanks! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/e0e84056/attachment.html From aland at burngreave.net Sun Jul 4 09:14:33 2010 From: aland at burngreave.net (Alan Dawson) Date: Sun, 4 Jul 2010 17:14:33 +0100 Subject: [Freeswitch-users] Receiving incoming SIP calls Message-ID: <20100704161433.GF29534@apple.rat.burntout.org> Hi, I'm new to VOIP, and freeswitch. I'm trying to enable certain internal extensions to receive incoming SIP calls, eg people can dial sip:1001 at my.domain and be able to receive those calls. but when they do this I get Rejected by acl "domains". log message. Any help,howtos,faq etc in resolving this appreciated. -- Alan Dawson -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100704/91920c2c/attachment.bin From sanjay.k.arora at gmail.com Sun Jul 4 09:43:43 2010 From: sanjay.k.arora at gmail.com (Sanjay Arora) Date: Sun, 4 Jul 2010 22:13:43 +0530 Subject: [Freeswitch-users] Request for suggestions - FreeSwitch Appliance Message-ID: Hello All I am looking for a freeswitch appliance, suitable for SMB. Something that can act as office pbx in addition to providing inbound & outbound VoIP calling services. 50 lines or less. Please advise if you are using any such appliance & your feedback or even if you know of one. Pointers/links/Comments welcome. With best regards. Sanjay. From brian at freeswitch.org Mon Jul 5 07:32:15 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 5 Jul 2010 09:32:15 -0500 Subject: [Freeswitch-users] Getting WTF1..N messages in the fs log In-Reply-To: References: Message-ID: <272B54DF-F5FC-4F83-B646-E7652F611D4B@freeswitch.org> UPDATE.. you're clearly not on the very latest code. /b On Jul 4, 2010, at 8:04 AM, Vitalii Colosov wrote: > Thanks Rupa, > > Neither freetdm nor openzap is being used. > > Maybe it is similar debug message from some other part of the code... > > Thank you, > Vitalie > From brian at freeswitch.org Mon Jul 5 08:04:15 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 5 Jul 2010 10:04:15 -0500 Subject: [Freeswitch-users] voice mail password In-Reply-To: References: Message-ID: <76C2C584-FD54-463E-A36B-55BF330F18D6@freeswitch.org> Did you happen to forget to reloadxml? /b On Jul 5, 2010, at 7:02 AM, wrote: > Hello, > > I have the following user as 1001.xml > > > > > > > > > > > > > > > > > > I can call to this extension and if it does not answer I can leave a message in its mailbox, I use the default configuration and voicemail is accessed in extension 4000. If I want to listen to the messages the password I have to dial "1001" not "0288", why is this so?, where is 1001 set as the password for the mailbox?, how do I change the XML configuration files to use 0288 or any other number I may choose? > > Thank you for your help. > > Luis. > From anthony.minessale at gmail.com Mon Jul 5 08:23:04 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 5 Jul 2010 10:23:04 -0500 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: you also need user_recurse_variables=false {user_recurse_variables=false,group_confirm_cancel_timeout=true} On Sat, Jul 3, 2010 at 8:22 AM, Nagalenoj H. wrote: > Here is the log got from the latest GIT source. > > http://pastebin.freeswitch.org/13347 > > > On Fri, Jul 2, 2010 at 10:15 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> update and reproduce that same log with latest GIT the version you are >> using has an issue. >> >> >> On Fri, Jul 2, 2010 at 12:25 AM, Nagalenoj H. wrote: >> >>> I've pasted the console log here, >>> >>> http://pastebin.freeswitch.org/13333 >>> >>> >>> On Wed, Jun 30, 2010 at 11:15 PM, Michael Collins wrote: >>> >>>> Can you supply a console log of these calls? >>>> -MC >>>> >>>> >>>> On Wed, Jun 30, 2010 at 7:37 AM, Nagalenoj H. wrote: >>>> >>>>> Dear Anthony, >>>>> I've tried using the group_confirm_cancel_timeout as per the >>>>> discussion we had in IRC. You wanted to used it as part of dial string and >>>>> not as a channel variable. >>>>> But, It doesn't work for me. >>>>> >>>>> Here is how I've given the commands and the script I've executed. Even >>>>> when I give group_confirm_cancel_timeout, the callee's leg is getting >>>>> disconnected after legtimeout. >>>>> >>>>> >>>>> connect >>>>> >>>>> sendmsg >>>>> call-command: execute >>>>> execute-app-name:answer >>>>> >>>>> sendmsg >>>>> call-command: execute >>>>> execute-app-name: set >>>>> execute-app-arg: group_confirm_key=exec >>>>> >>>>> sendmsg >>>>> call-command: execute >>>>> execute-app-name: set >>>>> execute-app-arg: group_confirm_file=perl /root/bridge.pl >>>>> >>>>> sendmsg >>>>> call-command: execute >>>>> execute-app-name: bridge >>>>> execute-app-arg: >>>>> {group_confirm_cancel_timeout=1}[leg_timeout=10]user/1005 >>>>> >>>>> >>>>> >>>>> bridge.pl: >>>>> #!/usr/bin/perl >>>>> use freeswitch; >>>>> >>>>> our $session; >>>>> freeswitch::consoleLog("info","Goint to get the digits"); >>>>> # To simulate the scenario I used sleep here. >>>>> sleep(30); >>>>> 1; >>>>> >>>>> Kindly tell me whats wrong in the above. >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> Regards, >>> Nagalenoj H. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -- > Regards, > Nagalenoj H. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/a220d8d5/attachment-0001.html From peder at networkoblivion.com Mon Jul 5 08:27:17 2010 From: peder at networkoblivion.com (Peder) Date: Mon, 5 Jul 2010 10:27:17 -0500 Subject: [Freeswitch-users] Request for suggestions - FreeSwitch Appliance In-Reply-To: References: Message-ID: <059a01cb1c56$8d673eb0$a835bc10$@com> https://www.cudatel.com/ We just got one to test and it is pretty cool. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sanjay Arora Sent: Sunday, July 04, 2010 11:44 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Request for suggestions - FreeSwitch Appliance Hello All I am looking for a freeswitch appliance, suitable for SMB. Something that can act as office pbx in addition to providing inbound & outbound VoIP calling services. 50 lines or less. Please advise if you are using any such appliance & your feedback or even if you know of one. Pointers/links/Comments welcome. With best regards. Sanjay. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Mon Jul 5 08:29:24 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 5 Jul 2010 10:29:24 -0500 Subject: [Freeswitch-users] Request for suggestions - FreeSwitch Appliance In-Reply-To: References: Message-ID: On Sun, Jul 4, 2010 at 11:43 AM, Sanjay Arora wrote: > Hello All > > I am looking for a freeswitch appliance, suitable for SMB. Something > that can act as office pbx in addition to providing inbound & outbound > VoIP calling services. 50 lines or less. > > Please advise if you are using any such appliance & your feedback or > even if you know of one. > > Pointers/links/Comments welcome. > > With best regards. > Sanjay. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/6d06f025/attachment.html From luixsansan at hotmail.com Mon Jul 5 08:29:27 2010 From: luixsansan at hotmail.com (luixsansan at hotmail.com) Date: Mon, 5 Jul 2010 17:29:27 +0200 Subject: [Freeswitch-users] voice mail password In-Reply-To: <76C2C584-FD54-463E-A36B-55BF330F18D6@freeswitch.org> References: <76C2C584-FD54-463E-A36B-55BF330F18D6@freeswitch.org> Message-ID: Hello, Since I am testing I can shut down and re-start freeswitch in order to reload all XML files, also I read somewhere, (I have read several documents to solve this problem without success) that after first installation the voicemail passwords are the same as the preconfigured extensions 1000-1019, actually it is, but nobody mention where this is set and how to change it. Thank you very much for your answer and help. regards. Luis. -------------------------------------------------- From: "Brian West" Sent: Monday, July 05, 2010 5:04 PM To: Subject: Re: [Freeswitch-users] voice mail password > Did you happen to forget to reloadxml? > > /b > > On Jul 5, 2010, at 7:02 AM, > wrote: > >> Hello, >> >> I have the following user as 1001.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> I can call to this extension and if it does not answer I can leave a >> message in its mailbox, I use the default configuration and voicemail is >> accessed in extension 4000. If I want to listen to the messages the >> password I have to dial "1001" not "0288", why is this so?, where is 1001 >> set as the password for the mailbox?, how do I change the XML >> configuration files to use 0288 or any other number I may choose? >> >> Thank you for your help. >> >> Luis. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jmesquita at freeswitch.org Mon Jul 5 09:00:02 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 5 Jul 2010 13:00:02 -0300 Subject: [Freeswitch-users] Call-ID format In-Reply-To: References: <98731.23464.qm@web30508.mail.mud.yahoo.com> <201007042133.21253.sos@sokhapkin.dyndns.org> <201007042147.05294.sos@sokhapkin.dyndns.org> Message-ID: Oops. JM On Mon, Jul 5, 2010 at 11:31 AM, Brian West wrote: > Does nobody ever bother reading the source code? Setting sip_call_id would > do the trick. > > /b > > > > On Jul 4, 2010, at 8:47 PM, Sergey Okhapkin wrote: > > > Hmm, both b-leg Call-id and channel's Unique-ID are the same - channel's > uuid. > > Correct me if I'm wrong. > > > > On Sunday 04 July 2010, Jo?o Mesquita wrote: > >> As far as I know, you can only change the Call-Id to the channel's > >> Unique-ID. There's a profile config for that, I suppose. > >> > >> JM > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/c50f7348/attachment.html From anthony.minessale at gmail.com Mon Jul 5 09:40:26 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 5 Jul 2010 11:40:26 -0500 Subject: [Freeswitch-users] Voicemail: can it advance to the next message when no user's input? In-Reply-To: References: Message-ID: probably $200 On Mon, Jul 5, 2010 at 9:11 AM, Yehavi Bourvine wrote: > We thought of doing it ourselves... How much bounty you think this > deserves? > > Thanks, Yehavi > > On 04/07/2010 22:42, "Jo?o Mesquita" wrote: > > Yes, but requires a patch on mod_voicemail code. > > Maybe suitable for a bounty? > > Regards, > Jo?o Mesquita > > > On Sun, Jul 4, 2010 at 9:08 AM, Yehavi Bourvine > wrote: > >> > >> > Hello, >> > >> > Our users are used to a voicemail system in which when you hear a >> voicemail messa... >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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 at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/12828158/attachment.html From jcasale at activenetwerx.com Mon Jul 5 10:04:11 2010 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 5 Jul 2010 17:04:11 +0000 Subject: [Freeswitch-users] FreeTDM and Digium analog cards In-Reply-To: References: Message-ID: >I don't understand your question. TDM refers to?http://en.wikipedia.org/wiki/Time-division_multiplexing > >T1/E1 use TDM. So what do you mean by digital variants? You make it sound like TDM cards and the digital variants you talk about are different >things. > >FreeTDM support analog and digital cards for both Sangoma and Digium. There is some PIKA support but it may be broken by now since few or no >people seems to use it. Moises, Thanks for taking the time to help. Sorry I wasn?t clear, what I meant by referring to digital cards was E1/T1 based cards versus the analog variants that function on the PSTN like a Digium TDM410. Tonight I am upgrading some hardware and will recompile fs, so I wondered if I should use this opportunity to move from OpenZAP to FreeTDM but I really couldn?t find a definitive answer wrt to its support for these cards, all the docs and examples explicitly refer to the digital cards. I guess looking at http://wiki.sangoma.com/wanpipe-api-freetdm it does show trunk_type definitions for FXO/S so I guess it does! Thanks! jlc From tony.tin at noahmedia.com.hk Mon Jul 5 11:12:03 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Tue, 6 Jul 2010 02:12:03 +0800 Subject: [Freeswitch-users] how to check whether openzap channel is idle In-Reply-To: References: Message-ID: Hi, Thanks a lot ! It works. However, it always use channel 1, is it possible to make it picking the channel in random. Regards, Tony On Mon, Jul 5, 2010 at 6:55 PM, lakshmanan ganapathy wrote: > Hi, > > If you say openzap/1/a/, then freeswitch will automatically use an > idle channel. > > > > > On Mon, Jul 5, 2010 at 12:18 PM, Tony Tin wrote: > >> Hi, >> >> Could anyone please tell me that how to check whether a openzap channel is >> in idle, so I can use it to originate a call. Thanks. >> >> Regards, >> Tony >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/f5d3402e/attachment.html From a.afzali2003 at gmail.com Mon Jul 5 11:18:05 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Mon, 5 Jul 2010 22:48:05 +0430 Subject: [Freeswitch-users] using FIFO to queue calls towards a gateway Message-ID: Hi FreeSWITCH, Suppose there is a small voice gateway that just has been equipped with a few FXO ports which connected to PSTN's analog lines. I'm going to use FIFO to manage call requests which going through this voice gateway. As I've learned about mod_fifo service, I think that the queue should have statically defined members (one for each port) which in their dial strings target that gateway (parametrized with the PSTN phone number by a session variable) and nothing else! Am I wrong about this? BEST -- afshin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/9992f87f/attachment.html From tgraziano at myitdepartment.net Mon Jul 5 11:19:00 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Mon, 5 Jul 2010 14:19:00 -0400 Subject: [Freeswitch-users] how to check whether openzap channel is idle In-Reply-To: References: Message-ID: You typically want the PSTN lines to come in in one direction (4,3,2,1 or 1,2,3,4) and outgoing calls hung the other direction to prevent glare. Wouldn't you? On Mon, Jul 5, 2010 at 2:12 PM, Tony Tin wrote: > Hi, > > Thanks a lot ! > > It works. However, it always use channel 1, is it possible to make it > picking the channel in random. > > Regards, > Tony > > > > On Mon, Jul 5, 2010 at 6:55 PM, lakshmanan ganapathy > wrote: > >> Hi, >> >> If you say openzap/1/a/, then freeswitch will automatically use an >> idle channel. >> >> >> >> >> On Mon, Jul 5, 2010 at 12:18 PM, Tony Tin wrote: >> >>> Hi, >>> >>> Could anyone please tell me that how to check whether a openzap channel >>> is in idle, so I can use it to originate a call. Thanks. >>> >>> Regards, >>> Tony >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano at voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano at myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk at voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/d0a92a72/attachment.html From tony.tin at noahmedia.com.hk Mon Jul 5 11:50:53 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Tue, 6 Jul 2010 02:50:53 +0800 Subject: [Freeswitch-users] how to check whether openzap channel is idle In-Reply-To: References: Message-ID: Thanks for the reply. Yes, that's what I'm worrying. The incoming order is 1,2,3,4, and seems that the outgoing order is the same. Any idea ? Regards, Tony On Tue, Jul 6, 2010 at 2:19 AM, Tony Graziano wrote: > You typically want the PSTN lines to come in in one direction (4,3,2,1 or > 1,2,3,4) and outgoing calls hung the other direction to prevent glare. > Wouldn't you? > > > On Mon, Jul 5, 2010 at 2:12 PM, Tony Tin wrote: > >> Hi, >> >> Thanks a lot ! >> >> It works. However, it always use channel 1, is it possible to make it >> picking the channel in random. >> >> Regards, >> Tony >> >> >> >> On Mon, Jul 5, 2010 at 6:55 PM, lakshmanan ganapathy < >> lakindia89 at gmail.com> wrote: >> >>> Hi, >>> >>> If you say openzap/1/a/, then freeswitch will automatically use >>> an idle channel. >>> >>> >>> >>> >>> On Mon, Jul 5, 2010 at 12:18 PM, Tony Tin wrote: >>> >>>> Hi, >>>> >>>> Could anyone please tell me that how to check whether a openzap channel >>>> is in idle, so I can use it to originate a call. Thanks. >>>> >>>> Regards, >>>> Tony >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgraziano at voice.myitdepartment.net > Fax: 434.984.8431 > > Email: tgraziano at myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpdesk at voice.myitdepartment.net > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/2a269858/attachment.html From jan.berger at video24.no Mon Jul 5 12:04:01 2010 From: jan.berger at video24.no (Jan Berger) Date: Mon, 5 Jul 2010 21:04:01 +0200 Subject: [Freeswitch-users] SIP2VXML Message-ID: <39ED5E8D0A5D42BCA019A827DE9F7A4C@dell9400> http://tools.ietf.org/html/rfc5552 Hi, the standard above describes a SIP 2 VXML interface. My question is can I do this with FreeSWITCH? Jan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/f650614f/attachment-0001.html From tony.tin at noahmedia.com.hk Mon Jul 5 12:06:42 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Tue, 6 Jul 2010 03:06:42 +0800 Subject: [Freeswitch-users] fail to bridge 2 openzap channels Message-ID: Hi, I'm trying to bridge 2 openzap channels, one is inbound call, another is outbound call. Below is the Lua script which originates the outbound call in the inbound call dialplan. If the "transfer" line is commented out, bridge is working fine. I suspect it is about call legs, but I can find the proper way to set then up. Could anyone please help, thanks. ------------------------------------------------------------------------------------- session_orig = freeswitch.Session("OpenZAP/2/a/98877666") while session:getState()~="CS_HANGUP" do os.execute("usleep 500000") if session_orig:answered() then break end end -- need to play message to called party before bridge -- comment out the below line, then bridge is working session_orig:transfer("orig_ext", "xml", "default") freeswitch.bridge(session, session_orig) ------------------------------------------------------------------------------------- Regards, Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/d5828143/attachment.html From asannucci at gmail.com Mon Jul 5 13:51:50 2010 From: asannucci at gmail.com (bakko) Date: Mon, 5 Jul 2010 22:51:50 +0200 Subject: [Freeswitch-users] Only Eutelia not working on NAT In-Reply-To: References: Message-ID: Hello, Are you tried to look at siptrace to see if the call arrive to FS? BR - Andrea - From anthony.minessale at gmail.com Mon Jul 5 15:06:31 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 5 Jul 2010 17:06:31 -0500 Subject: [Freeswitch-users] how to check whether openzap channel is idle In-Reply-To: References: Message-ID: use A instead. a means top down A means bottom up On Mon, Jul 5, 2010 at 1:50 PM, Tony Tin wrote: > Thanks for the reply. > > Yes, that's what I'm worrying. The incoming order is 1,2,3,4, and seems > that the outgoing order is the same. Any idea ? > > Regards, > Tony > > > > On Tue, Jul 6, 2010 at 2:19 AM, Tony Graziano < > tgraziano at myitdepartment.net> wrote: > >> You typically want the PSTN lines to come in in one direction (4,3,2,1 or >> 1,2,3,4) and outgoing calls hung the other direction to prevent glare. >> Wouldn't you? >> >> >> On Mon, Jul 5, 2010 at 2:12 PM, Tony Tin wrote: >> >>> Hi, >>> >>> Thanks a lot ! >>> >>> It works. However, it always use channel 1, is it possible to make it >>> picking the channel in random. >>> >>> Regards, >>> Tony >>> >>> >>> >>> On Mon, Jul 5, 2010 at 6:55 PM, lakshmanan ganapathy < >>> lakindia89 at gmail.com> wrote: >>> >>>> Hi, >>>> >>>> If you say openzap/1/a/, then freeswitch will automatically use >>>> an idle channel. >>>> >>>> >>>> >>>> >>>> On Mon, Jul 5, 2010 at 12:18 PM, Tony Tin wrote: >>>> >>>>> Hi, >>>>> >>>>> Could anyone please tell me that how to check whether a openzap channel >>>>> is in idle, so I can use it to originate a call. Thanks. >>>>> >>>>> Regards, >>>>> Tony >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> ====================== >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: tgraziano at voice.myitdepartment.net >> Fax: 434.984.8431 >> >> Email: tgraziano at myitdepartment.net >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: helpdesk at voice.myitdepartment.net >> Fax: 434.984.8427 >> >> Helpdesk Contract Customers: >> http://www.myitdepartment.net/gethelp/ >> >> Why do mathematicians always confuse Halloween and Christmas? >> Because 31 Oct = 25 Dec. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/c4ca236e/attachment.html From anthony.minessale at gmail.com Mon Jul 5 15:15:54 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 5 Jul 2010 17:15:54 -0500 Subject: [Freeswitch-users] fail to bridge 2 openzap channels In-Reply-To: References: Message-ID: transfer is a final action, the script must exit once it's been transferred. i think you need to explain what you really want to do. if it's just play a file use session:execute("playback", "file.wav") On Mon, Jul 5, 2010 at 2:06 PM, Tony Tin wrote: > Hi, > > I'm trying to bridge 2 openzap channels, one is inbound call, another is > outbound call. > > Below is the Lua script which originates the outbound call in the inbound > call dialplan. > > If the "transfer" line is commented out, bridge is working fine. I suspect > it is about call legs, but I can find the proper way to set then up. Could > anyone please help, thanks. > > > ------------------------------------------------------------------------------------- > > session_orig = freeswitch.Session("OpenZAP/2/a/98877666") > > while session:getState()~="CS_HANGUP" do > os.execute("usleep 500000") > if session_orig:answered() then > break > end > end > > -- need to play message to called party before bridge > -- comment out the below line, then bridge is working > session_orig:transfer("orig_ext", "xml", "default") > > freeswitch.bridge(session, session_orig) > > > ------------------------------------------------------------------------------------- > > Regards, > Tony > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/77d9a4b2/attachment.html From anthony.minessale at gmail.com Mon Jul 5 15:58:05 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 5 Jul 2010 17:58:05 -0500 Subject: [Freeswitch-users] SIP2VXML In-Reply-To: <39ED5E8D0A5D42BCA019A827DE9F7A4C@dell9400> References: <39ED5E8D0A5D42BCA019A827DE9F7A4C@dell9400> Message-ID: if you are patient, maybe..... =D Still how bout we just make a mod_vxml? Everyone keeps saying they want to make one for years now and nobody does. It's almost like people think with all the stuff FS can do out of the box, supporting this fly-by-night phenomenon called VXML that has failed to gain traction after 5 years is somehow too valuable to contribute lol We have all the tools onboard already to make it but I am guessing the thought is already crossing your mind based on the other thread about javascript =D On Mon, Jul 5, 2010 at 2:04 PM, Jan Berger wrote: > http://tools.ietf.org/html/rfc5552 > > > > Hi, the standard above describes a SIP 2 VXML interface. > > > > My question is can I do this with FreeSWITCH? > > > > Jan > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100705/dba87291/attachment-0001.html From tony.tin at noahmedia.com.hk Mon Jul 5 19:38:46 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Tue, 6 Jul 2010 10:38:46 +0800 Subject: [Freeswitch-users] how to check whether openzap channel is idle In-Reply-To: References: Message-ID: Thanks. I searched the wiki again, it is here, the keyword is "openzap bridge". http://wiki.freeswitch.org/wiki/OpenZAP#Dialplan_Configuration Regards, Tony On Tue, Jul 6, 2010 at 6:06 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > use A instead. > > a means top down > A means bottom up > > > On Mon, Jul 5, 2010 at 1:50 PM, Tony Tin wrote: > >> Thanks for the reply. >> >> Yes, that's what I'm worrying. The incoming order is 1,2,3,4, and seems >> that the outgoing order is the same. Any idea ? >> >> Regards, >> Tony >> >> >> >> On Tue, Jul 6, 2010 at 2:19 AM, Tony Graziano < >> tgraziano at myitdepartment.net> wrote: >> >>> You typically want the PSTN lines to come in in one direction (4,3,2,1 or >>> 1,2,3,4) and outgoing calls hung the other direction to prevent glare. >>> Wouldn't you? >>> >>> >>> On Mon, Jul 5, 2010 at 2:12 PM, Tony Tin wrote: >>> >>>> Hi, >>>> >>>> Thanks a lot ! >>>> >>>> It works. However, it always use channel 1, is it possible to make it >>>> picking the channel in random. >>>> >>>> Regards, >>>> Tony >>>> >>>> >>>> >>>> On Mon, Jul 5, 2010 at 6:55 PM, lakshmanan ganapathy < >>>> lakindia89 at gmail.com> wrote: >>>> >>>>> Hi, >>>>> >>>>> If you say openzap/1/a/, then freeswitch will automatically use >>>>> an idle channel. >>>>> >>>>> >>>>> >>>>> >>>>> On Mon, Jul 5, 2010 at 12:18 PM, Tony Tin wrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> Could anyone please tell me that how to check whether a openzap >>>>>> channel is in idle, so I can use it to originate a call. Thanks. >>>>>> >>>>>> Regards, >>>>>> Tony >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> ====================== >>> Tony Graziano, Manager >>> Telephone: 434.984.8430 >>> sip: tgraziano at voice.myitdepartment.net >>> Fax: 434.984.8431 >>> >>> Email: tgraziano at myitdepartment.net >>> >>> LAN/Telephony/Security and Control Systems Helpdesk: >>> Telephone: 434.984.8426 >>> sip: helpdesk at voice.myitdepartment.net >>> Fax: 434.984.8427 >>> >>> Helpdesk Contract Customers: >>> http://www.myitdepartment.net/gethelp/ >>> >>> Why do mathematicians always confuse Halloween and Christmas? >>> Because 31 Oct = 25 Dec. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/83868912/attachment.html From tony.tin at noahmedia.com.hk Mon Jul 5 19:51:17 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Tue, 6 Jul 2010 10:51:17 +0800 Subject: [Freeswitch-users] fail to bridge 2 openzap channels In-Reply-To: References: Message-ID: Thanks for your help. I'm building a IVRS. I want to make the outbound call act just like a inbound call which terminated at a specified extension, so the dialplan for that extension will be executed, which will play message, get dtmf, data logging and things like that. Is there another way to connect 2 channels except bridge, so 2 callers can talk to each other no matter it is a inbound or outboud call (example usage - one to one chat service). Regards, Tony On Tue, Jul 6, 2010 at 6:15 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > transfer is a final action, the script must exit once it's been > transferred. > i think you need to explain what you really want to do. > if it's just play a file use session:execute("playback", "file.wav") > > > On Mon, Jul 5, 2010 at 2:06 PM, Tony Tin wrote: > >> Hi, >> >> I'm trying to bridge 2 openzap channels, one is inbound call, another is >> outbound call. >> >> Below is the Lua script which originates the outbound call in the inbound >> call dialplan. >> >> If the "transfer" line is commented out, bridge is working fine. I suspect >> it is about call legs, but I can find the proper way to set then up. Could >> anyone please help, thanks. >> >> >> ------------------------------------------------------------------------------------- >> >> session_orig = freeswitch.Session("OpenZAP/2/a/98877666") >> >> while session:getState()~="CS_HANGUP" do >> os.execute("usleep 500000") >> if session_orig:answered() then >> break >> end >> end >> >> -- need to play message to called party before bridge >> -- comment out the below line, then bridge is working >> session_orig:transfer("orig_ext", "xml", "default") >> >> freeswitch.bridge(session, session_orig) >> >> >> ------------------------------------------------------------------------------------- >> >> Regards, >> Tony >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/b29f06a3/attachment.html From fs-list at communicatefreely.net Mon Jul 5 19:55:58 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 05 Jul 2010 22:55:58 -0400 Subject: [Freeswitch-users] Receiving incoming SIP calls In-Reply-To: <20100704161433.GF29534@apple.rat.burntout.org> References: <20100704161433.GF29534@apple.rat.burntout.org> Message-ID: <4C329B3E.1020704@communicatefreely.net> Hi Alan, There is normally an ACL applied to the external profile that only allows calls from gateways you set up in that SIP profile. My suggestion (what we do here for the same purpose), is to set up another SIP profile specifically for public SIP. In the profile, you can disable the ACL, and you can also specify a different context. You can create a very limited context that gives you a level of control over how those calls are handled. You may only want to match a subset of extensions, and you may also want to impose some limits (see mod_limit) on number of calls, number of attempts, and limit access to voicemail, etc. Alan Dawson wrote: > Hi, > > I'm new to VOIP, and freeswitch. > > I'm trying to enable certain internal extensions to receive incoming SIP calls, > eg people can dial sip:1001 at my.domain and be able to receive those calls. > but when they do this I get > > Rejected by acl "domains". > > log message. > > Any help,howtos,faq etc in resolving this appreciated. > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fs-list at communicatefreely.net Mon Jul 5 19:59:59 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 05 Jul 2010 22:59:59 -0400 Subject: [Freeswitch-users] Dynamic mod_conference configuration Message-ID: <4C329C2F.1040001@communicatefreely.net> Hello list, I am trying to set up our conferencing system to pull some preferences from a database. In the conference.conf.xml file, I have a few profiles that cover pretty much all the bases, but I want to be able to set the music on hold sound differently for each conference, without reloading XML. I tried using ${hold_music} as the moh_sound in the config file, but this doesn't seem to expand. Is there a way to dynamically set conference profile options with a channel variable? I read a bit on generating config with LUA. The wiki talks about dialplan and directory, but doesn't really say much about other module configs. Can I bind a lua script to mod_conference? Can someone give me a snippet of that binding if that's possible? Thanks for the help! -Tim From nagalenoj at gmail.com Mon Jul 5 23:58:17 2010 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Tue, 6 Jul 2010 12:28:17 +0530 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: Got core dump when I execute the bridge as execute-app-arg: {user_recurse_variables=false,group_confirm_cancel_timeout=true}[leg_timeout=10]user/1000 Console log is here, http://pastebin.freeswitch.org/13371 core dump backtrace is here, http://pastebin.freeswitch.org/13372 Also, tried after git pull, but the result is same. On Mon, Jul 5, 2010 at 8:53 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > you also need user_recurse_variables=false > > {user_recurse_variables=false,group_confirm_cancel_timeout=true} > > > > On Sat, Jul 3, 2010 at 8:22 AM, Nagalenoj H. wrote: > >> Here is the log got from the latest GIT source. >> >> http://pastebin.freeswitch.org/13347 >> >> >> On Fri, Jul 2, 2010 at 10:15 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> update and reproduce that same log with latest GIT the version you are >>> using has an issue. >>> >>> >>> On Fri, Jul 2, 2010 at 12:25 AM, Nagalenoj H. wrote: >>> >>>> I've pasted the console log here, >>>> >>>> http://pastebin.freeswitch.org/13333 >>>> >>>> >>>> On Wed, Jun 30, 2010 at 11:15 PM, Michael Collins wrote: >>>> >>>>> Can you supply a console log of these calls? >>>>> -MC >>>>> >>>>> >>>>> On Wed, Jun 30, 2010 at 7:37 AM, Nagalenoj H. wrote: >>>>> >>>>>> Dear Anthony, >>>>>> I've tried using the group_confirm_cancel_timeout as per the >>>>>> discussion we had in IRC. You wanted to used it as part of dial string and >>>>>> not as a channel variable. >>>>>> But, It doesn't work for me. >>>>>> >>>>>> Here is how I've given the commands and the script I've executed. Even >>>>>> when I give group_confirm_cancel_timeout, the callee's leg is getting >>>>>> disconnected after legtimeout. >>>>>> >>>>>> >>>>>> connect >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name:answer >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: set >>>>>> execute-app-arg: group_confirm_key=exec >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: set >>>>>> execute-app-arg: group_confirm_file=perl /root/bridge.pl >>>>>> >>>>>> sendmsg >>>>>> call-command: execute >>>>>> execute-app-name: bridge >>>>>> execute-app-arg: >>>>>> {group_confirm_cancel_timeout=1}[leg_timeout=10]user/1005 >>>>>> >>>>>> >>>>>> >>>>>> bridge.pl: >>>>>> #!/usr/bin/perl >>>>>> use freeswitch; >>>>>> >>>>>> our $session; >>>>>> freeswitch::consoleLog("info","Goint to get the digits"); >>>>>> # To simulate the scenario I used sleep here. >>>>>> sleep(30); >>>>>> 1; >>>>>> >>>>>> Kindly tell me whats wrong in the above. >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> -- >>>> Regards, >>>> Nagalenoj H. >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> -- >> Regards, >> Nagalenoj H. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/14460d9f/attachment.html From jan.berger at video24.no Tue Jul 6 00:13:24 2010 From: jan.berger at video24.no (Jan Berger) Date: Tue, 6 Jul 2010 09:13:24 +0200 Subject: [Freeswitch-users] SIP2VXML In-Reply-To: References: <39ED5E8D0A5D42BCA019A827DE9F7A4C@dell9400> Message-ID: <4DDF77F5D01C432A9C765C73BD0715EC@dell9400> Hi, Were doing both mod_vxml and mod_ccxml - and the project is open source on the ccxml/vxml side (www.video24.no ) and will integrate into FreeSWITCH asap. The stacks are written from scratch- only ccxml 1.0 and vxml 2.1 for now. We are currently 3 people - myself + 2 from UK. We have not done anything towards FS integration yet because we have a bit to do before we can connect. Jan _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 6. juli 2010 00:58 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP2VXML if you are patient, maybe..... =D Still how bout we just make a mod_vxml? Everyone keeps saying they want to make one for years now and nobody does. It's almost like people think with all the stuff FS can do out of the box, supporting this fly-by-night phenomenon called VXML that has failed to gain traction after 5 years is somehow too valuable to contribute lol We have all the tools onboard already to make it but I am guessing the thought is already crossing your mind based on the other thread about javascript =D On Mon, Jul 5, 2010 at 2:04 PM, Jan Berger wrote: http://tools.ietf.org/html/rfc5552 Hi, the standard above describes a SIP 2 VXML interface. My question is can I do this with FreeSWITCH? Jan _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/f30a56fe/attachment-0001.html From os at tenios.de Tue Jul 6 01:41:22 2010 From: os at tenios.de (=?iso-8859-1?Q?Oliver_Sch=F6nbeck?=) Date: Tue, 6 Jul 2010 10:41:22 +0200 Subject: [Freeswitch-users] sofia_contact response via socket Message-ID: <000c01cb1ce7$03a81000$0af83000$@de> Hi all, I recently noticed that the response send after a call to sofia_contact via socket connection lags the two \n at the end. Which leads to some Problems parsing the messages correctly. Is this behaviour intended? I?m using sofia_contact to check whether a user is currently registered or not so I am in the need to call this function via socket. Is there an alternative? Kind regards Oliver -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/7ff5fdf9/attachment.html From peder at networkoblivion.com Tue Jul 6 05:58:38 2010 From: peder at networkoblivion.com (Peder) Date: Tue, 6 Jul 2010 07:58:38 -0500 Subject: [Freeswitch-users] VM Force Password Change Message-ID: <016301cb1d0a$f38c2e80$daa48b80$@com> Is there a way to force a password change upon first login to a voicemail account? We set the password to 1234 and would like to force users to change it the first time they login to the account. Peder From elihay at savion.huji.ac.il Tue Jul 6 06:13:17 2010 From: elihay at savion.huji.ac.il (Eli Hayun) Date: Tue, 06 Jul 2010 16:13:17 +0300 Subject: [Freeswitch-users] Getting DTMF during Fifo In-Reply-To: References: <1278219174.2180.10.camel@localhost.localdomain> Message-ID: <1278421997.25511.4.camel@localhost.localdomain> On Sun, 2010-07-04 at 15:17 +0300, afshin afzali wrote: > bind_meta_app Thanks for your answer. The problem is that bind_meta_app requires * before the digits, and I need something else. Anyway, i solved it by entering a program to negotiate with the caller before entering the fifo, and using the exit key to enter into another program if he want to leave an answer. Eli From Nabble at slickdeals.endjunk.com Tue Jul 6 07:05:02 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Tue, 6 Jul 2010 07:05:02 -0700 (PDT) Subject: [Freeswitch-users] More questions migrating from Asterisk to FS Message-ID: <1278425102977-5260522.post@n2.nabble.com> I recently managed to get FS to run on my Seagate DockStar device. So far, I have no problem to configure my FS with a single Gizmo5 account, except losing the 1st two seconds at the beginning of a call connection (I am sure there is a fix to this, but haven't found one, yet). My first step is to move all my Gizmo5 accounts to my new FS system so that I can decommission my Asterisk PBX system. The question I have is if FS has a support for template with inherritance (like asterisk) so that I don't have to copy the same parameters for each Gizmo5 account, except the username/password. I apologize if this question had been asked before, but a search through mailing list couldn't did not seem to yield such a topic. Thank you. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/More-questions-migrating-from-Asterisk-to-FS-tp5260522p5260522.html Sent from the freeswitch-users mailing list archive at Nabble.com. From dswardstrom at remotelink.com Tue Jul 6 07:08:25 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Tue, 6 Jul 2010 07:08:25 -0700 (PDT) Subject: [Freeswitch-users] Dynamic mod_conference configuration In-Reply-To: <4C329C2F.1040001@communicatefreely.net> References: <4C329C2F.1040001@communicatefreely.net> Message-ID: <1278425305467-5260539.post@n2.nabble.com> Look at the JavaScript example: JavaScript Conference IVR @ http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR You should be able to port the JavaScript to Lua. I didn't know either, so just went with JavaScript because of all the nice examples. There are lots of hints of how to things with conferencing on that page and in the example. I actually moved much of the "sounds" out of the profiles and plan to provide them in the JavaScript code. On the other hand, if you want to have music on hold, you may need to use the Moderator flag. Since I am using it, I do get MOH until the Moderator arrives. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dynamic-mod-conference-configuration-tp5258560p5260539.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Tue Jul 6 07:07:54 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 6 Jul 2010 09:07:54 -0500 Subject: [Freeswitch-users] SIP2VXML In-Reply-To: <4DDF77F5D01C432A9C765C73BD0715EC@dell9400> References: <39ED5E8D0A5D42BCA019A827DE9F7A4C@dell9400> <4DDF77F5D01C432A9C765C73BD0715EC@dell9400> Message-ID: Feel free to ask for any assistance we may provide. Maybe accessable open source versions of the above might even spark more popularity for them. On Jul 6, 2010 2:21 AM, "Jan Berger" wrote: Hi, Were doing both mod_vxml and mod_ccxml ? and the project is open source on the ccxml/vxml side (www.video24.no) and will integrate into FreeSWITCH asap. The stacks are written from scratch? only ccxml 1.0 and vxml 2.1 for now. We are currently 3 people ? myself + 2 from UK. We have not done anything towards FS integration yet because we have a bit to do before we can connect. Jan ------------------------------ *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* 6. juli 2010 00:58 *To:* freeswitch-users at lists.freeswitch.org *Subject:* Re: [Freeswitch-users] SIP2VXML if you are patient, maybe..... =D Still how bout we just make a mod_vxml? Everyone keeps s... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/41ed20f6/attachment.html From anthony.minessale at gmail.com Tue Jul 6 08:26:40 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 6 Jul 2010 10:26:40 -0500 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: try latest GIT, that was an edge case and it's fixed. On Tue, Jul 6, 2010 at 1:58 AM, Nagalenoj H. wrote: > Got core dump when I execute the bridge as > execute-app-arg: > {user_recurse_variables=false,group_confirm_cancel_timeout=true}[leg_timeout=10]user/1000 > > Console log is here, > http://pastebin.freeswitch.org/13371 > > core dump backtrace is here, > http://pastebin.freeswitch.org/13372 > > Also, tried after git pull, but the result is same. > > > On Mon, Jul 5, 2010 at 8:53 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> you also need user_recurse_variables=false >> >> {user_recurse_variables=false,group_confirm_cancel_timeout=true} >> >> >> >> On Sat, Jul 3, 2010 at 8:22 AM, Nagalenoj H. wrote: >> >>> Here is the log got from the latest GIT source. >>> >>> http://pastebin.freeswitch.org/13347 >>> >>> >>> On Fri, Jul 2, 2010 at 10:15 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> update and reproduce that same log with latest GIT the version you are >>>> using has an issue. >>>> >>>> >>>> On Fri, Jul 2, 2010 at 12:25 AM, Nagalenoj H. wrote: >>>> >>>>> I've pasted the console log here, >>>>> >>>>> http://pastebin.freeswitch.org/13333 >>>>> >>>>> >>>>> On Wed, Jun 30, 2010 at 11:15 PM, Michael Collins wrote: >>>>> >>>>>> Can you supply a console log of these calls? >>>>>> -MC >>>>>> >>>>>> >>>>>> On Wed, Jun 30, 2010 at 7:37 AM, Nagalenoj H. wrote: >>>>>> >>>>>>> Dear Anthony, >>>>>>> I've tried using the group_confirm_cancel_timeout as per the >>>>>>> discussion we had in IRC. You wanted to used it as part of dial string and >>>>>>> not as a channel variable. >>>>>>> But, It doesn't work for me. >>>>>>> >>>>>>> Here is how I've given the commands and the script I've executed. >>>>>>> Even when I give group_confirm_cancel_timeout, the callee's leg is getting >>>>>>> disconnected after legtimeout. >>>>>>> >>>>>>> >>>>>>> connect >>>>>>> >>>>>>> sendmsg >>>>>>> call-command: execute >>>>>>> execute-app-name:answer >>>>>>> >>>>>>> sendmsg >>>>>>> call-command: execute >>>>>>> execute-app-name: set >>>>>>> execute-app-arg: group_confirm_key=exec >>>>>>> >>>>>>> sendmsg >>>>>>> call-command: execute >>>>>>> execute-app-name: set >>>>>>> execute-app-arg: group_confirm_file=perl /root/bridge.pl >>>>>>> >>>>>>> sendmsg >>>>>>> call-command: execute >>>>>>> execute-app-name: bridge >>>>>>> execute-app-arg: >>>>>>> {group_confirm_cancel_timeout=1}[leg_timeout=10]user/1005 >>>>>>> >>>>>>> >>>>>>> >>>>>>> bridge.pl: >>>>>>> #!/usr/bin/perl >>>>>>> use freeswitch; >>>>>>> >>>>>>> our $session; >>>>>>> freeswitch::consoleLog("info","Goint to get the digits"); >>>>>>> # To simulate the scenario I used sleep here. >>>>>>> sleep(30); >>>>>>> 1; >>>>>>> >>>>>>> Kindly tell me whats wrong in the above. >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> -- >>>>> Regards, >>>>> Nagalenoj H. >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> -- >>> Regards, >>> Nagalenoj H. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -- > Regards, > Nagalenoj H. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/9ced9863/attachment-0001.html From mustafa.pk at gmail.com Tue Jul 6 08:35:03 2010 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Tue, 6 Jul 2010 20:35:03 +0500 Subject: [Freeswitch-users] VM Force Password Change In-Reply-To: <016301cb1d0a$f38c2e80$daa48b80$@com> References: <016301cb1d0a$f38c2e80$daa48b80$@com> Message-ID: i don't think so, but you can definitely make it possible in dialplan. -m On Tue, Jul 6, 2010 at 5:58 PM, Peder wrote: > Is there a way to force a password change upon first login to a voicemail > account? ?We set the password to 1234 and would like to force users to > change it the first time they login to the account. > > Peder > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com From anthony.minessale at gmail.com Tue Jul 6 08:48:13 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 6 Jul 2010 10:48:13 -0500 Subject: [Freeswitch-users] More questions migrating from Asterisk to FS In-Reply-To: <1278425102977-5260522.post@n2.nabble.com> References: <1278425102977-5260522.post@n2.nabble.com> Message-ID: params put in the domain itself would apply to every user in that domain. so you can only put the id login and pass in your config if you wish On Tue, Jul 6, 2010 at 9:05 AM, mazilo wrote: > > I recently managed to get FS to run on my Seagate DockStar device. So far, > I > have no problem to configure my FS with a single Gizmo5 account, except > losing the 1st two seconds at the beginning of a call connection (I am sure > there is a fix to this, but haven't found one, yet). My first step is to > move all my Gizmo5 accounts to my new FS system so that I can decommission > my Asterisk PBX system. The question I have is if FS has a support for > template with inherritance (like asterisk) so that I don't have to copy the > same parameters for each Gizmo5 account, except the username/password. I > apologize if this question had been asked before, but a search through > mailing list couldn't did not seem to yield such a topic. > > Thank you. > > ----- > don't and stop are the ONLY two 4-letter words considered offensive to men, > but not when used together. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/More-questions-migrating-from-Asterisk-to-FS-tp5260522p5260522.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/770c5a59/attachment.html From peder at networkoblivion.com Tue Jul 6 08:54:53 2010 From: peder at networkoblivion.com (Peder) Date: Tue, 6 Jul 2010 10:54:53 -0500 Subject: [Freeswitch-users] VM Force Password Change In-Reply-To: References: <016301cb1d0a$f38c2e80$daa48b80$@com> Message-ID: <01bc01cb1d23$93213900$b963ab00$@com> Any idea how? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ghulam Mustafa Sent: Tuesday, July 06, 2010 10:35 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] VM Force Password Change i don't think so, but you can definitely make it possible in dialplan. -m On Tue, Jul 6, 2010 at 5:58 PM, Peder wrote: > Is there a way to force a password change upon first login to a voicemail > account? We set the password to 1234 and would like to force users to > change it the first time they login to the account. > > Peder > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mustafa.pk at gmail.com Tue Jul 6 09:12:14 2010 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Tue, 6 Jul 2010 21:12:14 +0500 Subject: [Freeswitch-users] VM Force Password Change In-Reply-To: <01bc01cb1d23$93213900$b963ab00$@com> References: <016301cb1d0a$f38c2e80$daa48b80$@com> <01bc01cb1d23$93213900$b963ab00$@com> Message-ID: i have not done something like this before, but you can create an event socket script to check first time login (possibly by checking _default_ password in db/xml) and then and then ask user to change the password. check it out http://wiki.freeswitch.org/wiki/IVR -m On Tue, Jul 6, 2010 at 8:54 PM, Peder wrote: > Any idea how? > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ghulam Mustafa > Sent: Tuesday, July 06, 2010 10:35 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] VM Force Password Change > > i don't think so, but you can definitely make it possible in dialplan. > > -m > > On Tue, Jul 6, 2010 at 5:58 PM, Peder wrote: >> Is there a way to force a password change upon first login to a voicemail >> account? ?We set the password to 1234 and would like to force users to >> change it the first time they login to the account. >> >> Peder >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Ghulam Mustafa > cell: +92 333.611.7681 > sip: cyrenity at ekiga.net > mail: mustafa.pk at gmail.com > web: cyrenity.wordpress.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com From Nabble at slickdeals.endjunk.com Tue Jul 6 10:20:48 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Tue, 6 Jul 2010 10:20:48 -0700 (PDT) Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: References: <1278204209127-5252219.post@n2.nabble.com> Message-ID: <1278436848678-5261486.post@n2.nabble.com> Anthony Minessale wrote: > Is this the only place with an error? Yes. Anthony Minessale wrote: > try altering the code so the if and else is replaced by just the contents > of the if. ie the switch va_sprintf line I tried your suggestion above and compilation went through to produce mod_sofia.so file with no problem and a portion of the excerpts is as shown below: make[8]: Entering directory `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.1_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' > Creating mod_sofia_la-sofia.lo > Compiling sofia.c ... > Creating mod_sofia_la-sofia_glue.lo > Compiling sofia_glue.c ... > Creating mod_sofia_la-sofia_presence.lo > Compiling sofia_presence.c ... > Creating mod_sofia_la-sofia_reg.lo > Compiling sofia_reg.c ... > Creating mod_sofia_la-sofia_sla.lo > Compiling sofia_sla.c ... > Creating mod_sofia_la-sip-dig.lo > Compiling sip-dig.c ... > Creating mod_sofia.la ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5261486.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Tue Jul 6 11:35:18 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 6 Jul 2010 13:35:18 -0500 Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <1278436848678-5261486.post@n2.nabble.com> References: <1278204209127-5252219.post@n2.nabble.com> <1278436848678-5261486.post@n2.nabble.com> Message-ID: can you try the latest git head to see if my workaround fixes it? On Tue, Jul 6, 2010 at 12:20 PM, mazilo wrote: > > > Anthony Minessale wrote: > > Is this the only place with an error? > Yes. > > > Anthony Minessale wrote: > > try altering the code so the if and else is replaced by just the contents > > of the if. ie the switch va_sprintf line > I tried your suggestion above and compilation went through to produce > mod_sofia.so file with no problem and a portion of the excerpts is as shown > below: > > make[8]: Entering directory > > `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.1_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' > > Creating mod_sofia_la-sofia.lo > > Compiling sofia.c ... > > Creating mod_sofia_la-sofia_glue.lo > > Compiling sofia_glue.c ... > > Creating mod_sofia_la-sofia_presence.lo > > Compiling sofia_presence.c ... > > Creating mod_sofia_la-sofia_reg.lo > > Compiling sofia_reg.c ... > > Creating mod_sofia_la-sofia_sla.lo > > Compiling sofia_sla.c ... > > Creating mod_sofia_la-sip-dig.lo > > Compiling sip-dig.c ... > > Creating mod_sofia.la > > > > ----- > don't and stop are the ONLY two 4-letter words considered offensive to men, > but not when used together. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5261486.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/245eecc1/attachment.html From brokendash at gmail.com Tue Jul 6 11:37:22 2010 From: brokendash at gmail.com (broken dash) Date: Tue, 6 Jul 2010 13:37:22 -0500 Subject: [Freeswitch-users] IPComms Provider Template Message-ID: I have been trying to setup a sip gateway using fusionpbx but for whatever reason the web gui isnt working 100% correctly.Does anyone have a template config for IPComms so I can manually include the XML into freeswitch? Thanks, Brian From steveayre at gmail.com Tue Jul 6 13:03:54 2010 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 6 Jul 2010 21:03:54 +0100 Subject: [Freeswitch-users] VM Force Password Change In-Reply-To: References: <016301cb1d0a$f38c2e80$daa48b80$@com> <01bc01cb1d23$93213900$b963ab00$@com> Message-ID: Try using mod_xml_curl to provide the user directory. voicemail password is a param in the user directory, you can generate that dynamically querying a database (e.g. MySQL) via a scripting language (e.g. PHP) using mod_xml_curl to get the current password (or a1 hash if you want to store it encrypted). You can have a dialplan extension that's executed prior to the extension that connects them to their voicemail if there's no password set that collects DTMF and then sends it to the webserver to update the database using Mod_curl http://wiki.freeswitch.org/wiki/Mod_voicemail#vm-password http://wiki.freeswitch.org/wiki/Mod_xml_curl http://wiki.freeswitch.org/wiki/Mod_curl On 6 July 2010 17:12, Ghulam Mustafa wrote: > i have not done something like this before, but you can create an > event socket script to check first time login (possibly by checking > _default_ password in db/xml) and then and then ask user to change the > password. > > check it out http://wiki.freeswitch.org/wiki/IVR > > > -m > > On Tue, Jul 6, 2010 at 8:54 PM, Peder wrote: > > Any idea how? > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ghulam Mustafa > > Sent: Tuesday, July 06, 2010 10:35 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] VM Force Password Change > > > > i don't think so, but you can definitely make it possible in dialplan. > > > > -m > > > > On Tue, Jul 6, 2010 at 5:58 PM, Peder wrote: > >> Is there a way to force a password change upon first login to a > voicemail > >> account? We set the password to 1234 and would like to force users to > >> change it the first time they login to the account. > >> > >> Peder > >> > >> > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Ghulam Mustafa > > cell: +92 333.611.7681 > > sip: cyrenity at ekiga.net > > mail: mustafa.pk at gmail.com > > web: cyrenity.wordpress.com > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Ghulam Mustafa > cell: +92 333.611.7681 > sip: cyrenity at ekiga.net > mail: mustafa.pk at gmail.com > web: cyrenity.wordpress.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/2f6a55c3/attachment.html From larclap at yahoo.com Tue Jul 6 13:41:03 2010 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 6 Jul 2010 13:41:03 -0700 Subject: [Freeswitch-users] 408 error? Message-ID: <00e501cb1d4b$8d349fa0$a79ddee0$@yahoo.com> I am getting a 408 error. I do not understand why. I am using git-1fba654 2010-07-01 on Centos 5.3. Could it be Flowroute? Thanks for any help, Lars http://pastebin.freeswitch.org/13389 From anthony.minessale at gmail.com Tue Jul 6 14:04:32 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 6 Jul 2010 16:04:32 -0500 Subject: [Freeswitch-users] 408 error? In-Reply-To: <00e501cb1d4b$8d349fa0$a79ddee0$@yahoo.com> References: <00e501cb1d4b$8d349fa0$a79ddee0$@yahoo.com> Message-ID: turn on the sip trace also in your log sofia profile siptrace on do that for every profile involved in the call On Tue, Jul 6, 2010 at 3:41 PM, Lars Zeb wrote: > I am getting a 408 error. I do not understand why. I am using git-1fba654 > 2010-07-01 on Centos 5.3. > > Could it be Flowroute? > > Thanks for any help, Lars > > http://pastebin.freeswitch.org/13389 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/120630ec/attachment.html From bjbrashier at gmail.com Tue Jul 6 14:06:44 2010 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 6 Jul 2010 15:06:44 -0600 Subject: [Freeswitch-users] bind_meta_app modification Message-ID: Our design needed something like bind_meta_app, but couldn't use the * key. So we made a modification to allow you to use different keys via an xml variable. I assume that this would also be useful to others. The specific change is in switch_ivr_async.c, but my understanding is that bind_meta_app is a dialplan tool... in which section should I submit this to jira? Bradley Brashier Virtual PBX From anthony.minessale at gmail.com Tue Jul 6 14:13:57 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 6 Jul 2010 16:13:57 -0500 Subject: [Freeswitch-users] bind_meta_app modification In-Reply-To: References: Message-ID: based on the file you are modifying it would be the core project. On Tue, Jul 6, 2010 at 4:06 PM, Bradley Brashier wrote: > Our design needed something like bind_meta_app, but couldn't use the * > key. So we made a modification to allow you to use different keys via > an xml variable. I assume that this would also be useful to others. > > The specific change is in switch_ivr_async.c, but my understanding is > that bind_meta_app is a dialplan tool... in which section should I > submit this to jira? > > Bradley Brashier > Virtual PBX > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/a1dc95cf/attachment.html From larclap at yahoo.com Tue Jul 6 14:43:13 2010 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 6 Jul 2010 14:43:13 -0700 Subject: [Freeswitch-users] 408 error? In-Reply-To: References: <00e501cb1d4b$8d349fa0$a79ddee0$@yahoo.com> Message-ID: <010501cb1d54$3c414540$b4c3cfc0$@yahoo.com> Seems as if Flowroute is the issue: "We are currently experiencing some termination issues and are working on resolving this. We apologize for the inconvenience." From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Tuesday, July 06, 2010 2:05 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] 408 error? turn on the sip trace also in your log sofia profile siptrace on do that for every profile involved in the call On Tue, Jul 6, 2010 at 3:41 PM, Lars Zeb wrote: I am getting a 408 error. I do not understand why. I am using git-1fba654 2010-07-01 on Centos 5.3. Could it be Flowroute? Thanks for any help, Lars http://pastebin.freeswitch.org/13389 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/9f8cefda/attachment-0001.html From xyangni at gmail.com Tue Jul 6 16:47:35 2010 From: xyangni at gmail.com (xuyan yang) Date: Wed, 7 Jul 2010 07:47:35 +0800 Subject: [Freeswitch-users] Only Eutelia not working on NAT In-Reply-To: References: Message-ID: Hi Andrea, I am not familiar with sip trace, but I have turned on the debug output and set log level to 9 in sofia_config.xml. When I placed a call FS did not show any debug or log events. I guess there may be some problem with my route's port forward function. But I actually solved this problem in a strange way. : set the ext-rtp-ip and ext-sip-ip in external.xml to a internal IP with autonet:192.168.1.8. I also found with surprise that even if I set those parameters to a internal IP of another PC (not the one running FS), it still works! On Tue, Jul 6, 2010 at 4:51 AM, bakko wrote: > Hello, > > Are you tried to look at siptrace to see if the call arrive to FS? > > BR > > - Andrea - > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/4526699b/attachment.html From brian at freeswitch.org Tue Jul 6 16:53:01 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 6 Jul 2010 18:53:01 -0500 Subject: [Freeswitch-users] Only Eutelia not working on NAT In-Reply-To: References: Message-ID: <687E1B1A-AF0B-4654-8463-1BA0210A452F@freeswitch.org> That makes little sense... its also "autonat" but those settings are when FreeSWITCH is behind nat and need to dynamically lie about the via header on a per request basis. It also uses the local-network-acl to tell what is local ie when NOT to lie and what is not.. ie when TO lie. /b On Jul 6, 2010, at 6:47 PM, xuyan yang wrote: > Hi Andrea, > > I am not familiar with sip trace, but I have turned on the debug output and set log level to 9 in sofia_config.xml. When I placed a call FS did not show any debug or log events. I guess there may be some problem with my route's port forward function. But I actually solved this problem in a strange way. > : > set the ext-rtp-ip and ext-sip-ip in external.xml to a internal IP with autonet:192.168.1.8. > I also found with surprise that even if I set those parameters to a internal IP of another PC (not the one running FS), it still works! > > From paul.gore.j at gmail.com Tue Jul 6 18:02:46 2010 From: paul.gore.j at gmail.com (paul gore) Date: Tue, 6 Jul 2010 21:02:46 -0400 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls Message-ID: Hi there, I am experimenting with FS on EC2, I like results, but stuck on weird audio issue - I followed FreeSwitch EC2 wiki article and modified internal profile and vars.xml accordingly, but unfortunately still cannot get it working. Incoming and outgoing calls made using a SIP phone to FS extensions work just fine. As well as calls to FS from PSTN. But calls to PSTN via gateways result in no audio at all, no ring, nothing, SIP signaling goes through OK. Sofia status profile shows correct values for Ext-RTP-IP for both profiles - my static public IP, RTP-IP shows local IP. Any thoughts on that? Anybody can share working profile configuration may be? Please help, I really need to get this going. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/1cb3d2eb/attachment.html From fs-list at communicatefreely.net Tue Jul 6 19:04:05 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Tue, 06 Jul 2010 22:04:05 -0400 Subject: [Freeswitch-users] VM Force Password Change In-Reply-To: References: <016301cb1d0a$f38c2e80$daa48b80$@com> <01bc01cb1d23$93213900$b963ab00$@com> Message-ID: <4C33E095.8080400@communicatefreely.net> If you use this method, what happens when a user tries to change their password from within the voicemail app? Is there some sort of hook in mod_voicemail that we can use to push the update to the database via the web server? -Tim Steven Ayre wrote: > Try using mod_xml_curl to provide the user directory. > > voicemail password is a param in the user directory, you can generate > that dynamically querying a database (e.g. MySQL) via a scripting > language (e.g. PHP) using mod_xml_curl to get the current password (or > a1 hash if you want to store it encrypted). > > You can have a dialplan extension that's executed prior to the extension > that connects them to their voicemail if there's no password set that > collects DTMF and then sends it to the webserver to update the database > using Mod_curl > > http://wiki.freeswitch.org/wiki/Mod_voicemail#vm-password > http://wiki.freeswitch.org/wiki/Mod_xml_curl > http://wiki.freeswitch.org/wiki/Mod_curl > > > > > On 6 July 2010 17:12, Ghulam Mustafa @gmail.com > wrote: > > i have not done something like this before, but you can create an > event socket script to check first time login (possibly by checking > _default_ password in db/xml) and then and then ask user to change the > password. > > check it out http://wiki.freeswitch.org/wiki/IVR > > > -m > > On Tue, Jul 6, 2010 at 8:54 PM, Peder > wrote: > > Any idea how? > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of > Ghulam Mustafa > > Sent: Tuesday, July 06, 2010 10:35 AM > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] VM Force Password Change > > > > i don't think so, but you can definitely make it possible in dialplan. > > > > -m > > > > On Tue, Jul 6, 2010 at 5:58 PM, Peder > wrote: > >> Is there a way to force a password change upon first login to a > voicemail > >> account? We set the password to 1234 and would like to force > users to > >> change it the first time they login to the account. > >> > >> Peder > >> > >> > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Ghulam Mustafa > > cell: +92 333.611.7681 > > sip: cyrenity at ekiga.net > > mail: mustafa.pk @gmail.com > > web: cyrenity.wordpress.com > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Ghulam Mustafa > cell: +92 333.611.7681 > sip: cyrenity at ekiga.net > mail: mustafa.pk @gmail.com > web: cyrenity.wordpress.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fs-list at communicatefreely.net Tue Jul 6 19:09:07 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Tue, 06 Jul 2010 22:09:07 -0400 Subject: [Freeswitch-users] mod_conference dynamic config options Message-ID: <4C33E1C3.4040508@communicatefreely.net> Hello list, I am trying to dynamically set the moh_sound variable in mod_conference, so that each customer can have a specific music on hold for their company conference room. Most of the options are common, so I have a few template profiles in the conference.conf.xml file. I can change the moh_sound variable to suit, but this would require me to create a profile for each room, then reload it. I would prefer to have this pull from a database. I'm using LUA in the dialplan to setup the conference based on the database values, but I can only see options for setting things like PIN, profile, etc. Is there any way I can set this or other settings with a dialplan variable? Failing that, can mod_conference be bound to a lua script that will give it an XML profile each time it is called? Thanks! -Tim From Nabble at slickdeals.endjunk.com Tue Jul 6 19:15:23 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Tue, 6 Jul 2010 19:15:23 -0700 (PDT) Subject: [Freeswitch-users] More questions migrating from Asterisk to FS In-Reply-To: References: <1278425102977-5260522.post@n2.nabble.com> Message-ID: <1278468923913-5263357.post@n2.nabble.com> Thank you for your speedy response. I am not sure I do understand what you said. Please kindly bear with me as FS is a new world to me. It sure will be nice to see some sample XML codes. I used this http://wiki.freeswitch.org/wiki/Provider_Configuration:_Gizmo Gizmo5 configuration. Let's say I have two Gizmo5 accounts 17471234567 and 17477654321. Then, the contents of my sip_profiles/external/gizmo5.xml file that list both gateways will have only different username/password and the rest parameters will be the same. It would be nice that the same parameters don't need to be relisted on each gateway. I hope this makes it more clear. Anthony Minessale wrote: > > params put in the domain itself would apply to every user in that domain. > so you can only put the id login and pass in your config if you wish > ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/More-questions-migrating-from-Asterisk-to-FS-tp5260522p5263357.html Sent from the freeswitch-users mailing list archive at Nabble.com. From Nabble at slickdeals.endjunk.com Tue Jul 6 19:21:52 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Tue, 6 Jul 2010 19:21:52 -0700 (PDT) Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: References: <1278204209127-5252219.post@n2.nabble.com> <1278436848678-5261486.post@n2.nabble.com> Message-ID: <1278469312148-5263368.post@n2.nabble.com> This will probably break all patches from OpenWRT for freeswitch-1.0.6. Could you make the patches against freeswitch-1.0.6? Anthony Minessale wrote: > > can you try the latest git head to see if my workaround fixes it? > ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5263368.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gerrit308 at gmail.com Tue Jul 6 20:14:25 2010 From: gerrit308 at gmail.com (humbr) Date: Tue, 6 Jul 2010 20:14:25 -0700 (PDT) Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <1278469312148-5263368.post@n2.nabble.com> References: <1278204209127-5252219.post@n2.nabble.com> <1278436848678-5261486.post@n2.nabble.com> <1278469312148-5263368.post@n2.nabble.com> Message-ID: <1278472465333-5263442.post@n2.nabble.com> Mazilo, wouldn't it be better to manually edit sofia.c with the 2 lines of code change and then generate a (temporary) patch for OpenWRT? Thi would solve the issue until the OpenWRT version re-aligns with the current git head of FS. Gerrit -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5263442.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gerrit308 at gmail.com Tue Jul 6 20:20:42 2010 From: gerrit308 at gmail.com (humbr) Date: Tue, 6 Jul 2010 20:20:42 -0700 (PDT) Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <1278472465333-5263442.post@n2.nabble.com> References: <1278204209127-5252219.post@n2.nabble.com> <1278436848678-5261486.post@n2.nabble.com> <1278469312148-5263368.post@n2.nabble.com> <1278472465333-5263442.post@n2.nabble.com> Message-ID: <1278472842929-5263450.post@n2.nabble.com> Just to make it easier:-) here is the link to http://fisheye.freeswitch.org/browse/freeswitch-git/src/mod/endpoints/mod_sofia/sofia.c?r2=088cee65f95c599914be73511095ce4dd9a7feba&r1=f2ea3ee315621f5c345a8fdade7e018990dc4c98 fisheye for the code change. Gerrit humbr wrote: > > Mazilo, wouldn't it be better to manually edit sofia.c with the 2 lines of > code change and then generate a (temporary) patch for OpenWRT? Thi would > solve the issue until the OpenWRT version re-aligns with the current git > head of FS. > > Gerrit > -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5263450.html Sent from the freeswitch-users mailing list archive at Nabble.com. From tony.tin at noahmedia.com.hk Tue Jul 6 22:35:47 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Wed, 7 Jul 2010 13:35:47 +0800 Subject: [Freeswitch-users] originate call hangup signal Message-ID: Hi, When I use OpenZAP channel to originate a call, after the called party hangup the phone. It takes freeswitch around 40 seconds to catch the hangup signal and stop the dial plan. I'm wondering whether there is a way to shorten that duration. Thanks. Regards, Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/5173866b/attachment.html From nagalenoj at gmail.com Tue Jul 6 22:47:42 2010 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Wed, 7 Jul 2010 11:17:42 +0530 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: After git pull, the behavior remains the same. When I check the console log, the leg has got disconnected after leg_timeout seconds but the session for the leg has got closed after the script exists. 2010-07-07 05:27:39.156085 [INFO] mod_dialplan_xml.c:331 Processing 1005->212 in context default 2010-07-07 05:27:39.194359 [NOTICE] mod_dptools.c:746 Channel [sofia/internal/1005 at 192.168.1.72] has been answered 2010-07-07 05:27:42.708162 [NOTICE] sofia.c:4875 Channel [sofia/internal/ sip:1000 at 192.168.6.114 ] has been answered 2010-07-07 05:27:49.020053 [INFO] mod_dptools.c:2393 Originate Failed. Cause: NO_ANSWER 2010-07-07 05:28:12.761902 [NOTICE] switch_core_session.c:1193 Session 6 (sofia/internal/sip:1000 at 192.168.6.114 ) Ended 2010-07-07 05:28:12.761902 [NOTICE] switch_core_session.c:1195 Close Channel sofia/internal/sip:1000 at 192.168.6.114 [CS_DESTROY] Here is the console log, http://pastebin.freeswitch.org/13395 On Tue, Jul 6, 2010 at 8:56 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > try latest GIT, that was an edge case and it's fixed. > > > On Tue, Jul 6, 2010 at 1:58 AM, Nagalenoj H. wrote: > >> Got core dump when I execute the bridge as >> execute-app-arg: >> {user_recurse_variables=false,group_confirm_cancel_timeout=true}[leg_timeout=10]user/1000 >> >> Console log is here, >> http://pastebin.freeswitch.org/13371 >> >> core dump backtrace is here, >> http://pastebin.freeswitch.org/13372 >> >> Also, tried after git pull, but the result is same. >> >> >> On Mon, Jul 5, 2010 at 8:53 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> you also need user_recurse_variables=false >>> >>> {user_recurse_variables=false,group_confirm_cancel_timeout=true} >>> >>> >>> >>> On Sat, Jul 3, 2010 at 8:22 AM, Nagalenoj H. wrote: >>> >>>> Here is the log got from the latest GIT source. >>>> >>>> http://pastebin.freeswitch.org/13347 >>>> >>>> >>>> On Fri, Jul 2, 2010 at 10:15 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> update and reproduce that same log with latest GIT the version you are >>>>> using has an issue. >>>>> >>>>> >>>>> On Fri, Jul 2, 2010 at 12:25 AM, Nagalenoj H. wrote: >>>>> >>>>>> I've pasted the console log here, >>>>>> >>>>>> http://pastebin.freeswitch.org/13333 >>>>>> >>>>>> >>>>>> On Wed, Jun 30, 2010 at 11:15 PM, Michael Collins >>>>> > wrote: >>>>>> >>>>>>> Can you supply a console log of these calls? >>>>>>> -MC >>>>>>> >>>>>>> >>>>>>> On Wed, Jun 30, 2010 at 7:37 AM, Nagalenoj H. wrote: >>>>>>> >>>>>>>> Dear Anthony, >>>>>>>> I've tried using the group_confirm_cancel_timeout as per the >>>>>>>> discussion we had in IRC. You wanted to used it as part of dial string and >>>>>>>> not as a channel variable. >>>>>>>> But, It doesn't work for me. >>>>>>>> >>>>>>>> Here is how I've given the commands and the script I've executed. >>>>>>>> Even when I give group_confirm_cancel_timeout, the callee's leg is getting >>>>>>>> disconnected after legtimeout. >>>>>>>> >>>>>>>> >>>>>>>> connect >>>>>>>> >>>>>>>> sendmsg >>>>>>>> call-command: execute >>>>>>>> execute-app-name:answer >>>>>>>> >>>>>>>> sendmsg >>>>>>>> call-command: execute >>>>>>>> execute-app-name: set >>>>>>>> execute-app-arg: group_confirm_key=exec >>>>>>>> >>>>>>>> sendmsg >>>>>>>> call-command: execute >>>>>>>> execute-app-name: set >>>>>>>> execute-app-arg: group_confirm_file=perl /root/bridge.pl >>>>>>>> >>>>>>>> sendmsg >>>>>>>> call-command: execute >>>>>>>> execute-app-name: bridge >>>>>>>> execute-app-arg: >>>>>>>> {group_confirm_cancel_timeout=1}[leg_timeout=10]user/1005 >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> bridge.pl: >>>>>>>> #!/usr/bin/perl >>>>>>>> use freeswitch; >>>>>>>> >>>>>>>> our $session; >>>>>>>> freeswitch::consoleLog("info","Goint to get the digits"); >>>>>>>> # To simulate the scenario I used sleep here. >>>>>>>> sleep(30); >>>>>>>> 1; >>>>>>>> >>>>>>>> Kindly tell me whats wrong in the above. >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> Nagalenoj H. >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 at hotmail.com >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> pstn:+19193869900 >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> -- >>>> Regards, >>>> Nagalenoj H. >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> -- >> Regards, >> Nagalenoj H. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/7317efe5/attachment.html From tony.tin at noahmedia.com.hk Wed Jul 7 01:48:40 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Wed, 7 Jul 2010 16:48:40 +0800 Subject: [Freeswitch-users] hangup does not break the Lua loop Message-ID: Hi, I'm writing dialplan with Lua script. There is a while loop in the dialplan, I found that the loop is not broke out even the call has been hung up. I'm wondering whether this is a normal behavior. Thanks. Regards, Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/53cc061c/attachment-0001.html From ali.stgt at gmail.com Wed Jul 7 05:03:11 2010 From: ali.stgt at gmail.com (=?UTF-8?B?RHVybXXFnyBBbGkgw5Z6dMO8cms=?=) Date: Wed, 7 Jul 2010 15:03:11 +0300 Subject: [Freeswitch-users] Sound not OK (Choppy Sound) while using ManagedSession (.NET/C#) Message-ID: Hello, I have successfully entegrated a dll-module (written udner .Net / c#) in fs which is loadable by the mod_managed component. Now, if I try to stream a wav file to the callee then the sound is very poor and choppy (also the file is played slower). The fs is running under Windows XP and tested under Windows 2003 with the same result. Streaming by using FreeSWITCH.Native.Api() works without problems, sound is perfect. Samples which I have tried: 1 - This works fine but delegate mechanism is missing there: FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); string uuid = fsApi.ExecuteString("create_uuid"); string apiResult = fsApi.Execute("originate", string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/Test/{2} &&playback({3}", uuid, callerID, phoneNumber, wavFile)); 2 - ManagedSession is integrated but we have sound problems: FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); string uuid = fsApi.ExecuteString("create_uuid"); string apiResult = fsApi.Execute("originate", string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/Test/{2} &park", uuid, callerID, phoneNumber)); FreeSWITCH.Native.ManagedSession blegSession = new FreeSWITCH.Native.ManagedSession(uuid); if (blegSession.IsAvailable) { ??? while (!blegSession.answered()) ??????? { ??? ??? blegSession.sleep(500, 1); ??? } ??????? if (blegSession.Ready() && blegSession.mediaReady()) ??????? { ??? ??? //blegSession.Answer(); ??? ??? //blegSession.Execute("playback",wavFile); ??????????????? blegSession.StreamFile(wavFile, 0); ??????? } ??????? blegSession.Hangup("Normal call clearing"); } Is there a mistake in my code or do I have forgotten something? Can you support me with some examples? Another issue is, that I am not able to make a origination directly by ManagedSession or CoreSession without using the FreeSWITCH.Native.Api(). How can I create an instance of a CoreSession object? Many thanks for you help in advance. Ali From brian at freeswitch.org Wed Jul 7 05:15:25 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 7 Jul 2010 07:15:25 -0500 Subject: [Freeswitch-users] hangup does not break the Lua loop In-Reply-To: References: Message-ID: <4B80CF4E-6336-4F2B-A118-ED87D6979F90@freeswitch.org> Post your code for your while loop... sounds like you're not checking session:ready to see if its still true. /b On Jul 7, 2010, at 3:48 AM, Tony Tin wrote: > Hi, > > I'm writing dialplan with Lua script. There is a while loop in the dialplan, I found that the loop is not broke out even the call has been hung up. I'm wondering whether this is a normal behavior. Thanks. > > Regards, > Tony From Nabble at slickdeals.endjunk.com Wed Jul 7 06:02:08 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Wed, 7 Jul 2010 06:02:08 -0700 (PDT) Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <1278472842929-5263450.post@n2.nabble.com> References: <1278204209127-5252219.post@n2.nabble.com> <1278436848678-5261486.post@n2.nabble.com> <1278469312148-5263368.post@n2.nabble.com> <1278472465333-5263442.post@n2.nabble.com> <1278472842929-5263450.post@n2.nabble.com> Message-ID: <1278507728660-5265110.post@n2.nabble.com> Thanks for the http://fisheye.freeswitch.org/browse/freeswitch-git/src/mod/endpoints/mod_sofia/sofia.c?r2=088cee65f95c599914be73511095ce4dd9a7feba&r1=f2ea3ee315621f5c345a8fdade7e018990dc4c98 fisheye link and that's what I was looking for. As you said, all I did was to change the following line: if (ap) { to: if (ap_ptr) { and the compilation crashed as shown below: make[7]: Entering directory `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.1_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' > make[8]: Entering directory > `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.1_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' > Creating mod_sofia_la-sofia.lo > Compiling sofia.c ... > sofia.c: In function 'logger': > sofia.c:1574: error: 'ap_ptr' undeclared (first use in this function) > sofia.c:1574: error: (Each undeclared identifier is reported only once > sofia.c:1574: error: for each function it appears in.) > make[8]: *** [mod_sofia_la-sofia.lo] Error 1 > make[8]: Leaving directory > `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.1_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' Perhaps, the patch won't work on a freeswitch-1.0.6. Oh well, at least we tried. humbr wrote: > > Just to make it easier:-) here is the link to > http://fisheye.freeswitch.org/browse/freeswitch-git/src/mod/endpoints/mod_sofia/sofia.c?r2=088cee65f95c599914be73511095ce4dd9a7feba&r1=f2ea3ee315621f5c345a8fdade7e018990dc4c98 > fisheye for the code change. > > Gerrit ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5265110.html Sent from the freeswitch-users mailing list archive at Nabble.com. From pjintheusa at gmail.com Wed Jul 7 06:30:25 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Wed, 7 Jul 2010 09:30:25 -0400 Subject: [Freeswitch-users] Sound not OK (Choppy Sound) while using ManagedSession (.NET/C#) In-Reply-To: References: Message-ID: In example two you are parking the call. You are then playing media into a parked call. I am not sure this is valid. According to the wiki "Please note that to retrieve a call that has been "parked", you'll have to bridgeto them or transfer the call to a valid location." - so you might need to transfer that parked call to a DialPlan App before you try and play media - or just not park it in the first place. I might be wrong - but that where I would start. On Wed, Jul 7, 2010 at 8:03 AM, Durmu? Ali ?zt?rk wrote: > Hello, > > I have successfully entegrated a dll-module (written udner .Net / c#) > in fs which is loadable by the mod_managed component. > > Now, if I try to stream a wav file to the callee then the sound is > very poor and choppy (also the file is played slower). > > The fs is running under Windows XP and tested under Windows 2003 with > the same result. > > Streaming by using FreeSWITCH.Native.Api() works without problems, > sound is perfect. > > Samples which I have tried: > > > > 1 - This works fine but delegate mechanism is missing there: > > FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); > > string uuid = fsApi.ExecuteString("create_uuid"); > string apiResult = fsApi.Execute("originate", > > string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/Test/{2} > &&playback({3}", uuid, callerID, phoneNumber, wavFile)); > > > > > 2 - ManagedSession is integrated but we have sound problems: > > FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); > > string uuid = fsApi.ExecuteString("create_uuid"); > > string apiResult = fsApi.Execute("originate", > > string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/Test/{2} > &park", uuid, callerID, phoneNumber)); > > FreeSWITCH.Native.ManagedSession blegSession = new > FreeSWITCH.Native.ManagedSession(uuid); > > if (blegSession.IsAvailable) > { > while (!blegSession.answered()) > { > blegSession.sleep(500, 1); > } > > if (blegSession.Ready() && blegSession.mediaReady()) > { > //blegSession.Answer(); > //blegSession.Execute("playback",wavFile); > blegSession.StreamFile(wavFile, 0); > } > > blegSession.Hangup("Normal call clearing"); > } > > > > > Is there a mistake in my code or do I have forgotten something? Can > you support me with some examples? > > > Another issue is, that I am not able to make a origination directly by > ManagedSession or CoreSession without using the > FreeSWITCH.Native.Api(). How can I create an instance of a CoreSession > object? > > > Many thanks for you help in advance. > > Ali > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/08245321/attachment.html From anthony.minessale at gmail.com Wed Jul 7 06:30:33 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Jul 2010 08:30:33 -0500 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: Message-ID: firewall or iptables On Tue, Jul 6, 2010 at 8:02 PM, paul gore wrote: > Hi there, > I am experimenting with FS on EC2, I like results, but stuck on weird audio > issue - I followed FreeSwitch EC2 wiki article and modified internal profile > and vars.xml accordingly, but unfortunately still cannot get it working. > Incoming and outgoing calls made using a SIP phone to FS extensions work > just fine. As well as calls to FS from PSTN. But calls to PSTN via gateways > result in no audio at all, no ring, nothing, SIP signaling goes through OK. > Sofia status profile shows correct values for Ext-RTP-IP for both profiles > - my static public IP, RTP-IP shows local IP. > Any thoughts on that? Anybody can share working profile configuration may > be? > Please help, I really need to get this going. > > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/b3010c82/attachment.html From sos at sokhapkin.dyndns.org Wed Jul 7 06:32:42 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 7 Jul 2010 09:32:42 -0400 Subject: [Freeswitch-users] mod_xml_curl question. Message-ID: <201007070932.42832.sos@sokhapkin.dyndns.org> I want to use mod_xml_curl to retrieve only dynamic dialplan sections, but to keep a large chunk of dialplan code in a static XML file to minimize the number of http requests. But when mod_xml_curl is loaded, dialplan search/execution starts with xml_curl execution. How to avoid that? I need static xml to be looked up/executed first, and retrieve dialplan context from web server only if the required dialplan context is not defined in static XML file. From anthony.minessale at gmail.com Wed Jul 7 06:34:58 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Jul 2010 08:34:58 -0500 Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <1278507728660-5265110.post@n2.nabble.com> References: <1278204209127-5252219.post@n2.nabble.com> <1278436848678-5261486.post@n2.nabble.com> <1278469312148-5263368.post@n2.nabble.com> <1278472465333-5263442.post@n2.nabble.com> <1278472842929-5263450.post@n2.nabble.com> <1278507728660-5265110.post@n2.nabble.com> Message-ID: you forgot part of the patch =D The following 2 lines should be the first 2 lines in the function. /* gcc 4.4 gets mad at us for testing if (ap) so let's try to work around it....*/ void *ap_ptr = (void *) (intptr_t) ap; On Wed, Jul 7, 2010 at 8:02 AM, mazilo wrote: > > Thanks for the > > http://fisheye.freeswitch.org/browse/freeswitch-git/src/mod/endpoints/mod_sofia/sofia.c?r2=088cee65f95c599914be73511095ce4dd9a7feba&r1=f2ea3ee315621f5c345a8fdade7e018990dc4c98 > fisheye link and that's what I was looking for. As you said, all I did was > to change the following line: > > > if (ap) { > to: > > if (ap_ptr) { > and the compilation crashed as shown below: > > make[7]: Entering directory > > `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.1_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' > > make[8]: Entering directory > > > `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.1_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' > > Creating mod_sofia_la-sofia.lo > > Compiling sofia.c ... > > sofia.c: In function 'logger': > > sofia.c:1574: error: 'ap_ptr' undeclared (first use in this function) > > sofia.c:1574: error: (Each undeclared identifier is reported only once > > sofia.c:1574: error: for each function it appears in.) > > make[8]: *** [mod_sofia_la-sofia.lo] Error 1 > > make[8]: Leaving directory > > > `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.1_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' > Perhaps, the patch won't work on a freeswitch-1.0.6. Oh well, at least we > tried. > > > humbr wrote: > > > > Just to make it easier:-) here is the link to > > > http://fisheye.freeswitch.org/browse/freeswitch-git/src/mod/endpoints/mod_sofia/sofia.c?r2=088cee65f95c599914be73511095ce4dd9a7feba&r1=f2ea3ee315621f5c345a8fdade7e018990dc4c98 > > fisheye for the code change. > > > > Gerrit > > > ----- > don't and stop are the ONLY two 4-letter words considered offensive to men, > but not when used together. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5265110.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/c8299503/attachment-0001.html From brian at freeswitch.org Wed Jul 7 06:36:55 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 7 Jul 2010 08:36:55 -0500 Subject: [Freeswitch-users] Sound not OK (Choppy Sound) while using ManagedSession (.NET/C#) In-Reply-To: References: Message-ID: <23C37CC1-6465-47D9-ACC2-C64F2F86826B@freeswitch.org> When a call is parked you can send commands to it to do anything you want. /b On Jul 7, 2010, at 8:30 AM, Phillip Jones wrote: > In example two you are parking the call. You are then playing media into a parked call. I am not sure this is valid. According to the wiki "Please note that to retrieve a call that has been "parked", you'll have to bridge to them or transfer the call to a valid location." - so you might need to transfer that parked call to a DialPlan App before you try and play media - or just not park it in the first place. > > I might be wrong - but that where I would start. From anthony.minessale at gmail.com Wed Jul 7 06:41:03 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Jul 2010 08:41:03 -0500 Subject: [Freeswitch-users] Sound not OK (Choppy Sound) while using ManagedSession (.NET/C#) In-Reply-To: References: Message-ID: instead of originate to park then sending the uuid to the constructor try supplying the dial string to the constructor which will place the call On Wed, Jul 7, 2010 at 8:30 AM, Phillip Jones wrote: > In example two you are parking the call. You are then playing media into a > parked call. I am not sure this is valid. According to the wiki "Please note > that to retrieve a call that has been "parked", you'll have to bridgeto them or transfer the call to a valid location." - so you might need to > transfer that parked call to a DialPlan App before you try and play media - > or just not park it in the first place. > > I might be wrong - but that where I would start. > > > On Wed, Jul 7, 2010 at 8:03 AM, Durmu? Ali ?zt?rk wrote: > >> Hello, >> >> I have successfully entegrated a dll-module (written udner .Net / c#) >> in fs which is loadable by the mod_managed component. >> >> Now, if I try to stream a wav file to the callee then the sound is >> very poor and choppy (also the file is played slower). >> >> The fs is running under Windows XP and tested under Windows 2003 with >> the same result. >> >> Streaming by using FreeSWITCH.Native.Api() works without problems, >> sound is perfect. >> >> Samples which I have tried: >> >> >> >> 1 - This works fine but delegate mechanism is missing there: >> >> FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); >> >> string uuid = fsApi.ExecuteString("create_uuid"); >> string apiResult = fsApi.Execute("originate", >> >> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/Test/{2} >> &&playback({3}", uuid, callerID, phoneNumber, wavFile)); >> >> >> >> >> 2 - ManagedSession is integrated but we have sound problems: >> >> FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); >> >> string uuid = fsApi.ExecuteString("create_uuid"); >> >> string apiResult = fsApi.Execute("originate", >> >> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/Test/{2} >> &park", uuid, callerID, phoneNumber)); >> >> FreeSWITCH.Native.ManagedSession blegSession = new >> FreeSWITCH.Native.ManagedSession(uuid); >> >> if (blegSession.IsAvailable) >> { >> while (!blegSession.answered()) >> { >> blegSession.sleep(500, 1); >> } >> >> if (blegSession.Ready() && blegSession.mediaReady()) >> { >> //blegSession.Answer(); >> //blegSession.Execute("playback",wavFile); >> blegSession.StreamFile(wavFile, 0); >> } >> >> blegSession.Hangup("Normal call clearing"); >> } >> >> >> >> >> Is there a mistake in my code or do I have forgotten something? Can >> you support me with some examples? >> >> >> Another issue is, that I am not able to make a origination directly by >> ManagedSession or CoreSession without using the >> FreeSWITCH.Native.Api(). How can I create an instance of a CoreSession >> object? >> >> >> Many thanks for you help in advance. >> >> Ali >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/a9385a4b/attachment.html From anthony.minessale at gmail.com Wed Jul 7 06:46:08 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Jul 2010 08:46:08 -0500 Subject: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout In-Reply-To: References: Message-ID: This is not at all the same. in your script you have to do one of the following: if they dialed the right digits answer the channel, if not hangup the channel. you are doing neither and originate is hanging up for you because you did not answer the channel. Can we be done with this thread now? it's trying my patience. On Wed, Jul 7, 2010 at 12:47 AM, Nagalenoj H. wrote: > After git pull, the behavior remains the same. > > When I check the console log, the leg has got disconnected after > leg_timeout seconds but the session for the leg has got closed after the > script exists. > > 2010-07-07 05:27:39.156085 [INFO] mod_dialplan_xml.c:331 Processing > 1005->212 in context default > 2010-07-07 05:27:39.194359 [NOTICE] mod_dptools.c:746 Channel > [sofia/internal/1005 at 192.168.1.72] has been answered > 2010-07-07 05:27:42.708162 [NOTICE] sofia.c:4875 Channel [sofia/internal/ > sip:1000 at 192.168.6.114 ] has been answered > 2010-07-07 05:27:49.020053 [INFO] mod_dptools.c:2393 Originate Failed. > Cause: NO_ANSWER > 2010-07-07 05:28:12.761902 [NOTICE] switch_core_session.c:1193 Session 6 > (sofia/internal/sip:1000 at 192.168.6.114 ) Ended > 2010-07-07 05:28:12.761902 [NOTICE] switch_core_session.c:1195 Close > Channel sofia/internal/sip:1000 at 192.168.6.114 [CS_DESTROY] > > Here is the console log, > http://pastebin.freeswitch.org/13395 > > > On Tue, Jul 6, 2010 at 8:56 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> try latest GIT, that was an edge case and it's fixed. >> >> >> On Tue, Jul 6, 2010 at 1:58 AM, Nagalenoj H. wrote: >> >>> Got core dump when I execute the bridge as >>> execute-app-arg: >>> {user_recurse_variables=false,group_confirm_cancel_timeout=true}[leg_timeout=10]user/1000 >>> >>> Console log is here, >>> http://pastebin.freeswitch.org/13371 >>> >>> core dump backtrace is here, >>> http://pastebin.freeswitch.org/13372 >>> >>> Also, tried after git pull, but the result is same. >>> >>> >>> On Mon, Jul 5, 2010 at 8:53 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> you also need user_recurse_variables=false >>>> >>>> {user_recurse_variables=false,group_confirm_cancel_timeout=true} >>>> >>>> >>>> >>>> On Sat, Jul 3, 2010 at 8:22 AM, Nagalenoj H. wrote: >>>> >>>>> Here is the log got from the latest GIT source. >>>>> >>>>> http://pastebin.freeswitch.org/13347 >>>>> >>>>> >>>>> On Fri, Jul 2, 2010 at 10:15 PM, Anthony Minessale < >>>>> anthony.minessale at gmail.com> wrote: >>>>> >>>>>> update and reproduce that same log with latest GIT the version you are >>>>>> using has an issue. >>>>>> >>>>>> >>>>>> On Fri, Jul 2, 2010 at 12:25 AM, Nagalenoj H. wrote: >>>>>> >>>>>>> I've pasted the console log here, >>>>>>> >>>>>>> http://pastebin.freeswitch.org/13333 >>>>>>> >>>>>>> >>>>>>> On Wed, Jun 30, 2010 at 11:15 PM, Michael Collins < >>>>>>> msc at freeswitch.org> wrote: >>>>>>> >>>>>>>> Can you supply a console log of these calls? >>>>>>>> -MC >>>>>>>> >>>>>>>> >>>>>>>> On Wed, Jun 30, 2010 at 7:37 AM, Nagalenoj H. wrote: >>>>>>>> >>>>>>>>> Dear Anthony, >>>>>>>>> I've tried using the group_confirm_cancel_timeout as per the >>>>>>>>> discussion we had in IRC. You wanted to used it as part of dial string and >>>>>>>>> not as a channel variable. >>>>>>>>> But, It doesn't work for me. >>>>>>>>> >>>>>>>>> Here is how I've given the commands and the script I've executed. >>>>>>>>> Even when I give group_confirm_cancel_timeout, the callee's leg is getting >>>>>>>>> disconnected after legtimeout. >>>>>>>>> >>>>>>>>> >>>>>>>>> connect >>>>>>>>> >>>>>>>>> sendmsg >>>>>>>>> call-command: execute >>>>>>>>> execute-app-name:answer >>>>>>>>> >>>>>>>>> sendmsg >>>>>>>>> call-command: execute >>>>>>>>> execute-app-name: set >>>>>>>>> execute-app-arg: group_confirm_key=exec >>>>>>>>> >>>>>>>>> sendmsg >>>>>>>>> call-command: execute >>>>>>>>> execute-app-name: set >>>>>>>>> execute-app-arg: group_confirm_file=perl /root/bridge.pl >>>>>>>>> >>>>>>>>> sendmsg >>>>>>>>> call-command: execute >>>>>>>>> execute-app-name: bridge >>>>>>>>> execute-app-arg: >>>>>>>>> {group_confirm_cancel_timeout=1}[leg_timeout=10]user/1005 >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> bridge.pl: >>>>>>>>> #!/usr/bin/perl >>>>>>>>> use freeswitch; >>>>>>>>> >>>>>>>>> our $session; >>>>>>>>> freeswitch::consoleLog("info","Goint to get the digits"); >>>>>>>>> # To simulate the scenario I used sleep here. >>>>>>>>> sleep(30); >>>>>>>>> 1; >>>>>>>>> >>>>>>>>> Kindly tell me whats wrong in the above. >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Regards, >>>>>>> Nagalenoj H. >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at 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 at hotmail.com >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>> IRC: irc.freenode.net #freeswitch >>>>>> >>>>>> FreeSWITCH Developer Conference >>>>>> sip:888 at conference.freeswitch.org >>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>> pstn:+19193869900 >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> -- >>>>> Regards, >>>>> Nagalenoj H. >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> -- >>> Regards, >>> Nagalenoj H. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -- > Regards, > Nagalenoj H. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/374741ab/attachment-0001.html From thakkar.jigar at gmail.com Mon Jul 5 23:31:52 2010 From: thakkar.jigar at gmail.com (Jigar Thakkar) Date: Tue, 6 Jul 2010 12:01:52 +0530 Subject: [Freeswitch-users] Two FreeSWITCH Box IVR Call Transfer Message-ID: Hi Guys, I am a beginner to FreeSWITCH. I have few questions. 1. What is the difference between BRIDGE, ORIGINATE and TRANSFER. 2. I have two FreeSWITCH box. BoxA and BoxB 1. User will make a call to BoxA; BoxA's IVR collect some information i.e. userid and pin. 2. Now BoxA will forward/transfer/bridge call to BoxB's IVR ( here i want to pass userid as channel variable) 3. Box B will ask for another pin i.e. pin2, verifies the same and pass whether pin2 is correct or not to BoxA (also transfers the call to boxA's ivr) 4. Box A shall play message to User whether this process is successful or not. And then call shall be hanged up. My work till date is 1. setup the two freeswitch ivr, box a and box b. 2. I wrote .net code to handle the individual role. i.e. to ask user id and pin 3. I facing the problem in forwarding a call from boxA to boxB along with the userID Please suggest the best way to implement it. I hope I have explained the scenario properly. Thanks & Regards, Jigs. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100706/d7e40883/attachment.html From thakkar.jigar at gmail.com Tue Jul 6 22:41:18 2010 From: thakkar.jigar at gmail.com (Jigar Thakkar) Date: Wed, 7 Jul 2010 11:11:18 +0530 Subject: [Freeswitch-users] Two FreeSWITCH Box IVR Call Transfer In-Reply-To: References: Message-ID: Hi, I have tried this to forward a call from BoxA to BoxB; 1. session.Execute("bridge", [user_id=123456]sofia/internal/5111 at BoxB) It forwards the call successfully; BoxB answer the call and does the process but here I don't see the variable "user_id" value I set. session.GetVariable("user_id") returns empty string. 2. I also tried session.Execute ("originate",[user_id=123456]sofia/internal/5111 at BoxB) In this case it creates a new Channel successfully but I don't see any call request to BoxB; It just creates a new channel successfully and after some time call gets cleared. Please Help me regarding this; i want to know what is the difference between Bridge and Originate; and Also Let me know the best possible scenario to implement the following mentioned scenario. Thanks & Regards, Jigs. On Tue, Jul 6, 2010 at 12:01 PM, Jigar Thakkar wrote: > Hi Guys, > > I am a beginner to FreeSWITCH. I have few questions. > > 1. What is the difference between BRIDGE, ORIGINATE and TRANSFER. > 2. I have two FreeSWITCH box. BoxA and BoxB > 1. User will make a call to BoxA; BoxA's IVR collect some information > i.e. userid and pin. > 2. Now BoxA will forward/transfer/bridge call to BoxB's IVR ( here i > want to pass userid as channel variable) > 3. Box B will ask for another pin i.e. pin2, verifies the same and pass > whether pin2 is correct or not to BoxA (also transfers the call to boxA's > ivr) > 4. Box A shall play message to User whether this process is successful > or not. And then call shall be hanged up. > > My work till date is > 1. setup the two freeswitch ivr, box a and box b. > 2. I wrote .net code to handle the individual role. i.e. to ask user id > and pin > 3. I facing the problem in forwarding a call from boxA to boxB along > with the userID > Please suggest the best way to implement it. I hope I have explained the > scenario properly. > > Thanks & Regards, > > Jigs. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/58fe4d3c/attachment.html From thakkar.jigar at gmail.com Wed Jul 7 06:00:19 2010 From: thakkar.jigar at gmail.com (Jigar Thakkar) Date: Wed, 7 Jul 2010 18:30:19 +0530 Subject: [Freeswitch-users] Two FreeSWITCH Box IVR Call Transfer In-Reply-To: References: Message-ID: I am sorry to send this message again. But i dont see my mails in archive so I changed membership settings and tried to send it again. Thanks. Jigs ---------- Forwarded message ---------- From: Jigar Thakkar Date: Wed, Jul 7, 2010 at 11:11 AM Subject: Re: Two FreeSWITCH Box IVR Call Transfer To: freeswitch-users at lists.freeswitch.org Hi, I have tried this to forward a call from BoxA to BoxB; 1. session.Execute("bridge", [user_id=123456]sofia/internal/5111 at BoxB) It forwards the call successfully; BoxB answer the call and does the process but here I don't see the variable "user_id" value I set. session.GetVariable("user_id") returns empty string. 2. I also tried session.Execute ("originate",[user_id=123456]sofia/internal/5111 at BoxB) In this case it creates a new Channel successfully but I don't see any call request to BoxB; It just creates a new channel successfully and after some time call gets cleared. Please Help me regarding this; i want to know what is the difference between Bridge and Originate; and Also Let me know the best possible scenario to implement the following mentioned scenario. Thanks & Regards, Jigs. On Tue, Jul 6, 2010 at 12:01 PM, Jigar Thakkar wrote: > Hi Guys, > > I am a beginner to FreeSWITCH. I have few questions. > > 1. What is the difference between BRIDGE, ORIGINATE and TRANSFER. > 2. I have two FreeSWITCH box. BoxA and BoxB > 1. User will make a call to BoxA; BoxA's IVR collect some information > i.e. userid and pin. > 2. Now BoxA will forward/transfer/bridge call to BoxB's IVR ( here i > want to pass userid as channel variable) > 3. Box B will ask for another pin i.e. pin2, verifies the same and pass > whether pin2 is correct or not to BoxA (also transfers the call to boxA's > ivr) > 4. Box A shall play message to User whether this process is successful > or not. And then call shall be hanged up. > > My work till date is > 1. setup the two freeswitch ivr, box a and box b. > 2. I wrote .net code to handle the individual role. i.e. to ask user id > and pin > 3. I facing the problem in forwarding a call from boxA to boxB along > with the userID > Please suggest the best way to implement it. I hope I have explained the > scenario properly. > > Thanks & Regards, > > Jigs. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/557ea4c7/attachment.html From saeedahmad1981 at gmail.com Wed Jul 7 07:02:37 2010 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Wed, 7 Jul 2010 16:02:37 +0200 Subject: [Freeswitch-users] Mod_h323 AliasAddress Error Message-ID: Dear All, I am trying to configure mod_h323 with freeswitch, while loading module i receive that error: 2010-07-07 16:01:02.027642 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_h26x] 2010-07-07 16:01:02.304187 [CONSOLE] mod_h323.cpp:74 Starting loading mod_h323 Assertion fail: Must have non-empty string in AliasAddress!, file h323ep.cxx, line 3586 bort, ore dump? I am following that wiki : http://wiki.freeswitch.org/wiki/Mod_h323 my h323.conf.xml is: I would appreciate any help. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/65cd5f8a/attachment.html From thakkar.jigar at gmail.com Wed Jul 7 07:07:14 2010 From: thakkar.jigar at gmail.com (Jigar Thakkar) Date: Wed, 7 Jul 2010 19:37:14 +0530 Subject: [Freeswitch-users] Two FreeSWITCH Box IVR Call Transfer In-Reply-To: References: Message-ID: In logs (Info) at BoxA before executing Bridge; I can see that it has set user_id=123456 (variable_user_id : [123456]) but I don't see this variable at BoxB. I also tried the same from dialplan; BoxA Here I can see the user_id is set like ( variable_user_id : [123456] ) BoxB Here in this Info i dont see user_id variable How can i do get the user_id variable value to BoxB? Please help. Thanks Jigs. On Wed, Jul 7, 2010 at 11:11 AM, Jigar Thakkar wrote: > Hi, > > I have tried this to forward a call from BoxA to BoxB; > > 1. session.Execute("bridge", [user_id=123456]sofia/internal/5111 at BoxB) > > It forwards the call successfully; BoxB answer the call and does the > process but here I don't see the variable "user_id" value I set. > session.GetVariable("user_id") returns empty string. > > 2. I also tried session.Execute > ("originate",[user_id=123456]sofia/internal/5111 at BoxB) > > In this case it creates a new Channel successfully but I don't see any > call request to BoxB; It just creates a new channel successfully and after > some time call gets cleared. > > Please Help me regarding this; i want to know what is the difference > between Bridge and Originate; and Also Let me know the best possible > scenario to implement the following mentioned scenario. > > Thanks & Regards, > > Jigs. > > > On Tue, Jul 6, 2010 at 12:01 PM, Jigar Thakkar wrote: > >> Hi Guys, >> >> I am a beginner to FreeSWITCH. I have few questions. >> >> 1. What is the difference between BRIDGE, ORIGINATE and TRANSFER. >> 2. I have two FreeSWITCH box. BoxA and BoxB >> 1. User will make a call to BoxA; BoxA's IVR collect some information >> i.e. userid and pin. >> 2. Now BoxA will forward/transfer/bridge call to BoxB's IVR ( here i >> want to pass userid as channel variable) >> 3. Box B will ask for another pin i.e. pin2, verifies the same and >> pass whether pin2 is correct or not to BoxA (also transfers the call to >> boxA's ivr) >> 4. Box A shall play message to User whether this process is >> successful or not. And then call shall be hanged up. >> >> My work till date is >> 1. setup the two freeswitch ivr, box a and box b. >> 2. I wrote .net code to handle the individual role. i.e. to ask user id >> and pin >> 3. I facing the problem in forwarding a call from boxA to boxB along >> with the userID >> Please suggest the best way to implement it. I hope I have explained the >> scenario properly. >> >> Thanks & Regards, >> >> Jigs. >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/207a1dd5/attachment-0001.html From david.ponzone at ipeva.fr Wed Jul 7 07:20:55 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 7 Jul 2010 16:20:55 +0200 Subject: [Freeswitch-users] G729 validator not working on Debian 4.0 Message-ID: Hi Everyone, I am trying to install the G729 codec on a Debian 4.0. On 2 different boxes, validator fails with: Floating point exception One box is a 64bit Debian 4.0 running on Intel(R) Core(TM)2 Duo CPU E6550 @ 2.33GHz. The other one is a 32 bit Debian 4.0 running on dual Intel(R) Xeon(TM) CPU 2.80GHz. Both boxes have been installed by totally different people, so there is no chance that they had been doing some identical strange things during install. Is there a known issue with Etch ? Do I have any chance to fix it ? Thank you David Ponzone Direction Technique email: david.ponzone at 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 pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/eefadff5/attachment.html From brian at freeswitch.org Wed Jul 7 07:24:00 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 7 Jul 2010 09:24:00 -0500 Subject: [Freeswitch-users] Two FreeSWITCH Box IVR Call Transfer In-Reply-To: References: Message-ID: <6DC3C0A0-2D97-4ADE-8E61-F5628FF53D93@freeswitch.org> You're going to have to pass it in a SIP header by setting a variable like sip_h_X-UserID=123456 then on the far side extract it out of the variables... variables from one FreeSWITCH to the next don't magically appear. /b On Jul 7, 2010, at 9:07 AM, Jigar Thakkar wrote: > How can i do get the user_id variable value to BoxB? > > Please help. > > Thanks > > Jigs. From brian at freeswitch.org Wed Jul 7 07:24:46 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 7 Jul 2010 09:24:46 -0500 Subject: [Freeswitch-users] G729 validator not working on Debian 4.0 In-Reply-To: References: Message-ID: <896AD182-9E40-44DC-852B-0C4B94A7062D@freeswitch.org> I know why and I need to re-run a build to fix that... its because of .hash vs .hash.gnu in linking. Suse suffers the same issue. /b On Jul 7, 2010, at 9:20 AM, David Ponzone wrote: > Hi Everyone, > > I am trying to install the G729 codec on a Debian 4.0. > On 2 different boxes, validator fails with: > > Floating point exception > > One box is a 64bit Debian 4.0 running on Intel(R) Core(TM)2 Duo CPU E6550 @ 2.33GHz. > The other one is a 32 bit Debian 4.0 running on dual Intel(R) Xeon(TM) CPU 2.80GHz. > Both boxes have been installed by totally different people, so there is no chance that they had been doing some identical strange things during install. > > Is there a known issue with Etch ? > Do I have any chance to fix it ? > > Thank you From dswardstrom at remotelink.com Wed Jul 7 07:40:56 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Wed, 7 Jul 2010 07:40:56 -0700 (PDT) Subject: [Freeswitch-users] Two FreeSWITCH Box IVR Call Transfer In-Reply-To: References: Message-ID: <1278513656974-5265595.post@n2.nabble.com> I have an untested solution to this issue. Note: I am using JavaScript code examples: The following will do the "divert"/"transfer" if you have a session. session.execute("deflect", dfltstr); When Sip is involved, FreeSwitch when using a SIP "Call Transfer" which asks some upstream system to send the call to some other system. When doing this, a new destination address can be supplied. This can include a "postd" parameter. I think postd stands for "post dial" and is a standard parameter that can be added to the destination address. RFC 3261, section 19.1.6 has the following example: Thus, tel:+358-555-1234567;postd=pp22 becomes sip:+358-555-1234567;postd=pp22 at foo.com;user=phone There should be a way to add a postd string to the new destination address. So you can put the "user id" into the postd string. When the divert/transfer occurs, the new system will get the call as if it was an originated call. Looking at the destination string, the presence or absence of postd can determine how the call should be handled. If a postd is provided, the value can be used as the "user id". Note: I was looking at conferencing so planned to pass the conference information this way. Another note: I don't think that "user=phone" is necessary. The systems seem to be designed to work with it provided or not provided. If/When I find time, I will code this up as a JavaScript example and add it to the Wiki. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Two-FreeSWITCH-Box-IVR-Call-Transfer-tp5265404p5265595.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Wed Jul 7 08:14:13 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 7 Jul 2010 10:14:13 -0500 Subject: [Freeswitch-users] G729 validator not working on Debian 4.0 In-Reply-To: References: Message-ID: <8816AD90-C267-497E-8F4F-6636EF0505A7@freeswitch.org> Can you please try http://files.freeswitch.org/g729/fsg729-139-installer Thanks, Brian On Jul 7, 2010, at 9:20 AM, David Ponzone wrote: > Hi Everyone, > > I am trying to install the G729 codec on a Debian 4.0. > On 2 different boxes, validator fails with: > > Floating point exception > > One box is a 64bit Debian 4.0 running on Intel(R) Core(TM)2 Duo CPU E6550 @ 2.33GHz. > The other one is a 32 bit Debian 4.0 running on dual Intel(R) Xeon(TM) CPU 2.80GHz. > Both boxes have been installed by totally different people, so there is no chance that they had been doing some identical strange things during install. > > Is there a known issue with Etch ? > Do I have any chance to fix it ? > > Thank you From mrene_lists at avgs.ca Wed Jul 7 08:41:53 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 7 Jul 2010 11:41:53 -0400 Subject: [Freeswitch-users] Mod_h323 AliasAddress Error In-Reply-To: References: Message-ID: Please report this on the bug tracker. See http://wiki.freeswitch.org/wiki/Reporting_Bugs#Reporting_A_Bug_With_JIRA Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-07-07, at 10:02 AM, Saeed Ahmed wrote: > Dear All, > > I am trying to configure mod_h323 with freeswitch, while loading module i receive that error: > > > 2010-07-07 16:01:02.027642 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_h26x] > 2010-07-07 16:01:02.304187 [CONSOLE] mod_h323.cpp:74 Starting loading mod_h323 > Assertion fail: Must have non-empty string in AliasAddress!, file h323ep.cxx, line 3586 > > bort, ore dump? > > > I am following that wiki : > > http://wiki.freeswitch.org/wiki/Mod_h323 > > my h323.conf.xml is: > > > > > > > > > > > > > > > > > > > > > I would appreciate any help. > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/d47334aa/attachment.html From robert.hadley at teotech.com Wed Jul 7 08:48:43 2010 From: robert.hadley at teotech.com (Robert Hadley) Date: Wed, 7 Jul 2010 08:48:43 -0700 Subject: [Freeswitch-users] VM Force Password Change In-Reply-To: <4C33E095.8080400@communicatefreely.net> References: <016301cb1d0a$f38c2e80$daa48b80$@com> <01bc01cb1d23$93213900$b963ab00$@com> <4C33E095.8080400@communicatefreely.net> Message-ID: <900160B5394F4506B1D77EF9EAE7D993@greyhawk.tonecommander.com> Hi Tim, Think of the vm-password in the lineId.xml file as an alternate or back-door password. Mod_voicemail stores the user's main voicemail password in its database (db/voicemail.db is the default), set via the VM IVR. It only uses the alternate if the main password check fails. You can set the lineId.xml file vm-password to "user-choose" to get part of the functionality you want: i.e. there is no alternate password set by default. Regards, Robert -----Original Message----- From: Tim St. Pierre [mailto:fs-list at communicatefreely.net] Sent: Tuesday, July 06, 2010 7:04 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] VM Force Password Change If you use this method, what happens when a user tries to change their password from within the voicemail app? Is there some sort of hook in mod_voicemail that we can use to push the update to the database via the web server? -Tim Steven Ayre wrote: > Try using mod_xml_curl to provide the user directory. > > voicemail password is a param in the user directory, you can generate > that dynamically querying a database (e.g. MySQL) via a scripting > language (e.g. PHP) using mod_xml_curl to get the current password (or > a1 hash if you want to store it encrypted). > > You can have a dialplan extension that's executed prior to the extension > that connects them to their voicemail if there's no password set that > collects DTMF and then sends it to the webserver to update the database > using Mod_curl > > http://wiki.freeswitch.org/wiki/Mod_voicemail#vm-password > http://wiki.freeswitch.org/wiki/Mod_xml_curl > http://wiki.freeswitch.org/wiki/Mod_curl > > > > > On 6 July 2010 17:12, Ghulam Mustafa @gmail.com > wrote: > > i have not done something like this before, but you can create an > event socket script to check first time login (possibly by checking > _default_ password in db/xml) and then and then ask user to change the > password. > > check it out http://wiki.freeswitch.org/wiki/IVR > > > -m > > On Tue, Jul 6, 2010 at 8:54 PM, Peder > wrote: > > Any idea how? > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of > Ghulam Mustafa > > Sent: Tuesday, July 06, 2010 10:35 AM > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] VM Force Password Change > > > > i don't think so, but you can definitely make it possible in dialplan. > > > > -m > > > > On Tue, Jul 6, 2010 at 5:58 PM, Peder > wrote: > >> Is there a way to force a password change upon first login to a > voicemail > >> account? We set the password to 1234 and would like to force > users to > >> change it the first time they login to the account. > >> > >> Peder > >> > >> > >> > >> From Nabble at slickdeals.endjunk.com Wed Jul 7 08:51:28 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Wed, 7 Jul 2010 08:51:28 -0700 (PDT) Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: References: <1278204209127-5252219.post@n2.nabble.com> <1278436848678-5261486.post@n2.nabble.com> <1278469312148-5263368.post@n2.nabble.com> <1278472465333-5263442.post@n2.nabble.com> <1278472842929-5263450.post@n2.nabble.com> <1278507728660-5265110.post@n2.nabble.com> Message-ID: <1278517888279-5265912.post@n2.nabble.com> Thanks for your quick response. I went ahead to add the *ap_ptr declaration, and another error message surfaced as shown below: make[7]: Entering directory `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.1_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' > make[8]: Entering directory > `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.1_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' > Creating mod_sofia_la-sofia.lo > Compiling sofia.c ... > sofia.c: In function 'logger': > sofia.c:1573: error: aggregate value used where an integer was expected > make[8]: *** [mod_sofia_la-sofia.lo] Error 1 I believe the problem with if(ap) line I encountered above has something to do with the gcc-4.3.3 with CodeSourcery enhancements compiler. If I use the plain gcc-4.3.3 compiler, if(ap) line won't cause an error in compilation. So long as I stay using the plain gcc-4.3.3, the compilation will successfully produce mod_sofia.so file. Anthony Minessale wrote: > > you forgot part of the patch =D > > The following 2 lines should be the first 2 lines in the function. > > /* gcc 4.4 gets mad at us for testing if (ap) so let's try to work around > it....*/ > void *ap_ptr = (void *) (intptr_t) ap; ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5265912.html Sent from the freeswitch-users mailing list archive at Nabble.com. From bottleman at icf.org.ru Wed Jul 7 09:02:04 2010 From: bottleman at icf.org.ru (Georgiewskiy Yuriy) Date: Wed, 7 Jul 2010 20:02:04 +0400 (MSD) Subject: [Freeswitch-users] Mod_h323 AliasAddress Error In-Reply-To: References: Message-ID: On 2010-07-07 16:02 +0200, Saeed Ahmed wrote freeswitch-users at lists.freeswi...: SA>Dear All, SA> SA>I am trying to configure mod_h323 with freeswitch, while loading module i SA>receive that error: SA> SA> SA>2010-07-07 16:01:02.027642 [CONSOLE] switch_loadable_module.c:944 SA>Successfully Loaded [mod_h26x] SA>2010-07-07 16:01:02.304187 [CONSOLE] mod_h323.cpp:74 Starting loading SA>mod_h323 SA>Assertion fail: Must have non-empty string in AliasAddress!, file SA>h323ep.cxx, line 3586 SA> SA>bort, ore dump? wiki is outdated, add ?? h323.conf.xml SA> SA>I am following that wiki : SA> SA>http://wiki.freeswitch.org/wiki/Mod_h323 SA> SA>my h323.conf.xml is: SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA>I would appreciate any help. SA> SA>Thanks SA> C ????????? With Best Regards ???????????? ????. Georgiewskiy Yuriy +7 4872 711666 +7 4872 711666 ???? +7 4872 711143 fax +7 4872 711143 ???????? ??? "?? ?? ??????" IT Service Ltd http://nkoort.ru http://nkoort.ru JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru YG129-RIPE YG129-RIPE From saeedahmad1981 at gmail.com Wed Jul 7 09:19:50 2010 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Wed, 7 Jul 2010 18:19:50 +0200 Subject: [Freeswitch-users] Mod_h323 AliasAddress Error In-Reply-To: References: Message-ID: Thanks Georgiewskiy, * * *I'll test it and will also update it on Wiki. * 2010/7/7 Georgiewskiy Yuriy > On 2010-07-07 16:02 +0200, Saeed Ahmed wrote > freeswitch-users at lists.freeswi...: > > SA>Dear All, > SA> > SA>I am trying to configure mod_h323 with freeswitch, while loading module > i > SA>receive that error: > SA> > SA> > SA>2010-07-07 16:01:02.027642 [CONSOLE] switch_loadable_module.c:944 > SA>Successfully Loaded [mod_h26x] > SA>2010-07-07 16:01:02.304187 [CONSOLE] mod_h323.cpp:74 Starting loading > SA>mod_h323 > SA>Assertion fail: Must have non-empty string in AliasAddress!, file > SA>h323ep.cxx, line 3586 > SA> > SA>bort, ore dump? > > wiki is outdated, add ?? > h323.conf.xml > > SA> > SA>I am following that wiki : > SA> > SA>http://wiki.freeswitch.org/wiki/Mod_h323 > SA> > SA>my h323.conf.xml is: > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA>I would appreciate any help. > SA> > SA>Thanks > SA> > > C ????????? With Best Regards > ???????????? ????. Georgiewskiy Yuriy > +7 4872 711666 +7 4872 711666 > ???? +7 4872 711143 fax +7 4872 711143 > ???????? ??? "?? ?? ??????" IT Service Ltd > http://nkoort.ru http://nkoort.ru > JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru > YG129-RIPE YG129-RIPE > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/96214054/attachment.html From anthony.minessale at gmail.com Wed Jul 7 09:31:39 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Jul 2010 11:31:39 -0500 Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <1278517888279-5265912.post@n2.nabble.com> References: <1278204209127-5252219.post@n2.nabble.com> <1278436848678-5261486.post@n2.nabble.com> <1278469312148-5263368.post@n2.nabble.com> <1278472465333-5263442.post@n2.nabble.com> <1278472842929-5263450.post@n2.nabble.com> <1278507728660-5265110.post@n2.nabble.com> <1278517888279-5265912.post@n2.nabble.com> Message-ID: ok i guess its hopeless then, On Wed, Jul 7, 2010 at 10:51 AM, mazilo wrote: > > Thanks for your quick response. I went ahead to add the *ap_ptr > declaration, > and another error message surfaced as shown below: > > make[7]: Entering directory > > `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.1_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' > > make[8]: Entering directory > > > `/opt/tmp/openwrt-svn-trunk-ARM/build_dir/target-arm_v5te_uClibc-0.9.30.1_eabi/freeswitch-1.0.6/src/mod/endpoints/mod_sofia' > > Creating mod_sofia_la-sofia.lo > > Compiling sofia.c ... > > sofia.c: In function 'logger': > > sofia.c:1573: error: aggregate value used where an integer was expected > > make[8]: *** [mod_sofia_la-sofia.lo] Error 1 > I believe the problem with if(ap) line I encountered above has something to > do with the gcc-4.3.3 with CodeSourcery enhancements compiler. If I use the > plain gcc-4.3.3 compiler, if(ap) line won't cause an error in compilation. > So long as I stay using the plain gcc-4.3.3, the compilation will > successfully produce mod_sofia.so file. > > Anthony Minessale wrote: > > > > you forgot part of the patch =D > > > > The following 2 lines should be the first 2 lines in the function. > > > > /* gcc 4.4 gets mad at us for testing if (ap) so let's try to work > around > > it....*/ > > void *ap_ptr = (void *) (intptr_t) ap; > > > ----- > don't and stop are the ONLY two 4-letter words considered offensive to men, > but not when used together. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5265912.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/f229f350/attachment.html From ali.stgt at gmail.com Wed Jul 7 09:38:07 2010 From: ali.stgt at gmail.com (=?UTF-8?B?RHVybXXFnyBBbGkgw5Z6dMO8cms=?=) Date: Wed, 7 Jul 2010 19:38:07 +0300 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 49, Issue 28 In-Reply-To: References: Message-ID: 2010/7/7 : Hello, Supplying the dial string to the ManagedSession constructor has solved our problem. We can originate the call and stream a media properly. Thanks a lot for your help. Ali ?zt?rk > ---------- Weitergeleitete Nachricht ---------- > From:?Brian West > To:?freeswitch-users at lists.freeswitch.org > Date:?Wed, 7 Jul 2010 08:36:55 -0500 > Subject:?Re: [Freeswitch-users] Sound not OK (Choppy Sound) while using ManagedSession (.NET/C#) > When a call is parked you can send commands to it to do anything you want. > > > /b > > On Jul 7, 2010, at 8:30 AM, Phillip Jones wrote: > >> In example two you are parking the call. You are then playing media into a parked call. I am not sure this is valid. According to the wiki "Please note that to retrieve a call that has been "parked", you'll have to bridge to them or transfer the call to a valid location." - so you might need to transfer that parked call to a DialPlan App before you try and play media - or just not park it in the first place. >> >> I might be wrong - but that where I would start. > > > > > > ---------- Weitergeleitete Nachricht ---------- > From:?Anthony Minessale > To:?freeswitch-users at lists.freeswitch.org > Date:?Wed, 7 Jul 2010 08:41:03 -0500 > Subject:?Re: [Freeswitch-users] Sound not OK (Choppy Sound) while using ManagedSession (.NET/C#) > instead of originate to park then sending the uuid to the constructor > try supplying the dial string to the constructor which will place the call > > On Wed, Jul 7, 2010 at 8:30 AM, Phillip Jones wrote: >> >> In example two you are parking the call. You are then playing media into a parked call. I am not sure this is valid. According to the wiki "Please note that to retrieve a call that has been "parked", you'll have to bridge to them or transfer the call to a valid location." - so you might need to transfer that parked call to a DialPlan App before you try and play media - or just not park it in the first place. >> >> I might be wrong - but that where I would start. >> >> On Wed, Jul 7, 2010 at 8:03 AM, Durmu? Ali ?zt?rk wrote: >>> >>> Hello, >>> >>> I have successfully entegrated a dll-module (written udner .Net / c#) >>> in fs which is loadable by the mod_managed component. >>> >>> Now, if I try to stream a wav file to the callee then the sound is >>> very poor and choppy (also the file is played slower). >>> >>> The fs is running under Windows XP and tested under Windows 2003 with >>> the same result. >>> >>> Streaming by using FreeSWITCH.Native.Api() works without problems, >>> sound is perfect. >>> >>> Samples which I have tried: >>> >>> >>> >>> 1 - This works fine but delegate mechanism is missing there: >>> >>> FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); >>> >>> string uuid = fsApi.ExecuteString("create_uuid"); >>> string apiResult = fsApi.Execute("originate", >>> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/Test/{2} >>> &&playback({3}", uuid, callerID, phoneNumber, wavFile)); >>> >>> >>> >>> >>> 2 - ManagedSession is integrated but we have sound problems: >>> >>> FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); >>> >>> string uuid = fsApi.ExecuteString("create_uuid"); >>> >>> string apiResult = fsApi.Execute("originate", >>> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/Test/{2} >>> &park", uuid, callerID, phoneNumber)); >>> >>> FreeSWITCH.Native.ManagedSession blegSession = new >>> FreeSWITCH.Native.ManagedSession(uuid); >>> >>> if (blegSession.IsAvailable) >>> { >>> ??? while (!blegSession.answered()) >>> ??????? { >>> ??? ??? blegSession.sleep(500, 1); >>> ??? } >>> >>> ??????? if (blegSession.Ready() && blegSession.mediaReady()) >>> ??????? { >>> ??? ??? //blegSession.Answer(); >>> ??? ??? //blegSession.Execute("playback",wavFile); >>> ??????????????? blegSession.StreamFile(wavFile, 0); >>> ??????? } >>> >>> ??????? blegSession.Hangup("Normal call clearing"); >>> } >>> >>> >>> >>> >>> Is there a mistake in my code or do I have forgotten something? Can >>> you support me with some examples? >>> >>> >>> Another issue is, that I am not able to make a origination directly by >>> ManagedSession or CoreSession without using the >>> FreeSWITCH.Native.Api(). How can I create an instance of a CoreSession >>> object? >>> >>> >>> Many thanks for you help in advance. >>> >>> Ali >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > ---------- Weitergeleitete Nachricht ---------- > From:?Anthony Minessale > To:?freeswitch-users at lists.freeswitch.org > Date:?Wed, 7 Jul 2010 08:46:08 -0500 > Subject:?Re: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout > This is not at all the same. > in your script you have to do one of the following: > if they dialed the right digits answer the channel, if not hangup the channel. > you are doing neither and originate is hanging up for you because you did not answer the channel. > Can we be done with this thread now? it's trying my?patience. > > > On Wed, Jul 7, 2010 at 12:47 AM, Nagalenoj H. wrote: >> >> After git pull, the behavior remains the same. >> >> When I check the console log, the leg has got disconnected after leg_timeout seconds but the session for the leg has got closed after the script exists. >> >> 2010-07-07 05:27:39.156085 [INFO] mod_dialplan_xml.c:331 Processing 1005->212 in context default >> 2010-07-07 05:27:39.194359 [NOTICE] mod_dptools.c:746 Channel [sofia/internal/1005 at 192.168.1.72] has been answered >> 2010-07-07 05:27:42.708162 [NOTICE] sofia.c:4875 Channel [sofia/internal/sip:1000 at 192.168.6.114] has been answered >> 2010-07-07 05:27:49.020053 [INFO] mod_dptools.c:2393 Originate Failed.? Cause: NO_ANSWER >> 2010-07-07 05:28:12.761902 [NOTICE] switch_core_session.c:1193 Session 6 (sofia/internal/sip:1000 at 192.168.6.114) Ended >> 2010-07-07 05:28:12.761902 [NOTICE] switch_core_session.c:1195 Close Channel sofia/internal/sip:1000 at 192.168.6.114 [CS_DESTROY] >> >> Here is the console log, >> http://pastebin.freeswitch.org/13395 >> >> On Tue, Jul 6, 2010 at 8:56 PM, Anthony Minessale wrote: >>> >>> try latest GIT, that was an edge case and it's fixed. >>> >>> On Tue, Jul 6, 2010 at 1:58 AM, Nagalenoj H. wrote: >>>> >>>> Got core dump when I execute the bridge as >>>> execute-app-arg: {user_recurse_variables=false,group_confirm_cancel_timeout=true}[leg_timeout=10]user/1000 >>>> >>>> Console log is here, >>>> http://pastebin.freeswitch.org/13371 >>>> >>>> core dump backtrace is here, >>>> http://pastebin.freeswitch.org/13372 >>>> >>>> Also, tried after git pull, but the result is same. >>>> >>>> On Mon, Jul 5, 2010 at 8:53 PM, Anthony Minessale wrote: >>>>> >>>>> you also need user_recurse_variables=false >>>>> {user_recurse_variables=false,group_confirm_cancel_timeout=true} >>>>> >>>>> >>>>> On Sat, Jul 3, 2010 at 8:22 AM, Nagalenoj H. wrote: >>>>>> >>>>>> Here is the log got from the latest GIT source. >>>>>> >>>>>> http://pastebin.freeswitch.org/13347 >>>>>> >>>>>> On Fri, Jul 2, 2010 at 10:15 PM, Anthony Minessale wrote: >>>>>>> >>>>>>> update and reproduce that same log with latest GIT the version you are using has an issue. >>>>>>> >>>>>>> On Fri, Jul 2, 2010 at 12:25 AM, Nagalenoj H. wrote: >>>>>>>> >>>>>>>> I've pasted the console log here, >>>>>>>> >>>>>>>> http://pastebin.freeswitch.org/13333 >>>>>>>> >>>>>>>> >>>>>>>> On Wed, Jun 30, 2010 at 11:15 PM, Michael Collins wrote: >>>>>>>>> >>>>>>>>> Can you supply a console log of these calls? >>>>>>>>> -MC >>>>>>>>> >>>>>>>>> On Wed, Jun 30, 2010 at 7:37 AM, Nagalenoj H. wrote: >>>>>>>>>> >>>>>>>>>> Dear Anthony, >>>>>>>>>> ??? I've tried using the group_confirm_cancel_timeout as per the discussion we had in IRC. You wanted to used it as part of dial string and not as a channel variable. >>>>>>>>>> ??? But, It doesn't work for me. >>>>>>>>>> >>>>>>>>>> Here is how I've given the commands and the script I've executed. Even when I give group_confirm_cancel_timeout, the callee's leg is getting disconnected after legtimeout. >>>>>>>>>> >>>>>>>>>> connect >>>>>>>>>> >>>>>>>>>> sendmsg >>>>>>>>>> call-command: execute >>>>>>>>>> execute-app-name:answer >>>>>>>>>> >>>>>>>>>> sendmsg >>>>>>>>>> call-command: execute >>>>>>>>>> execute-app-name: set >>>>>>>>>> execute-app-arg: group_confirm_key=exec >>>>>>>>>> >>>>>>>>>> sendmsg >>>>>>>>>> call-command: execute >>>>>>>>>> execute-app-name: set >>>>>>>>>> execute-app-arg: group_confirm_file=perl /root/bridge.pl >>>>>>>>>> >>>>>>>>>> sendmsg >>>>>>>>>> call-command: execute >>>>>>>>>> execute-app-name: bridge >>>>>>>>>> execute-app-arg: {group_confirm_cancel_timeout=1}[leg_timeout=10]user/1005 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> bridge.pl: >>>>>>>>>> #!/usr/bin/perl >>>>>>>>>> use freeswitch; >>>>>>>>>> >>>>>>>>>> our $session; >>>>>>>>>> freeswitch::consoleLog("info","Goint to get the digits"); >>>>>>>>>> # To simulate the scenario I used sleep here. >>>>>>>>>> sleep(30); >>>>>>>>>> 1; >>>>>>>>>> >>>>>>>>>> Kindly tell me whats wrong in the above. >>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Regards, >>>>>>>> Nagalenoj H. >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at 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 at hotmail.com >>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>>> IRC: irc.freenode.net #freeswitch >>>>>>> >>>>>>> FreeSWITCH Developer Conference >>>>>>> sip:888 at conference.freeswitch.org >>>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>>> pstn:+19193869900 >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> Nagalenoj H. >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 at hotmail.com >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> pstn:+19193869900 >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> -- >>>> Regards, >>>> Nagalenoj H. >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> -- >> Regards, >> Nagalenoj H. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From roger_salloum at shaw.ca Wed Jul 7 10:45:31 2010 From: roger_salloum at shaw.ca (Roger Salloum) Date: Wed, 07 Jul 2010 10:45:31 -0700 Subject: [Freeswitch-users] Channel variables from CLI Message-ID: Hi, Is there a way to set channel variables in the CLI? I have created a custom sql that uses channel variables to determine routes, and wanted to test it through the CLI. However, I can't find out how to set the channel variables in the CLI so that the lcr module will have that information to use in the sql statement. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/7f48f9a7/attachment.html From steve.d.ward at gmail.com Wed Jul 7 10:54:20 2010 From: steve.d.ward at gmail.com (Steven Ward) Date: Wed, 7 Jul 2010 13:54:20 -0400 Subject: [Freeswitch-users] Channel variables from CLI In-Reply-To: References: Message-ID: Perhaps this will do the trick for you: uuid_setvar [value] where represents the UUID of the channel and represents the channel variable name. On Wed, Jul 7, 2010 at 1:45 PM, Roger Salloum wrote: > Hi, > > Is there a way to set channel variables in the CLI? I have created a custom > sql that uses channel variables to determine routes, and wanted to test it > through the CLI. However, I can't find out how to set the channel variables > in the CLI so that the lcr module will have that information to use in the > sql statement. > > Thanks, > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/adb89a0d/attachment.html From paul.gore.j at gmail.com Wed Jul 7 11:30:07 2010 From: paul.gore.j at gmail.com (paul gore) Date: Wed, 7 Jul 2010 14:30:07 -0400 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: Message-ID: Firewall is configured according to the wiki, I also tried to open all udp ports, issue persists. Actually the problem became more complex - outgoing calls don't work with one particular termination provider, siptraffic.com , any ideas why? Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one? On 7/6/10, paul gore wrote: > Hi there, > I am experimenting with FS on EC2, I like results, but stuck on weird audio > issue - I followed FreeSwitch EC2 wiki article and modified internal > profile > and vars.xml accordingly, but unfortunately still cannot get it working. > Incoming and outgoing calls made using a SIP phone to FS extensions work > just fine. As well as calls to FS from PSTN. But calls to PSTN via gateways > result in no audio at all, no ring, nothing, SIP signaling goes through OK. > Sofia status profile shows correct values for Ext-RTP-IP for both profiles > - > my static public IP, RTP-IP shows local IP. > Any thoughts on that? Anybody can share working profile configuration may > be? > Please help, I really need to get this going. > > Thanks. > From anthony.minessale at gmail.com Wed Jul 7 11:37:24 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Jul 2010 13:37:24 -0500 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: Message-ID: not really, not with so little information. On Wed, Jul 7, 2010 at 1:30 PM, paul gore wrote: > Firewall is configured according to the wiki, I also tried to open all > udp ports, issue persists. > Actually the problem became more complex - outgoing calls don't work > with one particular termination provider, siptraffic.com , any ideas > why? > Outgoing calls also don't work when originating a call via js script > or via FS api. Any clues on that one? > > On 7/6/10, paul gore wrote: > > Hi there, > > I am experimenting with FS on EC2, I like results, but stuck on weird > audio > > issue - I followed FreeSwitch EC2 wiki article and modified internal > > profile > > and vars.xml accordingly, but unfortunately still cannot get it working. > > Incoming and outgoing calls made using a SIP phone to FS extensions work > > just fine. As well as calls to FS from PSTN. But calls to PSTN via > gateways > > result in no audio at all, no ring, nothing, SIP signaling goes through > OK. > > Sofia status profile shows correct values for Ext-RTP-IP for both > profiles > > - > > my static public IP, RTP-IP shows local IP. > > Any thoughts on that? Anybody can share working profile configuration may > > be? > > Please help, I really need to get this going. > > > > Thanks. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/a789b02d/attachment.html From sos at sokhapkin.dyndns.org Wed Jul 7 11:49:03 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 7 Jul 2010 14:49:03 -0400 Subject: [Freeswitch-users] mod_xml_curl question. In-Reply-To: <201007070932.42832.sos@sokhapkin.dyndns.org> References: <201007070932.42832.sos@sokhapkin.dyndns.org> Message-ID: <201007071449.03512.sos@sokhapkin.dyndns.org> I tried to add second dialplan binding to xml_curl.conf.xml pointing to a local xml file (with file: URL). But seems like I can't have more than 1 dialplan binding in xml_curl.conf.xml, only the first binding is executed. Is it by design or I'm doing something wrong? On Wednesday 07 July 2010, Sergey Okhapkin wrote: > I want to use mod_xml_curl to retrieve only dynamic dialplan sections, but > to keep a large chunk of dialplan code in a static XML file to minimize > the number of http requests. But when mod_xml_curl is loaded, dialplan > search/execution starts with xml_curl execution. How to avoid that? I need > static xml to be looked up/executed first, and retrieve dialplan context > from web server only if the required dialplan context is not defined in > static XML file. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jcasale at activenetwerx.com Wed Jul 7 12:55:31 2010 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Wed, 7 Jul 2010 19:55:31 +0000 Subject: [Freeswitch-users] mod_spandsp wiki error Message-ID: The wiki states in the Intro: The old mod_fax, mod_t38gateway, and many of the codecs in FreeSWITCH (mod_voipcodecs) have now merged to one module called mod_spandsp which takes advantage of all the DSP features found in the spandsp library including T.38 endpoint and gateway functionality. Yet the "Installation and configuration" states: mod_spandsp is compiled by default with the freeswitch source tree but is still in testing stage to use it just unload mod_voipcodecs and mod_spandsp and load mod_fax remember to change it in modules.conf.xml mod_spandsp require libtiff. Is that an error, should I fix this? If the old mod_fax, mod_t38gateway and ... modules have been renamed and mod_spandsp is in modules.conf.xml there is nothing to do then? Thanks, jlc From infos at madovsky.org Wed Jul 7 13:05:34 2010 From: infos at madovsky.org (Madovsky) Date: Wed, 7 Jul 2010 16:05:34 -0400 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls References: Message-ID: I had same problem from this provider without to explain why. One day it works, another day it doesn't, their support is crap... ----- Original Message ----- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, July 07, 2010 2:37 PM Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls not really, not with so little information. On Wed, Jul 7, 2010 at 1:30 PM, paul gore wrote: Firewall is configured according to the wiki, I also tried to open all udp ports, issue persists. Actually the problem became more complex - outgoing calls don't work with one particular termination provider, siptraffic.com , any ideas why? Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one? On 7/6/10, paul gore wrote: > Hi there, > I am experimenting with FS on EC2, I like results, but stuck on weird audio > issue - I followed FreeSwitch EC2 wiki article and modified internal > profile > and vars.xml accordingly, but unfortunately still cannot get it working. > Incoming and outgoing calls made using a SIP phone to FS extensions work > just fine. As well as calls to FS from PSTN. But calls to PSTN via gateways > result in no audio at all, no ring, nothing, SIP signaling goes through OK. > Sofia status profile shows correct values for Ext-RTP-IP for both profiles > - > my static public IP, RTP-IP shows local IP. > Any thoughts on that? Anybody can share working profile configuration may > be? > Please help, I really need to get this going. > > Thanks. > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/474db163/attachment-0001.html From sos at sokhapkin.dyndns.org Wed Jul 7 13:16:41 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 7 Jul 2010 16:16:41 -0400 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: Message-ID: <201007071616.41535.sos@sokhapkin.dyndns.org> What "doesn't work" means? It could be (and most likely is not) FS-related problem On Wednesday 07 July 2010, Madovsky wrote: > I had same problem from this provider without to explain why. > One day it works, another day it doesn't, their support is crap... > > ----- Original Message ----- > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Sent: Wednesday, July 07, 2010 2:37 PM > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing > calls > > > not really, not with so little information. > > > > On Wed, Jul 7, 2010 at 1:30 PM, paul gore wrote: > > Firewall is configured according to the wiki, I also tried to open all > udp ports, issue persists. > Actually the problem became more complex - outgoing calls don't work > with one particular termination provider, siptraffic.com , any ideas > why? > Outgoing calls also don't work when originating a call via js script > or via FS api. Any clues on that one? > > On 7/6/10, paul gore wrote: > > Hi there, > > I am experimenting with FS on EC2, I like results, but stuck on weird > > audio issue - I followed FreeSwitch EC2 wiki article and modified > > internal profile > > and vars.xml accordingly, but unfortunately still cannot get it > > working. Incoming and outgoing calls made using a SIP phone to FS > > extensions work just fine. As well as calls to FS from PSTN. But > > calls to PSTN via gateways result in no audio at all, no ring, > > nothing, SIP signaling goes through OK. Sofia status profile shows > > correct values for Ext-RTP-IP for both profiles - > > my static public IP, RTP-IP shows local IP. > > Any thoughts on that? Anybody can share working profile configuration > > may be? > > Please help, I really need to get this going. > > > > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > > --------------------------------------------------------------------------- > --- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From tgraziano at myitdepartment.net Wed Jul 7 13:23:42 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Wed, 7 Jul 2010 16:23:42 -0400 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: <201007071616.41535.sos@sokhapkin.dyndns.org> References: <201007071616.41535.sos@sokhapkin.dyndns.org> Message-ID: You should try from a standalone or local installation to ensure it works with this provider and your account before you attempt to run it on ec2 (imo). On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin wrote: > What "doesn't work" means? It could be (and most likely is not) FS-related > problem > > On Wednesday 07 July 2010, Madovsky wrote: > > I had same problem from this provider without to explain why. > > One day it works, another day it doesn't, their support is crap... > > > > ----- Original Message ----- > > From: Anthony Minessale > > To: freeswitch-users at lists.freeswitch.org > > Sent: Wednesday, July 07, 2010 2:37 PM > > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing > > calls > > > > > > not really, not with so little information. > > > > > > > > On Wed, Jul 7, 2010 at 1:30 PM, paul gore > wrote: > > > > Firewall is configured according to the wiki, I also tried to open > all > > udp ports, issue persists. > > Actually the problem became more complex - outgoing calls don't work > > with one particular termination provider, siptraffic.com , any ideas > > why? > > Outgoing calls also don't work when originating a call via js script > > or via FS api. Any clues on that one? > > > > On 7/6/10, paul gore wrote: > > > Hi there, > > > I am experimenting with FS on EC2, I like results, but stuck on > weird > > > audio issue - I followed FreeSwitch EC2 wiki article and modified > > > internal profile > > > and vars.xml accordingly, but unfortunately still cannot get it > > > working. Incoming and outgoing calls made using a SIP phone to FS > > > extensions work just fine. As well as calls to FS from PSTN. But > > > calls to PSTN via gateways result in no audio at all, no ring, > > > nothing, SIP signaling goes through OK. Sofia status profile shows > > > correct values for Ext-RTP-IP for both profiles - > > > my static public IP, RTP-IP shows local IP. > > > Any thoughts on that? Anybody can share working profile > configuration > > > may be? > > > Please help, I really need to get this going. > > > > > > Thanks. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > > > > --------------------------------------------------------------------------- > > --- > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano at voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano at myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk at voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/e797a2e5/attachment.html From paul.gore.j at gmail.com Wed Jul 7 13:36:43 2010 From: paul.gore.j at gmail.com (paul gore) Date: Wed, 7 Jul 2010 16:36:43 -0400 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: <201007071616.41535.sos@sokhapkin.dyndns.org> Message-ID: This provider does work on another box which is not natted as ec2. Most puzzling here though is why call originaion via api even not going via siptraffic still gets no audio. On 7/7/10, Tony Graziano wrote: > You should try from a standalone or local installation to ensure it works > with this provider and your account before you attempt to run it on ec2 > (imo). > > On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin > wrote: > >> What "doesn't work" means? It could be (and most likely is not) FS-related >> problem >> >> On Wednesday 07 July 2010, Madovsky wrote: >> > I had same problem from this provider without to explain why. >> > One day it works, another day it doesn't, their support is crap... >> > >> > ----- Original Message ----- >> > From: Anthony Minessale >> > To: freeswitch-users at lists.freeswitch.org >> > Sent: Wednesday, July 07, 2010 2:37 PM >> > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing >> > calls >> > >> > >> > not really, not with so little information. >> > >> > >> > >> > On Wed, Jul 7, 2010 at 1:30 PM, paul gore >> wrote: >> > >> > Firewall is configured according to the wiki, I also tried to open >> all >> > udp ports, issue persists. >> > Actually the problem became more complex - outgoing calls don't work >> > with one particular termination provider, siptraffic.com , any ideas >> > why? >> > Outgoing calls also don't work when originating a call via js script >> > or via FS api. Any clues on that one? >> > >> > On 7/6/10, paul gore wrote: >> > > Hi there, >> > > I am experimenting with FS on EC2, I like results, but stuck on >> weird >> > > audio issue - I followed FreeSwitch EC2 wiki article and modified >> > > internal profile >> > > and vars.xml accordingly, but unfortunately still cannot get it >> > > working. Incoming and outgoing calls made using a SIP phone to FS >> > > extensions work just fine. As well as calls to FS from PSTN. But >> > > calls to PSTN via gateways result in no audio at all, no ring, >> > > nothing, SIP signaling goes through OK. Sofia status profile shows >> > > correct values for Ext-RTP-IP for both profiles - >> > > my static public IP, RTP-IP shows local IP. >> > > Any thoughts on that? Anybody can share working profile >> configuration >> > > may be? >> > > Please help, I really need to get this going. >> > > >> > > Thanks. >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> > >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > >> > googletalk:conf+888 at conference.freeswitch.org >> > pstn:+19193869900 >> > >> > >> > >> > >> --------------------------------------------------------------------------- >> > --- >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgraziano at voice.myitdepartment.net > Fax: 434.984.8431 > > Email: tgraziano at myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpdesk at voice.myitdepartment.net > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. > From vetali100 at gmail.com Wed Jul 7 13:38:39 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Wed, 7 Jul 2010 23:38:39 +0300 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: <201007071616.41535.sos@sokhapkin.dyndns.org> Message-ID: Regarding the following: "Outgoing calls also don't work when originating a call via js script or via FS api. Any clues on that one?" What do you see in the log? CALL_REJECTED? If so, try adding your FS IP address to the list of allowed IPs on their site - looks like they changed something and you can't use (until they fix) only your login and password. If "don't work" means something else, then too less info. :-( Maybe you could provide some logs of the call attempts... Regards, Vitalie 2010/7/7 Tony Graziano > You should try from a standalone or local installation to ensure it works > with this provider and your account before you attempt to run it on ec2 > (imo). > > > On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin wrote: > >> What "doesn't work" means? It could be (and most likely is not) FS-related >> problem >> >> On Wednesday 07 July 2010, Madovsky wrote: >> > I had same problem from this provider without to explain why. >> > One day it works, another day it doesn't, their support is crap... >> > >> > ----- Original Message ----- >> > From: Anthony Minessale >> > To: freeswitch-users at lists.freeswitch.org >> > Sent: Wednesday, July 07, 2010 2:37 PM >> > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing >> > calls >> > >> > >> > not really, not with so little information. >> > >> > >> > >> > On Wed, Jul 7, 2010 at 1:30 PM, paul gore >> wrote: >> > >> > Firewall is configured according to the wiki, I also tried to open >> all >> > udp ports, issue persists. >> > Actually the problem became more complex - outgoing calls don't work >> > with one particular termination provider, siptraffic.com , any >> ideas >> > why? >> > Outgoing calls also don't work when originating a call via js script >> > or via FS api. Any clues on that one? >> > >> > On 7/6/10, paul gore wrote: >> > > Hi there, >> > > I am experimenting with FS on EC2, I like results, but stuck on >> weird >> > > audio issue - I followed FreeSwitch EC2 wiki article and modified >> > > internal profile >> > > and vars.xml accordingly, but unfortunately still cannot get it >> > > working. Incoming and outgoing calls made using a SIP phone to FS >> > > extensions work just fine. As well as calls to FS from PSTN. But >> > > calls to PSTN via gateways result in no audio at all, no ring, >> > > nothing, SIP signaling goes through OK. Sofia status profile shows >> > > correct values for Ext-RTP-IP for both profiles - >> > > my static public IP, RTP-IP shows local IP. >> > > Any thoughts on that? Anybody can share working profile >> configuration >> > > may be? >> > > Please help, I really need to get this going. >> > > >> > > Thanks. >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > googletalk:conf+888 at conference.freeswitch.org >> > pstn:+19193869900 >> > >> > >> > >> > >> --------------------------------------------------------------------------- >> > --- >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgraziano at voice.myitdepartment.net > Fax: 434.984.8431 > > Email: tgraziano at myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpdesk at voice.myitdepartment.net > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/46243e99/attachment-0001.html From vetali100 at gmail.com Wed Jul 7 14:07:34 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Thu, 8 Jul 2010 00:07:34 +0300 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: <201007071616.41535.sos@sokhapkin.dyndns.org> Message-ID: So "don't work" still means "no audio" and "sip is ok", clear... RTP packets are lost somewhere on the way: You can check where RTP packets are lost - between FS and siptraffic, or between FS and your SIP Client. 1.Install "ngrep" if you don't have it yet (on Ubuntu: apt-get install ngrep) 2.Run it: "ngrep port 5080" 3.Start a call and after few ngrep's messages, stop ngrep using Ctrl+C 4.Find the first INVITE line from your FS to siptraffic, it will contain your IP and port at the following lines: c=IN IP4 ****!!!YOUR-EC-IP!!!*****..t=0 0..m=audio ****!!!YOUR-RTP-PORT!!!**** RTP/AVP 5.Open new terminal window, and run "ngrep port ****!!!YOUR-RTP-PORT!!!****" 6. Wait 5 seconds 7. Stop ngrep using Ctrl+C 8. Hangup Now on the second terminal you should see a lot of line pairs like: YOUR-EC-IP -> SIPTRAFFIC-IP SIPTRAFFIC-IP -> YOUR-EC-IP If you see only one of the directions (e.g. only YOUR-EC-IP -> SIPTRAFFIC-IP), then some problem is between FS and Siptraffic. If you see both directions then problem is not here and most probably on the way from FS to your SIP Client or somewhere else (inside FS?) If so, try to investigate this part using port 5060 (same way as 5080). This analysis will narrow the problem a bit... Regads, Vitalie 2010/7/7 paul gore > This provider does work on another box which is not natted as ec2. > Most puzzling here though is why call originaion via api even not > going via siptraffic still gets no audio. > > On 7/7/10, Tony Graziano wrote: > > You should try from a standalone or local installation to ensure it works > > with this provider and your account before you attempt to run it on ec2 > > (imo). > > > > On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin > > wrote: > > > >> What "doesn't work" means? It could be (and most likely is not) > FS-related > >> problem > >> > >> On Wednesday 07 July 2010, Madovsky wrote: > >> > I had same problem from this provider without to explain why. > >> > One day it works, another day it doesn't, their support is crap... > >> > > >> > ----- Original Message ----- > >> > From: Anthony Minessale > >> > To: freeswitch-users at lists.freeswitch.org > >> > Sent: Wednesday, July 07, 2010 2:37 PM > >> > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on > outgoing > >> > calls > >> > > >> > > >> > not really, not with so little information. > >> > > >> > > >> > > >> > On Wed, Jul 7, 2010 at 1:30 PM, paul gore > >> wrote: > >> > > >> > Firewall is configured according to the wiki, I also tried to open > >> all > >> > udp ports, issue persists. > >> > Actually the problem became more complex - outgoing calls don't > work > >> > with one particular termination provider, siptraffic.com , any > ideas > >> > why? > >> > Outgoing calls also don't work when originating a call via js > script > >> > or via FS api. Any clues on that one? > >> > > >> > On 7/6/10, paul gore wrote: > >> > > Hi there, > >> > > I am experimenting with FS on EC2, I like results, but stuck on > >> weird > >> > > audio issue - I followed FreeSwitch EC2 wiki article and > modified > >> > > internal profile > >> > > and vars.xml accordingly, but unfortunately still cannot get it > >> > > working. Incoming and outgoing calls made using a SIP phone to > FS > >> > > extensions work just fine. As well as calls to FS from PSTN. But > >> > > calls to PSTN via gateways result in no audio at all, no ring, > >> > > nothing, SIP signaling goes through OK. Sofia status profile > shows > >> > > correct values for Ext-RTP-IP for both profiles - > >> > > my static public IP, RTP-IP shows local IP. > >> > > Any thoughts on that? Anybody can share working profile > >> configuration > >> > > may be? > >> > > Please help, I really need to get this going. > >> > > > >> > > Thanks. > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at 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 http://www.freeswitch.org/ > >> > ClueCon http://www.cluecon.com/ > >> > Twitter: http://twitter.com/FreeSWITCH_wire > >> > > >> > AIM: anthm > >> > MSN:anthony_minessale at hotmail.com > > > > >> > > >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >> > IRC: irc.freenode.net #freeswitch > >> > > >> > FreeSWITCH Developer Conference > >> > sip:888 at conference.freeswitch.org > > > > >> > > >> > googletalk:conf+888 at conference.freeswitch.org > > > > >> > pstn:+19193869900 > >> > > >> > > >> > > >> > > >> > --------------------------------------------------------------------------- > >> > --- > >> > > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at 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 at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > ====================== > > Tony Graziano, Manager > > Telephone: 434.984.8430 > > sip: tgraziano at voice.myitdepartment.net > > Fax: 434.984.8431 > > > > Email: tgraziano at myitdepartment.net > > > > LAN/Telephony/Security and Control Systems Helpdesk: > > Telephone: 434.984.8426 > > sip: helpdesk at voice.myitdepartment.net > > Fax: 434.984.8427 > > > > Helpdesk Contract Customers: > > http://www.myitdepartment.net/gethelp/ > > > > Why do mathematicians always confuse Halloween and Christmas? > > Because 31 Oct = 25 Dec. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/a9bf0e5f/attachment.html From sos at sokhapkin.dyndns.org Wed Jul 7 14:24:39 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 7 Jul 2010 17:24:39 -0400 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: Message-ID: <201007071724.40239.sos@sokhapkin.dyndns.org> May be it's just a FAS? Are you able to call that number with a different SIP client using the same call termination provider? On Wednesday 07 July 2010, Vitalii Colosov wrote: > So "don't work" still means "no audio" and "sip is ok", clear... > > RTP packets are lost somewhere on the way: > > You can check where RTP packets are lost - between FS and siptraffic, or > between FS and your SIP Client. > > 1.Install "ngrep" if you don't have it yet (on Ubuntu: apt-get install > ngrep) > 2.Run it: "ngrep port 5080" > 3.Start a call and after few ngrep's messages, stop ngrep using Ctrl+C > 4.Find the first INVITE line from your FS to siptraffic, it will contain > your IP and port at the following lines: > c=IN IP4 ****!!!YOUR-EC-IP!!!*****..t=0 0..m=audio > ****!!!YOUR-RTP-PORT!!!**** RTP/AVP > 5.Open new terminal window, and run "ngrep port > ****!!!YOUR-RTP-PORT!!!****" 6. Wait 5 seconds > 7. Stop ngrep using Ctrl+C > 8. Hangup > > Now on the second terminal you should see a lot of line pairs like: > YOUR-EC-IP -> SIPTRAFFIC-IP > SIPTRAFFIC-IP -> YOUR-EC-IP > > If you see only one of the directions (e.g. only YOUR-EC-IP -> > SIPTRAFFIC-IP), then some problem is between FS and Siptraffic. > > If you see both directions then problem is not here and most probably on > the way from FS to your SIP Client or somewhere else (inside FS?) > If so, try to investigate this part using port 5060 (same way as 5080). > > This analysis will narrow the problem a bit... > > Regads, > Vitalie > > > > 2010/7/7 paul gore > > > This provider does work on another box which is not natted as ec2. > > Most puzzling here though is why call originaion via api even not > > going via siptraffic still gets no audio. > > > > On 7/7/10, Tony Graziano wrote: > > > You should try from a standalone or local installation to ensure it > > > works with this provider and your account before you attempt to run it > > > on ec2 (imo). > > > > > > On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin > > > > > > wrote: > > >> What "doesn't work" means? It could be (and most likely is not) > > > > FS-related > > > > >> problem > > >> > > >> On Wednesday 07 July 2010, Madovsky wrote: > > >> > I had same problem from this provider without to explain why. > > >> > One day it works, another day it doesn't, their support is crap... > > >> > > > >> > ----- Original Message ----- > > >> > From: Anthony Minessale > > >> > To: freeswitch-users at lists.freeswitch.org > > >> > Sent: Wednesday, July 07, 2010 2:37 PM > > >> > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on > > > > outgoing > > > > >> > calls > > >> > > > >> > > > >> > not really, not with so little information. > > >> > > > >> > > > >> > > > >> > On Wed, Jul 7, 2010 at 1:30 PM, paul gore > > >> > > >> wrote: > > >> > Firewall is configured according to the wiki, I also tried to > > >> > open > > >> > > >> all > > >> > > >> > udp ports, issue persists. > > >> > Actually the problem became more complex - outgoing calls don't > > > > work > > > > >> > with one particular termination provider, siptraffic.com , any > > > > ideas > > > > >> > why? > > >> > Outgoing calls also don't work when originating a call via js > > > > script > > > > >> > or via FS api. Any clues on that one? > > >> > > > >> > On 7/6/10, paul gore wrote: > > >> > > Hi there, > > >> > > I am experimenting with FS on EC2, I like results, but stuck > > >> > > on > > >> > > >> weird > > >> > > >> > > audio issue - I followed FreeSwitch EC2 wiki article and > > > > modified > > > > >> > > internal profile > > >> > > and vars.xml accordingly, but unfortunately still cannot get > > >> > > it working. Incoming and outgoing calls made using a SIP phone > > >> > > to > > > > FS > > > > >> > > extensions work just fine. As well as calls to FS from PSTN. > > >> > > But calls to PSTN via gateways result in no audio at all, no > > >> > > ring, nothing, SIP signaling goes through OK. Sofia status > > >> > > profile > > > > shows > > > > >> > > correct values for Ext-RTP-IP for both profiles - > > >> > > my static public IP, RTP-IP shows local IP. > > >> > > Any thoughts on that? Anybody can share working profile > > >> > > >> configuration > > >> > > >> > > may be? > > >> > > Please help, I really need to get this going. > > >> > > > > >> > > Thanks. > > >> > > > >> > _______________________________________________ > > >> > FreeSWITCH-users mailing list > > >> > FreeSWITCH-users at 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 http://www.freeswitch.org/ > > >> > ClueCon http://www.cluecon.com/ > > >> > Twitter: http://twitter.com/FreeSWITCH_wire > > >> > > > >> > AIM: anthm > > >> > > > >> > MSN:anthony_minessale at hotmail.com > >> >m> > > > > >m> > > > > >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >> >essale at gmail.com> > > > > >com> > > > > >> > IRC: irc.freenode.net #freeswitch > > >> > > > >> > FreeSWITCH Developer Conference > > >> > > > >> > sip:888 at conference.freeswitch.org > >> >g> > > > > >g> > > > > >> > googletalk:conf+888 at conference.freeswitch.org > >> >8 at conference.freeswitch.org> > > > > >2B888 at conference.freeswitch.org> > > > > >> > pstn:+19193869900 > > > > ------------------------------------------------------------------------- > >-- > > > > >> > --- > > >> > > > >> > > > >> > _______________________________________________ > > >> > FreeSWITCH-users mailing list > > >> > FreeSWITCH-users at 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 at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE: > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > >> http://www.freeswitch.org > > > > > > -- > > > ====================== > > > Tony Graziano, Manager > > > Telephone: 434.984.8430 > > > sip: tgraziano at voice.myitdepartment.net > > > Fax: 434.984.8431 > > > > > > Email: tgraziano at myitdepartment.net > > > > > > LAN/Telephony/Security and Control Systems Helpdesk: > > > Telephone: 434.984.8426 > > > sip: helpdesk at voice.myitdepartment.net > > > Fax: 434.984.8427 > > > > > > Helpdesk Contract Customers: > > > http://www.myitdepartment.net/gethelp/ > > > > > > Why do mathematicians always confuse Halloween and Christmas? > > > Because 31 Oct = 25 Dec. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > From steveayre at gmail.com Wed Jul 7 14:29:31 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 7 Jul 2010 22:29:31 +0100 Subject: [Freeswitch-users] mod_spandsp wiki error In-Reply-To: References: Message-ID: Yes, it's out of date. Feel free to update it. On 7 July 2010 20:55, Joseph L. Casale wrote: > The wiki states in the Intro: > > The old mod_fax, mod_t38gateway, and many of the codecs in FreeSWITCH > (mod_voipcodecs) > have now merged to one module called mod_spandsp which takes advantage of > all the DSP > features found in the spandsp library including T.38 endpoint and gateway > functionality. > > Yet the "Installation and configuration" states: > > mod_spandsp is compiled by default with the freeswitch source tree but is > still in testing > stage to use it just unload mod_voipcodecs and mod_spandsp and load mod_fax > remember to change > it in modules.conf.xml mod_spandsp require libtiff. > > Is that an error, should I fix this? If the old mod_fax, mod_t38gateway and > ... modules > have been renamed and mod_spandsp is in modules.conf.xml there is nothing > to do then? > > Thanks, > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/5b166d3c/attachment.html From steveayre at gmail.com Wed Jul 7 14:31:13 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 7 Jul 2010 22:31:13 +0100 Subject: [Freeswitch-users] mod_xml_curl question. In-Reply-To: <201007071449.03512.sos@sokhapkin.dyndns.org> References: <201007070932.42832.sos@sokhapkin.dyndns.org> <201007071449.03512.sos@sokhapkin.dyndns.org> Message-ID: The 2nd (or 3rd or 4th etc) binding is for redundancy... it tries the first and only tries the 2nd if the first fails. So you can have a backup server in case the main one is offline. On 7 July 2010 19:49, Sergey Okhapkin wrote: > I tried to add second dialplan binding to xml_curl.conf.xml pointing to a > local xml file (with file: URL). But seems like I can't have more than 1 > dialplan binding in xml_curl.conf.xml, only the first binding is executed. > Is > it by design or I'm doing something wrong? > > On Wednesday 07 July 2010, Sergey Okhapkin wrote: > > I want to use mod_xml_curl to retrieve only dynamic dialplan sections, > but > > to keep a large chunk of dialplan code in a static XML file to minimize > > the number of http requests. But when mod_xml_curl is loaded, dialplan > > search/execution starts with xml_curl execution. How to avoid that? I > need > > static xml to be looked up/executed first, and retrieve dialplan context > > from web server only if the required dialplan context is not defined in > > static XML file. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/f4185541/attachment.html From steveayre at gmail.com Wed Jul 7 14:35:37 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 7 Jul 2010 22:35:37 +0100 Subject: [Freeswitch-users] Channel variables from CLI In-Reply-To: References: Message-ID: Try setting up an extension that sets the variables and calls lcr, then use originate from the cli to dial into that extension. If you're already using LCR for live traffic, you can use profiles to execute the existing query for normal calls and your custom sql just on a testing profile that you call explicitly: -Steve On 7 July 2010 18:45, Roger Salloum wrote: > Hi, > > Is there a way to set channel variables in the CLI? I have created a custom > sql that uses channel variables to determine routes, and wanted to test it > through the CLI. However, I can't find out how to set the channel variables > in the CLI so that the lcr module will have that information to use in the > sql statement. > > Thanks, > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/c6247cd3/attachment.html From paul.gore.j at gmail.com Wed Jul 7 14:50:45 2010 From: paul.gore.j at gmail.com (paul gore) Date: Wed, 7 Jul 2010 17:50:45 -0400 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: <201007071616.41535.sos@sokhapkin.dyndns.org> Message-ID: Seems like siptraffic uses 6 ip addresses for media, can that be the problem? Is there any setting in a gateway config xml which helps with that? I will do ngrep thing and update. On 7/7/10, paul gore wrote: > This provider does work on another box which is not natted as ec2. > Most puzzling here though is why call originaion via api even not > going via siptraffic still gets no audio. > > On 7/7/10, Tony Graziano wrote: >> You should try from a standalone or local installation to ensure it works >> with this provider and your account before you attempt to run it on ec2 >> (imo). >> >> On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin >> wrote: >> >>> What "doesn't work" means? It could be (and most likely is not) >>> FS-related >>> problem >>> >>> On Wednesday 07 July 2010, Madovsky wrote: >>> > I had same problem from this provider without to explain why. >>> > One day it works, another day it doesn't, their support is crap... >>> > >>> > ----- Original Message ----- >>> > From: Anthony Minessale >>> > To: freeswitch-users at lists.freeswitch.org >>> > Sent: Wednesday, July 07, 2010 2:37 PM >>> > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on >>> > outgoing >>> > calls >>> > >>> > >>> > not really, not with so little information. >>> > >>> > >>> > >>> > On Wed, Jul 7, 2010 at 1:30 PM, paul gore >>> wrote: >>> > >>> > Firewall is configured according to the wiki, I also tried to open >>> all >>> > udp ports, issue persists. >>> > Actually the problem became more complex - outgoing calls don't >>> > work >>> > with one particular termination provider, siptraffic.com , any >>> > ideas >>> > why? >>> > Outgoing calls also don't work when originating a call via js >>> > script >>> > or via FS api. Any clues on that one? >>> > >>> > On 7/6/10, paul gore wrote: >>> > > Hi there, >>> > > I am experimenting with FS on EC2, I like results, but stuck on >>> weird >>> > > audio issue - I followed FreeSwitch EC2 wiki article and >>> > modified >>> > > internal profile >>> > > and vars.xml accordingly, but unfortunately still cannot get it >>> > > working. Incoming and outgoing calls made using a SIP phone to >>> > FS >>> > > extensions work just fine. As well as calls to FS from PSTN. But >>> > > calls to PSTN via gateways result in no audio at all, no ring, >>> > > nothing, SIP signaling goes through OK. Sofia status profile >>> > shows >>> > > correct values for Ext-RTP-IP for both profiles - >>> > > my static public IP, RTP-IP shows local IP. >>> > > Any thoughts on that? Anybody can share working profile >>> configuration >>> > > may be? >>> > > Please help, I really need to get this going. >>> > > >>> > > Thanks. >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at 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 http://www.freeswitch.org/ >>> > ClueCon http://www.cluecon.com/ >>> > Twitter: http://twitter.com/FreeSWITCH_wire >>> > >>> > AIM: anthm >>> > >>> > MSN:anthony_minessale at hotmail.com >>> > >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> > IRC: irc.freenode.net #freeswitch >>> > >>> > FreeSWITCH Developer Conference >>> > >>> > sip:888 at conference.freeswitch.org >>> > >>> > googletalk:conf+888 at conference.freeswitch.org >>> > pstn:+19193869900 >>> > >>> > >>> > >>> > >>> --------------------------------------------------------------------------- >>> > --- >>> > >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> ====================== >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: tgraziano at voice.myitdepartment.net >> Fax: 434.984.8431 >> >> Email: tgraziano at myitdepartment.net >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: helpdesk at voice.myitdepartment.net >> Fax: 434.984.8427 >> >> Helpdesk Contract Customers: >> http://www.myitdepartment.net/gethelp/ >> >> Why do mathematicians always confuse Halloween and Christmas? >> Because 31 Oct = 25 Dec. >> > From sos at sokhapkin.dyndns.org Wed Jul 7 14:57:17 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 7 Jul 2010 17:57:17 -0400 Subject: [Freeswitch-users] mod_xml_curl question. In-Reply-To: References: <201007070932.42832.sos@sokhapkin.dyndns.org> <201007071449.03512.sos@sokhapkin.dyndns.org> Message-ID: <201007071757.17491.sos@sokhapkin.dyndns.org> I need the 1st one to be an XML file... It's always online :-) On Wednesday 07 July 2010, Steven Ayre wrote: > The 2nd (or 3rd or 4th etc) binding is for redundancy... it tries the first > and only tries the 2nd if the first fails. So you can have a backup server > in case the main one is offline. > > On 7 July 2010 19:49, Sergey Okhapkin wrote: > > I tried to add second dialplan binding to xml_curl.conf.xml pointing to a > > local xml file (with file: URL). But seems like I can't have more than 1 > > dialplan binding in xml_curl.conf.xml, only the first binding is > > executed. Is > > it by design or I'm doing something wrong? > > > > On Wednesday 07 July 2010, Sergey Okhapkin wrote: > > > I want to use mod_xml_curl to retrieve only dynamic dialplan sections, > > > > but > > > > > to keep a large chunk of dialplan code in a static XML file to > > > minimize the number of http requests. But when mod_xml_curl is loaded, > > > dialplan search/execution starts with xml_curl execution. How to avoid > > > that? I > > > > need > > > > > static xml to be looked up/executed first, and retrieve dialplan > > > context from web server only if the required dialplan context is not > > > defined in static XML file. > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user > > >s http://www.freeswitch.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > From jcasale at activenetwerx.com Wed Jul 7 15:14:06 2010 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Wed, 7 Jul 2010 22:14:06 +0000 Subject: [Freeswitch-users] mod_spandsp wiki error In-Reply-To: References: Message-ID: > Yes, it's out of date. Feel free to update it. Done, thanks for confirming. jlc From Nabble at slickdeals.endjunk.com Wed Jul 7 15:52:25 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Wed, 7 Jul 2010 15:52:25 -0700 (PDT) Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: References: <1278436848678-5261486.post@n2.nabble.com> <1278469312148-5263368.post@n2.nabble.com> <1278472465333-5263442.post@n2.nabble.com> <1278472842929-5263450.post@n2.nabble.com> <1278507728660-5265110.post@n2.nabble.com> <1278517888279-5265912.post@n2.nabble.com> Message-ID: <1278543145818-5267704.post@n2.nabble.com> Not really. At least, we gave it some tries. I believe it is not the problem on FS, but rather a problem to the gcc-4.3.3 compiler with CodeSourcery enhancements for OpenWRT. That said, if anyone wants to try, why not. Anthony Minessale wrote: > > ok i guess its hopeless then, ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5267704.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Wed Jul 7 16:12:15 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 7 Jul 2010 16:12:15 -0700 Subject: [Freeswitch-users] More questions migrating from Asterisk to FS In-Reply-To: <1278468923913-5263357.post@n2.nabble.com> References: <1278425102977-5260522.post@n2.nabble.com> <1278468923913-5263357.post@n2.nabble.com> Message-ID: On Tue, Jul 6, 2010 at 7:15 PM, mazilo wrote: > > Thank you for your speedy response. I am not sure I do understand what you > said. Please kindly bear with me as FS is a new world to me. It sure will > be > nice to see some sample XML codes. I used this > http://wiki.freeswitch.org/wiki/Provider_Configuration:_Gizmo Gizmo5 > configuration. Let's say I have two Gizmo5 accounts 17471234567 and > 17477654321. Then, the contents of my sip_profiles/external/gizmo5.xml file > that list both gateways will have only different username/password and the > rest parameters will be the same. It would be nice that the same parameters > don't need to be relisted on each gateway. I hope this makes it more clear. > I suppose that would be nice, but since these are static XML files your best bet is to create one, make sure it works, then make a copy. Edit the copy and you're done! Gateways have relatively few options compared to other configuration entities so having some sort of templating system doesn't save you a whole lot of time or energy. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/7de80e84/attachment.html From msc at freeswitch.org Wed Jul 7 16:14:23 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 7 Jul 2010 16:14:23 -0700 Subject: [Freeswitch-users] originate call hangup signal In-Reply-To: References: Message-ID: It depends on where the "hangup signal" comes from. Is this an analog line? If so, does the carrier provide disconnect supervision? It's entirely possible that the other end isn't doing a good job of telling FreeSWITCH that the call is over. -MC On Tue, Jul 6, 2010 at 10:35 PM, Tony Tin wrote: > Hi, > > When I use OpenZAP channel to originate a call, after the called party > hangup the phone. It takes freeswitch around 40 seconds to catch the hangup > signal and stop the dial plan. I'm wondering whether there is a way to > shorten that duration. Thanks. > > Regards, > Tony > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/b1c895d0/attachment.html From tony.tin at noahmedia.com.hk Wed Jul 7 19:34:54 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Thu, 8 Jul 2010 10:34:54 +0800 Subject: [Freeswitch-users] hangup does not break the Lua loop In-Reply-To: <4B80CF4E-6336-4F2B-A118-ED87D6979F90@freeswitch.org> References: <4B80CF4E-6336-4F2B-A118-ED87D6979F90@freeswitch.org> Message-ID: Thanks for the reply. It's "while true do" loop, I thought the hangup signal will break it. 1. session:ready will return true between call answered and hangup only, right ? 2. how does it compare to session:getState, "CS_HANGUP" or "CS_DESTROY" means call hung up ? Regards, Tony On Wed, Jul 7, 2010 at 8:15 PM, Brian West wrote: > Post your code for your while loop... sounds like you're not checking > session:ready to see if its still true. > > /b > > On Jul 7, 2010, at 3:48 AM, Tony Tin wrote: > > > Hi, > > > > I'm writing dialplan with Lua script. There is a while loop in the > dialplan, I found that the loop is not broke out even the call has been hung > up. I'm wondering whether this is a normal behavior. Thanks. > > > > Regards, > > Tony > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/129e4ac4/attachment.html From tony.tin at noahmedia.com.hk Wed Jul 7 19:39:23 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Thu, 8 Jul 2010 10:39:23 +0800 Subject: [Freeswitch-users] hangup does not break the Lua loop In-Reply-To: <4B80CF4E-6336-4F2B-A118-ED87D6979F90@freeswitch.org> References: <4B80CF4E-6336-4F2B-A118-ED87D6979F90@freeswitch.org> Message-ID: sorry, one more question how does session:ready compare to session:answered ? Thanks Tony On Wed, Jul 7, 2010 at 8:15 PM, Brian West wrote: > Post your code for your while loop... sounds like you're not checking > session:ready to see if its still true. > > /b > > On Jul 7, 2010, at 3:48 AM, Tony Tin wrote: > > > Hi, > > > > I'm writing dialplan with Lua script. There is a while loop in the > dialplan, I found that the loop is not broke out even the call has been hung > up. I'm wondering whether this is a normal behavior. Thanks. > > > > Regards, > > Tony > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/975beacc/attachment.html From tony.tin at noahmedia.com.hk Wed Jul 7 19:57:04 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Thu, 8 Jul 2010 10:57:04 +0800 Subject: [Freeswitch-users] originate call hangup signal In-Reply-To: References: Message-ID: Thanks for your help. It's a 4ESS IDSN and the carrier does provide disconnect supervision, is there any way to bypass it ? Regards, Tony On Thu, Jul 8, 2010 at 7:14 AM, Michael Collins wrote: > It depends on where the "hangup signal" comes from. Is this an analog line? > If so, does the carrier provide disconnect supervision? It's entirely > possible that the other end isn't doing a good job of telling FreeSWITCH > that the call is over. > > -MC > > On Tue, Jul 6, 2010 at 10:35 PM, Tony Tin wrote: > >> Hi, >> >> When I use OpenZAP channel to originate a call, after the called party >> hangup the phone. It takes freeswitch around 40 seconds to catch the hangup >> signal and stop the dial plan. I'm wondering whether there is a way to >> shorten that duration. Thanks. >> >> Regards, >> Tony >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/dfcc3073/attachment.html From msc at freeswitch.org Wed Jul 7 21:42:43 2010 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 7 Jul 2010 21:42:43 -0700 Subject: [Freeswitch-users] originate call hangup signal In-Reply-To: References: Message-ID: <2B4FFD98-7682-44AB-ADB8-9B9700B5AC32@freeswitch.org> Okay, next question: which PRI are you using? Is it Digium-based or Sangoma hardware? If the former then use the libpri method; the latter use freetdm. I think they're both covered on the wiki. You need to get an ISDN trace on the d-chan to see what is actually being sent to/from telco. -MC Sent from my iPhone On Jul 7, 2010, at 7:57 PM, Tony Tin wrote: > Thanks for your help. > > It's a 4ESS IDSN and the carrier does provide disconnect > supervision, is there any way to bypass it ? > > Regards, > Tony > > > On Thu, Jul 8, 2010 at 7:14 AM, Michael Collins > wrote: > It depends on where the "hangup signal" comes from. Is this an > analog line? If so, does the carrier provide disconnect supervision? > It's entirely possible that the other end isn't doing a good job of > telling FreeSWITCH that the call is over. > > -MC > > On Tue, Jul 6, 2010 at 10:35 PM, Tony Tin > wrote: > Hi, > > When I use OpenZAP channel to originate a call, after the called > party hangup the phone. It takes freeswitch around 40 seconds to > catch the hangup signal and stop the dial plan. I'm wondering > whether there is a way to shorten that duration. Thanks. > > Regards, > Tony > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100707/db6a0864/attachment-0001.html From sanms.zhang at gmail.com Thu Jul 8 00:57:46 2010 From: sanms.zhang at gmail.com (chi zhang) Date: Thu, 8 Jul 2010 15:57:46 +0800 Subject: [Freeswitch-users] Problem of FreeSwitch and T.38 Test In-Reply-To: References: <5ECC3A8A-CDF9-4EF4-AD87-A3FAEDFF26AB@freeswitch.org> Message-ID: Now, i have finished T.38 fax receive test with Zoiper, it works well. But, transfer a fax is still Not successful. Previous, i do simulate fax with sipP, but FS return 48(Disconnected after permitted retries). Accidentally, i found softphone: Zoiper has fax function, so retry fax with it, and in diaplan file: default.xml, 9178 was the receive fax number. So i call 9178 with Zoiper(register as 1000), fax receiving is perfect done. Then i test transfer fax function: call to 9179(configured by TX fax in default.xml), but FS return 2 (Timed out waiting for initial communication) , i have no idea about it, reason ? regards sammy 2010/7/2 chi zhang > I got it. > > -------------------log start-------------------------- > recv 576 bytes from udp/[192.168.26.39]:15060 at 01:41:26.500127: > ------------------------------------------------------------------------ > INVITE sip:666666 at 192.168.26.39:25060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 > From: 1000 ;tag=1 > To: 666666 > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: sip:1000 at 192.168.26.39:15060 > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: 184 > > v=0 > o=user1 3748 3748 IN IP4 192.168.26.39 > s=- > c=IN IP4 192.168.26.39 > t=0 0 > m=audio 6000 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11,16 > ------------------------------------------------------------------------ > send 294 bytes to udp/[192.168.26.39]:15060 at 01:41:26.500477: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 > From: 1000 ;tag=1 > To: 666666 > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Content-Length: 0 > > ------------------------------------------------------------------------ > 2010-07-02 09:41:26.498946 [DEBUG] sofia.c:5928 IP 192.168.26.39 Approved > by acl "192.168.26.0/24[] ". Access Granted. > [36m2010-07-02 09:41:26.498946 [NOTICE] switch_channel.c:776 New Channel > sofia/internal/1000 at 192.168.26.39:15060[f588c66c-48c8-4220-a944-8de287adb3ab] > 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_NEW > 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:320 > (sofia/internal/1000 at 192.168.26.39:15060) State NEW > 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4297 Channel sofia/internal/ > 1000 at 192.168.26.39:15060 entering state [received][100] > 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4308 Remote SDP: > v=0 > > o=user1 3748 3748 IN IP4 192.168.26.39 > > s=- > > c=IN IP4 192.168.26.39 > > t=0 0 > > m=audio 6000 RTP/AVP 8 101 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-11,16 > > > 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare > [PCMA:8:8000:20]/[G7221:115:32000:20] > 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare > [PCMA:8:8000:20]/[G7221:107:16000:20] > 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare > [PCMA:8:8000:20]/[G722:9:8000:20] > 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare > [PCMA:8:8000:20]/[PCMU:0:8000:20] > 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare > [PCMA:8:8000:20]/[PCMA:8:8000:20] > 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:2442 Set Codec > sofia/internal/1000 at 192.168.26.39:15060 PCMA/8000 20 ms 160 samples > 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3798 Set 2833 dtmf > send/recv payload to 101 > 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4455 (sofia/internal/ > 1000 at 192.168.26.39:15060) State Change CS_NEW -> CS_INIT > 2010-07-02 09:41:26.542951 [DEBUG] switch_core_session.c:1027 Send signal > sofia/internal/1000 at 192.168.26.39:15060 [BREAK] > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_INIT > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/1000 at 192.168.26.39:15060) State INIT > 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:83 sofia/internal/ > 1000 at 192.168.26.39:15060 SOFIA INIT > 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:119 (sofia/internal/ > 1000 at 192.168.26.39:15060) State Change CS_INIT -> CS_ROUTING > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send signal > sofia/internal/1000 at 192.168.26.39:15060 [BREAK] > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/1000 at 192.168.26.39:15060) State INIT going to sleep > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_ROUTING > 2010-07-02 09:41:26.544953 [DEBUG] switch_channel.c:1471 (sofia/internal/ > 1000 at 192.168.26.39:15060) Callstate Change DOWN -> RINGING > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING > 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:142 sofia/internal/ > 1000 at 192.168.26.39:15060 SOFIA ROUTING > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:77 > sofia/internal/1000 at 192.168.26.39:15060 Standard ROUTING > [32m2010-07-02 09:41:26.544953 [INFO] mod_dialplan_xml.c:331 Processing > 1000->666666 in context public > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->fax] > continue=false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] > destination_number(666666) =~ /^fax$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->4444] > continue=false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [4444] > destination_number(666666) =~ /^(4444)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->fax] > continue=false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] > destination_number(666666) =~ /^fax$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->666666] > continue=false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (PASS) [666666] > destination_number(666666) =~ /^(666666)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action answer() > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action > playback(silence_stream://2000) > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action > rxfax(/tmp/999.tiff) > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action hangup() > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:119 > (sofia/internal/1000 at 192.168.26.39:15060) State Change CS_ROUTING -> > CS_EXECUTE > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send signal > sofia/internal/1000 at 192.168.26.39:15060 [BREAK] > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING going to sleep > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_EXECUTE > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/1000 at 192.168.26.39:15060) State EXECUTE > 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:235 sofia/internal/ > 1000 at 192.168.26.39:15060 SOFIA EXECUTE > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:157 > sofia/internal/1000 at 192.168.26.39:15060 Standard EXECUTE > EXECUTE sofia/internal/1000 at 192.168.26.39:15060 answer() > 2010-07-02 09:41:26.580967 [DEBUG] sofia_glue.c:2682 AUDIO RTP > [sofia/internal/1000 at 192.168.26.39:15060] 192.168.26.39 port 22464 -> > 192.168.26.39 port 6000 codec: 8 ms: 20 > 2010-07-02 09:41:26.580967 [DEBUG] switch_rtp.c:1413 Starting timer [soft] > 160 bytes per 20ms > 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2892 Set 2833 dtmf send > payload to 101 > 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2897 Set 2833 dtmf receive > payload to 101 > 2010-07-02 09:41:26.582952 [DEBUG] mod_sofia.c:669 Local SDP > sofia/internal/1000 at 192.168.26.39:15060: > v=0 > o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=audio 22464 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > send 1091 bytes to udp/[192.168.26.39]:15060 at 01:41:26.584093: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 > From: 1000 ;tag=1 > To: 666666 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 249 > Remote-Party-ID: "666666" > >;party=calling;privacy=off;screen=no > > v=0 > o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=audio 22464 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ------------------------------------------------------------------------ > recv 369 bytes from udp/[192.168.26.39]:15060 at 01:41:26.584205: > ------------------------------------------------------------------------ > ACK sip:666666 at 192.168.26.39:25060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-4 > From: 1000 ;tag=1 > To: 666666 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 ACK > Contact: sip:1000 at 192.168.26.39:15060 > Max-Forwards: 70 > Subject: Performance Test > Content-Length: 0 > > ------------------------------------------------------------------------ > 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/ > 1000 at 192.168.26.39:15060 entering state [completed][200] > 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/ > 1000 at 192.168.26.39:15060 entering state [ready][200] > 2010-07-02 09:41:26.582952 [DEBUG] switch_core_session.c:647 Send signal > sofia/internal/1000 at 192.168.26.39:15060 [BREAK] > 2010-07-02 09:41:26.582952 [DEBUG] switch_channel.c:2494 (sofia/internal/ > 1000 at 192.168.26.39:15060) Callstate Change RINGING -> ACTIVE > [36m2010-07-02 09:41:26.582952 [NOTICE] mod_dptools.c:746 Channel > [sofia/internal/1000 at 192.168.26.39:15060] has been answered > EXECUTE sofia/internal/1000 at 192.168.26.39:15060playback(silence_stream://2000) > 2010-07-02 09:41:26.584953 [DEBUG] switch_ivr_play_say.c:1161 Codec > Activated L16 at 8000hz 1 channels 20ms > 2010-07-02 09:41:28.578954 [DEBUG] switch_ivr_play_say.c:1468 done playing > file > EXECUTE sofia/internal/1000 at 192.168.26.39:15060 rxfax(/tmp/999.tiff) > 2010-07-02 09:41:28.578954 [ERR] mod_spandsp.c:64 This is for fax test: > receive fax > 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:445 trans mode = 1 > 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:591 This is for fax > test: prag go to here!!! > 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1064 Raw read codec > activation Success L16 20000 > 2010-07-02 09:41:28.578954 [DEBUG] switch_core_codec.c:122 sofia/internal/ > 1000 at 192.168.26.39:15060 Push codec L16:10 > 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1080 Raw write codec > activation Success L16 > 2010-07-02 09:41:28.857958 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 > > send 956 bytes to udp/[192.168.21.76]:5060 at 01:41:29.754477: > ------------------------------------------------------------------------ > NOTIFY sip:1001 at 192.168.21.76:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:25060;rport;branch=z9hG4bKS989SFgmXvmNF > Max-Forwards: 70 > From: "1001" ;transport=UDP>;tag=vr6p3K1XK247r > To: "1001" ;tag=26647676 > Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. > CSeq: 132897310 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Event: message-summary > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Subscription-State: terminated;reason=timeout > Content-Type: application/simple-message-summary > Content-Length: 65 > > Messages-Waiting: no > Message-Account: sip:1001 at 192.168.26.39 > > ------------------------------------------------------------------------ > 2010-07-02 09:41:29.757979 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 > recv 389 bytes from udp/[192.168.21.76]:5060 at 01:41:29.758362: > ------------------------------------------------------------------------ > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP 192.168.26.39:25060 > ;rport=25060;branch=z9hG4bKS989SFgmXvmNF > To: "1001";tag=26647676 > From: "1001" ;transport=UDP>;tag=vr6p3K1XK247r > Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. > CSeq: 132897310 NOTIFY > Accept-Language: en > Content-Length: 0 > > ------------------------------------------------------------------------ > > 2010-07-02 09:41:36.557090 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 > recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:36.587253: > ------------------------------------------------------------------------ > INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: sip:666666 at 192.168.26.39:25060 > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: 345 > > v=0 > o=root 0 0 IN IP4 192.168.26.39 > s=Session SDP > c=IN IP4 192.168.26.39 > t=0 0 > m=image 49172 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval:0 > a=T38FaxTranscodingMMR:0 > a=T38FaxTranscodingJBIG:0 > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:200 > a=T38FaxMaxDatagram:72 > a=T38FaxUdpEC:t38UDPRedundancy > ------------------------------------------------------------------------ > send 312 bytes to udp/[192.168.26.39]:25060 at 01:41:36.587546: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 312 bytes from udp/[192.168.26.39]:25060 at 01:41:36.587641: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4297 Channel > sofia/internal/1000 at 192.168.26.39:15060 entering state [received][100] > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Content-Length: 0 > > ------------------------------------------------------------------------ > 2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4308 Remote SDP: > v=0 > > o=root 0 0 IN IP4 192.168.26.39 > > s=Session SDP > > c=IN IP4 192.168.26.39 > > t=0 0 > > m=image 49172 udptl t38 > > a=T38FaxVersion:0 > > a=T38MaxBitRate:9600 > > a=T38FaxFillBitRemoval:0 > > a=T38FaxTranscodingMMR:0 > > a=T38FaxTranscodingJBIG:0 > > a=T38FaxRateManagement:transferredTCF > > a=T38FaxMaxBuffer:200 > > a=T38FaxMaxDatagram:72 > > a=T38FaxUdpEC:t38UDPRedundancy > > > 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:744 T38FaxVersion = 0 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:745 T38MaxBitRate = > 9600 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:746 > T38FaxFillBitRemoval = 1 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:747 > T38FaxTranscodingMMR = 1 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:748 > T38FaxTranscodingJBIG = 1 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:749 > T38FaxRateManagement = 'transferredTCF' > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:750 T38FaxMaxBuffer = > 200 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:751 T38FaxMaxDatagram > = 72 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:752 T38FaxUdpEC = > 't38UDPRedundancy' > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:753 T38VendorInfo = '' > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:754 ip = > '192.168.26.39' > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:756 port = 49172 > 2010-07-02 09:41:36.597094 [DEBUG] mod_sofia.c:1232 IMAGE UDPTL CHANGING > DEST TO: [192.168.26.39:49172] > 2010-07-02 09:41:36.597094 [DEBUG] sofia_glue.c:122 sofia/internal/ > 1000 at 192.168.26.39:15060 image media sdp: > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:445 trans mode = 0 > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:36.597525: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > 2010-07-02 09:41:36.597094 [DEBUG] sofia.c:4297 Channel sofia/internal/ > 1000 at 192.168.26.39:15060 entering state [completed][200] > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:36.597657: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > 2010-07-02 09:41:36.607095 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 > recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:37.089156: > ------------------------------------------------------------------------ > INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: sip:666666 at 192.168.26.39:25060 > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: 345 > > v=0 > o=root 0 0 IN IP4 192.168.26.39 > s=Session SDP > c=IN IP4 192.168.26.39 > t=0 0 > m=image 49172 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval:0 > a=T38FaxTranscodingMMR:0 > a=T38FaxTranscodingJBIG:0 > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:200 > a=T38FaxMaxDatagram:72 > a=T38FaxUdpEC:t38UDPRedundancy > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.089302: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.089395: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.098259: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.098339: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:38.091157: > ------------------------------------------------------------------------ > INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: sip:666666 at 192.168.26.39:25060 > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: 345 > > v=0 > o=root 0 0 IN IP4 192.168.26.39 > s=Session SDP > c=IN IP4 192.168.26.39 > t=0 0 > m=image 49172 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval:0 > a=T38FaxTranscodingMMR:0 > a=T38FaxTranscodingJBIG:0 > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:200 > a=T38FaxMaxDatagram:72 > a=T38FaxUdpEC:t38UDPRedundancy > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.091305: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.091406: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.098260: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.098344: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > This is for fax test: dis 5This is for fax test: cause disconnect 4 > recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:40.093267: > ------------------------------------------------------------------------ > INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: sip:666666 at 192.168.26.39:25060 > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: 345 > > v=0 > o=root 0 0 IN IP4 192.168.26.39 > s=Session SDP > c=IN IP4 192.168.26.39 > t=0 0 > m=image 49172 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval:0 > a=T38FaxTranscodingMMR:0 > a=T38FaxTranscodingJBIG:0 > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:200 > a=T38FaxMaxDatagram:72 > a=T38FaxUdpEC:t38UDPRedundancy > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.093421: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.093541: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.098263: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.098418: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > 2010-07-02 09:41:41.606147 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 > recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:44.095284: > ------------------------------------------------------------------------ > INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: sip:666666 at 192.168.26.39:25060 > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: 345 > > v=0 > o=root 0 0 IN IP4 192.168.26.39 > s=Session SDP > c=IN IP4 192.168.26.39 > t=0 0 > m=image 49172 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval:0 > a=T38FaxTranscodingMMR:0 > a=T38FaxTranscodingJBIG:0 > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:200 > a=T38FaxMaxDatagram:72 > a=T38FaxUdpEC:t38UDPRedundancy > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.095457: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.095556: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.098263: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.098399: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > ------------------------------------------------------------------------ > 2010-07-02 09:41:46.607257 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:48.098287: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > > -----------------------------log end---------------------------------- > > > 2010/7/2 Brian West > > turn sip on >> >> sofia profile xxx siptrace on >> >> /b >> >> On Jul 1, 2010, at 8:23 PM, chi zhang wrote: >> >> > hi,brian >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/223cda48/attachment-0001.html From tony.tin at noahmedia.com.hk Thu Jul 8 01:01:46 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Thu, 8 Jul 2010 16:01:46 +0800 Subject: [Freeswitch-users] originate call hangup signal In-Reply-To: <2B4FFD98-7682-44AB-ADB8-9B9700B5AC32@freeswitch.org> References: <2B4FFD98-7682-44AB-ADB8-9B9700B5AC32@freeswitch.org> Message-ID: It's Digium TE220. I'm using the OpenZAP native stack, because I can not get the outbound call work with libpri compatibility stack. Attached is the freeswitch.log. I'm not sure whether it includes the d-chan trace, though I already enabled the "q931_dump". I originated a call to my mobile on freeswitch console with command "originate OpenZAP/2/A/98855404 6899", I answered the call then hung up, after around 30 seconds, I saw there is terminator event on the console and the call hangup. Thanks Regards, Tony On Thu, Jul 8, 2010 at 12:42 PM, Michael S Collins wrote: > Okay, next question: which PRI are you using? Is it Digium-based or Sangoma > hardware? If the former then use the libpri method; the latter use freetdm. > I think they're both covered on the wiki. You need to get an ISDN trace on > the d-chan to see what is actually being sent to/from telco. > > -MC > > Sent from my iPhone > > On Jul 7, 2010, at 7:57 PM, Tony Tin wrote: > > Thanks for your help. > > It's a 4ESS IDSN and the carrier does provide disconnect supervision, is > there any way to bypass it ? > > Regards, > Tony > > > On Thu, Jul 8, 2010 at 7:14 AM, Michael Collins < > msc at freeswitch.org> wrote: > >> It depends on where the "hangup signal" comes from. Is this an analog >> line? If so, does the carrier provide disconnect supervision? It's entirely >> possible that the other end isn't doing a good job of telling FreeSWITCH >> that the call is over. >> >> -MC >> >> On Tue, Jul 6, 2010 at 10:35 PM, Tony Tin < >> tony.tin at noahmedia.com.hk> wrote: >> >>> Hi, >>> >>> When I use OpenZAP channel to originate a call, after the called party >>> hangup the phone. It takes freeswitch around 40 seconds to catch the hangup >>> signal and stop the dial plan. I'm wondering whether there is a way to >>> shorten that duration. Thanks. >>> >>> Regards, >>> Tony >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/262816c3/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch.log Type: application/octet-stream Size: 24028 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/262816c3/attachment-0001.obj From sanms.zhang at gmail.com Thu Jul 8 01:05:33 2010 From: sanms.zhang at gmail.com (chi zhang) Date: Thu, 8 Jul 2010 16:05:33 +0800 Subject: [Freeswitch-users] Problem of FreeSwitch and T.38 Test In-Reply-To: References: <5ECC3A8A-CDF9-4EF4-AD87-A3FAEDFF26AB@freeswitch.org> Message-ID: This is the log: !!!---------------------------------log start-------------------------------------!!! recv 906 bytes from udp/[192.168.21.76]:5060 at 07:59:05.856966: ------------------------------------------------------------------------ INVITE sip:9179 at 192.168.26.39:25060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.21.76:5060 ;branch=z9hG4bK-d8754z-bc46e07b00e494a3-1---d8754z- Max-Forwards: 70 Contact: To: From: "1001";tag=9b6ed37d Call-ID: YTIxZDI4ZWNmZGViZjFjNmI5ZTEyYWJmNGE1ZjExZWI. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.6751 Content-Length: 329 v=0 o=Zoiper_user 0 0 IN IP4 192.168.21.76 s=Zoiper_session c=IN IP4 192.168.21.76 t=0 0 m=audio 8000 RTP/AVP 3 0 8 110 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ send 369 bytes to udp/[192.168.21.76]:5060 at 07:59:05.857299: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.21.76:5060 ;branch=z9hG4bK-d8754z-bc46e07b00e494a3-1---d8754z- From: "1001";tag=9b6ed37d To: Call-ID: YTIxZDI4ZWNmZGViZjFjNmI5ZTEyYWJmNGE1ZjExZWI. CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ 2010-07-08 15:59:05.857080 [DEBUG] sofia.c:5979 IP 192.168.21.76 Rejected by acl "192.168.26.0/24". Falling back to Digest auth. send 857 bytes to udp/[192.168.21.76]:5060 at 07:59:05.912630: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.21.76:5060 ;branch=z9hG4bK-d8754z-bc46e07b00e494a3-1---d8754z- From: "1001";tag=9b6ed37d To: ;tag=SKU88e5Sa1gSS Call-ID: YTIxZDI4ZWNmZGViZjFjNmI5ZTEyYWJmNGE1ZjExZWI. CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="192.168.26.39", nonce="fba8e3e1-c3d7-4faa-b367-7ba18ad5c276", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 390 bytes from udp/[192.168.21.76]:5060 at 07:59:05.931667: ------------------------------------------------------------------------ ACK sip:9179 at 192.168.26.39:25060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.21.76:5060 ;branch=z9hG4bK-d8754z-bc46e07b00e494a3-1---d8754z- Max-Forwards: 70 To: ;tag=SKU88e5Sa1gSS From: "1001";tag=9b6ed37d Call-ID: YTIxZDI4ZWNmZGViZjFjNmI5ZTEyYWJmNGE1ZjExZWI. CSeq: 1 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 1188 bytes from udp/[192.168.21.76]:5060 at 07:59:05.932411: ------------------------------------------------------------------------ INVITE sip:9179 at 192.168.26.39:25060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.21.76:5060 ;branch=z9hG4bK-d8754z-f0f812882a888ca8-1---d8754z- Max-Forwards: 70 Contact: To: From: "1001";tag=9b6ed37d Call-ID: YTIxZDI4ZWNmZGViZjFjNmI5ZTEyYWJmNGE1ZjExZWI. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Proxy-Authorization: Digest username="1001",realm="192.168.26.39",nonce="fba8e3e1-c3d7-4faa-b367-7ba18ad5c276",uri="sip:9179 at 192.168.26.39:25060 ;transport=UDP",response="02f8fac6c1ea37df7e41b1be478f46d2",cnonce="78fc6a89edc259bb930a8b2e2da67a14",nc=00000001,qop=auth,algorithm=MD5 User-Agent: Zoiper rev.6751 Content-Length: 329 v=0 o=Zoiper_user 0 0 IN IP4 192.168.21.76 s=Zoiper_session c=IN IP4 192.168.21.76 t=0 0 m=audio 8000 RTP/AVP 3 0 8 110 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ------------------------------------------------------------------------ send 369 bytes to udp/[192.168.21.76]:5060 at 07:59:05.932598: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.21.76:5060 ;branch=z9hG4bK-d8754z-f0f812882a888ca8-1---d8754z- From: "1001";tag=9b6ed37d To: Call-ID: YTIxZDI4ZWNmZGViZjFjNmI5ZTEyYWJmNGE1ZjExZWI. CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ 2010-07-08 15:59:05.932082 [DEBUG] sofia.c:5979 IP 192.168.21.76 Rejected by acl "192.168.26.0/24". Falling back to Digest auth. 2010-07-08 15:59:05.971086 [NOTICE] switch_channel.c:776 New Channel sofia/internal/1001 at 192.168.26.39:25060[d072ed1a-b182-4340-bbc8-e16923b4bed6] 2010-07-08 15:59:05.971086 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1001 at 192.168.26.39:25060) Running State Change CS_NEW 2010-07-08 15:59:05.971086 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1001 at 192.168.26.39:25060) State NEW 2010-07-08 15:59:06.037086 [DEBUG] sofia.c:6802 Setting NAT mode based on nat.auto 2010-07-08 15:59:06.037086 [DEBUG] sofia.c:4297 Channel sofia/internal/ 1001 at 192.168.26.39:25060 entering state [received][100] 2010-07-08 15:59:06.037086 [DEBUG] sofia.c:4308 Remote SDP: v=0 o=Zoiper_user 0 0 IN IP4 192.168.21.76 s=Zoiper_session c=IN IP4 192.168.21.76 t=0 0 m=audio 8000 RTP/AVP 3 0 8 110 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2010-07-08 15:59:06.037086 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [GSM:3:8000:20]/[G7221:115:32000:20] 2010-07-08 15:59:06.037086 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [GSM:3:8000:20]/[G7221:107:16000:20] 2010-07-08 15:59:06.037086 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [GSM:3:8000:20]/[G722:9:8000:20] 2010-07-08 15:59:06.037086 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [GSM:3:8000:20]/[PCMU:0:8000:20] 2010-07-08 15:59:06.037086 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [GSM:3:8000:20]/[PCMA:8:8000:20] 2010-07-08 15:59:06.037086 [DEBUG] sofia_glue.c:3859 Audio Codec Compare [GSM:3:8000:20]/[GSM:3:8000:20] 2010-07-08 15:59:06.037086 [DEBUG] sofia_glue.c:2442 Set Codec sofia/internal/1001 at 192.168.26.39:25060 GSM/8000 20 ms 160 samples 2010-07-08 15:59:06.037086 [DEBUG] sofia_glue.c:3798 Set 2833 dtmf send/recv payload to 101 2010-07-08 15:59:06.037086 [DEBUG] sofia.c:4455 (sofia/internal/ 1001 at 192.168.26.39:25060) State Change CS_NEW -> CS_INIT 2010-07-08 15:59:06.037086 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1001 at 192.168.26.39:25060 [BREAK] 2010-07-08 15:59:06.037086 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1001 at 192.168.26.39:25060) Running State Change CS_INIT 2010-07-08 15:59:06.037086 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1001 at 192.168.26.39:25060) State INIT 2010-07-08 15:59:06.037086 [DEBUG] mod_sofia.c:83 sofia/internal/ 1001 at 192.168.26.39:25060 SOFIA INIT 2010-07-08 15:59:06.037086 [DEBUG] mod_sofia.c:119 (sofia/internal/ 1001 at 192.168.26.39:25060) State Change CS_INIT -> CS_ROUTING 2010-07-08 15:59:06.037086 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1001 at 192.168.26.39:25060 [BREAK] 2010-07-08 15:59:06.037086 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1001 at 192.168.26.39:25060) State INIT going to sleep 2010-07-08 15:59:06.037086 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1001 at 192.168.26.39:25060) Running State Change CS_ROUTING 2010-07-08 15:59:06.037086 [DEBUG] switch_channel.c:1471 (sofia/internal/ 1001 at 192.168.26.39:25060) Callstate Change DOWN -> RINGING 2010-07-08 15:59:06.037086 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1001 at 192.168.26.39:25060) State ROUTING 2010-07-08 15:59:06.037086 [DEBUG] mod_sofia.c:142 sofia/internal/ 1001 at 192.168.26.39:25060 SOFIA ROUTING 2010-07-08 15:59:06.037086 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1001 at 192.168.26.39:25060 Standard ROUTING 2010-07-08 15:59:06.037086 [INFO] mod_dialplan_xml.c:331 Processing 1001->9179 in context default Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->unloop] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->tod_example] continue=true Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Action set(open=true) Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->holiday_example] continue=true Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Date/Time Match (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [global-intercept] destination_number(9179) =~ /^886$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [group-intercept] destination_number(9179) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [intercept-ext] destination_number(9179) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->redial] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [redial] destination_number(9179) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->global] continue=true Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Absolute Condition [global] Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [snom-demo-2] destination_number(9179) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [snom-demo-1] destination_number(9179) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [eavesdrop] destination_number(9179) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [eavesdrop] destination_number(9179) =~ /^779$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->call_return] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [call_return] destination_number(9179) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->del-group] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [del-group] destination_number(9179) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->add-group] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [add-group] destination_number(9179) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [call-group-simo] destination_number(9179) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [call-group-order] destination_number(9179) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [extension-intercom] destination_number(9179) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [Local_Extension] destination_number(9179) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->Local_Extension_Skinny] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [Local_Extension_Skinny] destination_number(9179) =~ /^(20[01][0-9])$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->group_dial_sales] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [group_dial_sales] destination_number(9179) =~ /^2000$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->group_dial_support] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [group_dial_support] destination_number(9179) =~ /^2001$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->group_dial_billing] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [group_dial_billing] destination_number(9179) =~ /^2002$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->operator] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [operator] destination_number(9179) =~ /^(operator|0)$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->vmain] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [vmain] destination_number(9179) =~ /^vmain$|^4000$|^\*98$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->sip_uri] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [sip_uri] destination_number(9179) =~ /^sip:(.*)$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->nb_conferences] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [nb_conferences] destination_number(9179) =~ /^(30\d{2})$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->wb_conferences] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [wb_conferences] destination_number(9179) =~ /^(31\d{2})$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->uwb_conferences] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [uwb_conferences] destination_number(9179) =~ /^(32\d{2})$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->cdquality_conferences] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [cdquality_conferences] destination_number(9179) =~ /^(33\d{2})$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->freeswitch_public_conf_via_sip] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(9179) =~ /^9(888|8888|1616|3232)$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->mad_boss_intercom] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [mad_boss_intercom] destination_number(9179) =~ /^0911$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->mad_boss_intercom] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [mad_boss_intercom] destination_number(9179) =~ /^0912$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->mad_boss] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [mad_boss] destination_number(9179) =~ /^0913$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->ivr_demo] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [ivr_demo] destination_number(9179) =~ /^5000$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->dynamic_conference] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [dynamic_conference] destination_number(9179) =~ /^5001$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->rtp_multicast_page] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [rtp_multicast_page] destination_number(9179) =~ /^pagegroup$|^7243$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->park] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [park] destination_number(9179) =~ /^5900$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->unpark] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [unpark] destination_number(9179) =~ /^5901$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->valet_park] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [valet_park] destination_number(9179) =~ /^(6000)$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->valet_park] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [valet_park] destination_number(9179) =~ /^(60\d[1-9])$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->park] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [park] destination_number(9179) =~ /park\+(\d+)/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->unpark] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [unpark] destination_number(9179) =~ /^parking$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->park] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [park] destination_number(9179) =~ /callpark/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->unpark] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [unpark] destination_number(9179) =~ /pickup/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->wait] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [wait] destination_number(9179) =~ /^wait$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->fax_receive] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (FAIL) [fax_receive] destination_number(9179) =~ /^9178$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 parsing [default->fax_transmit] continue=false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Regex (PASS) [fax_transmit] destination_number(9179) =~ /^9179$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Action txfax(/tmp/999.tif) Dialplan: sofia/internal/1001 at 192.168.26.39:25060 Action hangup() 2010-07-08 15:59:06.040087 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1001 at 192.168.26.39:25060) State Change CS_ROUTING -> CS_EXECUTE 2010-07-08 15:59:06.040087 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1001 at 192.168.26.39:25060 [BREAK] 2010-07-08 15:59:06.040087 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1001 at 192.168.26.39:25060) State ROUTING going to sleep 2010-07-08 15:59:06.040087 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1001 at 192.168.26.39:25060) Running State Change CS_EXECUTE 2010-07-08 15:59:06.040087 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1001 at 192.168.26.39:25060) State EXECUTE 2010-07-08 15:59:06.040087 [DEBUG] mod_sofia.c:235 sofia/internal/ 1001 at 192.168.26.39:25060 SOFIA EXECUTE 2010-07-08 15:59:06.040087 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1001 at 192.168.26.39:25060 Standard EXECUTE EXECUTE sofia/internal/1001 at 192.168.26.39:25060 set(open=true) 2010-07-08 15:59:06.040087 [DEBUG] mod_dptools.c:843 sofia/internal/ 1001 at 192.168.26.39:25060 SET [open]=[true] EXECUTE sofia/internal/1001 at 192.168.26.39:25060hash(insert/192.168.26.39-spymap/1001/d072ed1a-b182-4340-bbc8-e16923b4bed6) EXECUTE sofia/internal/1001 at 192.168.26.39:25060hash(insert/192.168.26.39-last_dial/1001/9179) EXECUTE sofia/internal/1001 at 192.168.26.39:25060hash(insert/192.168.26.39-last_dial/global/d072ed1a-b182-4340-bbc8-e16923b4bed6) EXECUTE sofia/internal/1001 at 192.168.26.39:25060 txfax(/tmp/999.tif) 2010-07-08 15:59:06.043087 [ERR] mod_spandsp.c:56 This is for fax test: transfer fax 2010-07-08 15:59:06.088091 [DEBUG] sofia_glue.c:2682 AUDIO RTP [sofia/internal/1001 at 192.168.26.39:25060] 192.168.26.39 port 31920 -> 192.168.21.76 port 8000 codec: 3 ms: 20 2010-07-08 15:59:06.088091 [DEBUG] switch_rtp.c:1413 Starting timer [soft] 160 bytes per 20ms 2010-07-08 15:59:06.091095 [DEBUG] sofia_glue.c:2892 Set 2833 dtmf send payload to 101 2010-07-08 15:59:06.091095 [DEBUG] sofia_glue.c:2897 Set 2833 dtmf receive payload to 101 2010-07-08 15:59:06.091095 [DEBUG] mod_sofia.c:669 Local SDP sofia/internal/ 1001 at 192.168.26.39:25060: v=0 o=FreeSWITCH 1278544026 1278544027 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=audio 31920 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2010-07-08 15:59:06.091095 [DEBUG] switch_core_session.c:647 Send signal sofia/internal/1001 at 192.168.26.39:25060 [BREAK] 2010-07-08 15:59:06.091095 [DEBUG] switch_channel.c:2494 (sofia/internal/ 1001 at 192.168.26.39:25060) Callstate Change RINGING -> ACTIVE 2010-07-08 15:59:06.091095 [NOTICE] mod_spandsp_fax.c:873 Channel [sofia/internal/1001 at 192.168.26.39:25060] has been answered send 1209 bytes to udp/[192.168.21.76]:5060 at 07:59:06.093433: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.21.76:5060 ;branch=z9hG4bK-d8754z-f0f812882a888ca8-1---d8754z- From: "1001";tag=9b6ed37d To: ;tag=tvm1aapX796BN Call-ID: YTIxZDI4ZWNmZGViZjFjNmI5ZTEyYWJmNGE1ZjExZWI. CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 1800;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 248 Remote-Party-ID: "9179" >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1278544026 1278544027 IN IP4 192.168.26.39 s=FreeSWITCH c=IN IP4 192.168.26.39 t=0 0 m=audio 31920 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2010-07-08 15:59:06.091095 [ERR] mod_spandsp_fax.c:445 trans mode = 1 2010-07-08 15:59:06.091095 [DEBUG] sofia.c:4297 Channel sofia/internal/ 1001 at 192.168.26.39:25060 entering state [completed][200] 2010-07-08 15:59:06.091095 [ERR] mod_spandsp_fax.c:591 This is for fax test: prag go to here!!! 2010-07-08 15:59:06.091095 [DEBUG] mod_spandsp_fax.c:1064 Raw read codec activation Success L16 20000 2010-07-08 15:59:06.091095 [DEBUG] switch_core_codec.c:122 sofia/internal/ 1001 at 192.168.26.39:25060 Push codec L16:10 2010-07-08 15:59:06.091095 [DEBUG] mod_spandsp_fax.c:1080 Raw write codec activation Success L16 recv 755 bytes from udp/[192.168.21.76]:5060 at 07:59:06.111565: ------------------------------------------------------------------------ ACK sip:9179 at 192.168.26.39:25060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.21.76:5060 ;branch=z9hG4bK-d8754z-d8c37a91bfb28de8-1---d8754z- Max-Forwards: 70 Contact: To: ;tag=tvm1aapX796BN From: "1001";tag=9b6ed37d Call-ID: YTIxZDI4ZWNmZGViZjFjNmI5ZTEyYWJmNGE1ZjExZWI. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="192.168.26.39",nonce="fba8e3e1-c3d7-4faa-b367-7ba18ad5c276",uri="sip:9179 at 192.168.26.39:25060 ;transport=UDP",response="02f8fac6c1ea37df7e41b1be478f46d2",cnonce="78fc6a89edc259bb930a8b2e2da67a14",nc=00000001,qop=auth,algorithm=MD5 User-Agent: Zoiper rev.6751 Content-Length: 0 ------------------------------------------------------------------------ 2010-07-08 15:59:06.109092 [DEBUG] sofia.c:4297 Channel sofia/internal/ 1001 at 192.168.26.39:25060 entering state [ready][200] 2010-07-08 15:59:06.163089 [DEBUG] switch_rtp.c:2517 Correct ip/port confirmed. 2010-07-08 16:00:07.342919 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 2010-07-08 16:00:07.342919 [DEBUG] mod_spandsp_fax.c:302 result = 2 2010-07-08 16:00:07.342919 [DEBUG] mod_spandsp_fax.c:304 ============================================================================== 2010-07-08 16:00:07.342919 [DEBUG] mod_spandsp_fax.c:316 Fax processing not successful - result (2) Timed out waiting for initial communication. 2010-07-08 16:00:07.342919 [DEBUG] mod_spandsp_fax.c:321 Remote station id: 2010-07-08 16:00:07.342919 [DEBUG] mod_spandsp_fax.c:322 Local station id: SpanDSP Fax Ident 2010-07-08 16:00:07.342919 [DEBUG] mod_spandsp_fax.c:323 Pages transferred: 0 2010-07-08 16:00:07.342919 [DEBUG] mod_spandsp_fax.c:325 Total fax pages: 0 2010-07-08 16:00:07.342919 [DEBUG] mod_spandsp_fax.c:326 Image resolution: 0x0 2010-07-08 16:00:07.342919 [DEBUG] mod_spandsp_fax.c:327 Transfer Rate: 14400 2010-07-08 16:00:07.342919 [DEBUG] mod_spandsp_fax.c:329 ECM status off 2010-07-08 16:00:07.342919 [DEBUG] mod_spandsp_fax.c:330 remote country: 2010-07-08 16:00:07.342919 [DEBUG] mod_spandsp_fax.c:331 remote vendor: 2010-07-08 16:00:07.342919 [DEBUG] mod_spandsp_fax.c:332 remote model: 2010-07-08 16:00:07.342919 [DEBUG] mod_spandsp_fax.c:334 ============================================================================== 2010-07-08 16:00:07.362919 [DEBUG] switch_core_codec.c:146 sofia/internal/ 1001 at 192.168.26.39:25060 Restore previous codec GSM:3. EXECUTE sofia/internal/1001 at 192.168.26.39:25060 hangup() 2010-07-08 16:00:07.362919 [DEBUG] switch_channel.c:2261 (sofia/internal/ 1001 at 192.168.26.39:25060) Callstate Change ACTIVE -> HANGUP 2010-07-08 16:00:07.362919 [NOTICE] mod_dptools.c:732 Hangup sofia/internal/ 1001 at 192.168.26.39:25060 [CS_EXECUTE] [NORMAL_CLEARING] 2010-07-08 16:00:07.362919 [DEBUG] switch_channel.c:2277 Send signal sofia/internal/1001 at 192.168.26.39:25060 [KILL] 2010-07-08 16:00:07.362919 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1001 at 192.168.26.39:25060 [BREAK] 2010-07-08 16:00:07.362919 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1001 at 192.168.26.39:25060) State EXECUTE going to sleep 2010-07-08 16:00:07.362919 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1001 at 192.168.26.39:25060) Running State Change CS_HANGUP 2010-07-08 16:00:07.362919 [DEBUG] switch_core_state_machine.c:500 (sofia/internal/1001 at 192.168.26.39:25060) State HANGUP 2010-07-08 16:00:07.362919 [DEBUG] mod_sofia.c:447 Channel sofia/internal/ 1001 at 192.168.26.39:25060 hanging up, cause: NORMAL_CLEARING 2010-07-08 16:00:07.420917 [DEBUG] mod_sofia.c:490 Sending BYE to sofia/internal/1001 at 192.168.26.39:25060 2010-07-08 16:00:07.422926 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1001 at 192.168.26.39:25060 Standard HANGUP, cause: NORMAL_CLEARING send 690 bytes to udp/[192.168.21.76]:5060 at 08:00:07.423127: ------------------------------------------------------------------------ BYE sip:1001 at 192.168.21.76:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.26.39:25060;rport;branch=z9hG4bKN7S88805r15eF Max-Forwards: 70 From: ;tag=tvm1aapX796BN To: "1001" ;tag=9b6ed37d Call-ID: YTIxZDI4ZWNmZGViZjFjNmI5ZTEyYWJmNGE1ZjExZWI. CSeq: 133169667 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE 2010-07-08 16:00:07.422926 [DEBUG] switch_core_state_machine.c:500 (sofia/internal/1001 at 192.168.26.39:25060) State HANGUP going to sleep Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ 2010-07-08 16:00:07.422926 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1001 at 192.168.26.39:25060) State Change CS_HANGUP -> CS_REPORTING 2010-07-08 16:00:07.422926 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1001 at 192.168.26.39:25060 [BREAK] 2010-07-08 16:00:07.422926 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1001 at 192.168.26.39:25060) Running State Change CS_REPORTING 2010-07-08 16:00:07.422926 [DEBUG] switch_core_state_machine.c:591 (sofia/internal/1001 at 192.168.26.39:25060) State REPORTING 2010-07-08 16:00:07.422926 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1001 at 192.168.26.39:25060 Standard REPORTING, cause: NORMAL_CLEARING 2010-07-08 16:00:07.422926 [DEBUG] switch_core_state_machine.c:591 (sofia/internal/1001 at 192.168.26.39:25060) State REPORTING going to sleep 2010-07-08 16:00:07.422926 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1001 at 192.168.26.39:25060) State Change CS_REPORTING -> CS_DESTROY 2010-07-08 16:00:07.422926 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1001 at 192.168.26.39:25060 [BREAK] 2010-07-08 16:00:07.422926 [DEBUG] switch_core_session.c:1175 Session 1 (sofia/internal/1001 at 192.168.26.39:25060) Locked, Waiting on external entities 2010-07-08 16:00:07.422926 [NOTICE] switch_core_session.c:1193 Session 1 (sofia/internal/1001 at 192.168.26.39:25060) Ended 2010-07-08 16:00:07.422926 [NOTICE] switch_core_session.c:1195 Close Channel sofia/internal/1001 at 192.168.26.39:25060 [CS_DESTROY] 2010-07-08 16:00:07.422926 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/1001 at 192.168.26.39:25060) Callstate Change HANGUP -> DOWN 2010-07-08 16:00:07.422926 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/1001 at 192.168.26.39:25060) Running State Change CS_DESTROY 2010-07-08 16:00:07.422926 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1001 at 192.168.26.39:25060) State DESTROY 2010-07-08 16:00:07.422926 [DEBUG] mod_sofia.c:352 sofia/internal/ 1001 at 192.168.26.39:25060 SOFIA DESTROY 2010-07-08 16:00:07.422926 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1001 at 192.168.26.39:25060 Standard DESTROY 2010-07-08 16:00:07.422926 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1001 at 192.168.26.39:25060) State DESTROY going to sleep recv 413 bytes from udp/[192.168.21.76]:5060 at 08:00:07.628924: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.39:25060 ;rport=25060;branch=z9hG4bKN7S88805r15eF Contact: To: "1001";tag=9b6ed37d From: ;tag=tvm1aapX796BN Call-ID: YTIxZDI4ZWNmZGViZjFjNmI5ZTEyYWJmNGE1ZjExZWI. CSeq: 133169667 BYE User-Agent: Zoiper rev.6751 Content-Length: 0 !!!---------------------------------log end-------------------------------------!!! 2010/7/8 chi zhang > Now, i have finished T.38 fax receive test with Zoiper, it works well. > But, transfer a fax is still Not successful. > Previous, i do simulate fax with sipP, but FS return 48(Disconnected > after permitted retries). > Accidentally, i found softphone: Zoiper has fax function, so retry fax > with it, and in diaplan file: default.xml, 9178 was the receive fax number. > So i call 9178 with Zoiper(register as 1000), fax receiving is perfect done. > Then i test transfer fax function: call to 9179(configured by TX fax in > default.xml), but FS return 2 (Timed out waiting for initial communication) > , i have no idea about it, reason ? > > regards > sammy > > > > 2010/7/2 chi zhang > > I got it. >> >> -------------------log start-------------------------- >> recv 576 bytes from udp/[192.168.26.39]:15060 at 01:41:26.500127: >> >> ------------------------------------------------------------------------ >> INVITE sip:666666 at 192.168.26.39:25060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >> From: 1000 ;tag=1 >> To: 666666 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:1000 at 192.168.26.39:15060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 184 >> >> v=0 >> o=user1 3748 3748 IN IP4 192.168.26.39 >> s=- >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=audio 6000 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-11,16 >> >> ------------------------------------------------------------------------ >> send 294 bytes to udp/[192.168.26.39]:15060 at 01:41:26.500477: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >> From: 1000 ;tag=1 >> To: 666666 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:26.498946 [DEBUG] sofia.c:5928 IP 192.168.26.39 Approved >> by acl "192.168.26.0/24[] ". Access >> Granted. >> [36m2010-07-02 09:41:26.498946 [NOTICE] switch_channel.c:776 New Channel >> sofia/internal/1000 at 192.168.26.39:15060[f588c66c-48c8-4220-a944-8de287adb3ab] >> 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_NEW >> 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:320 >> (sofia/internal/1000 at 192.168.26.39:15060) State NEW >> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4297 Channel sofia/internal/ >> 1000 at 192.168.26.39:15060 entering state [received][100] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4308 Remote SDP: >> v=0 >> >> o=user1 3748 3748 IN IP4 192.168.26.39 >> >> s=- >> >> c=IN IP4 192.168.26.39 >> >> t=0 0 >> >> m=audio 6000 RTP/AVP 8 101 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-11,16 >> >> >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >> [PCMA:8:8000:20]/[G7221:115:32000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >> [PCMA:8:8000:20]/[G7221:107:16000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >> [PCMA:8:8000:20]/[G722:9:8000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >> [PCMA:8:8000:20]/[PCMU:0:8000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >> [PCMA:8:8000:20]/[PCMA:8:8000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:2442 Set Codec >> sofia/internal/1000 at 192.168.26.39:15060 PCMA/8000 20 ms 160 samples >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3798 Set 2833 dtmf >> send/recv payload to 101 >> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4455 (sofia/internal/ >> 1000 at 192.168.26.39:15060) State Change CS_NEW -> CS_INIT >> 2010-07-02 09:41:26.542951 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_INIT >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 >> (sofia/internal/1000 at 192.168.26.39:15060) State INIT >> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:83 sofia/internal/ >> 1000 at 192.168.26.39:15060 SOFIA INIT >> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:119 (sofia/internal/ >> 1000 at 192.168.26.39:15060) State Change CS_INIT -> CS_ROUTING >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 >> (sofia/internal/1000 at 192.168.26.39:15060) State INIT going to sleep >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_ROUTING >> 2010-07-02 09:41:26.544953 [DEBUG] switch_channel.c:1471 (sofia/internal/ >> 1000 at 192.168.26.39:15060) Callstate Change DOWN -> RINGING >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING >> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:142 sofia/internal/ >> 1000 at 192.168.26.39:15060 SOFIA ROUTING >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:77 >> sofia/internal/1000 at 192.168.26.39:15060 Standard ROUTING >> [32m2010-07-02 09:41:26.544953 [INFO] mod_dialplan_xml.c:331 Processing >> 1000->666666 in context public >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->fax] >> continue=false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] >> destination_number(666666) =~ /^fax$/ break=on-false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->4444] >> continue=false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [4444] >> destination_number(666666) =~ /^(4444)$/ break=on-false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->fax] >> continue=false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] >> destination_number(666666) =~ /^fax$/ break=on-false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing >> [public->666666] continue=false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (PASS) [666666] >> destination_number(666666) =~ /^(666666)$/ break=on-false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action answer() >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action >> playback(silence_stream://2000) >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action >> rxfax(/tmp/999.tiff) >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action hangup() >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:119 >> (sofia/internal/1000 at 192.168.26.39:15060) State Change CS_ROUTING -> >> CS_EXECUTE >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING going to sleep >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_EXECUTE >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:348 >> (sofia/internal/1000 at 192.168.26.39:15060) State EXECUTE >> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:235 sofia/internal/ >> 1000 at 192.168.26.39:15060 SOFIA EXECUTE >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:157 >> sofia/internal/1000 at 192.168.26.39:15060 Standard EXECUTE >> EXECUTE sofia/internal/1000 at 192.168.26.39:15060 answer() >> 2010-07-02 09:41:26.580967 [DEBUG] sofia_glue.c:2682 AUDIO RTP >> [sofia/internal/1000 at 192.168.26.39:15060] 192.168.26.39 port 22464 -> >> 192.168.26.39 port 6000 codec: 8 ms: 20 >> 2010-07-02 09:41:26.580967 [DEBUG] switch_rtp.c:1413 Starting timer [soft] >> 160 bytes per 20ms >> 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2892 Set 2833 dtmf send >> payload to 101 >> 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2897 Set 2833 dtmf receive >> payload to 101 >> 2010-07-02 09:41:26.582952 [DEBUG] mod_sofia.c:669 Local SDP >> sofia/internal/1000 at 192.168.26.39:15060: >> v=0 >> o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=audio 22464 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> >> send 1091 bytes to udp/[192.168.26.39]:15060 at 01:41:26.584093: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >> From: 1000 ;tag=1 >> To: 666666 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 249 >> Remote-Party-ID: "666666" >> >;party=calling;privacy=off;screen=no >> >> v=0 >> o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=audio 22464 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> >> ------------------------------------------------------------------------ >> recv 369 bytes from udp/[192.168.26.39]:15060 at 01:41:26.584205: >> >> ------------------------------------------------------------------------ >> ACK sip:666666 at 192.168.26.39:25060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-4 >> From: 1000 ;tag=1 >> To: 666666 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 ACK >> Contact: sip:1000 at 192.168.26.39:15060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/ >> 1000 at 192.168.26.39:15060 entering state [completed][200] >> 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/ >> 1000 at 192.168.26.39:15060 entering state [ready][200] >> 2010-07-02 09:41:26.582952 [DEBUG] switch_core_session.c:647 Send signal >> sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >> 2010-07-02 09:41:26.582952 [DEBUG] switch_channel.c:2494 (sofia/internal/ >> 1000 at 192.168.26.39:15060) Callstate Change RINGING -> ACTIVE >> [36m2010-07-02 09:41:26.582952 [NOTICE] mod_dptools.c:746 Channel >> [sofia/internal/1000 at 192.168.26.39:15060] has been answered >> EXECUTE sofia/internal/1000 at 192.168.26.39:15060playback(silence_stream://2000) >> 2010-07-02 09:41:26.584953 [DEBUG] switch_ivr_play_say.c:1161 Codec >> Activated L16 at 8000hz 1 channels 20ms >> 2010-07-02 09:41:28.578954 [DEBUG] switch_ivr_play_say.c:1468 done playing >> file >> EXECUTE sofia/internal/1000 at 192.168.26.39:15060 rxfax(/tmp/999.tiff) >> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp.c:64 This is for fax test: >> receive fax >> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:445 trans mode = 1 >> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:591 This is for fax >> test: prag go to here!!! >> 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1064 Raw read codec >> activation Success L16 20000 >> 2010-07-02 09:41:28.578954 [DEBUG] switch_core_codec.c:122 sofia/internal/ >> 1000 at 192.168.26.39:15060 Push codec L16:10 >> 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1080 Raw write codec >> activation Success L16 >> 2010-07-02 09:41:28.857958 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 >> >> send 956 bytes to udp/[192.168.21.76]:5060 at 01:41:29.754477: >> >> ------------------------------------------------------------------------ >> NOTIFY sip:1001 at 192.168.21.76:5060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;rport;branch=z9hG4bKS989SFgmXvmNF >> Max-Forwards: 70 >> From: "1001" > ;transport=UDP>;tag=vr6p3K1XK247r >> To: "1001" ;tag=26647676 >> Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. >> CSeq: 132897310 NOTIFY >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Event: message-summary >> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer >> Subscription-State: terminated;reason=timeout >> Content-Type: application/simple-message-summary >> Content-Length: 65 >> >> Messages-Waiting: no >> Message-Account: sip:1001 at 192.168.26.39 >> >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:29.757979 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 >> recv 389 bytes from udp/[192.168.21.76]:5060 at 01:41:29.758362: >> >> ------------------------------------------------------------------------ >> SIP/2.0 481 Call/Transaction Does Not Exist >> Via: SIP/2.0/UDP 192.168.26.39:25060 >> ;rport=25060;branch=z9hG4bKS989SFgmXvmNF >> To: "1001";tag=26647676 >> From: "1001"> ;transport=UDP>;tag=vr6p3K1XK247r >> Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. >> CSeq: 132897310 NOTIFY >> Accept-Language: en >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> >> 2010-07-02 09:41:36.557090 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:36.587253: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 312 bytes to udp/[192.168.26.39]:25060 at 01:41:36.587546: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> recv 312 bytes from udp/[192.168.26.39]:25060 at 01:41:36.587641: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4297 Channel >> sofia/internal/1000 at 192.168.26.39:15060 entering state [received][100] >> >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4308 Remote SDP: >> v=0 >> >> o=root 0 0 IN IP4 192.168.26.39 >> >> s=Session SDP >> >> c=IN IP4 192.168.26.39 >> >> t=0 0 >> >> m=image 49172 udptl t38 >> >> a=T38FaxVersion:0 >> >> a=T38MaxBitRate:9600 >> >> a=T38FaxFillBitRemoval:0 >> >> a=T38FaxTranscodingMMR:0 >> >> a=T38FaxTranscodingJBIG:0 >> >> a=T38FaxRateManagement:transferredTCF >> >> a=T38FaxMaxBuffer:200 >> >> a=T38FaxMaxDatagram:72 >> >> a=T38FaxUdpEC:t38UDPRedundancy >> >> >> 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:744 T38FaxVersion = 0 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:745 T38MaxBitRate = >> 9600 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:746 >> T38FaxFillBitRemoval = 1 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:747 >> T38FaxTranscodingMMR = 1 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:748 >> T38FaxTranscodingJBIG = 1 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:749 >> T38FaxRateManagement = 'transferredTCF' >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:750 T38FaxMaxBuffer = >> 200 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:751 T38FaxMaxDatagram >> = 72 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:752 T38FaxUdpEC = >> 't38UDPRedundancy' >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:753 T38VendorInfo = >> '' >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:754 ip = >> '192.168.26.39' >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:756 port = 49172 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_sofia.c:1232 IMAGE UDPTL CHANGING >> DEST TO: [192.168.26.39:49172] >> 2010-07-02 09:41:36.597094 [DEBUG] sofia_glue.c:122 sofia/internal/ >> 1000 at 192.168.26.39:15060 image media sdp: >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:445 trans mode = 0 >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:36.597525: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:36.597094 [DEBUG] sofia.c:4297 Channel sofia/internal/ >> 1000 at 192.168.26.39:15060 entering state [completed][200] >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:36.597657: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:36.607095 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:37.089156: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.089302: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.089395: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.098259: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.098339: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:38.091157: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.091305: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.091406: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.098260: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.098344: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> This is for fax test: dis 5This is for fax test: cause disconnect 4 >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:40.093267: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.093421: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.093541: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.098263: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.098418: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:41.606147 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:44.095284: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.095457: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.095556: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.098263: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.098399: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:46.607257 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:48.098287: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> >> -----------------------------log end---------------------------------- >> >> >> 2010/7/2 Brian West >> >> turn sip on >>> >>> sofia profile xxx siptrace on >>> >>> /b >>> >>> On Jul 1, 2010, at 8:23 PM, chi zhang wrote: >>> >>> > hi,brian >>> > >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/94ac68c1/attachment-0001.html From david.ponzone at gmail.com Thu Jul 8 01:07:29 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 8 Jul 2010 10:07:29 +0200 Subject: [Freeswitch-users] Problem of FreeSwitch and T.38 Test In-Reply-To: References: <5ECC3A8A-CDF9-4EF4-AD87-A3FAEDFF26AB@freeswitch.org> Message-ID: <12B00ED2-34A6-414C-A0C6-590A963E3CD2@gmail.com> Chi, I am not sure to understand what you mean by "transfer fax function". Can you clear that up please ? Thanks David Ponzone Direction Technique email: david.ponzone at 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 pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 08/07/2010 ? 09:57, chi zhang a ?crit : > Now, i have finished T.38 fax receive test with Zoiper, it > works well. But, transfer a fax is still Not successful. > Previous, i do simulate fax with sipP, but FS return > 48(Disconnected after permitted retries). > Accidentally, i found softphone: Zoiper has fax function, so > retry fax with it, and in diaplan file: default.xml, 9178 was the > receive fax number. So i call 9178 with Zoiper(register as 1000), > fax receiving is perfect done. > Then i test transfer fax function: call to 9179(configured by TX > fax in default.xml), but FS return 2 (Timed out waiting for initial > communication) , i have no idea about it, reason ? > > regards > sammy > > > > 2010/7/2 chi zhang > I got it. > > -------------------log start-------------------------- > recv 576 bytes from udp/[192.168.26.39]:15060 at 01:41:26.500127: > > ------------------------------------------------------------------------ > INVITE sip:666666 at 192.168.26.39:25060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 > From: 1000 ;tag=1 > To: 666666 > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: sip:1000 at 192.168.26.39:15060 > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: 184 > > v=0 > o=user1 3748 3748 IN IP4 192.168.26.39 > s=- > c=IN IP4 192.168.26.39 > t=0 0 > m=audio 6000 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11,16 > > ------------------------------------------------------------------------ > send 294 bytes to udp/[192.168.26.39]:15060 at 01:41:26.500477: > > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 > From: 1000 ;tag=1 > To: 666666 > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Content-Length: 0 > > > ------------------------------------------------------------------------ > 2010-07-02 09:41:26.498946 [DEBUG] sofia.c:5928 IP 192.168.26.39 > Approved by acl "192.168.26.0/24[]". Access Granted. > [36m2010-07-02 09:41:26.498946 [NOTICE] switch_channel.c:776 New > Channel sofia/internal/1000 at 192.168.26.39:15060 [f588c66c-48c8-4220- > a944-8de287adb3ab] > 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39 > :15060) Running State Change CS_NEW > 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1000 at 192.168.26.39 > :15060) State NEW > 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4297 Channel sofia/internal/1000 at 192.168.26.39 > :15060 entering state [received][100] > 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4308 Remote SDP: > v=0 > > o=user1 3748 3748 IN IP4 192.168.26.39 > > s=- > > c=IN IP4 192.168.26.39 > > t=0 0 > > m=audio 6000 RTP/AVP 8 101 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-11,16 > > > 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec > Compare [PCMA:8:8000:20]/[G7221:115:32000:20] > 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec > Compare [PCMA:8:8000:20]/[G7221:107:16000:20] > 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec > Compare [PCMA:8:8000:20]/[G722:9:8000:20] > 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec > Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] > 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec > Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] > 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:2442 Set Codec sofia/internal/1000 at 192.168.26.39 > :15060 PCMA/8000 20 ms 160 samples > 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3798 Set 2833 dtmf > send/recv payload to 101 > 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4455 (sofia/internal/1000 at 192.168.26.39 > :15060) State Change CS_NEW -> CS_INIT > 2010-07-02 09:41:26.542951 [DEBUG] switch_core_session.c:1027 Send > signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39 > :15060) Running State Change CS_INIT > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.26.39 > :15060) State INIT > 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:83 sofia/internal/1000 at 192.168.26.39 > :15060 SOFIA INIT > 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:119 (sofia/internal/1000 at 192.168.26.39 > :15060) State Change CS_INIT -> CS_ROUTING > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send > signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.26.39 > :15060) State INIT going to sleep > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39 > :15060) Running State Change CS_ROUTING > 2010-07-02 09:41:26.544953 [DEBUG] switch_channel.c:1471 (sofia/internal/1000 at 192.168.26.39 > :15060) Callstate Change DOWN -> RINGING > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.26.39 > :15060) State ROUTING > 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:142 sofia/internal/1000 at 192.168.26.39 > :15060 SOFIA ROUTING > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1000 at 192.168.26.39 > :15060 Standard ROUTING > [32m2010-07-02 09:41:26.544953 [INFO] mod_dialplan_xml.c:331 > Processing 1000->666666 in context public > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public- > >fax] continue=false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] > destination_number(666666) =~ /^fax$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public- > >4444] continue=false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) > [4444] destination_number(666666) =~ /^(4444)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public- > >fax] continue=false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] > destination_number(666666) =~ /^fax$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public- > >666666] continue=false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (PASS) > [666666] destination_number(666666) =~ /^(666666)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action answer() > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action > playback(silence_stream://2000) > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action rxfax(/tmp/ > 999.tiff) > Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action hangup() > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1000 at 192.168.26.39 > :15060) State Change CS_ROUTING -> CS_EXECUTE > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send > signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.26.39 > :15060) State ROUTING going to sleep > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39 > :15060) Running State Change CS_EXECUTE > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1000 at 192.168.26.39 > :15060) State EXECUTE > 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:235 sofia/internal/1000 at 192.168.26.39 > :15060 SOFIA EXECUTE > 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1000 at 192.168.26.39 > :15060 Standard EXECUTE > EXECUTE sofia/internal/1000 at 192.168.26.39:15060 answer() > 2010-07-02 09:41:26.580967 [DEBUG] sofia_glue.c:2682 AUDIO RTP [sofia/internal/1000 at 192.168.26.39 > :15060] 192.168.26.39 port 22464 -> 192.168.26.39 port 6000 codec: 8 > ms: 20 > 2010-07-02 09:41:26.580967 [DEBUG] switch_rtp.c:1413 Starting timer > [soft] 160 bytes per 20ms > 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2892 Set 2833 dtmf > send payload to 101 > 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2897 Set 2833 dtmf > receive payload to 101 > 2010-07-02 09:41:26.582952 [DEBUG] mod_sofia.c:669 Local SDP sofia/internal/1000 at 192.168.26.39 > :15060: > v=0 > o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=audio 22464 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > send 1091 bytes to udp/[192.168.26.39]:15060 at 01:41:26.584093: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 > From: 1000 ;tag=1 > To: 666666 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, presence, dialog, line-seize, call- > info, sla, include-session-description, presence.winfo, message- > summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 249 > Remote-Party-ID: "666666" 666666 at 192.168.26.39>;party=calling;privacy=off;screen=no > > v=0 > o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=audio 22464 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > ------------------------------------------------------------------------ > recv 369 bytes from udp/[192.168.26.39]:15060 at 01:41:26.584205: > > ------------------------------------------------------------------------ > ACK sip:666666 at 192.168.26.39:25060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-4 > From: 1000 ;tag=1 > To: 666666 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 ACK > Contact: sip:1000 at 192.168.26.39:15060 > Max-Forwards: 70 > Subject: Performance Test > Content-Length: 0 > > > ------------------------------------------------------------------------ > 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/1000 at 192.168.26.39 > :15060 entering state [completed][200] > 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/1000 at 192.168.26.39 > :15060 entering state [ready][200] > 2010-07-02 09:41:26.582952 [DEBUG] switch_core_session.c:647 Send > signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] > 2010-07-02 09:41:26.582952 [DEBUG] switch_channel.c:2494 (sofia/internal/1000 at 192.168.26.39 > :15060) Callstate Change RINGING -> ACTIVE > [36m2010-07-02 09:41:26.582952 [NOTICE] mod_dptools.c:746 Channel [sofia/internal/1000 at 192.168.26.39 > :15060] has been answered > EXECUTE sofia/internal/1000 at 192.168.26.39:15060 > playback(silence_stream://2000) > 2010-07-02 09:41:26.584953 [DEBUG] switch_ivr_play_say.c:1161 Codec > Activated L16 at 8000hz 1 channels 20ms > 2010-07-02 09:41:28.578954 [DEBUG] switch_ivr_play_say.c:1468 done > playing file > EXECUTE sofia/internal/1000 at 192.168.26.39:15060 rxfax(/tmp/999.tiff) > 2010-07-02 09:41:28.578954 [ERR] mod_spandsp.c:64 This is for fax > test: receive fax > 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:445 trans mode = 1 > 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:591 This is for > fax test: prag go to here!!! > 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1064 Raw read > codec activation Success L16 20000 > 2010-07-02 09:41:28.578954 [DEBUG] switch_core_codec.c:122 sofia/internal/1000 at 192.168.26.39 > :15060 Push codec L16:10 > 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1080 Raw write > codec activation Success L16 > 2010-07-02 09:41:28.857958 [ERR] mod_spandsp_fax.c:1119 pvt- > >t38_mode = 0 > > send 956 bytes to udp/[192.168.21.76]:5060 at 01:41:29.754477: > > ------------------------------------------------------------------------ > NOTIFY sip:1001 at 192.168.21.76:5060 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.26.39:25060;rport;branch=z9hG4bKS989SFgmXvmNF > Max-Forwards: 70 > From: "1001" 1001 at 192.168.26.39:25060;transport=UDP>;tag=vr6p3K1XK247r > To: "1001" 1001 at 192.168.26.39:25060;transport=UDP>;tag=26647676 > Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. > CSeq: 132897310 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Event: message-summary > Allow-Events: talk, hold, presence, dialog, line-seize, call- > info, sla, include-session-description, presence.winfo, message- > summary, refer > Subscription-State: terminated;reason=timeout > Content-Type: application/simple-message-summary > Content-Length: 65 > > Messages-Waiting: no > Message-Account: sip:1001 at 192.168.26.39 > > > ------------------------------------------------------------------------ > 2010-07-02 09:41:29.757979 [ERR] mod_spandsp_fax.c:1119 pvt- > >t38_mode = 0 > recv 389 bytes from udp/[192.168.21.76]:5060 at 01:41:29.758362: > > ------------------------------------------------------------------------ > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP > 192.168.26.39:25060;rport=25060;branch=z9hG4bKS989SFgmXvmNF > To: "1001";tag=26647676 > From: "1001" 1001 at 192.168.26.39:25060;transport=UDP>;tag=vr6p3K1XK247r > Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. > CSeq: 132897310 NOTIFY > Accept-Language: en > Content-Length: 0 > > > ------------------------------------------------------------------------ > > 2010-07-02 09:41:36.557090 [ERR] mod_spandsp_fax.c:1119 pvt- > >t38_mode = 0 > recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:36.587253: > > ------------------------------------------------------------------------ > INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: sip:666666 at 192.168.26.39:25060 > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: 345 > > v=0 > o=root 0 0 IN IP4 192.168.26.39 > s=Session SDP > c=IN IP4 192.168.26.39 > t=0 0 > m=image 49172 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval:0 > a=T38FaxTranscodingMMR:0 > a=T38FaxTranscodingJBIG:0 > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:200 > a=T38FaxMaxDatagram:72 > a=T38FaxUdpEC:t38UDPRedundancy > > ------------------------------------------------------------------------ > send 312 bytes to udp/[192.168.26.39]:25060 at 01:41:36.587546: > > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Content-Length: 0 > > > ------------------------------------------------------------------------ > recv 312 bytes from udp/[192.168.26.39]:25060 at 01:41:36.587641: > > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4297 > Channel sofia/internal/1000 at 192.168.26.39:15060 entering state > [received][100] > > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Content-Length: 0 > > > ------------------------------------------------------------------------ > 2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4308 Remote SDP: > v=0 > > o=root 0 0 IN IP4 192.168.26.39 > > s=Session SDP > > c=IN IP4 192.168.26.39 > > t=0 0 > > m=image 49172 udptl t38 > > a=T38FaxVersion:0 > > a=T38MaxBitRate:9600 > > a=T38FaxFillBitRemoval:0 > > a=T38FaxTranscodingMMR:0 > > a=T38FaxTranscodingJBIG:0 > > a=T38FaxRateManagement:transferredTCF > > a=T38FaxMaxBuffer:200 > > a=T38FaxMaxDatagram:72 > > a=T38FaxUdpEC:t38UDPRedundancy > > > 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:1119 pvt- > >t38_mode = 0 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:744 > T38FaxVersion = 0 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:745 > T38MaxBitRate = 9600 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:746 > T38FaxFillBitRemoval = 1 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:747 > T38FaxTranscodingMMR = 1 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:748 > T38FaxTranscodingJBIG = 1 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:749 > T38FaxRateManagement = 'transferredTCF' > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:750 > T38FaxMaxBuffer = 200 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:751 > T38FaxMaxDatagram = 72 > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:752 T38FaxUdpEC > = 't38UDPRedundancy' > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:753 > T38VendorInfo = '' > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:754 ip = > '192.168.26.39' > 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:756 port = 49172 > 2010-07-02 09:41:36.597094 [DEBUG] mod_sofia.c:1232 IMAGE UDPTL > CHANGING DEST TO: [192.168.26.39:49172] > 2010-07-02 09:41:36.597094 [DEBUG] sofia_glue.c:122 sofia/internal/1000 at 192.168.26.39 > :15060 image media sdp: > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:445 trans mode = 0 > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:36.597525: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > 2010-07-02 09:41:36.597094 [DEBUG] sofia.c:4297 Channel sofia/internal/1000 at 192.168.26.39 > :15060 entering state [completed][200] > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:36.597657: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > 2010-07-02 09:41:36.607095 [ERR] mod_spandsp_fax.c:1119 pvt- > >t38_mode = 1 > recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:37.089156: > > ------------------------------------------------------------------------ > INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: sip:666666 at 192.168.26.39:25060 > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: 345 > > v=0 > o=root 0 0 IN IP4 192.168.26.39 > s=Session SDP > c=IN IP4 192.168.26.39 > t=0 0 > m=image 49172 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval:0 > a=T38FaxTranscodingMMR:0 > a=T38FaxTranscodingJBIG:0 > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:200 > a=T38FaxMaxDatagram:72 > a=T38FaxUdpEC:t38UDPRedundancy > > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.089302: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.089395: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.098259: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.098339: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:38.091157: > > ------------------------------------------------------------------------ > INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: sip:666666 at 192.168.26.39:25060 > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: 345 > > v=0 > o=root 0 0 IN IP4 192.168.26.39 > s=Session SDP > c=IN IP4 192.168.26.39 > t=0 0 > m=image 49172 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval:0 > a=T38FaxTranscodingMMR:0 > a=T38FaxTranscodingJBIG:0 > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:200 > a=T38FaxMaxDatagram:72 > a=T38FaxUdpEC:t38UDPRedundancy > > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.091305: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.091406: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.098260: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.098344: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > This is for fax test: dis 5This is for fax test: cause disconnect 4 > recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:40.093267: > > ------------------------------------------------------------------------ > INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: sip:666666 at 192.168.26.39:25060 > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: 345 > > v=0 > o=root 0 0 IN IP4 192.168.26.39 > s=Session SDP > c=IN IP4 192.168.26.39 > t=0 0 > m=image 49172 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval:0 > a=T38FaxTranscodingMMR:0 > a=T38FaxTranscodingJBIG:0 > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:200 > a=T38FaxMaxDatagram:72 > a=T38FaxUdpEC:t38UDPRedundancy > > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.093421: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.093541: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.098263: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.098418: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > 2010-07-02 09:41:41.606147 [ERR] mod_spandsp_fax.c:1119 pvt- > >t38_mode = 1 > recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:44.095284: > > ------------------------------------------------------------------------ > INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: sip:666666 at 192.168.26.39:25060 > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: 345 > > v=0 > o=root 0 0 IN IP4 192.168.26.39 > s=Session SDP > c=IN IP4 192.168.26.39 > t=0 0 > m=image 49172 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval:0 > a=T38FaxTranscodingMMR:0 > a=T38FaxTranscodingJBIG:0 > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:200 > a=T38FaxMaxDatagram:72 > a=T38FaxUdpEC:t38UDPRedundancy > > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.095457: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.095556: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.098263: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.098399: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > ------------------------------------------------------------------------ > 2010-07-02 09:41:46.607257 [ERR] mod_spandsp_fax.c:1119 pvt- > >t38_mode = 1 > send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:48.098287: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 > From: 666666 ;tag=1 > To: 1000 ;tag=85ySHt6rmaSyp > Call-ID: 1-16005 at 192.168.26.39 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 341 > > v=0 > o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 > s=FreeSWITCH > c=IN IP4 192.168.26.39 > t=0 0 > m=image 22464 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38VendorInfo:0 0 0 > > > -----------------------------log end---------------------------------- > > > 2010/7/2 Brian West > > turn sip on > > sofia profile xxx siptrace on > > /b > > On Jul 1, 2010, at 8:23 PM, chi zhang wrote: > > > hi,brian > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/3d88409b/attachment-0001.html From sanms.zhang at gmail.com Thu Jul 8 01:20:27 2010 From: sanms.zhang at gmail.com (chi zhang) Date: Thu, 8 Jul 2010 16:20:27 +0800 Subject: [Freeswitch-users] Problem of FreeSwitch and T.38 Test In-Reply-To: <12B00ED2-34A6-414C-A0C6-590A963E3CD2@gmail.com> References: <5ECC3A8A-CDF9-4EF4-AD87-A3FAEDFF26AB@freeswitch.org> <12B00ED2-34A6-414C-A0C6-590A963E3CD2@gmail.com> Message-ID: Hi, David transfer fax function means "FS send fax to a sip user" , previously , i use zoiper to send fax to FS, for FS ,this means "receive fax", now i want to realize send fax from FS to one sip user. regards sammy 2010/7/8 David Ponzone > Chi, > > I am not sure to understand what you mean by "transfer fax function". > Can you clear that up please ? > > Thanks > > David Ponzone Direction Technique > email: david.ponzone at 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 pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 08/07/2010 ? 09:57, chi zhang a ?crit : > > Now, i have finished T.38 fax receive test with Zoiper, it works well. > But, transfer a fax is still Not successful. > Previous, i do simulate fax with sipP, but FS return 48(Disconnected > after permitted retries). > Accidentally, i found softphone: Zoiper has fax function, so retry fax > with it, and in diaplan file: default.xml, 9178 was the receive fax number. > So i call 9178 with Zoiper(register as 1000), fax receiving is perfect done. > Then i test transfer fax function: call to 9179(configured by TX fax in > default.xml), but FS return 2 (Timed out waiting for initial communication) > , i have no idea about it, reason ? > > regards > sammy > > > > 2010/7/2 chi zhang > >> I got it. >> >> -------------------log start-------------------------- >> recv 576 bytes from udp/[192.168.26.39]:15060 at 01:41:26.500127: >> >> ------------------------------------------------------------------------ >> INVITE sip:666666 at 192.168.26.39:25060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >> From: 1000 ;tag=1 >> To: 666666 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:1000 at 192.168.26.39:15060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 184 >> >> v=0 >> o=user1 3748 3748 IN IP4 192.168.26.39 >> s=- >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=audio 6000 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-11,16 >> >> ------------------------------------------------------------------------ >> send 294 bytes to udp/[192.168.26.39]:15060 at 01:41:26.500477: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >> From: 1000 ;tag=1 >> To: 666666 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:26.498946 [DEBUG] sofia.c:5928 IP 192.168.26.39 Approved >> by acl "192.168.26.0/24[] ". Access >> Granted. >> [36m2010-07-02 09:41:26.498946 [NOTICE] switch_channel.c:776 New Channel >> sofia/internal/1000 at 192.168.26.39:15060[f588c66c-48c8-4220-a944-8de287adb3ab] >> 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_NEW >> 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:320 >> (sofia/internal/1000 at 192.168.26.39:15060) State NEW >> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4297 Channel sofia/internal/ >> 1000 at 192.168.26.39:15060 entering state [received][100] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4308 Remote SDP: >> v=0 >> >> o=user1 3748 3748 IN IP4 192.168.26.39 >> >> s=- >> >> c=IN IP4 192.168.26.39 >> >> t=0 0 >> >> m=audio 6000 RTP/AVP 8 101 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-11,16 >> >> >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >> [PCMA:8:8000:20]/[G7221:115:32000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >> [PCMA:8:8000:20]/[G7221:107:16000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >> [PCMA:8:8000:20]/[G722:9:8000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >> [PCMA:8:8000:20]/[PCMU:0:8000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >> [PCMA:8:8000:20]/[PCMA:8:8000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:2442 Set Codec >> sofia/internal/1000 at 192.168.26.39:15060 PCMA/8000 20 ms 160 samples >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3798 Set 2833 dtmf >> send/recv payload to 101 >> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4455 (sofia/internal/ >> 1000 at 192.168.26.39:15060) State Change CS_NEW -> CS_INIT >> 2010-07-02 09:41:26.542951 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_INIT >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 >> (sofia/internal/1000 at 192.168.26.39:15060) State INIT >> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:83 sofia/internal/ >> 1000 at 192.168.26.39:15060 SOFIA INIT >> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:119 (sofia/internal/ >> 1000 at 192.168.26.39:15060) State Change CS_INIT -> CS_ROUTING >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 >> (sofia/internal/1000 at 192.168.26.39:15060) State INIT going to sleep >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_ROUTING >> 2010-07-02 09:41:26.544953 [DEBUG] switch_channel.c:1471 (sofia/internal/ >> 1000 at 192.168.26.39:15060) Callstate Change DOWN -> RINGING >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING >> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:142 sofia/internal/ >> 1000 at 192.168.26.39:15060 SOFIA ROUTING >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:77 >> sofia/internal/1000 at 192.168.26.39:15060 Standard ROUTING >> [32m2010-07-02 09:41:26.544953 [INFO] mod_dialplan_xml.c:331 Processing >> 1000->666666 in context public >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->fax] >> continue=false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] >> destination_number(666666) =~ /^fax$/ break=on-false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->4444] >> continue=false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [4444] >> destination_number(666666) =~ /^(4444)$/ break=on-false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->fax] >> continue=false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] >> destination_number(666666) =~ /^fax$/ break=on-false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing >> [public->666666] continue=false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (PASS) [666666] >> destination_number(666666) =~ /^(666666)$/ break=on-false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action answer() >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action >> playback(silence_stream://2000) >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action >> rxfax(/tmp/999.tiff) >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action hangup() >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:119 >> (sofia/internal/1000 at 192.168.26.39:15060) State Change CS_ROUTING -> >> CS_EXECUTE >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send signal >> sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING going to sleep >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_EXECUTE >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:348 >> (sofia/internal/1000 at 192.168.26.39:15060) State EXECUTE >> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:235 sofia/internal/ >> 1000 at 192.168.26.39:15060 SOFIA EXECUTE >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:157 >> sofia/internal/1000 at 192.168.26.39:15060 Standard EXECUTE >> EXECUTE sofia/internal/1000 at 192.168.26.39:15060 answer() >> 2010-07-02 09:41:26.580967 [DEBUG] sofia_glue.c:2682 AUDIO RTP >> [sofia/internal/1000 at 192.168.26.39:15060] 192.168.26.39 port 22464 -> >> 192.168.26.39 port 6000 codec: 8 ms: 20 >> 2010-07-02 09:41:26.580967 [DEBUG] switch_rtp.c:1413 Starting timer [soft] >> 160 bytes per 20ms >> 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2892 Set 2833 dtmf send >> payload to 101 >> 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2897 Set 2833 dtmf receive >> payload to 101 >> 2010-07-02 09:41:26.582952 [DEBUG] mod_sofia.c:669 Local SDP >> sofia/internal/1000 at 192.168.26.39:15060: >> v=0 >> o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=audio 22464 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> >> send 1091 bytes to udp/[192.168.26.39]:15060 at 01:41:26.584093: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >> From: 1000 ;tag=1 >> To: 666666 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 249 >> Remote-Party-ID: "666666" >> >;party=calling;privacy=off;screen=no >> >> v=0 >> o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=audio 22464 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> >> ------------------------------------------------------------------------ >> recv 369 bytes from udp/[192.168.26.39]:15060 at 01:41:26.584205: >> >> ------------------------------------------------------------------------ >> ACK sip:666666 at 192.168.26.39:25060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-4 >> From: 1000 ;tag=1 >> To: 666666 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 ACK >> Contact: sip:1000 at 192.168.26.39:15060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/ >> 1000 at 192.168.26.39:15060 entering state [completed][200] >> 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/ >> 1000 at 192.168.26.39:15060 entering state [ready][200] >> 2010-07-02 09:41:26.582952 [DEBUG] switch_core_session.c:647 Send signal >> sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >> 2010-07-02 09:41:26.582952 [DEBUG] switch_channel.c:2494 (sofia/internal/ >> 1000 at 192.168.26.39:15060) Callstate Change RINGING -> ACTIVE >> [36m2010-07-02 09:41:26.582952 [NOTICE] mod_dptools.c:746 Channel >> [sofia/internal/1000 at 192.168.26.39:15060] has been answered >> EXECUTE sofia/internal/1000 at 192.168.26.39:15060playback(silence_stream://2000) >> 2010-07-02 09:41:26.584953 [DEBUG] switch_ivr_play_say.c:1161 Codec >> Activated L16 at 8000hz 1 channels 20ms >> 2010-07-02 09:41:28.578954 [DEBUG] switch_ivr_play_say.c:1468 done playing >> file >> EXECUTE sofia/internal/1000 at 192.168.26.39:15060 rxfax(/tmp/999.tiff) >> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp.c:64 This is for fax test: >> receive fax >> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:445 trans mode = 1 >> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:591 This is for fax >> test: prag go to here!!! >> 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1064 Raw read codec >> activation Success L16 20000 >> 2010-07-02 09:41:28.578954 [DEBUG] switch_core_codec.c:122 sofia/internal/ >> 1000 at 192.168.26.39:15060 Push codec L16:10 >> 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1080 Raw write codec >> activation Success L16 >> 2010-07-02 09:41:28.857958 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 >> >> send 956 bytes to udp/[192.168.21.76]:5060 at 01:41:29.754477: >> >> ------------------------------------------------------------------------ >> NOTIFY sip:1001 at 192.168.21.76:5060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;rport;branch=z9hG4bKS989SFgmXvmNF >> Max-Forwards: 70 >> From: "1001" > >;tag=vr6p3K1XK247r >> To: "1001" ;tag=26647676 >> Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. >> CSeq: 132897310 NOTIFY >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Event: message-summary >> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer >> Subscription-State: terminated;reason=timeout >> Content-Type: application/simple-message-summary >> Content-Length: 65 >> >> Messages-Waiting: no >> Message-Account: sip:1001 at 192.168.26.39 >> >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:29.757979 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 >> recv 389 bytes from udp/[192.168.21.76]:5060 at 01:41:29.758362: >> >> ------------------------------------------------------------------------ >> SIP/2.0 481 Call/Transaction Does Not Exist >> Via: SIP/2.0/UDP 192.168.26.39:25060 >> ;rport=25060;branch=z9hG4bKS989SFgmXvmNF >> To: "1001";tag=26647676 >> From: "1001"> >;tag=vr6p3K1XK247r >> Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. >> CSeq: 132897310 NOTIFY >> Accept-Language: en >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> >> 2010-07-02 09:41:36.557090 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:36.587253: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 312 bytes to udp/[192.168.26.39]:25060 at 01:41:36.587546: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> recv 312 bytes from udp/[192.168.26.39]:25060 at 01:41:36.587641: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4297 Channel >> sofia/internal/1000 at 192.168.26.39:15060 entering state [received][100] >> >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4308 Remote SDP: >> v=0 >> >> o=root 0 0 IN IP4 192.168.26.39 >> >> s=Session SDP >> >> c=IN IP4 192.168.26.39 >> >> t=0 0 >> >> m=image 49172 udptl t38 >> >> a=T38FaxVersion:0 >> >> a=T38MaxBitRate:9600 >> >> a=T38FaxFillBitRemoval:0 >> >> a=T38FaxTranscodingMMR:0 >> >> a=T38FaxTranscodingJBIG:0 >> >> a=T38FaxRateManagement:transferredTCF >> >> a=T38FaxMaxBuffer:200 >> >> a=T38FaxMaxDatagram:72 >> >> a=T38FaxUdpEC:t38UDPRedundancy >> >> >> 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:744 T38FaxVersion = 0 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:745 T38MaxBitRate = >> 9600 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:746 >> T38FaxFillBitRemoval = 1 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:747 >> T38FaxTranscodingMMR = 1 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:748 >> T38FaxTranscodingJBIG = 1 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:749 >> T38FaxRateManagement = 'transferredTCF' >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:750 T38FaxMaxBuffer = >> 200 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:751 T38FaxMaxDatagram >> = 72 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:752 T38FaxUdpEC = >> 't38UDPRedundancy' >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:753 T38VendorInfo = >> '' >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:754 ip = >> '192.168.26.39' >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:756 port = 49172 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_sofia.c:1232 IMAGE UDPTL CHANGING >> DEST TO: [192.168.26.39:49172] >> 2010-07-02 09:41:36.597094 [DEBUG] sofia_glue.c:122 sofia/internal/ >> 1000 at 192.168.26.39:15060 image media sdp: >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:445 trans mode = 0 >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:36.597525: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:36.597094 [DEBUG] sofia.c:4297 Channel sofia/internal/ >> 1000 at 192.168.26.39:15060 entering state [completed][200] >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:36.597657: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:36.607095 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:37.089156: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.089302: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.089395: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.098259: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.098339: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:38.091157: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.091305: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.091406: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.098260: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.098344: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> This is for fax test: dis 5This is for fax test: cause disconnect 4 >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:40.093267: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.093421: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.093541: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.098263: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.098418: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:41.606147 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:44.095284: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.095457: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.095556: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.098263: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.098399: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:46.607257 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:48.098287: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> >> -----------------------------log end---------------------------------- >> >> >> 2010/7/2 Brian West >> >> turn sip on >>> >>> sofia profile xxx siptrace on >>> >>> /b >>> >>> On Jul 1, 2010, at 8:23 PM, chi zhang wrote: >>> >>> > hi,brian >>> > >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/2275c4db/attachment-0001.html From david.ponzone at gmail.com Thu Jul 8 01:32:27 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 8 Jul 2010 10:32:27 +0200 Subject: [Freeswitch-users] Problem of FreeSwitch and T.38 Test In-Reply-To: References: <5ECC3A8A-CDF9-4EF4-AD87-A3FAEDFF26AB@freeswitch.org> <12B00ED2-34A6-414C-A0C6-590A963E3CD2@gmail.com> Message-ID: Ok, so if I understood correctly, you call a number on FS, and you expect to receive a fax on that call. Zopier is ok, but it's far from a perfect piece of software. Perhaps it is not able to receive a fax when it calls out. Could you try to call Zoiper from FS before sending the fax to Zoiper ? David Ponzone Direction Technique email: david.ponzone at 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 pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 08/07/2010 ? 10:20, chi zhang a ?crit : > Hi, David > > transfer fax function means "FS send fax to a sip user" , > previously , i use zoiper to send fax to FS, for FS ,this means > "receive fax", now i want to realize send fax from FS to one sip user. > > regards > sammy > > 2010/7/8 David Ponzone > Chi, > > I am not sure to understand what you mean by "transfer fax function". > Can you clear that up please ? > > Thanks > > David Ponzone Direction Technique > email: david.ponzone at 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 pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou > falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le > d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > Le 08/07/2010 ? 09:57, chi zhang a ?crit : > >> Now, i have finished T.38 fax receive test with Zoiper, it >> works well. But, transfer a fax is still Not successful. >> Previous, i do simulate fax with sipP, but FS return >> 48(Disconnected after permitted retries). >> Accidentally, i found softphone: Zoiper has fax function, so >> retry fax with it, and in diaplan file: default.xml, 9178 was the >> receive fax number. So i call 9178 with Zoiper(register as 1000), >> fax receiving is perfect done. >> Then i test transfer fax function: call to 9179(configured by >> TX fax in default.xml), but FS return 2 (Timed out waiting for >> initial communication) , i have no idea about it, reason ? >> >> regards >> sammy >> >> >> >> 2010/7/2 chi zhang >> I got it. >> >> -------------------log start-------------------------- >> recv 576 bytes from udp/[192.168.26.39]:15060 at 01:41:26.500127: >> >> ------------------------------------------------------------------------ >> INVITE sip:666666 at 192.168.26.39:25060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >> From: 1000 ;tag=1 >> To: 666666 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:1000 at 192.168.26.39:15060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 184 >> >> v=0 >> o=user1 3748 3748 IN IP4 192.168.26.39 >> s=- >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=audio 6000 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-11,16 >> >> ------------------------------------------------------------------------ >> send 294 bytes to udp/[192.168.26.39]:15060 at 01:41:26.500477: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >> From: 1000 ;tag=1 >> To: 666666 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:26.498946 [DEBUG] sofia.c:5928 IP 192.168.26.39 >> Approved by acl "192.168.26.0/24[]". Access Granted. >> [36m2010-07-02 09:41:26.498946 [NOTICE] switch_channel.c:776 New >> Channel sofia/internal/1000 at 192.168.26.39:15060 [f588c66c-48c8-4220- >> a944-8de287adb3ab] >> 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39 >> :15060) Running State Change CS_NEW >> 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1000 at 192.168.26.39 >> :15060) State NEW >> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4297 Channel sofia/internal/1000 at 192.168.26.39 >> :15060 entering state [received][100] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4308 Remote SDP: >> v=0 >> >> o=user1 3748 3748 IN IP4 192.168.26.39 >> >> s=- >> >> c=IN IP4 192.168.26.39 >> >> t=0 0 >> >> m=audio 6000 RTP/AVP 8 101 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-11,16 >> >> >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec >> Compare [PCMA:8:8000:20]/[G7221:115:32000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec >> Compare [PCMA:8:8000:20]/[G7221:107:16000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec >> Compare [PCMA:8:8000:20]/[G722:9:8000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec >> Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec >> Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:2442 Set Codec sofia/internal/1000 at 192.168.26.39 >> :15060 PCMA/8000 20 ms 160 samples >> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3798 Set 2833 dtmf >> send/recv payload to 101 >> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4455 (sofia/internal/1000 at 192.168.26.39 >> :15060) State Change CS_NEW -> CS_INIT >> 2010-07-02 09:41:26.542951 [DEBUG] switch_core_session.c:1027 Send >> signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39 >> :15060) Running State Change CS_INIT >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.26.39 >> :15060) State INIT >> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:83 sofia/internal/1000 at 192.168.26.39 >> :15060 SOFIA INIT >> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:119 (sofia/internal/1000 at 192.168.26.39 >> :15060) State Change CS_INIT -> CS_ROUTING >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send >> signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.26.39 >> :15060) State INIT going to sleep >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39 >> :15060) Running State Change CS_ROUTING >> 2010-07-02 09:41:26.544953 [DEBUG] switch_channel.c:1471 (sofia/internal/1000 at 192.168.26.39 >> :15060) Callstate Change DOWN -> RINGING >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.26.39 >> :15060) State ROUTING >> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:142 sofia/internal/1000 at 192.168.26.39 >> :15060 SOFIA ROUTING >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1000 at 192.168.26.39 >> :15060 Standard ROUTING >> [32m2010-07-02 09:41:26.544953 [INFO] mod_dialplan_xml.c:331 >> Processing 1000->666666 in context public >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public- >> >fax] continue=false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) >> [fax] destination_number(666666) =~ /^fax$/ break=on-false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public- >> >4444] continue=false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) >> [4444] destination_number(666666) =~ /^(4444)$/ break=on-false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public- >> >fax] continue=false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) >> [fax] destination_number(666666) =~ /^fax$/ break=on-false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public- >> >666666] continue=false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (PASS) >> [666666] destination_number(666666) =~ /^(666666)$/ break=on-false >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action answer() >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action >> playback(silence_stream://2000) >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action rxfax(/tmp/ >> 999.tiff) >> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action hangup() >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1000 at 192.168.26.39 >> :15060) State Change CS_ROUTING -> CS_EXECUTE >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send >> signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.26.39 >> :15060) State ROUTING going to sleep >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.26.39 >> :15060) Running State Change CS_EXECUTE >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1000 at 192.168.26.39 >> :15060) State EXECUTE >> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:235 sofia/internal/1000 at 192.168.26.39 >> :15060 SOFIA EXECUTE >> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1000 at 192.168.26.39 >> :15060 Standard EXECUTE >> EXECUTE sofia/internal/1000 at 192.168.26.39:15060 answer() >> 2010-07-02 09:41:26.580967 [DEBUG] sofia_glue.c:2682 AUDIO RTP [sofia/internal/1000 at 192.168.26.39 >> :15060] 192.168.26.39 port 22464 -> 192.168.26.39 port 6000 codec: >> 8 ms: 20 >> 2010-07-02 09:41:26.580967 [DEBUG] switch_rtp.c:1413 Starting timer >> [soft] 160 bytes per 20ms >> 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2892 Set 2833 dtmf >> send payload to 101 >> 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2897 Set 2833 dtmf >> receive payload to 101 >> 2010-07-02 09:41:26.582952 [DEBUG] mod_sofia.c:669 Local SDP sofia/internal/1000 at 192.168.26.39 >> :15060: >> v=0 >> o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=audio 22464 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> >> send 1091 bytes to udp/[192.168.26.39]:15060 at 01:41:26.584093: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >> From: 1000 ;tag=1 >> To: 666666 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, hold, presence, dialog, line-seize, call- >> info, sla, include-session-description, presence.winfo, message- >> summary, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 249 >> Remote-Party-ID: "666666" > 666666 at 192.168.26.39>;party=calling;privacy=off;screen=no >> >> v=0 >> o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=audio 22464 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> >> ------------------------------------------------------------------------ >> recv 369 bytes from udp/[192.168.26.39]:15060 at 01:41:26.584205: >> >> ------------------------------------------------------------------------ >> ACK sip:666666 at 192.168.26.39:25060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-4 >> From: 1000 ;tag=1 >> To: 666666 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 ACK >> Contact: sip:1000 at 192.168.26.39:15060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/1000 at 192.168.26.39 >> :15060 entering state [completed][200] >> 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/1000 at 192.168.26.39 >> :15060 entering state [ready][200] >> 2010-07-02 09:41:26.582952 [DEBUG] switch_core_session.c:647 Send >> signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >> 2010-07-02 09:41:26.582952 [DEBUG] switch_channel.c:2494 (sofia/internal/1000 at 192.168.26.39 >> :15060) Callstate Change RINGING -> ACTIVE >> [36m2010-07-02 09:41:26.582952 [NOTICE] mod_dptools.c:746 Channel [sofia/internal/1000 at 192.168.26.39 >> :15060] has been answered >> EXECUTE sofia/internal/1000 at 192.168.26.39:15060 >> playback(silence_stream://2000) >> 2010-07-02 09:41:26.584953 [DEBUG] switch_ivr_play_say.c:1161 Codec >> Activated L16 at 8000hz 1 channels 20ms >> 2010-07-02 09:41:28.578954 [DEBUG] switch_ivr_play_say.c:1468 done >> playing file >> EXECUTE sofia/internal/1000 at 192.168.26.39:15060 rxfax(/tmp/999.tiff) >> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp.c:64 This is for fax >> test: receive fax >> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:445 trans mode = 1 >> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:591 This is for >> fax test: prag go to here!!! >> 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1064 Raw read >> codec activation Success L16 20000 >> 2010-07-02 09:41:28.578954 [DEBUG] switch_core_codec.c:122 sofia/internal/1000 at 192.168.26.39 >> :15060 Push codec L16:10 >> 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1080 Raw write >> codec activation Success L16 >> 2010-07-02 09:41:28.857958 [ERR] mod_spandsp_fax.c:1119 pvt- >> >t38_mode = 0 >> >> send 956 bytes to udp/[192.168.21.76]:5060 at 01:41:29.754477: >> >> ------------------------------------------------------------------------ >> NOTIFY sip:1001 at 192.168.21.76:5060 SIP/2.0 >> Via: SIP/2.0/UDP >> 192.168.26.39:25060;rport;branch=z9hG4bKS989SFgmXvmNF >> Max-Forwards: 70 >> From: "1001" > 1001 at 192.168.26.39:25060;transport=UDP>;tag=vr6p3K1XK247r >> To: "1001" > 1001 at 192.168.26.39:25060;transport=UDP>;tag=26647676 >> Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. >> CSeq: 132897310 NOTIFY >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Event: message-summary >> Allow-Events: talk, hold, presence, dialog, line-seize, call- >> info, sla, include-session-description, presence.winfo, message- >> summary, refer >> Subscription-State: terminated;reason=timeout >> Content-Type: application/simple-message-summary >> Content-Length: 65 >> >> Messages-Waiting: no >> Message-Account: sip:1001 at 192.168.26.39 >> >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:29.757979 [ERR] mod_spandsp_fax.c:1119 pvt- >> >t38_mode = 0 >> recv 389 bytes from udp/[192.168.21.76]:5060 at 01:41:29.758362: >> >> ------------------------------------------------------------------------ >> SIP/2.0 481 Call/Transaction Does Not Exist >> Via: SIP/2.0/UDP >> 192.168.26.39:25060;rport=25060;branch=z9hG4bKS989SFgmXvmNF >> To: "1001"> 1001 at 192.168.26.39:25060;transport=UDP>;tag=26647676 >> From: "1001"> 1001 at 192.168.26.39:25060;transport=UDP>;tag=vr6p3K1XK247r >> Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. >> CSeq: 132897310 NOTIFY >> Accept-Language: en >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> >> 2010-07-02 09:41:36.557090 [ERR] mod_spandsp_fax.c:1119 pvt- >> >t38_mode = 0 >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:36.587253: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 312 bytes to udp/[192.168.26.39]:25060 at 01:41:36.587546: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> recv 312 bytes from udp/[192.168.26.39]:25060 at 01:41:36.587641: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4297 >> Channel sofia/internal/1000 at 192.168.26.39:15060 entering state >> [received][100] >> >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4308 Remote SDP: >> v=0 >> >> o=root 0 0 IN IP4 192.168.26.39 >> >> s=Session SDP >> >> c=IN IP4 192.168.26.39 >> >> t=0 0 >> >> m=image 49172 udptl t38 >> >> a=T38FaxVersion:0 >> >> a=T38MaxBitRate:9600 >> >> a=T38FaxFillBitRemoval:0 >> >> a=T38FaxTranscodingMMR:0 >> >> a=T38FaxTranscodingJBIG:0 >> >> a=T38FaxRateManagement:transferredTCF >> >> a=T38FaxMaxBuffer:200 >> >> a=T38FaxMaxDatagram:72 >> >> a=T38FaxUdpEC:t38UDPRedundancy >> >> >> 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:1119 pvt- >> >t38_mode = 0 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:744 >> T38FaxVersion = 0 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:745 >> T38MaxBitRate = 9600 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:746 >> T38FaxFillBitRemoval = 1 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:747 >> T38FaxTranscodingMMR = 1 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:748 >> T38FaxTranscodingJBIG = 1 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:749 >> T38FaxRateManagement = 'transferredTCF' >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:750 >> T38FaxMaxBuffer = 200 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:751 >> T38FaxMaxDatagram = 72 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:752 >> T38FaxUdpEC = 't38UDPRedundancy' >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:753 >> T38VendorInfo = '' >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:754 ip = >> '192.168.26.39' >> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:756 port = 49172 >> 2010-07-02 09:41:36.597094 [DEBUG] mod_sofia.c:1232 IMAGE UDPTL >> CHANGING DEST TO: [192.168.26.39:49172] >> 2010-07-02 09:41:36.597094 [DEBUG] sofia_glue.c:122 sofia/internal/1000 at 192.168.26.39 >> :15060 image media sdp: >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:445 trans mode = 0 >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:36.597525: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:36.597094 [DEBUG] sofia.c:4297 Channel sofia/internal/1000 at 192.168.26.39 >> :15060 entering state [completed][200] >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:36.597657: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:36.607095 [ERR] mod_spandsp_fax.c:1119 pvt- >> >t38_mode = 1 >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:37.089156: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.089302: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.089395: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.098259: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.098339: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:38.091157: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.091305: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.091406: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.098260: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.098344: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> This is for fax test: dis 5This is for fax test: cause disconnect 4 >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:40.093267: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.093421: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.093541: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.098263: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.098418: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:41.606147 [ERR] mod_spandsp_fax.c:1119 pvt- >> >t38_mode = 1 >> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:44.095284: >> >> ------------------------------------------------------------------------ >> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: sip:666666 at 192.168.26.39:25060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 345 >> >> v=0 >> o=root 0 0 IN IP4 192.168.26.39 >> s=Session SDP >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 49172 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval:0 >> a=T38FaxTranscodingMMR:0 >> a=T38FaxTranscodingJBIG:0 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:200 >> a=T38FaxMaxDatagram:72 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.095457: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.095556: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.098263: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.098399: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> ------------------------------------------------------------------------ >> 2010-07-02 09:41:46.607257 [ERR] mod_spandsp_fax.c:1119 pvt- >> >t38_mode = 1 >> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:48.098287: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >> From: 666666 ;tag=1 >> To: 1000 ;tag=85ySHt6rmaSyp >> Call-ID: 1-16005 at 192.168.26.39 >> CSeq: 2 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 341 >> >> v=0 >> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >> s=FreeSWITCH >> c=IN IP4 192.168.26.39 >> t=0 0 >> m=image 22464 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38VendorInfo:0 0 0 >> >> >> -----------------------------log >> end---------------------------------- >> >> >> 2010/7/2 Brian West >> >> turn sip on >> >> sofia profile xxx siptrace on >> >> /b >> >> On Jul 1, 2010, at 8:23 PM, chi zhang wrote: >> >> > hi,brian >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/5e89f56d/attachment-0001.html From vetali100 at gmail.com Thu Jul 8 01:56:42 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Thu, 8 Jul 2010 11:56:42 +0300 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: <201007071616.41535.sos@sokhapkin.dyndns.org> Message-ID: There should not be any problem if they use multiple IP addresses for media stream, as long as siptraffic's SDP contains correct IP that will participate in RTPs exchange... Do note that media IP can be different from sip IP, and it is OK, if implemented properly... 2010/7/8 paul gore > Seems like siptraffic uses 6 ip addresses for media, can that be the > problem? Is there any setting in a gateway config xml which helps with > that? > I will do ngrep thing and update. > > On 7/7/10, paul gore wrote: > > This provider does work on another box which is not natted as ec2. > > Most puzzling here though is why call originaion via api even not > > going via siptraffic still gets no audio. > > > > On 7/7/10, Tony Graziano wrote: > >> You should try from a standalone or local installation to ensure it > works > >> with this provider and your account before you attempt to run it on ec2 > >> (imo). > >> > >> On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin > >> wrote: > >> > >>> What "doesn't work" means? It could be (and most likely is not) > >>> FS-related > >>> problem > >>> > >>> On Wednesday 07 July 2010, Madovsky wrote: > >>> > I had same problem from this provider without to explain why. > >>> > One day it works, another day it doesn't, their support is crap... > >>> > > >>> > ----- Original Message ----- > >>> > From: Anthony Minessale > >>> > To: freeswitch-users at lists.freeswitch.org > >>> > Sent: Wednesday, July 07, 2010 2:37 PM > >>> > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on > >>> > outgoing > >>> > calls > >>> > > >>> > > >>> > not really, not with so little information. > >>> > > >>> > > >>> > > >>> > On Wed, Jul 7, 2010 at 1:30 PM, paul gore > >>> wrote: > >>> > > >>> > Firewall is configured according to the wiki, I also tried to > open > >>> all > >>> > udp ports, issue persists. > >>> > Actually the problem became more complex - outgoing calls don't > >>> > work > >>> > with one particular termination provider, siptraffic.com , any > >>> > ideas > >>> > why? > >>> > Outgoing calls also don't work when originating a call via js > >>> > script > >>> > or via FS api. Any clues on that one? > >>> > > >>> > On 7/6/10, paul gore wrote: > >>> > > Hi there, > >>> > > I am experimenting with FS on EC2, I like results, but stuck on > >>> weird > >>> > > audio issue - I followed FreeSwitch EC2 wiki article and > >>> > modified > >>> > > internal profile > >>> > > and vars.xml accordingly, but unfortunately still cannot get it > >>> > > working. Incoming and outgoing calls made using a SIP phone to > >>> > FS > >>> > > extensions work just fine. As well as calls to FS from PSTN. > But > >>> > > calls to PSTN via gateways result in no audio at all, no ring, > >>> > > nothing, SIP signaling goes through OK. Sofia status profile > >>> > shows > >>> > > correct values for Ext-RTP-IP for both profiles - > >>> > > my static public IP, RTP-IP shows local IP. > >>> > > Any thoughts on that? Anybody can share working profile > >>> configuration > >>> > > may be? > >>> > > Please help, I really need to get this going. > >>> > > > >>> > > Thanks. > >>> > > >>> > _______________________________________________ > >>> > FreeSWITCH-users mailing list > >>> > FreeSWITCH-users at 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 http://www.freeswitch.org/ > >>> > ClueCon http://www.cluecon.com/ > >>> > Twitter: http://twitter.com/FreeSWITCH_wire > >>> > > >>> > AIM: anthm > >>> > > >>> > MSN:anthony_minessale at hotmail.com > > > > >>> > > >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >>> > IRC: irc.freenode.net #freeswitch > >>> > > >>> > FreeSWITCH Developer Conference > >>> > > >>> > sip:888 at conference.freeswitch.org > > > > >>> > > >>> > googletalk:conf+888 at conference.freeswitch.org > > > > >>> > pstn:+19193869900 > >>> > > >>> > > >>> > > >>> > > >>> > --------------------------------------------------------------------------- > >>> > --- > >>> > > >>> > > >>> > _______________________________________________ > >>> > FreeSWITCH-users mailing list > >>> > FreeSWITCH-users at 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 at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> ====================== > >> Tony Graziano, Manager > >> Telephone: 434.984.8430 > >> sip: tgraziano at voice.myitdepartment.net > >> Fax: 434.984.8431 > >> > >> Email: tgraziano at myitdepartment.net > >> > >> LAN/Telephony/Security and Control Systems Helpdesk: > >> Telephone: 434.984.8426 > >> sip: helpdesk at voice.myitdepartment.net > >> Fax: 434.984.8427 > >> > >> Helpdesk Contract Customers: > >> http://www.myitdepartment.net/gethelp/ > >> > >> Why do mathematicians always confuse Halloween and Christmas? > >> Because 31 Oct = 25 Dec. > >> > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/c57596df/attachment.html From sanms.zhang at gmail.com Thu Jul 8 02:10:52 2010 From: sanms.zhang at gmail.com (chi zhang) Date: Thu, 8 Jul 2010 17:10:52 +0800 Subject: [Freeswitch-users] Problem of FreeSwitch and T.38 Test In-Reply-To: References: <5ECC3A8A-CDF9-4EF4-AD87-A3FAEDFF26AB@freeswitch.org> <12B00ED2-34A6-414C-A0C6-590A963E3CD2@gmail.com> Message-ID: I just use FS's CLI to call zoiper from FS, it works OK CLI cmd : originate sofia/internal/1001 at 192.168.26.39:25060 1001 i think if i want to do send fax from FS, i need one softphone can receive fax, zoiper can do this or not? That is a question... 2010/7/8 David Ponzone > Ok, so if I understood correctly, you call a number on FS, and you expect > to receive a fax on that call. > Zopier is ok, but it's far from a perfect piece of software. > Perhaps it is not able to receive a fax when it calls out. > Could you try to call Zoiper from FS before sending the fax to Zoiper ? > > David Ponzone Direction Technique > email: david.ponzone at 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 pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 08/07/2010 ? 10:20, chi zhang a ?crit : > > Hi, David > > transfer fax function means "FS send fax to a sip user" , previously , > i use zoiper to send fax to FS, for FS ,this means "receive fax", now i > want to realize send fax from FS to one sip user. > > regards > sammy > > 2010/7/8 David Ponzone > >> Chi, >> >> I am not sure to understand what you mean by "transfer fax function". >> Can you clear that up please ? >> >> Thanks >> >> David Ponzone Direction Technique >> email: david.ponzone at 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 pi?ces jointes sont confidentiels et ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autoris?e est interdite. Tout message ?lectronique est susceptible >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur.* >> * >> * >> >> >> >> Le 08/07/2010 ? 09:57, chi zhang a ?crit : >> >> Now, i have finished T.38 fax receive test with Zoiper, it works >> well. But, transfer a fax is still Not successful. >> Previous, i do simulate fax with sipP, but FS return 48(Disconnected >> after permitted retries). >> Accidentally, i found softphone: Zoiper has fax function, so retry fax >> with it, and in diaplan file: default.xml, 9178 was the receive fax number. >> So i call 9178 with Zoiper(register as 1000), fax receiving is perfect done. >> Then i test transfer fax function: call to 9179(configured by TX fax >> in default.xml), but FS return 2 (Timed out waiting for initial >> communication) , i have no idea about it, reason ? >> >> regards >> sammy >> >> >> >> 2010/7/2 chi zhang >> >>> I got it. >>> >>> -------------------log start-------------------------- >>> recv 576 bytes from udp/[192.168.26.39]:15060 at 01:41:26.500127: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:666666 at 192.168.26.39:25060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >>> From: 1000 ;tag=1 >>> To: 666666 >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: sip:1000 at 192.168.26.39:15060 >>> Max-Forwards: 70 >>> Subject: Performance Test >>> Content-Type: application/sdp >>> Content-Length: 184 >>> >>> v=0 >>> o=user1 3748 3748 IN IP4 192.168.26.39 >>> s=- >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=audio 6000 RTP/AVP 8 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-11,16 >>> >>> ------------------------------------------------------------------------ >>> send 294 bytes to udp/[192.168.26.39]:15060 at 01:41:26.500477: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >>> From: 1000 ;tag=1 >>> To: 666666 >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:26.498946 [DEBUG] sofia.c:5928 IP 192.168.26.39 Approved >>> by acl "192.168.26.0/24[] ". Access >>> Granted. >>> [36m2010-07-02 09:41:26.498946 [NOTICE] switch_channel.c:776 New Channel >>> sofia/internal/1000 at 192.168.26.39:15060[f588c66c-48c8-4220-a944-8de287adb3ab] >>> 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_NEW >>> 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:320 >>> (sofia/internal/1000 at 192.168.26.39:15060) State NEW >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4297 Channel sofia/internal/ >>> 1000 at 192.168.26.39:15060 entering state [received][100] >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4308 Remote SDP: >>> v=0 >>> >>> o=user1 3748 3748 IN IP4 192.168.26.39 >>> >>> s=- >>> >>> c=IN IP4 192.168.26.39 >>> >>> t=0 0 >>> >>> m=audio 6000 RTP/AVP 8 101 >>> >>> a=rtpmap:8 PCMA/8000 >>> >>> a=rtpmap:101 telephone-event/8000 >>> >>> a=fmtp:101 0-11,16 >>> >>> >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >>> [PCMA:8:8000:20]/[G7221:115:32000:20] >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >>> [PCMA:8:8000:20]/[G7221:107:16000:20] >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >>> [PCMA:8:8000:20]/[G722:9:8000:20] >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >>> [PCMA:8:8000:20]/[PCMU:0:8000:20] >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >>> [PCMA:8:8000:20]/[PCMA:8:8000:20] >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:2442 Set Codec >>> sofia/internal/1000 at 192.168.26.39:15060 PCMA/8000 20 ms 160 samples >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3798 Set 2833 dtmf >>> send/recv payload to 101 >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4455 (sofia/internal/ >>> 1000 at 192.168.26.39:15060) State Change CS_NEW -> CS_INIT >>> 2010-07-02 09:41:26.542951 [DEBUG] switch_core_session.c:1027 Send signal >>> sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_INIT >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/internal/1000 at 192.168.26.39:15060) State INIT >>> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:83 sofia/internal/ >>> 1000 at 192.168.26.39:15060 SOFIA INIT >>> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:119 (sofia/internal/ >>> 1000 at 192.168.26.39:15060) State Change CS_INIT -> CS_ROUTING >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send signal >>> sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/internal/1000 at 192.168.26.39:15060) State INIT going to sleep >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change >>> CS_ROUTING >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_channel.c:1471 (sofia/internal/ >>> 1000 at 192.168.26.39:15060) Callstate Change DOWN -> RINGING >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING >>> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:142 sofia/internal/ >>> 1000 at 192.168.26.39:15060 SOFIA ROUTING >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:77 >>> sofia/internal/1000 at 192.168.26.39:15060 Standard ROUTING >>> [32m2010-07-02 09:41:26.544953 [INFO] mod_dialplan_xml.c:331 Processing >>> 1000->666666 in context public >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->fax] >>> continue=false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] >>> destination_number(666666) =~ /^fax$/ break=on-false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->4444] >>> continue=false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [4444] >>> destination_number(666666) =~ /^(4444)$/ break=on-false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->fax] >>> continue=false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] >>> destination_number(666666) =~ /^fax$/ break=on-false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing >>> [public->666666] continue=false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (PASS) [666666] >>> destination_number(666666) =~ /^(666666)$/ break=on-false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action answer() >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action >>> playback(silence_stream://2000) >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action >>> rxfax(/tmp/999.tiff) >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action hangup() >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:119 >>> (sofia/internal/1000 at 192.168.26.39:15060) State Change CS_ROUTING -> >>> CS_EXECUTE >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send signal >>> sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING going to sleep >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change >>> CS_EXECUTE >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:348 >>> (sofia/internal/1000 at 192.168.26.39:15060) State EXECUTE >>> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:235 sofia/internal/ >>> 1000 at 192.168.26.39:15060 SOFIA EXECUTE >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:157 >>> sofia/internal/1000 at 192.168.26.39:15060 Standard EXECUTE >>> EXECUTE sofia/internal/1000 at 192.168.26.39:15060 answer() >>> 2010-07-02 09:41:26.580967 [DEBUG] sofia_glue.c:2682 AUDIO RTP >>> [sofia/internal/1000 at 192.168.26.39:15060] 192.168.26.39 port 22464 -> >>> 192.168.26.39 port 6000 codec: 8 ms: 20 >>> 2010-07-02 09:41:26.580967 [DEBUG] switch_rtp.c:1413 Starting timer >>> [soft] 160 bytes per 20ms >>> 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2892 Set 2833 dtmf send >>> payload to 101 >>> 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2897 Set 2833 dtmf >>> receive payload to 101 >>> 2010-07-02 09:41:26.582952 [DEBUG] mod_sofia.c:669 Local SDP >>> sofia/internal/1000 at 192.168.26.39:15060: >>> v=0 >>> o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=audio 22464 RTP/AVP 8 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=silenceSupp:off - - - - >>> a=ptime:20 >>> a=sendrecv >>> >>> send 1091 bytes to udp/[192.168.26.39]:15060 at 01:41:26.584093: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >>> From: 1000 ;tag=1 >>> To: 666666 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, >>> sla, include-session-description, presence.winfo, message-summary, refer >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 249 >>> Remote-Party-ID: "666666" >>> >;party=calling;privacy=off;screen=no >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=audio 22464 RTP/AVP 8 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=silenceSupp:off - - - - >>> a=ptime:20 >>> >>> ------------------------------------------------------------------------ >>> recv 369 bytes from udp/[192.168.26.39]:15060 at 01:41:26.584205: >>> >>> ------------------------------------------------------------------------ >>> ACK sip:666666 at 192.168.26.39:25060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-4 >>> From: 1000 ;tag=1 >>> To: 666666 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 ACK >>> Contact: sip:1000 at 192.168.26.39:15060 >>> Max-Forwards: 70 >>> Subject: Performance Test >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/ >>> 1000 at 192.168.26.39:15060 entering state [completed][200] >>> 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/ >>> 1000 at 192.168.26.39:15060 entering state [ready][200] >>> 2010-07-02 09:41:26.582952 [DEBUG] switch_core_session.c:647 Send signal >>> sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >>> 2010-07-02 09:41:26.582952 [DEBUG] switch_channel.c:2494 (sofia/internal/ >>> 1000 at 192.168.26.39:15060) Callstate Change RINGING -> ACTIVE >>> [36m2010-07-02 09:41:26.582952 [NOTICE] mod_dptools.c:746 Channel >>> [sofia/internal/1000 at 192.168.26.39:15060] has been answered >>> EXECUTE sofia/internal/1000 at 192.168.26.39:15060playback(silence_stream://2000) >>> 2010-07-02 09:41:26.584953 [DEBUG] switch_ivr_play_say.c:1161 Codec >>> Activated L16 at 8000hz 1 channels 20ms >>> 2010-07-02 09:41:28.578954 [DEBUG] switch_ivr_play_say.c:1468 done >>> playing file >>> EXECUTE sofia/internal/1000 at 192.168.26.39:15060 rxfax(/tmp/999.tiff) >>> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp.c:64 This is for fax test: >>> receive fax >>> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:445 trans mode = 1 >>> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:591 This is for fax >>> test: prag go to here!!! >>> 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1064 Raw read codec >>> activation Success L16 20000 >>> 2010-07-02 09:41:28.578954 [DEBUG] switch_core_codec.c:122 >>> sofia/internal/1000 at 192.168.26.39:15060 Push codec L16:10 >>> 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1080 Raw write codec >>> activation Success L16 >>> 2010-07-02 09:41:28.857958 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 >>> >>> send 956 bytes to udp/[192.168.21.76]:5060 at 01:41:29.754477: >>> >>> ------------------------------------------------------------------------ >>> NOTIFY sip:1001 at 192.168.21.76:5060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:25060 >>> ;rport;branch=z9hG4bKS989SFgmXvmNF >>> Max-Forwards: 70 >>> From: "1001" >> >;tag=vr6p3K1XK247r >>> To: "1001" ;tag=26647676 >>> Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. >>> CSeq: 132897310 NOTIFY >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Event: message-summary >>> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, >>> sla, include-session-description, presence.winfo, message-summary, refer >>> Subscription-State: terminated;reason=timeout >>> Content-Type: application/simple-message-summary >>> Content-Length: 65 >>> >>> Messages-Waiting: no >>> Message-Account: sip:1001 at 192.168.26.39 >>> >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:29.757979 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 >>> recv 389 bytes from udp/[192.168.21.76]:5060 at 01:41:29.758362: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 481 Call/Transaction Does Not Exist >>> Via: SIP/2.0/UDP 192.168.26.39:25060 >>> ;rport=25060;branch=z9hG4bKS989SFgmXvmNF >>> To: "1001";tag=26647676 >>> From: "1001">> >;tag=vr6p3K1XK247r >>> Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. >>> CSeq: 132897310 NOTIFY >>> Accept-Language: en >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> >>> 2010-07-02 09:41:36.557090 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 >>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:36.587253: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: sip:666666 at 192.168.26.39:25060 >>> Max-Forwards: 70 >>> Subject: Performance Test >>> Content-Type: application/sdp >>> Content-Length: 345 >>> >>> v=0 >>> o=root 0 0 IN IP4 192.168.26.39 >>> s=Session SDP >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 49172 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:9600 >>> a=T38FaxFillBitRemoval:0 >>> a=T38FaxTranscodingMMR:0 >>> a=T38FaxTranscodingJBIG:0 >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:200 >>> a=T38FaxMaxDatagram:72 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> >>> ------------------------------------------------------------------------ >>> send 312 bytes to udp/[192.168.26.39]:25060 at 01:41:36.587546: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> recv 312 bytes from udp/[192.168.26.39]:25060 at 01:41:36.587641: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4297 Channel >>> sofia/internal/1000 at 192.168.26.39:15060 entering state [received][100] >>> >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4308 Remote SDP: >>> v=0 >>> >>> o=root 0 0 IN IP4 192.168.26.39 >>> >>> s=Session SDP >>> >>> c=IN IP4 192.168.26.39 >>> >>> t=0 0 >>> >>> m=image 49172 udptl t38 >>> >>> a=T38FaxVersion:0 >>> >>> a=T38MaxBitRate:9600 >>> >>> a=T38FaxFillBitRemoval:0 >>> >>> a=T38FaxTranscodingMMR:0 >>> >>> a=T38FaxTranscodingJBIG:0 >>> >>> a=T38FaxRateManagement:transferredTCF >>> >>> a=T38FaxMaxBuffer:200 >>> >>> a=T38FaxMaxDatagram:72 >>> >>> a=T38FaxUdpEC:t38UDPRedundancy >>> >>> >>> 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 0 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:744 T38FaxVersion = >>> 0 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:745 T38MaxBitRate = >>> 9600 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:746 >>> T38FaxFillBitRemoval = 1 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:747 >>> T38FaxTranscodingMMR = 1 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:748 >>> T38FaxTranscodingJBIG = 1 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:749 >>> T38FaxRateManagement = 'transferredTCF' >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:750 T38FaxMaxBuffer >>> = 200 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:751 >>> T38FaxMaxDatagram = 72 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:752 T38FaxUdpEC = >>> 't38UDPRedundancy' >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:753 T38VendorInfo = >>> '' >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:754 ip = >>> '192.168.26.39' >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:756 port = 49172 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_sofia.c:1232 IMAGE UDPTL CHANGING >>> DEST TO: [192.168.26.39:49172] >>> 2010-07-02 09:41:36.597094 [DEBUG] sofia_glue.c:122 sofia/internal/ >>> 1000 at 192.168.26.39:15060 image media sdp: >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:445 trans mode = 0 >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:36.597525: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:36.597094 [DEBUG] sofia.c:4297 Channel sofia/internal/ >>> 1000 at 192.168.26.39:15060 entering state [completed][200] >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:36.597657: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:36.607095 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 >>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:37.089156: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: sip:666666 at 192.168.26.39:25060 >>> Max-Forwards: 70 >>> Subject: Performance Test >>> Content-Type: application/sdp >>> Content-Length: 345 >>> >>> v=0 >>> o=root 0 0 IN IP4 192.168.26.39 >>> s=Session SDP >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 49172 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:9600 >>> a=T38FaxFillBitRemoval:0 >>> a=T38FaxTranscodingMMR:0 >>> a=T38FaxTranscodingJBIG:0 >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:200 >>> a=T38FaxMaxDatagram:72 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.089302: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.089395: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.098259: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.098339: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:38.091157: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: sip:666666 at 192.168.26.39:25060 >>> Max-Forwards: 70 >>> Subject: Performance Test >>> Content-Type: application/sdp >>> Content-Length: 345 >>> >>> v=0 >>> o=root 0 0 IN IP4 192.168.26.39 >>> s=Session SDP >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 49172 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:9600 >>> a=T38FaxFillBitRemoval:0 >>> a=T38FaxTranscodingMMR:0 >>> a=T38FaxTranscodingJBIG:0 >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:200 >>> a=T38FaxMaxDatagram:72 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.091305: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.091406: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.098260: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.098344: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> This is for fax test: dis 5This is for fax test: cause disconnect 4 >>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:40.093267: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: sip:666666 at 192.168.26.39:25060 >>> Max-Forwards: 70 >>> Subject: Performance Test >>> Content-Type: application/sdp >>> Content-Length: 345 >>> >>> v=0 >>> o=root 0 0 IN IP4 192.168.26.39 >>> s=Session SDP >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 49172 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:9600 >>> a=T38FaxFillBitRemoval:0 >>> a=T38FaxTranscodingMMR:0 >>> a=T38FaxTranscodingJBIG:0 >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:200 >>> a=T38FaxMaxDatagram:72 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.093421: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.093541: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.098263: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.098418: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:41.606147 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 >>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:44.095284: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: sip:666666 at 192.168.26.39:25060 >>> Max-Forwards: 70 >>> Subject: Performance Test >>> Content-Type: application/sdp >>> Content-Length: 345 >>> >>> v=0 >>> o=root 0 0 IN IP4 192.168.26.39 >>> s=Session SDP >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 49172 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:9600 >>> a=T38FaxFillBitRemoval:0 >>> a=T38FaxTranscodingMMR:0 >>> a=T38FaxTranscodingJBIG:0 >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:200 >>> a=T38FaxMaxDatagram:72 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.095457: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.095556: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.098263: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.098399: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:46.607257 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = 1 >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:48.098287: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> >>> -----------------------------log end---------------------------------- >>> >>> >>> 2010/7/2 Brian West >>> >>> turn sip on >>>> >>>> sofia profile xxx siptrace on >>>> >>>> /b >>>> >>>> On Jul 1, 2010, at 8:23 PM, chi zhang wrote: >>>> >>>> > hi,brian >>>> > >>>> >>>> >>>> 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/250837a1/attachment-0001.html From david.ponzone at gmail.com Thu Jul 8 02:30:35 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 8 Jul 2010 11:30:35 +0200 Subject: [Freeswitch-users] Problem of FreeSwitch and T.38 Test In-Reply-To: References: <5ECC3A8A-CDF9-4EF4-AD87-A3FAEDFF26AB@freeswitch.org> <12B00ED2-34A6-414C-A0C6-590A963E3CD2@gmail.com> Message-ID: As far as I remember, Zoiper can receive fax. For the rest, I think you lost me somewhere... From the beginning, I was thinking you was trying to receive a fax with Zoiper, but I see you wonder if it can receive fax. What I said earlier is just that it's possible Zopier can receive only on inbound call. Earlier, you was trying to receive a fax from FS to Zoiper during an outbound call from Zoiper to FS, which is not the normal way (it does not happen this way in real life). David Ponzone Direction Technique email: david.ponzone at 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 pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 08/07/2010 ? 11:10, chi zhang a ?crit : > I just use FS's CLI to call zoiper from FS, it works OK > > CLI cmd : originate sofia/internal/1001 at 192.168.26.39:25060 1001 > > i think if i want to do send fax from FS, i need one softphone can > receive fax, zoiper can do this or not? That is a question... > > > 2010/7/8 David Ponzone > Ok, so if I understood correctly, you call a number on FS, and you > expect to receive a fax on that call. > Zopier is ok, but it's far from a perfect piece of software. > Perhaps it is not able to receive a fax when it calls out. > Could you try to call Zoiper from FS before sending the fax to > Zoiper ? > > David Ponzone Direction Technique > email: david.ponzone at 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 pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou > falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le > d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > Le 08/07/2010 ? 10:20, chi zhang a ?crit : > >> Hi, David >> >> transfer fax function means "FS send fax to a sip user" , >> previously , i use zoiper to send fax to FS, for FS ,this means >> "receive fax", now i want to realize send fax from FS to one sip >> user. >> >> regards >> sammy >> >> 2010/7/8 David Ponzone >> Chi, >> >> I am not sure to understand what you mean by "transfer fax function". >> Can you clear that up please ? >> >> Thanks >> >> David Ponzone Direction Technique >> email: david.ponzone at 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 pi?ces jointes sont confidentiels et >> ?tablis ? l'intention exclusive de ses destinataires. Toute >> utilisation ou diffusion non autoris?e est interdite. Tout message >> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute >> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou >> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de >> le d?truire imm?diatement et d'avertir l'exp?diteur. >> >> >> >> >> Le 08/07/2010 ? 09:57, chi zhang a ?crit : >> >>> Now, i have finished T.38 fax receive test with Zoiper, it >>> works well. But, transfer a fax is still Not successful. >>> Previous, i do simulate fax with sipP, but FS return >>> 48(Disconnected after permitted retries). >>> Accidentally, i found softphone: Zoiper has fax function, so >>> retry fax with it, and in diaplan file: default.xml, 9178 was the >>> receive fax number. So i call 9178 with Zoiper(register as 1000), >>> fax receiving is perfect done. >>> Then i test transfer fax function: call to 9179(configured by >>> TX fax in default.xml), but FS return 2 (Timed out waiting for >>> initial communication) , i have no idea about it, reason ? >>> >>> regards >>> sammy >>> >>> >>> >>> 2010/7/2 chi zhang >>> I got it. >>> >>> -------------------log start-------------------------- >>> recv 576 bytes from udp/[192.168.26.39]:15060 at 01:41:26.500127: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:666666 at 192.168.26.39:25060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >>> From: 1000 ;tag=1 >>> To: 666666 >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: sip:1000 at 192.168.26.39:15060 >>> Max-Forwards: 70 >>> Subject: Performance Test >>> Content-Type: application/sdp >>> Content-Length: 184 >>> >>> v=0 >>> o=user1 3748 3748 IN IP4 192.168.26.39 >>> s=- >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=audio 6000 RTP/AVP 8 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-11,16 >>> >>> ------------------------------------------------------------------------ >>> send 294 bytes to udp/[192.168.26.39]:15060 at 01:41:26.500477: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >>> From: 1000 ;tag=1 >>> To: 666666 >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:26.498946 [DEBUG] sofia.c:5928 IP 192.168.26.39 >>> Approved by acl "192.168.26.0/24[]". Access Granted. >>> [36m2010-07-02 09:41:26.498946 [NOTICE] switch_channel.c:776 New >>> Channel sofia/internal/1000 at 192.168.26.39:15060 >>> [f588c66c-48c8-4220-a944-8de287adb3ab] >>> 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change >>> CS_NEW >>> 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:320 >>> (sofia/internal/1000 at 192.168.26.39:15060) State NEW >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4297 Channel sofia/internal/1000 at 192.168.26.39 >>> :15060 entering state [received][100] >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4308 Remote SDP: >>> v=0 >>> >>> o=user1 3748 3748 IN IP4 192.168.26.39 >>> >>> s=- >>> >>> c=IN IP4 192.168.26.39 >>> >>> t=0 0 >>> >>> m=audio 6000 RTP/AVP 8 101 >>> >>> a=rtpmap:8 PCMA/8000 >>> >>> a=rtpmap:101 telephone-event/8000 >>> >>> a=fmtp:101 0-11,16 >>> >>> >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec >>> Compare [PCMA:8:8000:20]/[G7221:115:32000:20] >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec >>> Compare [PCMA:8:8000:20]/[G7221:107:16000:20] >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec >>> Compare [PCMA:8:8000:20]/[G722:9:8000:20] >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec >>> Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec >>> Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:2442 Set Codec sofia/internal/1000 at 192.168.26.39 >>> :15060 PCMA/8000 20 ms 160 samples >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3798 Set 2833 dtmf >>> send/recv payload to 101 >>> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4455 (sofia/internal/1000 at 192.168.26.39 >>> :15060) State Change CS_NEW -> CS_INIT >>> 2010-07-02 09:41:26.542951 [DEBUG] switch_core_session.c:1027 Send >>> signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change >>> CS_INIT >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/internal/1000 at 192.168.26.39:15060) State INIT >>> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:83 sofia/internal/1000 at 192.168.26.39 >>> :15060 SOFIA INIT >>> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:119 (sofia/internal/1000 at 192.168.26.39 >>> :15060) State Change CS_INIT -> CS_ROUTING >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send >>> signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/internal/1000 at 192.168.26.39:15060) State INIT going to sleep >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change >>> CS_ROUTING >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_channel.c:1471 (sofia/internal/1000 at 192.168.26.39 >>> :15060) Callstate Change DOWN -> RINGING >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING >>> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:142 sofia/internal/1000 at 192.168.26.39 >>> :15060 SOFIA ROUTING >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1000 at 192.168.26.39 >>> :15060 Standard ROUTING >>> [32m2010-07-02 09:41:26.544953 [INFO] mod_dialplan_xml.c:331 >>> Processing 1000->666666 in context public >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public- >>> >fax] continue=false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) >>> [fax] destination_number(666666) =~ /^fax$/ break=on-false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public- >>> >4444] continue=false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) >>> [4444] destination_number(666666) =~ /^(4444)$/ break=on-false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public- >>> >fax] continue=false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) >>> [fax] destination_number(666666) =~ /^fax$/ break=on-false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public- >>> >666666] continue=false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (PASS) >>> [666666] destination_number(666666) =~ /^(666666)$/ break=on-false >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action answer() >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action >>> playback(silence_stream://2000) >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action rxfax(/ >>> tmp/999.tiff) >>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action hangup() >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:119 >>> (sofia/internal/1000 at 192.168.26.39:15060) State Change CS_ROUTING - >>> > CS_EXECUTE >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send >>> signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING going to >>> sleep >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change >>> CS_EXECUTE >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:348 >>> (sofia/internal/1000 at 192.168.26.39:15060) State EXECUTE >>> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:235 sofia/internal/1000 at 192.168.26.39 >>> :15060 SOFIA EXECUTE >>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1000 at 192.168.26.39 >>> :15060 Standard EXECUTE >>> EXECUTE sofia/internal/1000 at 192.168.26.39:15060 answer() >>> 2010-07-02 09:41:26.580967 [DEBUG] sofia_glue.c:2682 AUDIO RTP [sofia/internal/1000 at 192.168.26.39 >>> :15060] 192.168.26.39 port 22464 -> 192.168.26.39 port 6000 codec: >>> 8 ms: 20 >>> 2010-07-02 09:41:26.580967 [DEBUG] switch_rtp.c:1413 Starting >>> timer [soft] 160 bytes per 20ms >>> 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2892 Set 2833 dtmf >>> send payload to 101 >>> 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2897 Set 2833 dtmf >>> receive payload to 101 >>> 2010-07-02 09:41:26.582952 [DEBUG] mod_sofia.c:669 Local SDP sofia/internal/1000 at 192.168.26.39 >>> :15060: >>> v=0 >>> o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=audio 22464 RTP/AVP 8 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=silenceSupp:off - - - - >>> a=ptime:20 >>> a=sendrecv >>> >>> send 1091 bytes to udp/[192.168.26.39]:15060 at 01:41:26.584093: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >>> From: 1000 ;tag=1 >>> To: 666666 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Allow-Events: talk, hold, presence, dialog, line-seize, call- >>> info, sla, include-session-description, presence.winfo, message- >>> summary, refer >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 249 >>> Remote-Party-ID: "666666" >> 666666 at 192.168.26.39>;party=calling;privacy=off;screen=no >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=audio 22464 RTP/AVP 8 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=silenceSupp:off - - - - >>> a=ptime:20 >>> >>> ------------------------------------------------------------------------ >>> recv 369 bytes from udp/[192.168.26.39]:15060 at 01:41:26.584205: >>> >>> ------------------------------------------------------------------------ >>> ACK sip:666666 at 192.168.26.39:25060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-4 >>> From: 1000 ;tag=1 >>> To: 666666 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 ACK >>> Contact: sip:1000 at 192.168.26.39:15060 >>> Max-Forwards: 70 >>> Subject: Performance Test >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/1000 at 192.168.26.39 >>> :15060 entering state [completed][200] >>> 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/1000 at 192.168.26.39 >>> :15060 entering state [ready][200] >>> 2010-07-02 09:41:26.582952 [DEBUG] switch_core_session.c:647 Send >>> signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >>> 2010-07-02 09:41:26.582952 [DEBUG] switch_channel.c:2494 (sofia/internal/1000 at 192.168.26.39 >>> :15060) Callstate Change RINGING -> ACTIVE >>> [36m2010-07-02 09:41:26.582952 [NOTICE] mod_dptools.c:746 Channel [sofia/internal/1000 at 192.168.26.39 >>> :15060] has been answered >>> EXECUTE sofia/internal/1000 at 192.168.26.39:15060 >>> playback(silence_stream://2000) >>> 2010-07-02 09:41:26.584953 [DEBUG] switch_ivr_play_say.c:1161 >>> Codec Activated L16 at 8000hz 1 channels 20ms >>> 2010-07-02 09:41:28.578954 [DEBUG] switch_ivr_play_say.c:1468 done >>> playing file >>> EXECUTE sofia/internal/1000 at 192.168.26.39:15060 rxfax(/tmp/999.tiff) >>> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp.c:64 This is for fax >>> test: receive fax >>> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:445 trans mode >>> = 1 >>> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:591 This is for >>> fax test: prag go to here!!! >>> 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1064 Raw read >>> codec activation Success L16 20000 >>> 2010-07-02 09:41:28.578954 [DEBUG] switch_core_codec.c:122 sofia/internal/1000 at 192.168.26.39 >>> :15060 Push codec L16:10 >>> 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1080 Raw >>> write codec activation Success L16 >>> 2010-07-02 09:41:28.857958 [ERR] mod_spandsp_fax.c:1119 pvt- >>> >t38_mode = 0 >>> >>> send 956 bytes to udp/[192.168.21.76]:5060 at 01:41:29.754477: >>> >>> ------------------------------------------------------------------------ >>> NOTIFY sip:1001 at 192.168.21.76:5060 SIP/2.0 >>> Via: SIP/2.0/UDP >>> 192.168.26.39:25060;rport;branch=z9hG4bKS989SFgmXvmNF >>> Max-Forwards: 70 >>> From: "1001" >> 1001 at 192.168.26.39:25060;transport=UDP>;tag=vr6p3K1XK247r >>> To: "1001" >> 1001 at 192.168.26.39:25060;transport=UDP>;tag=26647676 >>> Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. >>> CSeq: 132897310 NOTIFY >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Event: message-summary >>> Allow-Events: talk, hold, presence, dialog, line-seize, call- >>> info, sla, include-session-description, presence.winfo, message- >>> summary, refer >>> Subscription-State: terminated;reason=timeout >>> Content-Type: application/simple-message-summary >>> Content-Length: 65 >>> >>> Messages-Waiting: no >>> Message-Account: sip:1001 at 192.168.26.39 >>> >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:29.757979 [ERR] mod_spandsp_fax.c:1119 pvt- >>> >t38_mode = 0 >>> recv 389 bytes from udp/[192.168.21.76]:5060 at 01:41:29.758362: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 481 Call/Transaction Does Not Exist >>> Via: SIP/2.0/UDP >>> 192.168.26.39:25060;rport=25060;branch=z9hG4bKS989SFgmXvmNF >>> To: "1001">> 1001 at 192.168.26.39:25060;transport=UDP>;tag=26647676 >>> From: "1001">> 1001 at 192.168.26.39:25060;transport=UDP>;tag=vr6p3K1XK247r >>> Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. >>> CSeq: 132897310 NOTIFY >>> Accept-Language: en >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> >>> 2010-07-02 09:41:36.557090 [ERR] mod_spandsp_fax.c:1119 pvt- >>> >t38_mode = 0 >>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:36.587253: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: sip:666666 at 192.168.26.39:25060 >>> Max-Forwards: 70 >>> Subject: Performance Test >>> Content-Type: application/sdp >>> Content-Length: 345 >>> >>> v=0 >>> o=root 0 0 IN IP4 192.168.26.39 >>> s=Session SDP >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 49172 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:9600 >>> a=T38FaxFillBitRemoval:0 >>> a=T38FaxTranscodingMMR:0 >>> a=T38FaxTranscodingJBIG:0 >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:200 >>> a=T38FaxMaxDatagram:72 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> >>> ------------------------------------------------------------------------ >>> send 312 bytes to udp/[192.168.26.39]:25060 at 01:41:36.587546: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> recv 312 bytes from udp/[192.168.26.39]:25060 at 01:41:36.587641: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4297 >>> Channel sofia/internal/1000 at 192.168.26.39:15060 entering state >>> [received][100] >>> >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4308 Remote SDP: >>> v=0 >>> >>> o=root 0 0 IN IP4 192.168.26.39 >>> >>> s=Session SDP >>> >>> c=IN IP4 192.168.26.39 >>> >>> t=0 0 >>> >>> m=image 49172 udptl t38 >>> >>> a=T38FaxVersion:0 >>> >>> a=T38MaxBitRate:9600 >>> >>> a=T38FaxFillBitRemoval:0 >>> >>> a=T38FaxTranscodingMMR:0 >>> >>> a=T38FaxTranscodingJBIG:0 >>> >>> a=T38FaxRateManagement:transferredTCF >>> >>> a=T38FaxMaxBuffer:200 >>> >>> a=T38FaxMaxDatagram:72 >>> >>> a=T38FaxUdpEC:t38UDPRedundancy >>> >>> >>> 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:1119 pvt- >>> >t38_mode = 0 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:744 >>> T38FaxVersion = 0 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:745 >>> T38MaxBitRate = 9600 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:746 >>> T38FaxFillBitRemoval = 1 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:747 >>> T38FaxTranscodingMMR = 1 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:748 >>> T38FaxTranscodingJBIG = 1 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:749 >>> T38FaxRateManagement = 'transferredTCF' >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:750 >>> T38FaxMaxBuffer = 200 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:751 >>> T38FaxMaxDatagram = 72 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:752 >>> T38FaxUdpEC = 't38UDPRedundancy' >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:753 >>> T38VendorInfo = '' >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:754 ip = >>> '192.168.26.39' >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:756 port = >>> 49172 >>> 2010-07-02 09:41:36.597094 [DEBUG] mod_sofia.c:1232 IMAGE UDPTL >>> CHANGING DEST TO: [192.168.26.39:49172] >>> 2010-07-02 09:41:36.597094 [DEBUG] sofia_glue.c:122 sofia/internal/1000 at 192.168.26.39 >>> :15060 image media sdp: >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:445 trans mode >>> = 0 >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:36.597525: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:36.597094 [DEBUG] sofia.c:4297 Channel sofia/internal/1000 at 192.168.26.39 >>> :15060 entering state [completed][200] >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:36.597657: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:36.607095 [ERR] mod_spandsp_fax.c:1119 pvt- >>> >t38_mode = 1 >>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:37.089156: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: sip:666666 at 192.168.26.39:25060 >>> Max-Forwards: 70 >>> Subject: Performance Test >>> Content-Type: application/sdp >>> Content-Length: 345 >>> >>> v=0 >>> o=root 0 0 IN IP4 192.168.26.39 >>> s=Session SDP >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 49172 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:9600 >>> a=T38FaxFillBitRemoval:0 >>> a=T38FaxTranscodingMMR:0 >>> a=T38FaxTranscodingJBIG:0 >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:200 >>> a=T38FaxMaxDatagram:72 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.089302: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.089395: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.098259: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.098339: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:38.091157: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: sip:666666 at 192.168.26.39:25060 >>> Max-Forwards: 70 >>> Subject: Performance Test >>> Content-Type: application/sdp >>> Content-Length: 345 >>> >>> v=0 >>> o=root 0 0 IN IP4 192.168.26.39 >>> s=Session SDP >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 49172 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:9600 >>> a=T38FaxFillBitRemoval:0 >>> a=T38FaxTranscodingMMR:0 >>> a=T38FaxTranscodingJBIG:0 >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:200 >>> a=T38FaxMaxDatagram:72 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.091305: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.091406: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.098260: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.098344: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> This is for fax test: dis 5This is for fax test: cause disconnect 4 >>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:40.093267: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: sip:666666 at 192.168.26.39:25060 >>> Max-Forwards: 70 >>> Subject: Performance Test >>> Content-Type: application/sdp >>> Content-Length: 345 >>> >>> v=0 >>> o=root 0 0 IN IP4 192.168.26.39 >>> s=Session SDP >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 49172 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:9600 >>> a=T38FaxFillBitRemoval:0 >>> a=T38FaxTranscodingMMR:0 >>> a=T38FaxTranscodingJBIG:0 >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:200 >>> a=T38FaxMaxDatagram:72 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.093421: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.093541: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.098263: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.098418: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:41.606147 [ERR] mod_spandsp_fax.c:1119 pvt- >>> >t38_mode = 1 >>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:44.095284: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: sip:666666 at 192.168.26.39:25060 >>> Max-Forwards: 70 >>> Subject: Performance Test >>> Content-Type: application/sdp >>> Content-Length: 345 >>> >>> v=0 >>> o=root 0 0 IN IP4 192.168.26.39 >>> s=Session SDP >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 49172 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:9600 >>> a=T38FaxFillBitRemoval:0 >>> a=T38FaxTranscodingMMR:0 >>> a=T38FaxTranscodingJBIG:0 >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:200 >>> a=T38FaxMaxDatagram:72 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.095457: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.095556: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.098263: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.098399: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> ------------------------------------------------------------------------ >>> 2010-07-02 09:41:46.607257 [ERR] mod_spandsp_fax.c:1119 pvt- >>> >t38_mode = 1 >>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:48.098287: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>> From: 666666 ;tag=1 >>> To: 1000 ;tag=85ySHt6rmaSyp >>> Call-ID: 1-16005 at 192.168.26.39 >>> CSeq: 2 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>> Supported: timer, precondition, path, replaces >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 341 >>> >>> v=0 >>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>> s=FreeSWITCH >>> c=IN IP4 192.168.26.39 >>> t=0 0 >>> m=image 22464 udptl t38 >>> a=T38FaxVersion:0 >>> a=T38MaxBitRate:14400 >>> a=T38FaxFillBitRemoval >>> a=T38FaxRateManagement:transferredTCF >>> a=T38FaxMaxBuffer:2000 >>> a=T38FaxMaxDatagram:400 >>> a=T38FaxUdpEC:t38UDPRedundancy >>> a=T38VendorInfo:0 0 0 >>> >>> >>> -----------------------------log >>> end---------------------------------- >>> >>> >>> 2010/7/2 Brian West >>> >>> turn sip on >>> >>> sofia profile xxx siptrace on >>> >>> /b >>> >>> On Jul 1, 2010, at 8:23 PM, chi zhang wrote: >>> >>> > hi,brian >>> > >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/ac17db41/attachment-0001.html From steveayre at gmail.com Thu Jul 8 02:37:12 2010 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 8 Jul 2010 10:37:12 +0100 Subject: [Freeswitch-users] hangup does not break the Lua loop In-Reply-To: References: <4B80CF4E-6336-4F2B-A118-ED87D6979F90@freeswitch.org> Message-ID: session:ready - checks whether the session is still active (true anytime between call starts and hangup) session:answered - checks whether the session is flagged as answered (true anytime after the call has been answered) I don't think session:answered returns false after the call has hung up, while session:ready will. Also session:answered won't be true until the call is answered, but session:ready will be. -Steve On 8 July 2010 03:39, Tony Tin wrote: > sorry, one more question > > how does session:ready compare to session:answered ? > > Thanks > > Tony > > > On Wed, Jul 7, 2010 at 8:15 PM, Brian West wrote: > >> Post your code for your while loop... sounds like you're not checking >> session:ready to see if its still true. >> >> /b >> >> On Jul 7, 2010, at 3:48 AM, Tony Tin wrote: >> >> > Hi, >> > >> > I'm writing dialplan with Lua script. There is a while loop in the >> dialplan, I found that the loop is not broke out even the call has been hung >> up. I'm wondering whether this is a normal behavior. Thanks. >> > >> > Regards, >> > Tony >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/449029da/attachment.html From javieraristizabal at gmail.com Thu Jul 8 05:06:04 2010 From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=) Date: Thu, 8 Jul 2010 07:06:04 -0500 Subject: [Freeswitch-users] hangup does not break the Lua loop In-Reply-To: References: <4B80CF4E-6336-4F2B-A118-ED87D6979F90@freeswitch.org> Message-ID: Tony, you can try: while (var and session:ready) do ........ end On Thu, Jul 8, 2010 at 4:37 AM, Steven Ayre wrote: > session:ready - checks whether the session is still active (true anytime > between call starts and hangup) > > session:answered - checks whether the session is flagged as answered (true > anytime after the call has been answered) > > I don't think session:answered returns false after the call has hung up, > while session:ready will. Also session:answered won't be true until the call > is answered, but session:ready will be. > > -Steve > > > > > On 8 July 2010 03:39, Tony Tin wrote: > >> sorry, one more question >> >> how does session:ready compare to session:answered ? >> >> Thanks >> >> Tony >> >> >> On Wed, Jul 7, 2010 at 8:15 PM, Brian West wrote: >> >>> Post your code for your while loop... sounds like you're not checking >>> session:ready to see if its still true. >>> >>> /b >>> >>> On Jul 7, 2010, at 3:48 AM, Tony Tin wrote: >>> >>> > Hi, >>> > >>> > I'm writing dialplan with Lua script. There is a while loop in the >>> dialplan, I found that the loop is not broke out even the call has been hung >>> up. I'm wondering whether this is a normal behavior. Thanks. >>> > >>> > Regards, >>> > Tony >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Javier Aristiz?bal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/6a1053e9/attachment.html From yurazilot1 at list.ru Thu Jul 8 03:11:35 2010 From: yurazilot1 at list.ru (viewpoint) Date: Thu, 08 Jul 2010 14:11:35 +0400 Subject: [Freeswitch-users] FS HA-cluster Message-ID: Hello. I currently have a project where I'm researching how to establish a clustered platform wieth failover time ~= some milliseconds. At present we have: 2 identical servers FS, wieth established Pacemaker (http://www.clusterlabs.org/wiki/Install#From_Source) - heartbeat. On the one hand it is possible to tell that all works, as, at refusal of the first server, the second server receives cluster-ip. A problem that FreeSWITCH it is necessary to restart (or simply to start) that it began to work with the new IP-address. As the result, a switching total time makes 10-20 sec which basic part is necessary on launch FreeSwitch. any ideas? Thanks. From anthony.minessale at gmail.com Thu Jul 8 06:20:57 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 8 Jul 2010 08:20:57 -0500 Subject: [Freeswitch-users] hangup does not break the Lua loop In-Reply-To: References: <4B80CF4E-6336-4F2B-A118-ED87D6979F90@freeswitch.org> Message-ID: Also session:ready will return false if the call is being transferred. Bottom line is you should always be checking session:ready on any loops and periodicly throughout your script and exit asap if it returns false. On Jul 8, 2010 7:13 AM, "Javier Aristiz?bal" wrote: Tony, you can try: while (var and session:ready) do ........ end On Thu, Jul 8, 2010 at 4:37 AM, Steven Ayre wrote: > > session:ready - checks... -- Javier Aristiz?bal _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/3ec28bcf/attachment.html From tzury.by at reguluslabs.com Thu Jul 8 06:28:32 2010 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Thu, 8 Jul 2010 16:28:32 +0300 Subject: [Freeswitch-users] enforcing RTP payload numbers for codecs Message-ID: Hi, I just download the latest, build and brought up all is fine. However, our clients are connected via IP over GPRS(UMTS) and thus we are using Speex codec. Is there a way I can enforce speex to have 97 or 98 instead of 99? Currently as it looks in the vars.xml speex is 99. see dump (taken from default vars.xml) thanks Tzury RTP Dynamic Payload Numbers currently used in FreeSWITCH and what for. 96 - AMR 97 - iLBC (30) 98 - iLBC (20) 99 - Speex 8kHz, 16kHz, 32kHz 100 - From anthony.minessale at gmail.com Thu Jul 8 06:32:00 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 8 Jul 2010 08:32:00 -0500 Subject: [Freeswitch-users] More questions migrating from Asterisk to FS In-Reply-To: <1278468923913-5263357.post@n2.nabble.com> References: <1278425102977-5260522.post@n2.nabble.com> <1278468923913-5263357.post@n2.nabble.com> Message-ID: Any or you wish to be the same for all users can be added to the rather than the and they will be inherited to every user. On Jul 6, 2010 9:19 PM, "mazilo" wrote: Thank you for your speedy response. I am not sure I do understand what you said. Please kindly bear with me as FS is a new world to me. It sure will be nice to see some sample XML codes. I used this http://wiki.freeswitch.org/wiki/Provider_Configuration:_Gizmo Gizmo5 configuration. Let's say I have two Gizmo5 accounts 17471234567 and 17477654321. Then, the contents of my sip_profiles/external/gizmo5.xml file that list both gateways will have only different username/password and the rest parameters will be the same. It would be nice that the same parameters don't need to be relisted on each gateway. I hope this makes it more clear. Anthony Minessale wrote: > > params put in the domain itself would apply to every user in that do... ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used ... View this message in context: http://freeswitch-users.2379917.n2.nabble.com/More-questions-migrating-from-Asterisk-to-FS-tp5260522p5263357.html Sent from the freeswitch-users mailing list archive at Nabble.com. ________________________________... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/084f2678/attachment-0001.html From anthony.minessale at gmail.com Thu Jul 8 06:33:03 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 8 Jul 2010 08:33:03 -0500 Subject: [Freeswitch-users] More questions migrating from Asterisk to FS In-Reply-To: <1278468923913-5263357.post@n2.nabble.com> References: <1278425102977-5260522.post@n2.nabble.com> <1278468923913-5263357.post@n2.nabble.com> Message-ID: Doh (using my phone, typo) On Jul 6, 2010 9:19 PM, "mazilo" wrote: Thank you for your speedy response. I am not sure I do understand what you said. Please kindly bear with me as FS is a new world to me. It sure will be nice to see some sample XML codes. I used this http://wiki.freeswitch.org/wiki/Provider_Configuration:_Gizmo Gizmo5 configuration. Let's say I have two Gizmo5 accounts 17471234567 and 17477654321. Then, the contents of my sip_profiles/external/gizmo5.xml file that list both gateways will have only different username/password and the rest parameters will be the same. It would be nice that the same parameters don't need to be relisted on each gateway. I hope this makes it more clear. Anthony Minessale wrote: > > params put in the domain itself would apply to every user in that do... ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used ... View this message in context: http://freeswitch-users.2379917.n2.nabble.com/More-questions-migrating-from-Asterisk-to-FS-tp5260522p5263357.html Sent from the freeswitch-users mailing list archive at Nabble.com. ________________________________... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/de04e3b1/attachment.html From tony.tin at noahmedia.com.hk Thu Jul 8 06:34:34 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Thu, 8 Jul 2010 21:34:34 +0800 Subject: [Freeswitch-users] hangup does not break the Lua loop In-Reply-To: References: <4B80CF4E-6336-4F2B-A118-ED87D6979F90@freeswitch.org> Message-ID: Thanks for replies, really appreciated ! Tony On Thu, Jul 8, 2010 at 9:20 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Also session:ready will return false if the call is being transferred. > Bottom line is you should always be checking session:ready on any loops and > periodicly throughout your script and exit asap if it returns false. > > On Jul 8, 2010 7:13 AM, "Javier Aristiz?bal" > wrote: > > Tony, you can try: > > while (var and session:ready) do > ........ > end > > > > > > On Thu, Jul 8, 2010 at 4:37 AM, Steven Ayre wrote: > > > > session:ready - checks... > -- > Javier Aristiz?bal > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/e0c748b2/attachment.html From moises.silva at gmail.com Thu Jul 8 06:51:58 2010 From: moises.silva at gmail.com (Moises Silva) Date: Thu, 8 Jul 2010 09:51:58 -0400 Subject: [Freeswitch-users] enforcing RTP payload numbers for codecs In-Reply-To: References: Message-ID: Hey Tzury, I don't think you can change that as of now unless you modify src/mod/codecs/mod_speex/mod_speex.c There is an ianacode array at the bottom filled with 99. Now, the inevitable question is, why do you need to change it? Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com On Thu, Jul 8, 2010 at 9:28 AM, Tzury Bar Yochay wrote: > Hi, > > I just download the latest, build and brought up all is fine. > However, our clients are connected via IP over GPRS(UMTS) and thus we > are using Speex codec. > > Is there a way I can enforce speex to have 97 or 98 instead of 99? > Currently as it looks in the vars.xml speex is 99. > see dump (taken from default vars.xml) > > thanks > Tzury > RTP Dynamic Payload Numbers currently used in FreeSWITCH and what > for. > > 96 - AMR > 97 - iLBC (30) > 98 - iLBC (20) > 99 - Speex 8kHz, 16kHz, 32kHz > 100 - > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/6b58b42f/attachment.html From rupa at rupa.com Thu Jul 8 06:53:38 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 8 Jul 2010 08:53:38 -0500 Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: References: Message-ID: instead of stopping/starting FS, why not stop/start just the appropriate sofia profile? That should be much faster. You can script it using fs_cli -x "sofia ..." On Thu, Jul 8, 2010 at 5:11 AM, viewpoint wrote: > Hello. > > I currently have a project where I'm researching how to establish a > clustered platform wieth failover time ~= some milliseconds. > > At present we have: 2 identical servers FS, wieth established Pacemaker ( > http://www.clusterlabs.org/wiki/Install#From_Source) - heartbeat. > On the one hand it is possible to tell that all works, as, at refusal of > the first server, the second server receives cluster-ip. A problem that > FreeSWITCH it is necessary to restart (or simply to start) that it began to > work with the new IP-address. As the result, a switching total time makes > 10-20 sec which basic part is necessary on launch FreeSwitch. > > any ideas? > > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/c2e3574e/attachment.html From Nabble at slickdeals.endjunk.com Thu Jul 8 07:01:03 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Thu, 8 Jul 2010 07:01:03 -0700 (PDT) Subject: [Freeswitch-users] More questions migrating from Asterisk to FS In-Reply-To: References: <1278425102977-5260522.post@n2.nabble.com> <1278468923913-5263357.post@n2.nabble.com> Message-ID: <1278597663871-5270225.post@n2.nabble.com> I am sorry that I am completey lost with such a cryptic statement. Could you kindly elaborate it? Thanks. Anthony Minessale wrote: > > Doh (using my phone, typo) ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/More-questions-migrating-from-Asterisk-to-FS-tp5260522p5270225.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Jul 8 07:04:25 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jul 2010 09:04:25 -0500 Subject: [Freeswitch-users] enforcing RTP payload numbers for codecs In-Reply-To: References: Message-ID: <671D3268-0CE1-40A5-AA22-764BC1E293B7@freeswitch.org> Ok here is what I have to rant about... the thing is called an RTP map because it maps numbers to names in the SDP. INVITE inbound says 99/speex the 200OK outbound says 98/speex, so you are to send to 98 but receive on 99 for the codec speex... the issue here is NOTHING BLOODY DOES THIS RIGHT. How about we start opening bugs on these issues with the vendors of the products you actually paid money to. :P /b On Jul 8, 2010, at 8:28 AM, Tzury Bar Yochay wrote: > Hi, > > I just download the latest, build and brought up all is fine. > However, our clients are connected via IP over GPRS(UMTS) and thus we > are using Speex codec. > > Is there a way I can enforce speex to have 97 or 98 instead of 99? > Currently as it looks in the vars.xml speex is 99. > see dump (taken from default vars.xml) > > thanks > Tzury > RTP Dynamic Payload Numbers currently used in FreeSWITCH and what for. > > 96 - AMR > 97 - iLBC (30) > 98 - iLBC (20) > 99 - Speex 8kHz, 16kHz, 32kHz > 100 - From anthony.minessale at gmail.com Thu Jul 8 07:08:32 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 8 Jul 2010 09:08:32 -0500 Subject: [Freeswitch-users] enforcing RTP payload numbers for codecs In-Reply-To: References: Message-ID: No, you can't force codecs in the dynamic range and if properly implemented, it won't matter what number either side chooses. On Jul 8, 2010 8:34 AM, "Tzury Bar Yochay" wrote: Hi, I just download the latest, build and brought up all is fine. However, our clients are connected via IP over GPRS(UMTS) and thus we are using Speex codec. Is there a way I can enforce speex to have 97 or 98 instead of 99? Currently as it looks in the vars.xml speex is 99. see dump (taken from default vars.xml) thanks Tzury RTP Dynamic Payload Numbers currently used in FreeSWITCH and what for. 96 - AMR 97 - iLBC (30) 98 - iLBC (20) 99 - Speex 8kHz, 16kHz, 32kHz 100 - _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/8ae841cc/attachment-0001.html From steveayre at gmail.com Thu Jul 8 07:09:05 2010 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 8 Jul 2010 15:09:05 +0100 Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: References: Message-ID: Or use sysctl to set net.ipv4.ip_nonlocal_bind = 1 That will allow FS to bind to the IP even when it does not have that IP. It won't receive traffic until the machine takes control of that IP, but will allow FS to start up before that time so that it'll start handling calls as soon as it fails over. You could also look at an approach like this http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS if you have another couple of servers available. -Steve On 8 July 2010 14:53, Rupa Schomaker wrote: > instead of stopping/starting FS, why not stop/start just the appropriate > sofia profile? That should be much faster. You can script it using fs_cli > -x "sofia ..." > > > On Thu, Jul 8, 2010 at 5:11 AM, viewpoint wrote: > >> Hello. >> >> I currently have a project where I'm researching how to establish a >> clustered platform wieth failover time ~= some milliseconds. >> >> At present we have: 2 identical servers FS, wieth established Pacemaker ( >> http://www.clusterlabs.org/wiki/Install#From_Source) - heartbeat. >> On the one hand it is possible to tell that all works, as, at refusal of >> the first server, the second server receives cluster-ip. A problem that >> FreeSWITCH it is necessary to restart (or simply to start) that it began to >> work with the new IP-address. As the result, a switching total time makes >> 10-20 sec which basic part is necessary on launch FreeSwitch. >> >> any ideas? >> >> Thanks. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/0137ea64/attachment.html From klejch+freeswitch at netbox.cz Thu Jul 8 07:21:40 2010 From: klejch+freeswitch at netbox.cz (Vladimir Klejch) Date: Thu, 8 Jul 2010 16:21:40 +0200 (CEST) Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: References: Message-ID: Hi On linux you can use /proc/sys/net/ipv4/ip_nonlocal_bind to bind to non local address on start of FS and then you don't need any restart of FS or reload of sofia profile if this addres is active on the node or you can have this floating address on dummy iface (?or lo ) and then use right settings in /proc/sys/net/ipv4/conf/eth?/arp_filter,arp_ignore and then FS can use this address to bind on start and you don't need restart of FS?or relaod of sofia profile ... Kleo On Thu, 8 Jul 2010, viewpoint wrote: > Hello. > > I currently have a project where I'm researching how to establish a > clustered platform wieth failover time ~= some milliseconds. > > At present we have: 2 identical servers FS, wieth established Pacemaker > (http://www.clusterlabs.org/wiki/Install#From_Source) - heartbeat. On > the one hand it is possible to tell that all works, as, at refusal of > the first server, the second server receives cluster-ip. A problem that > FreeSWITCH it is necessary to restart (or simply to start) that it began > to work with the new IP-address. As the result, a switching total time > makes 10-20 sec which basic part is necessary on launch FreeSwitch. > > any ideas? > > Thanks. -- klejch+freeswitch at netbox.cz From tzury.by at reguluslabs.com Thu Jul 8 07:28:16 2010 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Thu, 8 Jul 2010 17:28:16 +0300 Subject: [Freeswitch-users] enforcing RTP payload numbers for codecs In-Reply-To: References: Message-ID: Brian and Anthony, Dears, I am afraid it is a bug in teh FS See, if during SDP the client declare on payload 98 and fs declare on 99. Then client expect receiving 99 while sending 98. That is fine. However, when both clients are the same (as in our case) since FS is not trans-coding clients receives 98 in given that same sip client cannot work with previous versions of FS where there Speex was 98 (and 97 in older) On Thu, Jul 8, 2010 at 5:08 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > No, you can't force codecs in the dynamic range and if properly > implemented, it won't matter what number either side chooses. > > On Jul 8, 2010 8:34 AM, "Tzury Bar Yochay" > wrote: > > Hi, > > I just download the latest, build and brought up all is fine. > However, our clients are connected via IP over GPRS(UMTS) and thus we > are using Speex codec. > > Is there a way I can enforce speex to have 97 or 98 instead of 99? > Currently as it looks in the vars.xml speex is 99. > see dump (taken from default vars.xml) > > thanks > Tzury > RTP Dynamic Payload Numbers currently used in FreeSWITCH and what > for. > > 96 - AMR > 97 - iLBC (30) > 98 - iLBC (20) > 99 - Speex 8kHz, 16kHz, 32kHz > 100 - > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Tzury Bar Yochay tzury.by at reguluslabs.com + 972 52 5133399 twitter.com/tzury -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/ad60cd65/attachment.html From tzury.by at reguluslabs.com Thu Jul 8 07:40:33 2010 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Thu, 8 Jul 2010 17:40:33 +0300 Subject: [Freeswitch-users] enforcing RTP payload numbers for codecs In-Reply-To: References: Message-ID: If I could have enforce FS to transcode that session between those two clients it would have work properly (currently experiencing half duplex). On Thu, Jul 8, 2010 at 5:31 PM, Tzury Bar Yochay wrote: > BTW, > > Off topic though, I would be glad to hire you for our FS works. I was > looking for that for long enough and glad you brought that up ;-). > > > On Thu, Jul 8, 2010 at 5:28 PM, Tzury Bar Yochay > wrote: > >> Brian and Anthony, >> Dears, >> >> I am afraid it is a bug in teh FS >> See, if during SDP the client declare on payload 98 and fs declare on 99. >> Then client expect receiving 99 while sending 98. That is fine. >> However, when both clients are the same (as in our case) since FS is not >> trans-coding >> clients receives 98 in >> given that same sip client cannot work with previous versions of FS >> where there Speex was 98 (and 97 in older) >> >> >> On Thu, Jul 8, 2010 at 5:08 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> No, you can't force codecs in the dynamic range and if properly >>> implemented, it won't matter what number either side chooses. >>> >>> On Jul 8, 2010 8:34 AM, "Tzury Bar Yochay" >>> wrote: >>> >>> Hi, >>> >>> I just download the latest, build and brought up all is fine. >>> However, our clients are connected via IP over GPRS(UMTS) and thus we >>> are using Speex codec. >>> >>> Is there a way I can enforce speex to have 97 or 98 instead of 99? >>> Currently as it looks in the vars.xml speex is 99. >>> see dump (taken from default vars.xml) >>> >>> thanks >>> Tzury >>> RTP Dynamic Payload Numbers currently used in FreeSWITCH and what >>> for. >>> >>> 96 - AMR >>> 97 - iLBC (30) >>> 98 - iLBC (20) >>> 99 - Speex 8kHz, 16kHz, 32kHz >>> 100 - >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Tzury Bar Yochay >> >> tzury.by at reguluslabs.com >> + 972 52 5133399 >> twitter.com/tzury >> > > > > -- > Tzury Bar Yochay > > tzury.by at reguluslabs.com > + 972 52 5133399 > twitter.com/tzury > -- Tzury Bar Yochay tzury.by at reguluslabs.com + 972 52 5133399 twitter.com/tzury -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/fc5efb96/attachment.html From voipnewbie123 at gmail.com Thu Jul 8 07:10:27 2010 From: voipnewbie123 at gmail.com (Voip Newbie) Date: Thu, 8 Jul 2010 19:10:27 +0500 Subject: [Freeswitch-users] Freeswitch - wholesale setup Message-ID: Dear List, I am new to voip and to freeswitch and I am trying to learn how to setup a wholesale scenario using FreeSwitch. I have very little experience in running wholesale voip business on voipswitch. I know all or most of the answers to my questions are already either in wiki or in list archive, and I am really sorry to email the list asking for help but with the little knowledge I have its way over my head and I failed to compile the information in right way to achieve any result, please forgive me if these questions bother any of you at all. Here is what I wish to achieve with FreeSwitch: 1) Setup username/password less accounts for customers with IP authentication. One customer can have multiple IPs. Customer can send traffic using SIP or H323 protocol. A prefix will be assigned to customer for sending traffic. Eg. 1234 + Country code + Area code + Number. 2) Customer can be post paid or per paid, so need to disable customer's ability to call when assigned credit limit is reached. 3) Setup providers (Gateways) which do not provide username/password for authentication and do not require FreeSwitch to register with them, FreeSwitch IP will be allowed to send traffic directly with a 3-4 digit prefix. Provider can be on H323 or SIP either and can have multiple IPs (1 primary and other for fail-over) 4) Customer and providers need access to CDR. So we need to configure Freeswitch the way that it can store CDRs in MySQL database, that database can be accessed by a web application to show CDR on web. *Hardware and OS info:* CentOS 5.4 (Linux 2.6.18-164.15.1.el5PAE on i686) Intel(R) Xeon(R) CPU E5420 @ 2.50GHz, 4 cores 4 GB RAM. 1 GIGABIT NIC with a Public IP address *Progress so far:* 1) FreeSwitch installed with installation procedure at FreeSwitch Wiki ( http://wiki.freeswitch.org/wiki/Installation_Guide#Download_Source_Tarball) 2) Registered and called already created extensions (1001,1002) from x-lite, called echo extensions and everything worked fine. *Questions:* 1) I can see there are 2 H323 mods available for FreeSwitch, which one is better to use in production. a) http://wiki.freeswitch.org/wiki/FreeSwitch_H323 b) http://wiki.freeswitch.org/wiki/Mod_h323 2) How and where (location in freeswitch conf) can I create customers, add IP addresses to authorize without username and password, assign a prefix to the customer? An example would be nice. 3) How and where (location in freeswitch conf) I can create gateways. An example would be nice. 4) Where to create dialplan to route customer calls to provider. An example would be nice. 5) How to manage credit assigned to customers, and how to bill the calls? 6) How to configure Freeswitch so that it can dump CDRs to a mysql database. 7) According to FreeSwitch feature list, it supports g723 and g729 in pass-througe mode, so this means if both customer and providers support g729 & g723, calls will pass? 8) Is there any limit on g729 and g723 calls in pass-through mood? 9) Approx how many concurrent calls can FreeSwitch support in H323 and in SIP based on the hardware info given above. Any pointers, help, links, examples will be highly appreciated. Thanks in advance. -Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/c66b6a72/attachment-0001.html From sanms.zhang at gmail.com Thu Jul 8 08:15:01 2010 From: sanms.zhang at gmail.com (chi zhang) Date: Thu, 8 Jul 2010 23:15:01 +0800 Subject: [Freeswitch-users] Problem of FreeSwitch and T.38 Test In-Reply-To: References: <5ECC3A8A-CDF9-4EF4-AD87-A3FAEDFF26AB@freeswitch.org> <12B00ED2-34A6-414C-A0C6-590A963E3CD2@gmail.com> Message-ID: Hi, David Thanks for your patient...:) In fact, i want to do two things. 1st is: one sip user of FS receives a fax one PSTN user. 2st is: one sip user of FS sends a fax to PSTN user. In my test environment, i can only do this test in Voip system without T38 gateway nor real fax machine. So i can only simulate the fax receive/send process with two softphone. Previously, i do one sip call from zoiper(1000) to 9178( receive fax's dialplan of default.xml ). It can be regarded as the receive fax process of FS. Now, i want to debug the send fax source of FS, such as: user 2000 SEND fax to user 1000, and 1000 receives the fax and saves it. 2010/7/8 David Ponzone > As far as I remember, Zoiper can receive fax. > > For the rest, I think you lost me somewhere... > From the beginning, I was thinking you was trying to receive a fax with > Zoiper, but I see you wonder if it can receive fax. > What I said earlier is just that it's possible Zopier can receive only on > inbound call. > Earlier, you was trying to receive a fax from FS to Zoiper during an > outbound call from Zoiper to FS, which is not the normal way (it does not > happen this way in real life). > > David Ponzone Direction Technique > email: david.ponzone at 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 pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 08/07/2010 ? 11:10, chi zhang a ?crit : > > I just use FS's CLI to call zoiper from FS, it works OK > > CLI cmd : originate sofia/internal/1001 at 192.168.26.39:25060 1001 > > i think if i want to do send fax from FS, i need one softphone can receive > fax, zoiper can do this or not? That is a question... > > > 2010/7/8 David Ponzone > >> Ok, so if I understood correctly, you call a number on FS, and you expect >> to receive a fax on that call. >> Zopier is ok, but it's far from a perfect piece of software. >> Perhaps it is not able to receive a fax when it calls out. >> Could you try to call Zoiper from FS before sending the fax to Zoiper ? >> >> David Ponzone Direction Technique >> email: david.ponzone at 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 pi?ces jointes sont confidentiels et ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autoris?e est interdite. Tout message ?lectronique est susceptible >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur.* >> * >> * >> >> >> >> Le 08/07/2010 ? 10:20, chi zhang a ?crit : >> >> Hi, David >> >> transfer fax function means "FS send fax to a sip user" , previously , >> i use zoiper to send fax to FS, for FS ,this means "receive fax", now i >> want to realize send fax from FS to one sip user. >> >> regards >> sammy >> >> 2010/7/8 David Ponzone >> >>> Chi, >>> >>> I am not sure to understand what you mean by "transfer fax function". >>> Can you clear that up please ? >>> >>> Thanks >>> >>> David Ponzone Direction Technique >>> email: david.ponzone at 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 pi?ces jointes sont confidentiels et ?tablis ? >>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>> non autoris?e est interdite. Tout message ?lectronique est susceptible >>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >>> l'exp?diteur.* >>> * >>> * >>> >>> >>> >>> Le 08/07/2010 ? 09:57, chi zhang a ?crit : >>> >>> Now, i have finished T.38 fax receive test with Zoiper, it works >>> well. But, transfer a fax is still Not successful. >>> Previous, i do simulate fax with sipP, but FS return 48(Disconnected >>> after permitted retries). >>> Accidentally, i found softphone: Zoiper has fax function, so retry >>> fax with it, and in diaplan file: default.xml, 9178 was the receive fax >>> number. So i call 9178 with Zoiper(register as 1000), fax receiving is >>> perfect done. >>> Then i test transfer fax function: call to 9179(configured by TX fax >>> in default.xml), but FS return 2 (Timed out waiting for initial >>> communication) , i have no idea about it, reason ? >>> >>> regards >>> sammy >>> >>> >>> >>> 2010/7/2 chi zhang >>> >>>> I got it. >>>> >>>> -------------------log start-------------------------- >>>> recv 576 bytes from udp/[192.168.26.39]:15060 at 01:41:26.500127: >>>> >>>> ------------------------------------------------------------------------ >>>> INVITE sip:666666 at 192.168.26.39:25060 SIP/2.0 >>>> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >>>> From: 1000 ;tag=1 >>>> To: 666666 >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: sip:1000 at 192.168.26.39:15060 >>>> Max-Forwards: 70 >>>> Subject: Performance Test >>>> Content-Type: application/sdp >>>> Content-Length: 184 >>>> >>>> v=0 >>>> o=user1 3748 3748 IN IP4 192.168.26.39 >>>> s=- >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=audio 6000 RTP/AVP 8 101 >>>> a=rtpmap:8 PCMA/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-11,16 >>>> >>>> ------------------------------------------------------------------------ >>>> send 294 bytes to udp/[192.168.26.39]:15060 at 01:41:26.500477: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 100 Trying >>>> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >>>> From: 1000 ;tag=1 >>>> To: 666666 >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Content-Length: 0 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> 2010-07-02 09:41:26.498946 [DEBUG] sofia.c:5928 IP 192.168.26.39 >>>> Approved by acl "192.168.26.0/24[] ". >>>> Access Granted. >>>> [36m2010-07-02 09:41:26.498946 [NOTICE] switch_channel.c:776 New >>>> Channel sofia/internal/1000 at 192.168.26.39:15060[f588c66c-48c8-4220-a944-8de287adb3ab] >>>> 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_NEW >>>> 2010-07-02 09:41:26.500953 [DEBUG] switch_core_state_machine.c:320 >>>> (sofia/internal/1000 at 192.168.26.39:15060) State NEW >>>> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4297 Channel sofia/internal/ >>>> 1000 at 192.168.26.39:15060 entering state [received][100] >>>> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4308 Remote SDP: >>>> v=0 >>>> >>>> o=user1 3748 3748 IN IP4 192.168.26.39 >>>> >>>> s=- >>>> >>>> c=IN IP4 192.168.26.39 >>>> >>>> t=0 0 >>>> >>>> m=audio 6000 RTP/AVP 8 101 >>>> >>>> a=rtpmap:8 PCMA/8000 >>>> >>>> a=rtpmap:101 telephone-event/8000 >>>> >>>> a=fmtp:101 0-11,16 >>>> >>>> >>>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >>>> [PCMA:8:8000:20]/[G7221:115:32000:20] >>>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >>>> [PCMA:8:8000:20]/[G7221:107:16000:20] >>>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >>>> [PCMA:8:8000:20]/[G722:9:8000:20] >>>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >>>> [PCMA:8:8000:20]/[PCMU:0:8000:20] >>>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3859 Audio Codec Compare >>>> [PCMA:8:8000:20]/[PCMA:8:8000:20] >>>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:2442 Set Codec >>>> sofia/internal/1000 at 192.168.26.39:15060 PCMA/8000 20 ms 160 samples >>>> 2010-07-02 09:41:26.542951 [DEBUG] sofia_glue.c:3798 Set 2833 dtmf >>>> send/recv payload to 101 >>>> 2010-07-02 09:41:26.542951 [DEBUG] sofia.c:4455 (sofia/internal/ >>>> 1000 at 192.168.26.39:15060) State Change CS_NEW -> CS_INIT >>>> 2010-07-02 09:41:26.542951 [DEBUG] switch_core_session.c:1027 Send >>>> signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >>>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change CS_INIT >>>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 >>>> (sofia/internal/1000 at 192.168.26.39:15060) State INIT >>>> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:83 sofia/internal/ >>>> 1000 at 192.168.26.39:15060 SOFIA INIT >>>> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:119 (sofia/internal/ >>>> 1000 at 192.168.26.39:15060) State Change CS_INIT -> CS_ROUTING >>>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send >>>> signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >>>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:338 >>>> (sofia/internal/1000 at 192.168.26.39:15060) State INIT going to sleep >>>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change >>>> CS_ROUTING >>>> 2010-07-02 09:41:26.544953 [DEBUG] switch_channel.c:1471 >>>> (sofia/internal/1000 at 192.168.26.39:15060) Callstate Change DOWN -> >>>> RINGING >>>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 >>>> (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING >>>> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:142 sofia/internal/ >>>> 1000 at 192.168.26.39:15060 SOFIA ROUTING >>>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:77 >>>> sofia/internal/1000 at 192.168.26.39:15060 Standard ROUTING >>>> [32m2010-07-02 09:41:26.544953 [INFO] mod_dialplan_xml.c:331 Processing >>>> 1000->666666 in context public >>>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->fax] >>>> continue=false >>>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] >>>> destination_number(666666) =~ /^fax$/ break=on-false >>>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing >>>> [public->4444] continue=false >>>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [4444] >>>> destination_number(666666) =~ /^(4444)$/ break=on-false >>>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing [public->fax] >>>> continue=false >>>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (FAIL) [fax] >>>> destination_number(666666) =~ /^fax$/ break=on-false >>>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 parsing >>>> [public->666666] continue=false >>>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Regex (PASS) [666666] >>>> destination_number(666666) =~ /^(666666)$/ break=on-false >>>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action answer() >>>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action >>>> playback(silence_stream://2000) >>>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action >>>> rxfax(/tmp/999.tiff) >>>> Dialplan: sofia/internal/1000 at 192.168.26.39:15060 Action hangup() >>>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:119 >>>> (sofia/internal/1000 at 192.168.26.39:15060) State Change CS_ROUTING -> >>>> CS_EXECUTE >>>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_session.c:1027 Send >>>> signal sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >>>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:341 >>>> (sofia/internal/1000 at 192.168.26.39:15060) State ROUTING going to sleep >>>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/1000 at 192.168.26.39:15060) Running State Change >>>> CS_EXECUTE >>>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:348 >>>> (sofia/internal/1000 at 192.168.26.39:15060) State EXECUTE >>>> 2010-07-02 09:41:26.544953 [DEBUG] mod_sofia.c:235 sofia/internal/ >>>> 1000 at 192.168.26.39:15060 SOFIA EXECUTE >>>> 2010-07-02 09:41:26.544953 [DEBUG] switch_core_state_machine.c:157 >>>> sofia/internal/1000 at 192.168.26.39:15060 Standard EXECUTE >>>> EXECUTE sofia/internal/1000 at 192.168.26.39:15060 answer() >>>> 2010-07-02 09:41:26.580967 [DEBUG] sofia_glue.c:2682 AUDIO RTP >>>> [sofia/internal/1000 at 192.168.26.39:15060] 192.168.26.39 port 22464 -> >>>> 192.168.26.39 port 6000 codec: 8 ms: 20 >>>> 2010-07-02 09:41:26.580967 [DEBUG] switch_rtp.c:1413 Starting timer >>>> [soft] 160 bytes per 20ms >>>> 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2892 Set 2833 dtmf send >>>> payload to 101 >>>> 2010-07-02 09:41:26.582952 [DEBUG] sofia_glue.c:2897 Set 2833 dtmf >>>> receive payload to 101 >>>> 2010-07-02 09:41:26.582952 [DEBUG] mod_sofia.c:669 Local SDP >>>> sofia/internal/1000 at 192.168.26.39:15060: >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=audio 22464 RTP/AVP 8 101 >>>> a=rtpmap:8 PCMA/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=silenceSupp:off - - - - >>>> a=ptime:20 >>>> a=sendrecv >>>> >>>> send 1091 bytes to udp/[192.168.26.39]:15060 at 01:41:26.584093: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-0 >>>> From: 1000 ;tag=1 >>>> To: 666666 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, >>>> sla, include-session-description, presence.winfo, message-summary, refer >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 249 >>>> Remote-Party-ID: "666666" >>>> >;party=calling;privacy=off;screen=no >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012423 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=audio 22464 RTP/AVP 8 101 >>>> a=rtpmap:8 PCMA/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=silenceSupp:off - - - - >>>> a=ptime:20 >>>> >>>> ------------------------------------------------------------------------ >>>> recv 369 bytes from udp/[192.168.26.39]:15060 at 01:41:26.584205: >>>> >>>> ------------------------------------------------------------------------ >>>> ACK sip:666666 at 192.168.26.39:25060 SIP/2.0 >>>> Via: SIP/2.0/UDP 192.168.26.39:15060;branch=z9hG4bK-16005-1-4 >>>> From: 1000 ;tag=1 >>>> To: 666666 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 ACK >>>> Contact: sip:1000 at 192.168.26.39:15060 >>>> Max-Forwards: 70 >>>> Subject: Performance Test >>>> Content-Length: 0 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/ >>>> 1000 at 192.168.26.39:15060 entering state [completed][200] >>>> 2010-07-02 09:41:26.582952 [DEBUG] sofia.c:4297 Channel sofia/internal/ >>>> 1000 at 192.168.26.39:15060 entering state [ready][200] >>>> 2010-07-02 09:41:26.582952 [DEBUG] switch_core_session.c:647 Send signal >>>> sofia/internal/1000 at 192.168.26.39:15060 [BREAK] >>>> 2010-07-02 09:41:26.582952 [DEBUG] switch_channel.c:2494 >>>> (sofia/internal/1000 at 192.168.26.39:15060) Callstate Change RINGING -> >>>> ACTIVE >>>> [36m2010-07-02 09:41:26.582952 [NOTICE] mod_dptools.c:746 Channel >>>> [sofia/internal/1000 at 192.168.26.39:15060] has been answered >>>> EXECUTE sofia/internal/1000 at 192.168.26.39:15060playback(silence_stream://2000) >>>> 2010-07-02 09:41:26.584953 [DEBUG] switch_ivr_play_say.c:1161 Codec >>>> Activated L16 at 8000hz 1 channels 20ms >>>> 2010-07-02 09:41:28.578954 [DEBUG] switch_ivr_play_say.c:1468 done >>>> playing file >>>> EXECUTE sofia/internal/1000 at 192.168.26.39:15060 rxfax(/tmp/999.tiff) >>>> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp.c:64 This is for fax test: >>>> receive fax >>>> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:445 trans mode = 1 >>>> 2010-07-02 09:41:28.578954 [ERR] mod_spandsp_fax.c:591 This is for fax >>>> test: prag go to here!!! >>>> 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1064 Raw read codec >>>> activation Success L16 20000 >>>> 2010-07-02 09:41:28.578954 [DEBUG] switch_core_codec.c:122 >>>> sofia/internal/1000 at 192.168.26.39:15060 Push codec L16:10 >>>> 2010-07-02 09:41:28.578954 [DEBUG] mod_spandsp_fax.c:1080 Raw write >>>> codec activation Success L16 >>>> 2010-07-02 09:41:28.857958 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = >>>> 0 >>>> >>>> send 956 bytes to udp/[192.168.21.76]:5060 at 01:41:29.754477: >>>> >>>> ------------------------------------------------------------------------ >>>> NOTIFY sip:1001 at 192.168.21.76:5060 SIP/2.0 >>>> Via: SIP/2.0/UDP 192.168.26.39:25060 >>>> ;rport;branch=z9hG4bKS989SFgmXvmNF >>>> Max-Forwards: 70 >>>> From: "1001" >>> >;tag=vr6p3K1XK247r >>>> To: "1001" ;tag=26647676 >>>> Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. >>>> CSeq: 132897310 NOTIFY >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Event: message-summary >>>> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, >>>> sla, include-session-description, presence.winfo, message-summary, refer >>>> Subscription-State: terminated;reason=timeout >>>> Content-Type: application/simple-message-summary >>>> Content-Length: 65 >>>> >>>> Messages-Waiting: no >>>> Message-Account: sip:1001 at 192.168.26.39 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> 2010-07-02 09:41:29.757979 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = >>>> 0 >>>> recv 389 bytes from udp/[192.168.21.76]:5060 at 01:41:29.758362: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 481 Call/Transaction Does Not Exist >>>> Via: SIP/2.0/UDP 192.168.26.39:25060 >>>> ;rport=25060;branch=z9hG4bKS989SFgmXvmNF >>>> To: "1001";tag=26647676 >>>> From: "1001">>> >;tag=vr6p3K1XK247r >>>> Call-ID: ZTljY2QxZGQ2NGEwOWQwNzRmMDMzYWM2ZDA4NzczNTY. >>>> CSeq: 132897310 NOTIFY >>>> Accept-Language: en >>>> Content-Length: 0 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> 2010-07-02 09:41:36.557090 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = >>>> 0 >>>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:36.587253: >>>> >>>> ------------------------------------------------------------------------ >>>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: sip:666666 at 192.168.26.39:25060 >>>> Max-Forwards: 70 >>>> Subject: Performance Test >>>> Content-Type: application/sdp >>>> Content-Length: 345 >>>> >>>> v=0 >>>> o=root 0 0 IN IP4 192.168.26.39 >>>> s=Session SDP >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 49172 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:9600 >>>> a=T38FaxFillBitRemoval:0 >>>> a=T38FaxTranscodingMMR:0 >>>> a=T38FaxTranscodingJBIG:0 >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:200 >>>> a=T38FaxMaxDatagram:72 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> >>>> ------------------------------------------------------------------------ >>>> send 312 bytes to udp/[192.168.26.39]:25060 at 01:41:36.587546: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 100 Trying >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Content-Length: 0 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> recv 312 bytes from udp/[192.168.26.39]:25060 at 01:41:36.587641: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 100 Trying >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4297 Channel >>>> sofia/internal/1000 at 192.168.26.39:15060 entering state [received][100] >>>> >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Content-Length: 0 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> 2010-07-02 09:41:36.587092 [DEBUG] sofia.c:4308 Remote SDP: >>>> v=0 >>>> >>>> o=root 0 0 IN IP4 192.168.26.39 >>>> >>>> s=Session SDP >>>> >>>> c=IN IP4 192.168.26.39 >>>> >>>> t=0 0 >>>> >>>> m=image 49172 udptl t38 >>>> >>>> a=T38FaxVersion:0 >>>> >>>> a=T38MaxBitRate:9600 >>>> >>>> a=T38FaxFillBitRemoval:0 >>>> >>>> a=T38FaxTranscodingMMR:0 >>>> >>>> a=T38FaxTranscodingJBIG:0 >>>> >>>> a=T38FaxRateManagement:transferredTCF >>>> >>>> a=T38FaxMaxBuffer:200 >>>> >>>> a=T38FaxMaxDatagram:72 >>>> >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> >>>> >>>> 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = >>>> 0 >>>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:744 T38FaxVersion = >>>> 0 >>>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:745 T38MaxBitRate = >>>> 9600 >>>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:746 >>>> T38FaxFillBitRemoval = 1 >>>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:747 >>>> T38FaxTranscodingMMR = 1 >>>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:748 >>>> T38FaxTranscodingJBIG = 1 >>>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:749 >>>> T38FaxRateManagement = 'transferredTCF' >>>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:750 T38FaxMaxBuffer >>>> = 200 >>>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:751 >>>> T38FaxMaxDatagram = 72 >>>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:752 T38FaxUdpEC = >>>> 't38UDPRedundancy' >>>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:753 T38VendorInfo = >>>> '' >>>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:754 ip = >>>> '192.168.26.39' >>>> 2010-07-02 09:41:36.597094 [DEBUG] mod_spandsp_fax.c:756 port = 49172 >>>> 2010-07-02 09:41:36.597094 [DEBUG] mod_sofia.c:1232 IMAGE UDPTL CHANGING >>>> DEST TO: [192.168.26.39:49172] >>>> 2010-07-02 09:41:36.597094 [DEBUG] sofia_glue.c:122 sofia/internal/ >>>> 1000 at 192.168.26.39:15060 image media sdp: >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> 2010-07-02 09:41:36.597094 [ERR] mod_spandsp_fax.c:445 trans mode = 0 >>>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:36.597525: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> 2010-07-02 09:41:36.597094 [DEBUG] sofia.c:4297 Channel sofia/internal/ >>>> 1000 at 192.168.26.39:15060 entering state [completed][200] >>>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:36.597657: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> 2010-07-02 09:41:36.607095 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = >>>> 1 >>>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:37.089156: >>>> >>>> ------------------------------------------------------------------------ >>>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: sip:666666 at 192.168.26.39:25060 >>>> Max-Forwards: 70 >>>> Subject: Performance Test >>>> Content-Type: application/sdp >>>> Content-Length: 345 >>>> >>>> v=0 >>>> o=root 0 0 IN IP4 192.168.26.39 >>>> s=Session SDP >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 49172 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:9600 >>>> a=T38FaxFillBitRemoval:0 >>>> a=T38FaxTranscodingMMR:0 >>>> a=T38FaxTranscodingJBIG:0 >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:200 >>>> a=T38FaxMaxDatagram:72 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> >>>> ------------------------------------------------------------------------ >>>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.089302: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.089395: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:37.098259: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:37.098339: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:38.091157: >>>> >>>> ------------------------------------------------------------------------ >>>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: sip:666666 at 192.168.26.39:25060 >>>> Max-Forwards: 70 >>>> Subject: Performance Test >>>> Content-Type: application/sdp >>>> Content-Length: 345 >>>> >>>> v=0 >>>> o=root 0 0 IN IP4 192.168.26.39 >>>> s=Session SDP >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 49172 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:9600 >>>> a=T38FaxFillBitRemoval:0 >>>> a=T38FaxTranscodingMMR:0 >>>> a=T38FaxTranscodingJBIG:0 >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:200 >>>> a=T38FaxMaxDatagram:72 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> >>>> ------------------------------------------------------------------------ >>>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.091305: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.091406: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:38.098260: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:38.098344: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> This is for fax test: dis 5This is for fax test: cause disconnect 4 >>>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:40.093267: >>>> >>>> ------------------------------------------------------------------------ >>>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: sip:666666 at 192.168.26.39:25060 >>>> Max-Forwards: 70 >>>> Subject: Performance Test >>>> Content-Type: application/sdp >>>> Content-Length: 345 >>>> >>>> v=0 >>>> o=root 0 0 IN IP4 192.168.26.39 >>>> s=Session SDP >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 49172 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:9600 >>>> a=T38FaxFillBitRemoval:0 >>>> a=T38FaxTranscodingMMR:0 >>>> a=T38FaxTranscodingJBIG:0 >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:200 >>>> a=T38FaxMaxDatagram:72 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> >>>> ------------------------------------------------------------------------ >>>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.093421: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.093541: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:40.098263: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:40.098418: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> 2010-07-02 09:41:41.606147 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = >>>> 1 >>>> recv 737 bytes from udp/[192.168.26.39]:15060 at 01:41:44.095284: >>>> >>>> ------------------------------------------------------------------------ >>>> INVITE sip:1000 at 192.168.26.39:15060 SIP/2.0 >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: sip:666666 at 192.168.26.39:25060 >>>> Max-Forwards: 70 >>>> Subject: Performance Test >>>> Content-Type: application/sdp >>>> Content-Length: 345 >>>> >>>> v=0 >>>> o=root 0 0 IN IP4 192.168.26.39 >>>> s=Session SDP >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 49172 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:9600 >>>> a=T38FaxFillBitRemoval:0 >>>> a=T38FaxTranscodingMMR:0 >>>> a=T38FaxTranscodingJBIG:0 >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:200 >>>> a=T38FaxMaxDatagram:72 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> >>>> ------------------------------------------------------------------------ >>>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.095457: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.095556: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:44.098263: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> recv 952 bytes from udp/[192.168.26.39]:25060 at 01:41:44.098399: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> ------------------------------------------------------------------------ >>>> 2010-07-02 09:41:46.607257 [ERR] mod_spandsp_fax.c:1119 pvt->t38_mode = >>>> 1 >>>> send 952 bytes to udp/[192.168.26.39]:25060 at 01:41:48.098287: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 200 OK >>>> Via: SIP/2.0/UDP 192.168.26.39:25060;branch=z9hG4bK-16005-1-7 >>>> From: 666666 ;tag=1 >>>> To: 1000 ;tag=85ySHt6rmaSyp >>>> Call-ID: 1-16005 at 192.168.26.39 >>>> CSeq: 2 INVITE >>>> Contact: >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >>>> Supported: timer, precondition, path, replaces >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 341 >>>> >>>> v=0 >>>> o=FreeSWITCH 1278012422 1278012424 IN IP4 192.168.26.39 >>>> s=FreeSWITCH >>>> c=IN IP4 192.168.26.39 >>>> t=0 0 >>>> m=image 22464 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:14400 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38VendorInfo:0 0 0 >>>> >>>> >>>> -----------------------------log end---------------------------------- >>>> >>>> >>>> 2010/7/2 Brian West >>>> >>>> turn sip on >>>>> >>>>> sofia profile xxx siptrace on >>>>> >>>>> /b >>>>> >>>>> On Jul 1, 2010, at 8:23 PM, chi zhang wrote: >>>>> >>>>> > hi,brian >>>>> > >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/48047be8/attachment-0001.html From peder at networkoblivion.com Thu Jul 8 08:22:36 2010 From: peder at networkoblivion.com (Peder) Date: Thu, 8 Jul 2010 10:22:36 -0500 Subject: [Freeswitch-users] Freeswitch - wholesale setup In-Reply-To: References: Message-ID: <07d601cb1eb1$64f05db0$2ed11910$@com> Um, so you want to setup to be a wholesale SIP provider, but everything on the wiki is way too technical for you? Your best bet is to email consulting at freeswitch.org and pay for help. If you ask a specific question you will get answers, but essentially saying "I don't know anything, please set it all up for me" is going to result in no help at all. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Voip Newbie Sent: Thursday, July 08, 2010 9:10 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch - wholesale setup Dear List, I am new to voip and to freeswitch and I am trying to learn how to setup a wholesale scenario using FreeSwitch. I have very little experience in running wholesale voip business on voipswitch. I know all or most of the answers to my questions are already either in wiki or in list archive, and I am really sorry to email the list asking for help but with the little knowledge I have its way over my head and I failed to compile the information in right way to achieve any result, please forgive me if these questions bother any of you at all. Here is what I wish to achieve with FreeSwitch: 1) Setup username/password less accounts for customers with IP authentication. One customer can have multiple IPs. Customer can send traffic using SIP or H323 protocol. A prefix will be assigned to customer for sending traffic. Eg. 1234 + Country code + Area code + Number. 2) Customer can be post paid or per paid, so need to disable customer's ability to call when assigned credit limit is reached. 3) Setup providers (Gateways) which do not provide username/password for authentication and do not require FreeSwitch to register with them, FreeSwitch IP will be allowed to send traffic directly with a 3-4 digit prefix. Provider can be on H323 or SIP either and can have multiple IPs (1 primary and other for fail-over) 4) Customer and providers need access to CDR. So we need to configure Freeswitch the way that it can store CDRs in MySQL database, that database can be accessed by a web application to show CDR on web. Hardware and OS info: CentOS 5.4 (Linux 2.6.18-164.15.1.el5PAE on i686) Intel(R) Xeon(R) CPU E5420 @ 2.50GHz, 4 cores 4 GB RAM. 1 GIGABIT NIC with a Public IP address Progress so far: 1) FreeSwitch installed with installation procedure at FreeSwitch Wiki (http://wiki.freeswitch.org/wiki/Installation_Guide#Download_Source_Tarball) 2) Registered and called already created extensions (1001,1002) from x-lite, called echo extensions and everything worked fine. Questions: 1) I can see there are 2 H323 mods available for FreeSwitch, which one is better to use in production. a) http://wiki.freeswitch.org/wiki/FreeSwitch_H323 b) http://wiki.freeswitch.org/wiki/Mod_h323 2) How and where (location in freeswitch conf) can I create customers, add IP addresses to authorize without username and password, assign a prefix to the customer? An example would be nice. 3) How and where (location in freeswitch conf) I can create gateways. An example would be nice. 4) Where to create dialplan to route customer calls to provider. An example would be nice. 5) How to manage credit assigned to customers, and how to bill the calls? 6) How to configure Freeswitch so that it can dump CDRs to a mysql database. 7) According to FreeSwitch feature list, it supports g723 and g729 in pass-througe mode, so this means if both customer and providers support g729 & g723, calls will pass? 8) Is there any limit on g729 and g723 calls in pass-through mood? 9) Approx how many concurrent calls can FreeSwitch support in H323 and in SIP based on the hardware info given above. Any pointers, help, links, examples will be highly appreciated. Thanks in advance. -Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/a45a9ac0/attachment.html From rupa at rupa.com Thu Jul 8 08:29:56 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 8 Jul 2010 10:29:56 -0500 Subject: [Freeswitch-users] Freeswitch - wholesale setup In-Reply-To: References: Message-ID: This is really an extensive list. Nearly all of this is on the wiki. If that is too much, a email response to pages on the wiki isn't going to be of much help. Perhaps you need a consultant? On Thu, Jul 8, 2010 at 9:10 AM, Voip Newbie wrote: > Dear List, > > I am new to voip and to freeswitch and I am trying to learn how to setup a > wholesale scenario using FreeSwitch. I have very little experience in > running wholesale voip business on voipswitch. > > I know all or most of the answers to my questions are already either in > wiki or in list archive, and I am really sorry to email the list asking for > help but with the little knowledge I have its way over my head and I failed > to compile the information in right way to achieve any result, please > forgive me if these questions bother any of you at all. > > Here is what I wish to achieve with FreeSwitch: > > 1) Setup username/password less accounts for customers with IP > authentication. One customer can have multiple IPs. Customer can send > traffic using SIP or H323 protocol. A prefix will be assigned to customer > for sending traffic. Eg. 1234 + Country code + Area code + Number. > 2) Customer can be post paid or per paid, so need to disable customer's > ability to call when assigned credit limit is reached. > 3) Setup providers (Gateways) which do not provide username/password for > authentication and do not require FreeSwitch to register with them, > FreeSwitch IP will be allowed to send traffic directly with a 3-4 digit > prefix. Provider can be on H323 or SIP either and can have multiple IPs (1 > primary and other for fail-over) > 4) Customer and providers need access to CDR. So we need to configure > Freeswitch the way that it can store CDRs in MySQL database, that database > can be accessed by a web application to show CDR on web. > > *Hardware and OS info:* > CentOS 5.4 (Linux 2.6.18-164.15.1.el5PAE on i686) > Intel(R) Xeon(R) CPU E5420 @ 2.50GHz, 4 cores > 4 GB RAM. > 1 GIGABIT NIC with a Public IP address > > *Progress so far:* > 1) FreeSwitch installed with installation procedure at FreeSwitch Wiki ( > http://wiki.freeswitch.org/wiki/Installation_Guide#Download_Source_Tarball > ) > 2) Registered and called already created extensions (1001,1002) from > x-lite, called echo extensions and everything worked fine. > > *Questions:* > 1) I can see there are 2 H323 mods available for FreeSwitch, which one is > better to use in production. > a) http://wiki.freeswitch.org/wiki/FreeSwitch_H323 > b) http://wiki.freeswitch.org/wiki/Mod_h323 > 2) How and where (location in freeswitch conf) can I create customers, add > IP addresses to authorize without username and password, assign a prefix to > the customer? An example would be nice. > 3) How and where (location in freeswitch conf) I can create gateways. An > example would be nice. > 4) Where to create dialplan to route customer calls to provider. An example > would be nice. > 5) How to manage credit assigned to customers, and how to bill the calls? > 6) How to configure Freeswitch so that it can dump CDRs to a mysql > database. > 7) According to FreeSwitch feature list, it supports g723 and g729 in > pass-througe mode, so this means if both customer and providers support g729 > & g723, calls will pass? > 8) Is there any limit on g729 and g723 calls in pass-through mood? > 9) Approx how many concurrent calls can FreeSwitch support in H323 and in > SIP based on the hardware info given above. > > Any pointers, help, links, examples will be highly appreciated. > > Thanks in advance. > > -Eric > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/3cbebbe0/attachment.html From infos at madovsky.org Thu Jul 8 09:04:02 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 8 Jul 2010 12:04:02 -0400 Subject: [Freeswitch-users] Freeswitch - wholesale setup References: <07d601cb1eb1$64f05db0$2ed11910$@com> Message-ID: <79C486FA6812401A81187C7C40D611E3@MOBILEE1705> dear Eric, you need to learn from WIKI or pay a consultant. Regards Franck ----- Original Message ----- From: Peder To: freeswitch-users at lists.freeswitch.org Sent: Thursday, July 08, 2010 11:22 AM Subject: Re: [Freeswitch-users] Freeswitch - wholesale setup Um, so you want to setup to be a wholesale SIP provider, but everything on the wiki is way too technical for you? Your best bet is to email consulting at freeswitch.org and pay for help. If you ask a specific question you will get answers, but essentially saying "I don't know anything, please set it all up for me" is going to result in no help at all. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Voip Newbie Sent: Thursday, July 08, 2010 9:10 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch - wholesale setup Dear List, I am new to voip and to freeswitch and I am trying to learn how to setup a wholesale scenario using FreeSwitch. I have very little experience in running wholesale voip business on voipswitch. I know all or most of the answers to my questions are already either in wiki or in list archive, and I am really sorry to email the list asking for help but with the little knowledge I have its way over my head and I failed to compile the information in right way to achieve any result, please forgive me if these questions bother any of you at all. Here is what I wish to achieve with FreeSwitch: 1) Setup username/password less accounts for customers with IP authentication. One customer can have multiple IPs. Customer can send traffic using SIP or H323 protocol. A prefix will be assigned to customer for sending traffic. Eg. 1234 + Country code + Area code + Number. 2) Customer can be post paid or per paid, so need to disable customer's ability to call when assigned credit limit is reached. 3) Setup providers (Gateways) which do not provide username/password for authentication and do not require FreeSwitch to register with them, FreeSwitch IP will be allowed to send traffic directly with a 3-4 digit prefix. Provider can be on H323 or SIP either and can have multiple IPs (1 primary and other for fail-over) 4) Customer and providers need access to CDR. So we need to configure Freeswitch the way that it can store CDRs in MySQL database, that database can be accessed by a web application to show CDR on web. Hardware and OS info: CentOS 5.4 (Linux 2.6.18-164.15.1.el5PAE on i686) Intel(R) Xeon(R) CPU E5420 @ 2.50GHz, 4 cores 4 GB RAM. 1 GIGABIT NIC with a Public IP address Progress so far: 1) FreeSwitch installed with installation procedure at FreeSwitch Wiki (http://wiki.freeswitch.org/wiki/Installation_Guide#Download_Source_Tarball) 2) Registered and called already created extensions (1001,1002) from x-lite, called echo extensions and everything worked fine. Questions: 1) I can see there are 2 H323 mods available for FreeSwitch, which one is better to use in production. a) http://wiki.freeswitch.org/wiki/FreeSwitch_H323 b) http://wiki.freeswitch.org/wiki/Mod_h323 2) How and where (location in freeswitch conf) can I create customers, add IP addresses to authorize without username and password, assign a prefix to the customer? An example would be nice. 3) How and where (location in freeswitch conf) I can create gateways. An example would be nice. 4) Where to create dialplan to route customer calls to provider. An example would be nice. 5) How to manage credit assigned to customers, and how to bill the calls? 6) How to configure Freeswitch so that it can dump CDRs to a mysql database. 7) According to FreeSwitch feature list, it supports g723 and g729 in pass-througe mode, so this means if both customer and providers support g729 & g723, calls will pass? 8) Is there any limit on g729 and g723 calls in pass-through mood? 9) Approx how many concurrent calls can FreeSwitch support in H323 and in SIP based on the hardware info given above. Any pointers, help, links, examples will be highly appreciated. Thanks in advance. -Eric ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/405835fa/attachment-0001.html From anthony.minessale at gmail.com Thu Jul 8 09:54:24 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 8 Jul 2010 11:54:24 -0500 Subject: [Freeswitch-users] enforcing RTP payload numbers for codecs In-Reply-To: References: Message-ID: The server is required to send the client on the payload it requested regardless of what the server chooses. RFC 3264: Once the answerer has sent the answer, it MUST be prepared to receive media for any recvonly streams described by that answer. It MUST be prepared to send and receive media for any sendrecv streams in the answer, and it MAY send media immediately. The answerer MUST be prepared to receive media for recvonly or sendrecv streams using any media formats listed for those streams in the answer, and it MAY send media immediately. When sending media, it SHOULD use a packetization interval equal to the value of the ptime attribute in the offer, if any was present. It SHOULD send media using a bandwidth no higher than the value of the bandwidth attribute in the offer, if any was present. The answerer MUST send using a media format in the offer that is also listed in the answer, and SHOULD send using the most preferred media format in the offer that is also listed in the answer. In the case of RTP, it MUST use the payload type numbers from the offer, even if they differ from those in the answer. What is pretty amusing though is from the same RFC: In the case of RTP, if a particular codec was referenced with a specific payload type number in the offer, that same payload type number SHOULD be used for that codec in the answer. Even if the same payload type number is used, the answer MUST contain rtpmap attributes to define the payload type mappings for dynamic payload types, and SHOULD contain mappings for static payload types. The media formats in the "m=" line MUST be listed in order of preference, with the first format listed being preferred. In this case, preferred means that the offerer SHOULD use the format with the highest preference from the answer. So SHOULD means they have no nerve to enforce it because it's ambiguous and we can choose not to do it. The MUST in the first example is what we follow. MAYBE its possible to find a way to prefer the callers pt like we SHOULD, we'll have to think about it. On Thu, Jul 8, 2010 at 9:28 AM, Tzury Bar Yochay wrote: > Brian and Anthony, > Dears, > > I am afraid it is a bug in teh FS > See, if during SDP the client declare on payload 98 and fs declare on 99. > Then client expect receiving 99 while sending 98. That is fine. > However, when both clients are the same (as in our case) since FS is not > trans-coding > clients receives 98 in > given that same sip client cannot work with previous versions of FS > where there Speex was 98 (and 97 in older) > > > On Thu, Jul 8, 2010 at 5:08 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> No, you can't force codecs in the dynamic range and if properly >> implemented, it won't matter what number either side chooses. >> >> On Jul 8, 2010 8:34 AM, "Tzury Bar Yochay" >> wrote: >> >> Hi, >> >> I just download the latest, build and brought up all is fine. >> However, our clients are connected via IP over GPRS(UMTS) and thus we >> are using Speex codec. >> >> Is there a way I can enforce speex to have 97 or 98 instead of 99? >> Currently as it looks in the vars.xml speex is 99. >> see dump (taken from default vars.xml) >> >> thanks >> Tzury >> RTP Dynamic Payload Numbers currently used in FreeSWITCH and what >> for. >> >> 96 - AMR >> 97 - iLBC (30) >> 98 - iLBC (20) >> 99 - Speex 8kHz, 16kHz, 32kHz >> 100 - >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Tzury Bar Yochay > > tzury.by at reguluslabs.com > + 972 52 5133399 > twitter.com/tzury > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/858c2e12/attachment.html From msc at freeswitch.org Thu Jul 8 09:57:46 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 8 Jul 2010 09:57:46 -0700 Subject: [Freeswitch-users] originate call hangup signal In-Reply-To: References: <2B4FFD98-7682-44AB-ADB8-9B9700B5AC32@freeswitch.org> Message-ID: Interesting. I've had more success with the libpri stack than the native stack because the native stack is definitely a work in progress. I assisted Mike J in getting 5ESS support in the native stack, but we definitely don't have explicit 4ESS support. I think you are better off using the libpri method and debugging it than trying to get the native PRI stack to work on a 4ESS connection. Hop on #openzap on irc.freenode.net and ask for some assistance there. We have a few guys who are familiar with PRI and libpri, etc. who might be able to help you figure out what's going on. -MC On Thu, Jul 8, 2010 at 1:01 AM, Tony Tin wrote: > It's Digium TE220. I'm using the OpenZAP native stack, because I can not > get the outbound call work with libpri compatibility stack. > > Attached is the freeswitch.log. I'm not sure whether it includes the d-chan > trace, though I already enabled the "q931_dump". > > I originated a call to my mobile on freeswitch console with command > "originate OpenZAP/2/A/98855404 6899", I answered the call then hung up, > after around 30 seconds, I saw there is terminator event on the console and > the call hangup. > > Thanks > > Regards, > Tony > > > > > On Thu, Jul 8, 2010 at 12:42 PM, Michael S Collins wrote: > >> Okay, next question: which PRI are you using? Is it Digium-based or >> Sangoma hardware? If the former then use the libpri method; the latter use >> freetdm. I think they're both covered on the wiki. You need to get an ISDN >> trace on the d-chan to see what is actually being sent to/from telco. >> >> -MC >> >> Sent from my iPhone >> >> On Jul 7, 2010, at 7:57 PM, Tony Tin wrote: >> >> Thanks for your help. >> >> It's a 4ESS IDSN and the carrier does provide disconnect supervision, is >> there any way to bypass it ? >> >> Regards, >> Tony >> >> >> On Thu, Jul 8, 2010 at 7:14 AM, Michael Collins < >> msc at freeswitch.org> wrote: >> >>> It depends on where the "hangup signal" comes from. Is this an analog >>> line? If so, does the carrier provide disconnect supervision? It's entirely >>> possible that the other end isn't doing a good job of telling FreeSWITCH >>> that the call is over. >>> >>> -MC >>> >>> On Tue, Jul 6, 2010 at 10:35 PM, Tony Tin < >>> tony.tin at noahmedia.com.hk> wrote: >>> >>>> Hi, >>>> >>>> When I use OpenZAP channel to originate a call, after the called party >>>> hangup the phone. It takes freeswitch around 40 seconds to catch the hangup >>>> signal and stop the dial plan. I'm wondering whether there is a way to >>>> shorten that duration. Thanks. >>>> >>>> Regards, >>>> Tony >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> >>>> FreeSWITCH-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/3d6da77f/attachment-0001.html From tzury.by at reguluslabs.com Thu Jul 8 10:59:16 2010 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Thu, 8 Jul 2010 20:59:16 +0300 Subject: [Freeswitch-users] enforcing RTP payload numbers for codecs In-Reply-To: References: Message-ID: Anthony, This is what happened according to our log and wireshark analysis. I will read through the RFC again see which side is taking it wrong (FS or PJSIP). Either way, thank you and all others for time and energy you have spent answering this and other posts on the mailing list, let alone FreeSWITCH development and maintenance. Tzury Client caller say in the SDP speex uses PT=98. FS answer in SDP speex will use PT=98, and then the ring tone is played with PT=98 and the caller hears the ringback well. FS invite destination with SDP speex PT=99 (value defined in the code). Client answers SDP speex uses PT=98. Answer client sends speex with PT=99 (according the FS request) and caller send speex PT=98 (according to FS request). Answer waits to get speex with PT=98, and caller waits to get speex with PT=98 (the same it was with the ringtone). Answer side hears caller, but caller doesn't hear client. If FS would have declare PT=(the value requested from the caller), the answer side would have send the PT equal to what was agreed with the FS first leg of the call. On Thu, Jul 8, 2010 at 7:54 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > The server is required to send the client on the payload it requested > regardless of what the server chooses. > > RFC 3264: > Once the answerer has sent the answer, it MUST be prepared to receive > media for any recvonly streams described by that answer. It MUST be > prepared to send and receive media for any sendrecv streams in the > answer, and it MAY send media immediately. The answerer MUST be > prepared to receive media for recvonly or sendrecv streams using any > media formats listed for those streams in the answer, and it MAY send > media immediately. When sending media, it SHOULD use a packetization > interval equal to the value of the ptime attribute in the offer, if > any was present. It SHOULD send media using a bandwidth no higher > than the value of the bandwidth attribute in the offer, if any was > present. The answerer MUST send using a media format in the offer > that is also listed in the answer, and SHOULD send using the most > preferred media format in the offer that is also listed in the answer. > In the case of RTP, it MUST use the payload type numbers > from the offer, even if they differ from those in the answer. > > > What is pretty amusing though is from the same RFC: > > > In the case of RTP, if a particular codec was referenced with a > specific payload type number in the offer, that same payload type > number SHOULD be used for that codec in the answer. Even if the same > payload type number is used, the answer MUST contain rtpmap > attributes to define the payload type mappings for dynamic payload > types, and SHOULD contain mappings for static payload types. The > media formats in the "m=" line MUST be listed in order of preference, > with the first format listed being preferred. In this case, > preferred means that the offerer SHOULD use the format with the > highest preference from the answer. > > > So SHOULD means they have no nerve to enforce it because it's ambiguous and > we can choose not to do it. > The MUST in the first example is what we follow. > > MAYBE its possible to find a way to prefer the callers pt like we SHOULD, > we'll have to think about it. > > > > > > > On Thu, Jul 8, 2010 at 9:28 AM, Tzury Bar Yochay > wrote: > >> Brian and Anthony, >> Dears, >> >> I am afraid it is a bug in teh FS >> See, if during SDP the client declare on payload 98 and fs declare on 99. >> Then client expect receiving 99 while sending 98. That is fine. >> However, when both clients are the same (as in our case) since FS is not >> trans-coding >> clients receives 98 in >> given that same sip client cannot work with previous versions of FS >> where there Speex was 98 (and 97 in older) >> >> >> On Thu, Jul 8, 2010 at 5:08 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> No, you can't force codecs in the dynamic range and if properly >>> implemented, it won't matter what number either side chooses. >>> >>> On Jul 8, 2010 8:34 AM, "Tzury Bar Yochay" >>> wrote: >>> >>> Hi, >>> >>> I just download the latest, build and brought up all is fine. >>> However, our clients are connected via IP over GPRS(UMTS) and thus we >>> are using Speex codec. >>> >>> Is there a way I can enforce speex to have 97 or 98 instead of 99? >>> Currently as it looks in the vars.xml speex is 99. >>> see dump (taken from default vars.xml) >>> >>> thanks >>> Tzury >>> RTP Dynamic Payload Numbers currently used in FreeSWITCH and what >>> for. >>> >>> 96 - AMR >>> 97 - iLBC (30) >>> 98 - iLBC (20) >>> 99 - Speex 8kHz, 16kHz, 32kHz >>> 100 - >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Tzury Bar Yochay >> >> tzury.by at reguluslabs.com >> + 972 52 5133399 >> twitter.com/tzury >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Tzury Bar Yochay tzury.by at reguluslabs.com + 972 52 5133399 twitter.com/tzury -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/094401bc/attachment.html From luixsansan at hotmail.com Thu Jul 8 12:03:18 2010 From: luixsansan at hotmail.com (luixsansan at hotmail.com) Date: Thu, 8 Jul 2010 21:03:18 +0200 Subject: [Freeswitch-users] dtmf in Skype Message-ID: Hello, I have FS ver. 1.0.6, Windows Vista Ultimate and Skype 4.2.0.169 for Windows. (user1) On a laptop I have Windows Vista Home and Skype 4.1.0.179 (user2) When I make a call from user2 to user1 I listen the IVR menu and see the activity on FS console. The problem is that when I click on the skype's phone keyboard I can not listen to any DTMF signal and user1 is not receiving DTMF signals ( the IVR menu is not interrupted). Is there a configuration change I have to do in order to make skype send/receive dtmf signals or do I need another type of skype client? The info in the document for mod_skypopen is not very clear. Thank you. Luis. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/8253ae8b/attachment-0001.html From gmaruzz at celliax.org Thu Jul 8 12:47:46 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 8 Jul 2010 21:47:46 +0200 Subject: [Freeswitch-users] dtmf in Skype In-Reply-To: References: Message-ID: that is not true ;) Is very clear, indeed (I wrote it ;) ) look for dtmf in all the wiki page, you'll find some parameter to be used. If it's still unclear, please rewrite here, and describe fully what you'r doing and what is the problem -giovanni On Thu, Jul 8, 2010 at 9:03 PM, wrote: > Hello, > > I have FS ver. 1.0.6, Windows Vista Ultimate?and Skype 4.2.0.169 for > Windows. (user1) > > On a laptop I have Windows Vista Home and Skype 4.1.0.179 (user2) > > When I make a call from user2 to user1 I listen the IVR?menu and see the > activity on FS console. The problem is that when I click on the skype's > phone keyboard?I can not listen to any DTMF?signal and user1 is not > receiving DTMF signals ( the IVR menu is not interrupted). > > Is there a configuration change I have to do in order to make skype > send/receive dtmf signals or do I need another type of skype client? > > The info in the document for mod_skypopen is not very clear. > > > Thank you. > > Luis. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From paul.gore.j at gmail.com Thu Jul 8 16:31:22 2010 From: paul.gore.j at gmail.com (paul gore) Date: Thu, 8 Jul 2010 19:31:22 -0400 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: <201007071616.41535.sos@sokhapkin.dyndns.org> Message-ID: I got ngrep trace for port 5060 while making a call to a US number via siptraffic.com from X-Lite phone. I canceled the call after ~ 20 sec., I heard no audio not even ringing. Is there anything in this trace which can help identify the problem? 10.194.206.102:5060 - is my local EC2 IP 184.72.206.204:5060 - is my public EC2 IP 77.72.169.128:5060 - siptraffic.com proxy IP Thanks! 67.33.160.119:18294 -> 10.194.206.102:5060 INVITE sip:45517705678570 at myserver.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f 524431af92cef56-1--d87543-;rport..Max-Forwards: 70..Contact: < sip:4000002 at 67.33.160.119:18027>..To: "45517 709248570">..From: "4000002" >;tag=5f1ec15f..Call- ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE , REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest user name="4000002",realm="myserver.com ",nonce="cf9019cc-f44a-4568-97d1-e9883fb1821f",uri="sip:45517705678570 at v ersafon.com ",response="57da72527524e0e065c9a3221bfadd38",cnonce="140f655ff3427f6ba3767ab7040231f3",nc=0000 0001,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp 41150..Content-Length: 417....v=0..o=- 8 2 IN IP4 192.168.0.8..s=CounterPath X-Lite 3.0..c=IN IP4 192.168.0.8..t=0 0..m=audio 46298 RTP/AVP 107 119 100 106 0 105 98 8 101..a=alt:1 1 : tomYv1/D Yont/s+3 192.168.0.8 46298..a=fmtp:101 0-15..a=rtpmap:107 BV32/16000..a=rtpmap:119 BV32-FEC/16000..a=rtpmap:100 SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap :105 SPEEX-FEC/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv.. # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;r port=18294;received=67.33.160.119..From: "4000002" < sip:4000002 at myserver.com >;tag=5f1ec15f..To: "455177092 48570" >..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 I NVITE..User-Agent: myserver..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 INVITE sip:0017705678570 at sip.siptraffic.comSIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9h G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793 at 184.72.206.204 >;tag=18853e82KDe7j. .To: >..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 2 INVITE..Contact: ..User-A gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.. Supported: timer, precondition, path, replaces..Allow-Events: talk, refer..Content-Type: application/sdp.. Content-Disposition: session..Content-Length: 295..X-FS-Support: update_display..Remote-Party-ID: "4000002 " >;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1278518039 1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8 3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/80 00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20.. # U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..From : "4000002" >;tag=18853e82KDe7j..To: m>;tag=20113ac4c230cd6412168..Contact: sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,C ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: application/sdp..Content-Length: 198....v=0..o=C ARRIER 1278549617 1278549617 IN IP4 77.72.168.40..s=SIP Call..c=IN IP4 77.72.168.40..t=0 0..m=audio 57672 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=ptime:20.. # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1- -d87543-;rport=18294;received=67.33.160.119..From: "4000002" < sip:4000002 at myserver.com >;tag=5f1ec15f..To: "45517705678570" >;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjk zMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Contact: ..User-A gent: myserver..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message- summary, refer..Content-Type: application/sdp..Content-Disposition: session..Content-Length: 251..Remote-P arty-ID: "45517705678570" >;party=calling;privacy=off;screen=no....v=0.. o=FreeSWITCH 1278530815 1278530816 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m= audio 18788 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=sil enceSupp:off - - - -..a=ptime:20.. # U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..From : "4000002" >;tag=18853e82KDe7j..To: m>;tag=20113ac4c230cd6412168..Contact: sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,C ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: application/sdp..Content-Length: 204....v=0..o=C ARRIER 1278549619 1278549619 IN IP4 208.167.230.118..s=SIP Call..c=IN IP4 208.167.230.118..t=0 0..m=audio 57786 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=ptime:20.. # U 67.33.160.119:18294 -> 10.194.206.102:5060 .... # U 67.33.160.119:18294 -> 10.194.206.102:5060 CANCEL sip:45517705678570 at myserver.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f 524431af92cef56-1--d87543-;rport..To: "45517705678570"< sip:45517705678570 at myserver.com >..From: "4000002">;tag=5f1ec15f..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 CANC EL..Proxy-Authorization: Digest username="4000002",realm="myserver.com ",nonce="cf9019cc-f44a-4568-97d1-e98 83fb1821f",uri="sip:45517705678570 at myserver.com ",response="46c7e289f7490c807565c561699b03d6",cnonce="a226c 55446c605ee229f045602b29135",nc=00000002,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp 41 150..Content-Length: 0.... # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;rport =18294;received=67.33.160.119..From: "4000002" >;tag=5f1ec15f..To: "4551770567857 0" >;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkY mY...CSeq: 2 CANCEL..Content-Length: 0.... # U 10.194.206.102:5060 -> 67.33.160.119:18294 SIP/2.0 487 Request Terminated..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524431af92cef56- 1--d87543-;rport=18294;received=67.33.160.119..From: "4000002" < sip:4000002 at myserver.com >;tag=5f1ec15f..To : "45517705678570" >;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmN jkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..User-Agent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSA GE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, repla ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presen ce.winfo, message-summary, refer..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 CANCEL sip:0017705678570 at sip.siptraffic.comSIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9h G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793 at 184.72.206.204 >;tag=18853e82KDe7j. .To: >..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 2 CANCEL..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length: 0.... # U 67.33.160.119:18294 -> 10.194.206.102:5060 ACK sip:45517705678570 at myserver.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 ;branch=z9hG4bK-d87543-f524 431af92cef56-1--d87543-;rport..To: "45517705678570" < sip:45517705678570 at myserver.com >;tag=BXgB1FZBUZ3Da..F rom: "4000002">;tag=5f1ec15f..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYm Y...CSeq: 2 ACK..Content-Length: 0.... # U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From: "4000002" >;tag=18853e82KDe7j..To: >..Contact: s ip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 CANCEL ..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSA GE..Content-Length: 0.... # U 77.72.169.128:5060 -> 10.194.206.102:5080 SIP/2.0 487 Request terminated..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4bKBU626KBp16t5Q..Fr om: "4000002" >;tag=18853e82KDe7j..To: ..Contact: sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OP TIONS,INFO,MESSAGE..Content-Length: 0.... # U 10.194.206.102:5080 -> 77.72.169.128:5060 ACK sip:0017705678570 at sip.siptraffic.comSIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9hG4b KBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793 at 184.72.206.204 >;tag=18853e82KDe7j..To : >..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 A CK..Content-Length: 0.... # On Wed, Jul 7, 2010 at 5:50 PM, paul gore wrote: > Seems like siptraffic uses 6 ip addresses for media, can that be the > problem? Is there any setting in a gateway config xml which helps with > that? > I will do ngrep thing and update. > > On 7/7/10, paul gore wrote: > > This provider does work on another box which is not natted as ec2. > > Most puzzling here though is why call originaion via api even not > > going via siptraffic still gets no audio. > > > > On 7/7/10, Tony Graziano wrote: > >> You should try from a standalone or local installation to ensure it > works > >> with this provider and your account before you attempt to run it on ec2 > >> (imo). > >> > >> On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin > >> wrote: > >> > >>> What "doesn't work" means? It could be (and most likely is not) > >>> FS-related > >>> problem > >>> > >>> On Wednesday 07 July 2010, Madovsky wrote: > >>> > I had same problem from this provider without to explain why. > >>> > One day it works, another day it doesn't, their support is crap... > >>> > > >>> > ----- Original Message ----- > >>> > From: Anthony Minessale > >>> > To: freeswitch-users at lists.freeswitch.org > >>> > Sent: Wednesday, July 07, 2010 2:37 PM > >>> > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on > >>> > outgoing > >>> > calls > >>> > > >>> > > >>> > not really, not with so little information. > >>> > > >>> > > >>> > > >>> > On Wed, Jul 7, 2010 at 1:30 PM, paul gore > >>> wrote: > >>> > > >>> > Firewall is configured according to the wiki, I also tried to > open > >>> all > >>> > udp ports, issue persists. > >>> > Actually the problem became more complex - outgoing calls don't > >>> > work > >>> > with one particular termination provider, siptraffic.com , any > >>> > ideas > >>> > why? > >>> > Outgoing calls also don't work when originating a call via js > >>> > script > >>> > or via FS api. Any clues on that one? > >>> > > >>> > On 7/6/10, paul gore wrote: > >>> > > Hi there, > >>> > > I am experimenting with FS on EC2, I like results, but stuck on > >>> weird > >>> > > audio issue - I followed FreeSwitch EC2 wiki article and > >>> > modified > >>> > > internal profile > >>> > > and vars.xml accordingly, but unfortunately still cannot get it > >>> > > working. Incoming and outgoing calls made using a SIP phone to > >>> > FS > >>> > > extensions work just fine. As well as calls to FS from PSTN. > But > >>> > > calls to PSTN via gateways result in no audio at all, no ring, > >>> > > nothing, SIP signaling goes through OK. Sofia status profile > >>> > shows > >>> > > correct values for Ext-RTP-IP for both profiles - > >>> > > my static public IP, RTP-IP shows local IP. > >>> > > Any thoughts on that? Anybody can share working profile > >>> configuration > >>> > > may be? > >>> > > Please help, I really need to get this going. > >>> > > > >>> > > Thanks. > >>> > > >>> > _______________________________________________ > >>> > FreeSWITCH-users mailing list > >>> > FreeSWITCH-users at 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 http://www.freeswitch.org/ > >>> > ClueCon http://www.cluecon.com/ > >>> > Twitter: http://twitter.com/FreeSWITCH_wire > >>> > > >>> > AIM: anthm > >>> > > >>> > MSN:anthony_minessale at hotmail.com > > > > >>> > > >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >>> > IRC: irc.freenode.net #freeswitch > >>> > > >>> > FreeSWITCH Developer Conference > >>> > > >>> > sip:888 at conference.freeswitch.org > > > > >>> > > >>> > googletalk:conf+888 at conference.freeswitch.org > > > > >>> > pstn:+19193869900 > >>> > > >>> > > >>> > > >>> > > >>> > --------------------------------------------------------------------------- > >>> > --- > >>> > > >>> > > >>> > _______________________________________________ > >>> > FreeSWITCH-users mailing list > >>> > FreeSWITCH-users at 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 at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> ====================== > >> Tony Graziano, Manager > >> Telephone: 434.984.8430 > >> sip: tgraziano at voice.myitdepartment.net > >> Fax: 434.984.8431 > >> > >> Email: tgraziano at myitdepartment.net > >> > >> LAN/Telephony/Security and Control Systems Helpdesk: > >> Telephone: 434.984.8426 > >> sip: helpdesk at voice.myitdepartment.net > >> Fax: 434.984.8427 > >> > >> Helpdesk Contract Customers: > >> http://www.myitdepartment.net/gethelp/ > >> > >> Why do mathematicians always confuse Halloween and Christmas? > >> Because 31 Oct = 25 Dec. > >> > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/2727d960/attachment-0001.html From brian at freeswitch.org Thu Jul 8 16:53:54 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jul 2010 18:53:54 -0500 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: <201007071616.41535.sos@sokhapkin.dyndns.org> Message-ID: this sip trace is less than useful can you get a pcap file and put it where someone can wget it? /b On Jul 8, 2010, at 6:31 PM, paul gore wrote: > I got ngrep trace for port 5060 while making a call to a US number via siptraffic.com from X-Lite phone. I canceled the call after ~ 20 sec., I heard no audio not even ringing. > Is there anything in this trace which can help identify the problem? > > 10.194.206.102:5060 - is my local EC2 IP > 184.72.206.204:5060 - is my public EC2 IP > 77.72.169.128:5060 - siptraffic.com proxy IP > > Thanks! From roger_salloum at shaw.ca Thu Jul 8 20:03:58 2010 From: roger_salloum at shaw.ca (Roger Salloum) Date: Thu, 8 Jul 2010 20:03:58 -0700 Subject: [Freeswitch-users] Contact Header Modification Message-ID: Is it possible to modify the contact header such that it is the caller id? I noticed sip_contact_user but it does not seen to effect the outbound leg of the call. The documentation specifically mentions that it is only for the internal SIP contact so I don't believe it does what I was hoping it would. I know you are able to set it to a static value, but I was hoping to be able to set it on a call per call basis. Thanks From brian at freeswitch.org Thu Jul 8 20:11:13 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jul 2010 22:11:13 -0500 Subject: [Freeswitch-users] Contact Header Modification In-Reply-To: References: Message-ID: <38CF20CB-1731-4D7D-96E6-BC3212CFF0C7@freeswitch.org> So you want to change the contact part on an outbound invite? can you elaborate on WHY? /b On Jul 8, 2010, at 10:03 PM, Roger Salloum wrote: > Is it possible to modify the contact header such that it is the caller id? I noticed sip_contact_user but it does not seen to effect the outbound leg of the call. The documentation specifically mentions that it is only for the internal SIP contact so I don't believe it does what I was hoping it would. I know you are able to set it to a static value, but I was hoping to be able to set it on a call per call basis. > > Thanks From woof at iwoof.org Thu Jul 8 21:26:30 2010 From: woof at iwoof.org (Andy Spitzer) Date: Fri, 09 Jul 2010 00:26:30 -0400 Subject: [Freeswitch-users] SIP2VXML In-Reply-To: References: <39ED5E8D0A5D42BCA019A827DE9F7A4C@dell9400> Message-ID: Woof! On Mon, 05 Jul 2010 18:58:05 -0400, Anthony Minessale wrote: > It's almost like people think with all the stuff FS can do out of the > box, supporting this fly-by-night phenomenon called VXML that has failed > to gain traction after 5 years is somehow too valuable to contribute lol VoiceXML started 11 years ago, in 1999. And, still, "VoiceXML sucks" (my signature phrase for all posts about VoiceXML. Google it.) --Woof! From yurazilot1 at list.ru Thu Jul 8 22:01:22 2010 From: yurazilot1 at list.ru (viewpoint) Date: Fri, 09 Jul 2010 09:01:22 +0400 Subject: [Freeswitch-users] =?koi8-r?b?RlMgSEEtY2x1c3Rlcg==?= In-Reply-To: References: Message-ID: It has appeared so simply... All works! Thank you! Thu, 8 Jul 2010 08:53:38 -0500 ?????? ?? Rupa Schomaker : > instead of stopping/starting FS, why not stop/start just the appropriate sofia profile? ?That should be much faster. ?You can script it using fs_cli -x "sofia ..." > > > On Thu, Jul 8, 2010 at 5:11 AM, viewpoint wrote: > Hello. > > I currently have a project where I'm researching how to establish a clustered platform wieth failover time ~= some milliseconds. > > At present we have: 2 identical servers FS, wieth established Pacemaker (http://www.clusterlabs.org/wiki/Install#From_Source) - heartbeat. > On the one hand it is possible to tell that all works, as, at refusal of the first server, the second server receives cluster-ip. A problem that FreeSWITCH it is necessary to restart (or simply to start) that it began to work with the new IP-address. As the result, a switching total time makes 10-20 sec which basic part is necessary on launch FreeSwitch. > > any ideas? > > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ? ? ???? ???? - http://my.mail.ru/list/yurazilot1/ From kouxiaodong at gmail.com Thu Jul 8 22:11:51 2010 From: kouxiaodong at gmail.com (k xd) Date: Fri, 9 Jul 2010 13:11:51 +0800 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: <201007071616.41535.sos@sokhapkin.dyndns.org> Message-ID: I ever met same issue in EC2. Modify the sip_profile configuration file like "internal.xml" Replace the below item with actual ip address: Thanks, Will On Fri, Jul 9, 2010 at 7:31 AM, paul gore wrote: > I got ngrep trace for port 5060 while making a call to a US number via > siptraffic.com from X-Lite phone. I canceled the call after ~ 20 sec., I > heard no audio not even ringing. > Is there anything in this trace which can help identify the problem? > > 10.194.206.102:5060 - is my local EC2 IP > 184.72.206.204:5060 - is my public EC2 IP > 77.72.169.128:5060 - siptraffic.com proxy IP > > Thanks! > > > > 67.33.160.119:18294 -> 10.194.206.102:5060 > INVITE sip:45517705678570 at myserver.comSIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 > ;branch=z9hG4bK-d87543-f > 524431af92cef56-1--d87543-;rport..Max-Forwards: 70..Contact: < > sip:4000002 at 67.33.160.119:18027>..To: "45517 > 709248570">..From: > "4000002" > >;tag=5f1ec15f..Call- > ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Allow: > INVITE, ACK, CANCEL, OPTIONS, BYE > , REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: > application/sdp..Proxy-Authorization: Digest user > name="4000002",realm="myserver.com > ",nonce="cf9019cc-f44a-4568-97d1-e9883fb1821f",uri="sip:45517705678570 at v > ersafon.com > ",response="57da72527524e0e065c9a3221bfadd38",cnonce="140f655ff3427f6ba3767ab7040231f3",nc=0000 > 0001,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp > 41150..Content-Length: 417....v=0..o=- > 8 2 IN IP4 192.168.0.8..s=CounterPath X-Lite 3.0..c=IN IP4 > 192.168.0.8..t=0 0..m=audio 46298 RTP/AVP 107 > 119 100 106 0 105 98 8 101..a=alt:1 1 : tomYv1/D Yont/s+3 192.168.0.8 > 46298..a=fmtp:101 0-15..a=rtpmap:107 > BV32/16000..a=rtpmap:119 BV32-FEC/16000..a=rtpmap:100 > SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap > :105 SPEEX-FEC/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:101 > telephone-event/8000..a=sendrecv.. > # > U 10.194.206.102:5060 -> 67.33.160.119:18294 > SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.0.8:29486 > ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;r > port=18294;received=67.33.160.119..From: "4000002" < > sip:4000002 at myserver.com >;tag=5f1ec15f..To: > "455177092 > 48570" >..Call-ID: > ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 I > NVITE..User-Agent: myserver..Content-Length: 0.... > # > U 10.194.206.102:5080 -> 77.72.169.128:5060 > INVITE sip:0017705678570 at sip.siptraffic.comSIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 > ;rport;branch=z9h > G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < > sip:0014444295793 at 184.72.206.204 > >;tag=18853e82KDe7j. > .To: >..Call-ID: > 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 > 2 INVITE..Contact: ;transport=udp;gw=voicetrading.com>..User-A > gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, > UPDATE, INFO, REGISTER, REFER, NOTIFY.. > Supported: timer, precondition, path, replaces..Allow-Events: talk, > refer..Content-Type: application/sdp.. > Content-Disposition: session..Content-Length: 295..X-FS-Support: > update_display..Remote-Party-ID: "4000002 > " >;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH > 1278518039 > 1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 > 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8 > 3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 > GSM/8000..a=rtpmap:101 telephone-event/80 > 00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20.. > # > U 77.72.169.128:5060 -> 10.194.206.102:5080 > SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 > ;rport;branch=z9hG4bKBU626KBp16t5Q..From > : "4000002" >;tag=18853e82KDe7j..To: > > m>;tag=20113ac4c230cd6412168..Contact: > sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c > 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip > Registrar/Proxy Server)..Allow: ACK,BYE,C > ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: > application/sdp..Content-Length: 198....v=0..o=C > ARRIER 1278549617 1278549617 IN IP4 77.72.168.40..s=SIP Call..c=IN IP4 > 77.72.168.40..t=0 0..m=audio 57672 > RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 > telephone-event/8000..a=ptime:20.. > # > U 10.194.206.102:5060 -> 67.33.160.119:18294 > SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.8:29486 > ;branch=z9hG4bK-d87543-f524431af92cef56-1- > -d87543-;rport=18294;received=67.33.160.119..From: "4000002" < > sip:4000002 at myserver.com >;tag=5f1ec15f..To: > "45517705678570" >;tag=BXgB1FZBUZ3Da..Call-ID: > ZDAzODE0Y2JkZjYzODE5NmVmNjk > zMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Contact: > ..User-A > gent: myserver..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, > OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, > precondition, path, replaces..Allow-Events: > talk, presence, dialog, line-seize, call-info, sla, > include-session-description, presence.winfo, message- > summary, refer..Content-Type: application/sdp..Content-Disposition: > session..Content-Length: 251..Remote-P > arty-ID: "45517705678570" > >;party=calling;privacy=off;screen=no....v=0.. > o=FreeSWITCH 1278530815 1278530816 IN IP4 > 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m= > audio 18788 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 > telephone-event/8000..a=fmtp:101 0-16..a=sil > enceSupp:off - - - -..a=ptime:20.. > # > > > > U 77.72.169.128:5060 -> 10.194.206.102:5080 > SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 > ;rport;branch=z9hG4bKBU626KBp16t5Q..From > : "4000002" >;tag=18853e82KDe7j..To: > > m>;tag=20113ac4c230cd6412168..Contact: > sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c > 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip > Registrar/Proxy Server)..Allow: ACK,BYE,C > ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: > application/sdp..Content-Length: 204....v=0..o=C > ARRIER 1278549619 1278549619 IN IP4 208.167.230.118..s=SIP Call..c=IN IP4 > 208.167.230.118..t=0 0..m=audio > 57786 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 > telephone-event/8000..a=ptime:20.. > # > > U 67.33.160.119:18294 -> 10.194.206.102:5060 > .... > # > U 67.33.160.119:18294 -> 10.194.206.102:5060 > CANCEL sip:45517705678570 at myserver.comSIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 > ;branch=z9hG4bK-d87543-f > 524431af92cef56-1--d87543-;rport..To: "45517705678570"< > sip:45517705678570 at myserver.com >..From: > "4000002" ip:4000002 at myserver.com >;tag=5f1ec15f..Call-ID: > ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 CANC > EL..Proxy-Authorization: Digest username="4000002",realm="myserver.com > ",nonce="cf9019cc-f44a-4568-97d1-e98 > 83fb1821f",uri="sip:45517705678570 at myserver.com > ",response="46c7e289f7490c807565c561699b03d6",cnonce="a226c > > 55446c605ee229f045602b29135",nc=00000002,qop=auth,algorithm=MD5..User-Agent: > X-Lite release 1011s stamp 41 > 150..Content-Length: 0.... > # > U 10.194.206.102:5060 -> 67.33.160.119:18294 > SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.8:29486 > ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;rport > =18294;received=67.33.160.119..From: "4000002" >;tag=5f1ec15f..To: > "4551770567857 > 0" >;tag=BXgB1FZBUZ3Da..Call-ID: > ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkY > mY...CSeq: 2 CANCEL..Content-Length: 0.... > # > U 10.194.206.102:5060 -> 67.33.160.119:18294 > SIP/2.0 487 Request Terminated..Via: SIP/2.0/UDP 192.168.0.8:29486 > ;branch=z9hG4bK-d87543-f524431af92cef56- > 1--d87543-;rport=18294;received=67.33.160.119..From: "4000002" < > sip:4000002 at myserver.com >;tag=5f1ec15f..To > : "45517705678570" >;tag=BXgB1FZBUZ3Da..Call-ID: > ZDAzODE0Y2JkZjYzODE5NmVmN > jkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..User-Agent: myserver..Allow: INVITE, > ACK, BYE, CANCEL, OPTIONS, MESSA > GE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: > timer, precondition, path, repla > ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, > include-session-description, presen > ce.winfo, message-summary, refer..Content-Length: 0.... > # > U 10.194.206.102:5080 -> 77.72.169.128:5060 > CANCEL sip:0017705678570 at sip.siptraffic.comSIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 > ;rport;branch=z9h > G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < > sip:0014444295793 at 184.72.206.204 > >;tag=18853e82KDe7j. > .To: >..Call-ID: > 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 > 2 CANCEL..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length: > 0.... > # > > U 67.33.160.119:18294 -> 10.194.206.102:5060 > ACK sip:45517705678570 at myserver.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 > ;branch=z9hG4bK-d87543-f524 > 431af92cef56-1--d87543-;rport..To: "45517705678570" < > sip:45517705678570 at myserver.com > >;tag=BXgB1FZBUZ3Da..F > rom: "4000002">;tag=5f1ec15f..Call-ID: > ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYm > Y...CSeq: 2 ACK..Content-Length: 0.... > # > > U 77.72.169.128:5060 -> 10.194.206.102:5080 > SIP/2.0 200 Ok..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From: > "4000002" ip:0014444295793 at 184.72.206.204 >;tag=18853e82KDe7j..To: > >..Contact: > s > ip:0017705678570 at 77.72.169.128:5060..Call-ID: > 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 CANCEL > ..Server: (Very nice Sip Registrar/Proxy Server)..Allow: > ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSA > GE..Content-Length: 0.... > # > U 77.72.169.128:5060 -> 10.194.206.102:5080 > SIP/2.0 487 Request terminated..Via: SIP/2.0/UDP 184.72.206.204:5080 > ;rport;branch=z9hG4bKBU626KBp16t5Q..Fr > om: "4000002" >;tag=18853e82KDe7j..To: > com>..Contact: sip:0017705678570 at 77.72.169.128:5060..Call-ID: > 37f59333-04cc-122e-c381-12313b06cd32..CSeq: > 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: > ACK,BYE,CANCEL,INVITE,REGISTER,OP > TIONS,INFO,MESSAGE..Content-Length: 0.... > # > U 10.194.206.102:5080 -> 77.72.169.128:5060 > ACK sip:0017705678570 at sip.siptraffic.comSIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 > ;rport;branch=z9hG4b > KBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < > sip:0014444295793 at 184.72.206.204 > >;tag=18853e82KDe7j..To > : >..Call-ID: > 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 A > CK..Content-Length: 0.... > # > > > > > > > > > On Wed, Jul 7, 2010 at 5:50 PM, paul gore wrote: > >> Seems like siptraffic uses 6 ip addresses for media, can that be the >> problem? Is there any setting in a gateway config xml which helps with >> that? >> I will do ngrep thing and update. >> >> On 7/7/10, paul gore wrote: >> > This provider does work on another box which is not natted as ec2. >> > Most puzzling here though is why call originaion via api even not >> > going via siptraffic still gets no audio. >> > >> > On 7/7/10, Tony Graziano wrote: >> >> You should try from a standalone or local installation to ensure it >> works >> >> with this provider and your account before you attempt to run it on ec2 >> >> (imo). >> >> >> >> On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin >> >> wrote: >> >> >> >>> What "doesn't work" means? It could be (and most likely is not) >> >>> FS-related >> >>> problem >> >>> >> >>> On Wednesday 07 July 2010, Madovsky wrote: >> >>> > I had same problem from this provider without to explain why. >> >>> > One day it works, another day it doesn't, their support is crap... >> >>> > >> >>> > ----- Original Message ----- >> >>> > From: Anthony Minessale >> >>> > To: freeswitch-users at lists.freeswitch.org >> >>> > Sent: Wednesday, July 07, 2010 2:37 PM >> >>> > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on >> >>> > outgoing >> >>> > calls >> >>> > >> >>> > >> >>> > not really, not with so little information. >> >>> > >> >>> > >> >>> > >> >>> > On Wed, Jul 7, 2010 at 1:30 PM, paul gore >> >>> wrote: >> >>> > >> >>> > Firewall is configured according to the wiki, I also tried to >> open >> >>> all >> >>> > udp ports, issue persists. >> >>> > Actually the problem became more complex - outgoing calls don't >> >>> > work >> >>> > with one particular termination provider, siptraffic.com , any >> >>> > ideas >> >>> > why? >> >>> > Outgoing calls also don't work when originating a call via js >> >>> > script >> >>> > or via FS api. Any clues on that one? >> >>> > >> >>> > On 7/6/10, paul gore wrote: >> >>> > > Hi there, >> >>> > > I am experimenting with FS on EC2, I like results, but stuck >> on >> >>> weird >> >>> > > audio issue - I followed FreeSwitch EC2 wiki article and >> >>> > modified >> >>> > > internal profile >> >>> > > and vars.xml accordingly, but unfortunately still cannot get >> it >> >>> > > working. Incoming and outgoing calls made using a SIP phone to >> >>> > FS >> >>> > > extensions work just fine. As well as calls to FS from PSTN. >> But >> >>> > > calls to PSTN via gateways result in no audio at all, no ring, >> >>> > > nothing, SIP signaling goes through OK. Sofia status profile >> >>> > shows >> >>> > > correct values for Ext-RTP-IP for both profiles - >> >>> > > my static public IP, RTP-IP shows local IP. >> >>> > > Any thoughts on that? Anybody can share working profile >> >>> configuration >> >>> > > may be? >> >>> > > Please help, I really need to get this going. >> >>> > > >> >>> > > Thanks. >> >> >>> > >> >>> > _______________________________________________ >> >>> > FreeSWITCH-users mailing list >> >>> > FreeSWITCH-users at 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 http://www.freeswitch.org/ >> >>> > ClueCon http://www.cluecon.com/ >> >>> > Twitter: http://twitter.com/FreeSWITCH_wire >> >>> > >> >>> > AIM: anthm >> >>> > >> >>> > MSN:anthony_minessale at hotmail.com >> >> > >> >>> > >> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> > >> >>> > IRC: irc.freenode.net #freeswitch >> >>> > >> >>> > FreeSWITCH Developer Conference >> >>> > >> >>> > sip:888 at conference.freeswitch.org >> >> > >> >>> > >> >>> > googletalk:conf+888 at conference.freeswitch.org >> >> > >> >>> > pstn:+19193869900 >> >>> > >> >>> > >> >>> > >> >>> > >> >>> >> --------------------------------------------------------------------------- >> >>> > --- >> >> >>> > >> >>> > >> >>> > _______________________________________________ >> >>> > FreeSWITCH-users mailing list >> >>> > FreeSWITCH-users at 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 at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >> >> >> >> >> >> >> -- >> >> ====================== >> >> Tony Graziano, Manager >> >> Telephone: 434.984.8430 >> >> sip: tgraziano at voice.myitdepartment.net >> >> Fax: 434.984.8431 >> >> >> >> Email: tgraziano at myitdepartment.net >> >> >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> >> Telephone: 434.984.8426 >> >> sip: helpdesk at voice.myitdepartment.net >> >> Fax: 434.984.8427 >> >> >> >> Helpdesk Contract Customers: >> >> http://www.myitdepartment.net/gethelp/ >> >> >> >> Why do mathematicians always confuse Halloween and Christmas? >> >> Because 31 Oct = 25 Dec. >> >> >> > >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/5e7fb4d9/attachment-0001.html From jan.berger at video24.no Fri Jul 9 00:11:38 2010 From: jan.berger at video24.no (Jan Berger) Date: Fri, 9 Jul 2010 09:11:38 +0200 Subject: [Freeswitch-users] SIP2VXML In-Reply-To: References: <39ED5E8D0A5D42BCA019A827DE9F7A4C@dell9400> Message-ID: <908418189EF44BB98023F68CD865A864@dell9400> So why does it "suck"? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andy Spitzer Sent: 9. juli 2010 06:27 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP2VXML Woof! On Mon, 05 Jul 2010 18:58:05 -0400, Anthony Minessale wrote: > It's almost like people think with all the stuff FS can do out of the > box, supporting this fly-by-night phenomenon called VXML that has failed > to gain traction after 5 years is somehow too valuable to contribute lol VoiceXML started 11 years ago, in 1999. And, still, "VoiceXML sucks" (my signature phrase for all posts about VoiceXML. Google it.) --Woof! _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From yurazilot1 at list.ru Fri Jul 9 00:36:57 2010 From: yurazilot1 at list.ru (viewpoint) Date: Fri, 09 Jul 2010 11:36:57 +0400 Subject: [Freeswitch-users] =?koi8-r?b?RlMgSEEtY2x1c3Rlcg==?= In-Reply-To: References: Message-ID: Many thanks for fast and substantial answers! Has established ipv4.ip_nonlocal_bind=1. Now there is no necessity to reboot FS or sofia profile. As a result it has turned out: 1) Two servers FS which kernel works with the shared db through unix ODBC 2) It Is realised Cluster-IP by installations Pacemaker (http://www.clusterlabs.org/wiki/Install#From_Source) As server FS work with share db, there is an impression that at switching Cluster-IP with active on a passive, passive server can pick up session begun on an active server (the necessary data about current sessions takes from the share db). Whether So it? Testing has shown that at switching Cluster-IP on reserve FS, that does not process session which have begun on first FS :( Thu, 8 Jul 2010 16:21:40 +0200 (CEST) ?????? ?? Vladimir Klejch : > > Hi > > On linux you can use /proc/sys/net/ipv4/ip_nonlocal_bind to bind to non local > address on start of FS and then you don't need any restart of FS or > reload of sofia profile if this addres is active on the node > > or you can have this floating address on dummy iface (?or lo ) and then > use right settings in /proc/sys/net/ipv4/conf/eth?/arp_filter,arp_ignore > and then FS can use this address to bind on start and you don't need > restart of FS?or relaod of sofia profile ... > > > > Kleo > > On Thu, 8 Jul 2010, viewpoint wrote: > > > Hello. > > > > I currently have a project where I'm researching how to establish a > > clustered platform wieth failover time ~= some milliseconds. > > > > At present we have: 2 identical servers FS, wieth established Pacemaker > > (http://www.clusterlabs.org/wiki/Install#From_Source) - heartbeat. On > > the one hand it is possible to tell that all works, as, at refusal of > > the first server, the second server receives cluster-ip. A problem that > > FreeSWITCH it is necessary to restart (or simply to start) that it began > > to work with the new IP-address. As the result, a switching total time > > makes 10-20 sec which basic part is necessary on launch FreeSwitch. > > > > any ideas? > > > > Thanks. > > > -- > klejch+freeswitch at netbox.cz > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ? ? ???? ???? - http://my.mail.ru/list/yurazilot1/ From vetali100 at gmail.com Fri Jul 9 00:57:40 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Fri, 9 Jul 2010 10:57:40 +0300 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: <201007071616.41535.sos@sokhapkin.dyndns.org> Message-ID: You have provided only SIP log. Where is the media analysis? You should have run another ngrep to see if media traffic is OK between FS and siptraffic: "ngrep port 31564" I took the port value from the following line: c=IN IP4 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP >From the following SIP request: U 10.194.206.102:5080 -> 77.72.169.128:5060 INVITE sip:0017705678570 at sip.siptraffic.comSIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 ;rport;branch=z9h G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < sip:0014444295793 at 184.72.206.204 >;tag=18853e82KDe7j. .To: >..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 2 INVITE..Contact: ..User-A gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY.. Supported: timer, precondition, path, replaces..Allow-Events: talk, refer..Content-Type: application/sdp.. Content-Disposition: session..Content-Length: 295..X-FS-Support: update_display..Remote-Party-ID: "4000002 " >;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1278518039 1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8 3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/80 00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20.. # 2010/7/9 k xd > I ever met same issue in EC2. > > Modify the sip_profile configuration file like "internal.xml" > Replace the below item with actual ip address: > > > Thanks, > Will > > On Fri, Jul 9, 2010 at 7:31 AM, paul gore wrote: > >> I got ngrep trace for port 5060 while making a call to a US number via >> siptraffic.com from X-Lite phone. I canceled the call after ~ 20 sec., I >> heard no audio not even ringing. >> Is there anything in this trace which can help identify the problem? >> >> 10.194.206.102:5060 - is my local EC2 IP >> 184.72.206.204:5060 - is my public EC2 IP >> 77.72.169.128:5060 - siptraffic.com proxy IP >> >> Thanks! >> >> >> >> 67.33.160.119:18294 -> 10.194.206.102:5060 >> INVITE sip:45517705678570 at myserver.comSIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 >> ;branch=z9hG4bK-d87543-f >> 524431af92cef56-1--d87543-;rport..Max-Forwards: 70..Contact: < >> sip:4000002 at 67.33.160.119:18027>..To: "45517 >> 709248570">..From: >> "4000002" >> >;tag=5f1ec15f..Call- >> ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Allow: >> INVITE, ACK, CANCEL, OPTIONS, BYE >> , REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: >> application/sdp..Proxy-Authorization: Digest user >> name="4000002",realm="myserver.com >> ",nonce="cf9019cc-f44a-4568-97d1-e9883fb1821f",uri="sip:45517705678570 at v >> ersafon.com >> ",response="57da72527524e0e065c9a3221bfadd38",cnonce="140f655ff3427f6ba3767ab7040231f3",nc=0000 >> 0001,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp >> 41150..Content-Length: 417....v=0..o=- >> 8 2 IN IP4 192.168.0.8..s=CounterPath X-Lite 3.0..c=IN IP4 >> 192.168.0.8..t=0 0..m=audio 46298 RTP/AVP 107 >> 119 100 106 0 105 98 8 101..a=alt:1 1 : tomYv1/D Yont/s+3 192.168.0.8 >> 46298..a=fmtp:101 0-15..a=rtpmap:107 >> BV32/16000..a=rtpmap:119 BV32-FEC/16000..a=rtpmap:100 >> SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap >> :105 SPEEX-FEC/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:101 >> telephone-event/8000..a=sendrecv.. >> # >> U 10.194.206.102:5060 -> 67.33.160.119:18294 >> SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.0.8:29486 >> ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;r >> port=18294;received=67.33.160.119..From: "4000002" < >> sip:4000002 at myserver.com >;tag=5f1ec15f..To: >> "455177092 >> 48570" >..Call-ID: >> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 I >> NVITE..User-Agent: myserver..Content-Length: 0.... >> # >> U 10.194.206.102:5080 -> 77.72.169.128:5060 >> INVITE sip:0017705678570 at sip.siptraffic.comSIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 >> ;rport;branch=z9h >> G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < >> sip:0014444295793 at 184.72.206.204 >> >;tag=18853e82KDe7j. >> .To: >..Call-ID: >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 >> 2 INVITE..Contact: > ;transport=udp;gw=voicetrading.com>..User-A >> gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, >> UPDATE, INFO, REGISTER, REFER, NOTIFY.. >> Supported: timer, precondition, path, replaces..Allow-Events: talk, >> refer..Content-Type: application/sdp.. >> Content-Disposition: session..Content-Length: 295..X-FS-Support: >> update_display..Remote-Party-ID: "4000002 >> " >;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH >> 1278518039 >> 1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 >> 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8 >> 3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 >> GSM/8000..a=rtpmap:101 telephone-event/80 >> 00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20.. >> # >> U 77.72.169.128:5060 -> 10.194.206.102:5080 >> SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 >> ;rport;branch=z9hG4bKBU626KBp16t5Q..From >> : "4000002" >;tag=18853e82KDe7j..To: >> >> m>;tag=20113ac4c230cd6412168..Contact: >> sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c >> 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip >> Registrar/Proxy Server)..Allow: ACK,BYE,C >> ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: >> application/sdp..Content-Length: 198....v=0..o=C >> ARRIER 1278549617 1278549617 IN IP4 77.72.168.40..s=SIP Call..c=IN IP4 >> 77.72.168.40..t=0 0..m=audio 57672 >> RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 >> telephone-event/8000..a=ptime:20.. >> # >> U 10.194.206.102:5060 -> 67.33.160.119:18294 >> SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.8:29486 >> ;branch=z9hG4bK-d87543-f524431af92cef56-1- >> -d87543-;rport=18294;received=67.33.160.119..From: "4000002" < >> sip:4000002 at myserver.com >;tag=5f1ec15f..To: >> "45517705678570" >;tag=BXgB1FZBUZ3Da..Call-ID: >> ZDAzODE0Y2JkZjYzODE5NmVmNjk >> zMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Contact: >> ..User-A >> gent: myserver..Accept: application/sdp..Allow: INVITE, ACK, BYE, >> CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, >> precondition, path, replaces..Allow-Events: >> talk, presence, dialog, line-seize, call-info, sla, >> include-session-description, presence.winfo, message- >> summary, refer..Content-Type: application/sdp..Content-Disposition: >> session..Content-Length: 251..Remote-P >> arty-ID: "45517705678570" >> >;party=calling;privacy=off;screen=no....v=0.. >> o=FreeSWITCH 1278530815 1278530816 IN IP4 >> 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m= >> audio 18788 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 >> telephone-event/8000..a=fmtp:101 0-16..a=sil >> enceSupp:off - - - -..a=ptime:20.. >> # >> >> >> >> U 77.72.169.128:5060 -> 10.194.206.102:5080 >> SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 >> ;rport;branch=z9hG4bKBU626KBp16t5Q..From >> : "4000002" >;tag=18853e82KDe7j..To: >> >> m>;tag=20113ac4c230cd6412168..Contact: >> sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c >> 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip >> Registrar/Proxy Server)..Allow: ACK,BYE,C >> ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: >> application/sdp..Content-Length: 204....v=0..o=C >> ARRIER 1278549619 1278549619 IN IP4 208.167.230.118..s=SIP Call..c=IN >> IP4 208.167.230.118..t=0 0..m=audio >> 57786 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 >> telephone-event/8000..a=ptime:20.. >> # >> >> U 67.33.160.119:18294 -> 10.194.206.102:5060 >> .... >> # >> U 67.33.160.119:18294 -> 10.194.206.102:5060 >> CANCEL sip:45517705678570 at myserver.comSIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 >> ;branch=z9hG4bK-d87543-f >> 524431af92cef56-1--d87543-;rport..To: "45517705678570"< >> sip:45517705678570 at myserver.com >..From: >> "4000002"> ip:4000002 at myserver.com >;tag=5f1ec15f..Call-ID: >> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 CANC >> EL..Proxy-Authorization: Digest username="4000002",realm="myserver.com >> ",nonce="cf9019cc-f44a-4568-97d1-e98 >> 83fb1821f",uri="sip:45517705678570 at myserver.com >> ",response="46c7e289f7490c807565c561699b03d6",cnonce="a226c >> >> 55446c605ee229f045602b29135",nc=00000002,qop=auth,algorithm=MD5..User-Agent: >> X-Lite release 1011s stamp 41 >> 150..Content-Length: 0.... >> # >> U 10.194.206.102:5060 -> 67.33.160.119:18294 >> SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.8:29486 >> ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;rport >> =18294;received=67.33.160.119..From: "4000002" < >> sip:4000002 at myserver.com >;tag=5f1ec15f..To: >> "4551770567857 >> 0" >;tag=BXgB1FZBUZ3Da..Call-ID: >> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkY >> mY...CSeq: 2 CANCEL..Content-Length: 0.... >> # >> U 10.194.206.102:5060 -> 67.33.160.119:18294 >> SIP/2.0 487 Request Terminated..Via: SIP/2.0/UDP 192.168.0.8:29486 >> ;branch=z9hG4bK-d87543-f524431af92cef56- >> 1--d87543-;rport=18294;received=67.33.160.119..From: "4000002" < >> sip:4000002 at myserver.com >;tag=5f1ec15f..To >> : "45517705678570" >;tag=BXgB1FZBUZ3Da..Call-ID: >> ZDAzODE0Y2JkZjYzODE5NmVmN >> jkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..User-Agent: myserver..Allow: >> INVITE, ACK, BYE, CANCEL, OPTIONS, MESSA >> GE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, >> SUBSCRIBE..Supported: timer, precondition, path, repla >> ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, >> include-session-description, presen >> ce.winfo, message-summary, refer..Content-Length: 0.... >> # >> U 10.194.206.102:5080 -> 77.72.169.128:5060 >> CANCEL sip:0017705678570 at sip.siptraffic.comSIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 >> ;rport;branch=z9h >> G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < >> sip:0014444295793 at 184.72.206.204 >> >;tag=18853e82KDe7j. >> .To: >..Call-ID: >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 >> 2 CANCEL..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length: >> 0.... >> # >> >> U 67.33.160.119:18294 -> 10.194.206.102:5060 >> ACK sip:45517705678570 at myserver.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486 >> ;branch=z9hG4bK-d87543-f524 >> 431af92cef56-1--d87543-;rport..To: "45517705678570" < >> sip:45517705678570 at myserver.com >> >;tag=BXgB1FZBUZ3Da..F >> rom: "4000002">;tag=5f1ec15f..Call-ID: >> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYm >> Y...CSeq: 2 ACK..Content-Length: 0.... >> # >> >> U 77.72.169.128:5060 -> 10.194.206.102:5080 >> SIP/2.0 200 Ok..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From: >> "4000002" > ip:0014444295793 at 184.72.206.204 >;tag=18853e82KDe7j..To: >> >..Contact: >> s >> ip:0017705678570 at 77.72.169.128:5060..Call-ID: >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 CANCEL >> ..Server: (Very nice Sip Registrar/Proxy Server)..Allow: >> ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSA >> GE..Content-Length: 0.... >> # >> U 77.72.169.128:5060 -> 10.194.206.102:5080 >> SIP/2.0 487 Request terminated..Via: SIP/2.0/UDP 184.72.206.204:5080 >> ;rport;branch=z9hG4bKBU626KBp16t5Q..Fr >> om: "4000002" >;tag=18853e82KDe7j..To: >> > com>..Contact: sip:0017705678570 at 77.72.169.128:5060..Call-ID: >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: >> 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: >> ACK,BYE,CANCEL,INVITE,REGISTER,OP >> TIONS,INFO,MESSAGE..Content-Length: 0.... >> # >> U 10.194.206.102:5080 -> 77.72.169.128:5060 >> ACK sip:0017705678570 at sip.siptraffic.comSIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080 >> ;rport;branch=z9hG4b >> KBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < >> sip:0014444295793 at 184.72.206.204 >> >;tag=18853e82KDe7j..To >> : >..Call-ID: >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 A >> CK..Content-Length: 0.... >> # >> >> >> >> >> >> >> >> >> On Wed, Jul 7, 2010 at 5:50 PM, paul gore wrote: >> >>> Seems like siptraffic uses 6 ip addresses for media, can that be the >>> problem? Is there any setting in a gateway config xml which helps with >>> that? >>> I will do ngrep thing and update. >>> >>> On 7/7/10, paul gore wrote: >>> > This provider does work on another box which is not natted as ec2. >>> > Most puzzling here though is why call originaion via api even not >>> > going via siptraffic still gets no audio. >>> > >>> > On 7/7/10, Tony Graziano wrote: >>> >> You should try from a standalone or local installation to ensure it >>> works >>> >> with this provider and your account before you attempt to run it on >>> ec2 >>> >> (imo). >>> >> >>> >> On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin >>> >> wrote: >>> >> >>> >>> What "doesn't work" means? It could be (and most likely is not) >>> >>> FS-related >>> >>> problem >>> >>> >>> >>> On Wednesday 07 July 2010, Madovsky wrote: >>> >>> > I had same problem from this provider without to explain why. >>> >>> > One day it works, another day it doesn't, their support is crap... >>> >>> > >>> >>> > ----- Original Message ----- >>> >>> > From: Anthony Minessale >>> >>> > To: freeswitch-users at lists.freeswitch.org >>> >>> > Sent: Wednesday, July 07, 2010 2:37 PM >>> >>> > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on >>> >>> > outgoing >>> >>> > calls >>> >>> > >>> >>> > >>> >>> > not really, not with so little information. >>> >>> > >>> >>> > >>> >>> > >>> >>> > On Wed, Jul 7, 2010 at 1:30 PM, paul gore >> > >>> >>> wrote: >>> >>> > >>> >>> > Firewall is configured according to the wiki, I also tried to >>> open >>> >>> all >>> >>> > udp ports, issue persists. >>> >>> > Actually the problem became more complex - outgoing calls don't >>> >>> > work >>> >>> > with one particular termination provider, siptraffic.com , any >>> >>> > ideas >>> >>> > why? >>> >>> > Outgoing calls also don't work when originating a call via js >>> >>> > script >>> >>> > or via FS api. Any clues on that one? >>> >>> > >>> >>> > On 7/6/10, paul gore wrote: >>> >>> > > Hi there, >>> >>> > > I am experimenting with FS on EC2, I like results, but stuck >>> on >>> >>> weird >>> >>> > > audio issue - I followed FreeSwitch EC2 wiki article and >>> >>> > modified >>> >>> > > internal profile >>> >>> > > and vars.xml accordingly, but unfortunately still cannot get >>> it >>> >>> > > working. Incoming and outgoing calls made using a SIP phone >>> to >>> >>> > FS >>> >>> > > extensions work just fine. As well as calls to FS from PSTN. >>> But >>> >>> > > calls to PSTN via gateways result in no audio at all, no >>> ring, >>> >>> > > nothing, SIP signaling goes through OK. Sofia status profile >>> >>> > shows >>> >>> > > correct values for Ext-RTP-IP for both profiles - >>> >>> > > my static public IP, RTP-IP shows local IP. >>> >>> > > Any thoughts on that? Anybody can share working profile >>> >>> configuration >>> >>> > > may be? >>> >>> > > Please help, I really need to get this going. >>> >>> > > >>> >>> > > Thanks. >>> >>> >>> > >>> >>> > _______________________________________________ >>> >>> > FreeSWITCH-users mailing list >>> >>> > FreeSWITCH-users at 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 http://www.freeswitch.org/ >>> >>> > ClueCon http://www.cluecon.com/ >>> >>> > Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> > >>> >>> > AIM: anthm >>> >>> > >>> >>> > MSN:anthony_minessale at hotmail.com >>> >>> > >>> >>> > >>> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> >>> > >>> >>> > IRC: irc.freenode.net #freeswitch >>> >>> > >>> >>> > FreeSWITCH Developer Conference >>> >>> > >>> >>> > sip:888 at conference.freeswitch.org >>> >>> > >>> >>> > >>> >>> > googletalk:conf+888 at conference.freeswitch.org >>> >>> > >>> >>> > pstn:+19193869900 >>> >>> > >>> >>> > >>> >>> > >>> >>> > >>> >>> >>> --------------------------------------------------------------------------- >>> >>> > --- >>> >>> >>> > >>> >>> > >>> >>> > _______________________________________________ >>> >>> > FreeSWITCH-users mailing list >>> >>> > FreeSWITCH-users at 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 at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >>> >> >>> >> >>> >> >>> >> -- >>> >> ====================== >>> >> Tony Graziano, Manager >>> >> Telephone: 434.984.8430 >>> >> sip: tgraziano at voice.myitdepartment.net >>> >> Fax: 434.984.8431 >>> >> >>> >> Email: tgraziano at myitdepartment.net >>> >> >>> >> LAN/Telephony/Security and Control Systems Helpdesk: >>> >> Telephone: 434.984.8426 >>> >> sip: helpdesk at voice.myitdepartment.net >>> >> Fax: 434.984.8427 >>> >> >>> >> Helpdesk Contract Customers: >>> >> http://www.myitdepartment.net/gethelp/ >>> >> >>> >> Why do mathematicians always confuse Halloween and Christmas? >>> >> Because 31 Oct = 25 Dec. >>> >> >>> > >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/33b9f7af/attachment-0001.html From david.ponzone at gmail.com Fri Jul 9 02:39:52 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Fri, 9 Jul 2010 11:39:52 +0200 Subject: [Freeswitch-users] Contact Header Modification In-Reply-To: References: Message-ID: Perhaps what you're looking for is the following gateway param: David Ponzone Direction Technique email: david.ponzone at 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 pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 09/07/2010 ? 05:03, Roger Salloum a ?crit : > Is it possible to modify the contact header such that it is the > caller id? I noticed sip_contact_user but it does not seen to effect > the outbound leg of the call. The documentation specifically > mentions that it is only for the internal SIP contact so I don't > believe it does what I was hoping it would. I know you are able to > set it to a static value, but I was hoping to be able to set it on a > call per call basis. > > Thanks > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/e5842c46/attachment.html From luixsansan at hotmail.com Fri Jul 9 03:47:04 2010 From: luixsansan at hotmail.com (luixsansan at hotmail.com) Date: Fri, 9 Jul 2010 12:47:04 +0200 Subject: [Freeswitch-users] dtmf in Skype In-Reply-To: References: Message-ID: Hello, I use the sample config file then, when I call from another Skype Client in other computer to user1 extension 5000 answers and I listen to the IVR demo, I pulse an extension or a number of the menu options but nothing happens, I keep listening the menu until FS hangs up for timeout. by the way, about Dialplan in the document I can read "There are a few simple examples given in the "default.xml" dialplan located in mod_skypopen/configs/" I do not see any dialplan example. Regards. Luis. -------------------------------------------------- From: "Giovanni Maruzzelli" Sent: Thursday, July 08, 2010 9:47 PM To: Subject: Re: [Freeswitch-users] dtmf in Skype > that is not true ;) Is very clear, indeed (I wrote it ;) ) > > look for dtmf in all the wiki page, you'll find some parameter to be used. > > If it's still unclear, please rewrite here, and describe fully what > you'r doing and what is the problem > > -giovanni > > On Thu, Jul 8, 2010 at 9:03 PM, wrote: >> Hello, >> >> I have FS ver. 1.0.6, Windows Vista Ultimate and Skype 4.2.0.169 for >> Windows. (user1) >> >> On a laptop I have Windows Vista Home and Skype 4.1.0.179 (user2) >> >> When I make a call from user2 to user1 I listen the IVR menu and see the >> activity on FS console. The problem is that when I click on the skype's >> phone keyboard I can not listen to any DTMF signal and user1 is not >> receiving DTMF signals ( the IVR menu is not interrupted). >> >> Is there a configuration change I have to do in order to make skype >> send/receive dtmf signals or do I need another type of skype client? >> >> The info in the document for mod_skypopen is not very clear. >> >> >> Thank you. >> >> Luis. >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gmaruzz at celliax.org Fri Jul 9 04:22:09 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 9 Jul 2010 13:22:09 +0200 Subject: [Freeswitch-users] dtmf in Skype In-Reply-To: References: Message-ID: On Fri, Jul 9, 2010 at 12:47 PM, wrote: config file is ok > > then, when I call from another Skype Client in other computer to user1 > extension 5000 answers and I listen to the IVR demo, I pulse an extension or > a number of the menu options but nothing happens, I keep listening the menu > until FS hangs up for timeout. this is very strange, because the skype clients sends the dtmf as out of band signaling, so it is always seen by skypopen (eg: no fiddling with audio dtmf recognition). can you please open a Jira issue? and please do this: 1) if you have put mod_skypopen in modules.conf.xml, please comment it out (let's not start it when we don't see it ;) ) 2) shutdown and restart FS 3) from its command line: 4) "fsctl loglevel 7" 5) "console loglevel 7" 6) "load mod_skypopen" 7) receive the call and press some dtmf in the remote skype client keypad and attach to the Jira all the output, since the beginning (attach as a file, would be better than paste as comment. > -------------------------------------------------- > From: "Giovanni Maruzzelli" > Sent: Thursday, July 08, 2010 9:47 PM > To: > Subject: Re: [Freeswitch-users] dtmf in Skype > >> that is not true ;) Is very clear, indeed (I wrote it ;) ) >> >> look for dtmf in all the wiki page, you'll find some parameter to be used. >> >> If it's still unclear, please rewrite here, and describe fully what >> you'r doing and what is the problem >> >> -giovanni >> >> On Thu, Jul 8, 2010 at 9:03 PM, ? wrote: >>> Hello, >>> >>> I have FS ver. 1.0.6, Windows Vista Ultimate and Skype 4.2.0.169 for >>> Windows. (user1) >>> >>> On a laptop I have Windows Vista Home and Skype 4.1.0.179 (user2) >>> >>> When I make a call from user2 to user1 I listen the IVR menu and see the >>> activity on FS console. The problem is that when I click on the skype's >>> phone keyboard I can not listen to any DTMF signal and user1 is not >>> receiving DTMF signals ( the IVR menu is not interrupted). >>> >>> Is there a configuration change I have to do in order to make skype >>> send/receive dtmf signals or do I need another type of skype client? >>> >>> The info in the document for mod_skypopen is not very clear. >>> >>> >>> Thank you. >>> >>> Luis. >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From thakkar.jigar at gmail.com Fri Jul 9 04:35:26 2010 From: thakkar.jigar at gmail.com (Jigar Thakkar) Date: Fri, 9 Jul 2010 17:05:26 +0530 Subject: [Freeswitch-users] Two FreeSWITCH Box IVR Call Transfer In-Reply-To: <1278513656974-5265595.post@n2.nabble.com> References: <1278513656974-5265595.post@n2.nabble.com> Message-ID: Thanks Brian, It worked. @David, I like to explore the way you suggested. But you may have a look at what Brian told. Is it secure to transfer data in header ? If not then what can be the way to make it secure? Thanks & Regards, Jigar. On Wed, Jul 7, 2010 at 8:10 PM, David Swardstrom wrote: > > I have an untested solution to this issue. > Note: I am using JavaScript code examples: > The following will do the "divert"/"transfer" if you have a session. > session.execute("deflect", dfltstr); > > When Sip is involved, FreeSwitch when using a SIP "Call Transfer" which > asks > some upstream system to send the call to some other system. > > When doing this, a new destination address can be supplied. > This can include a "postd" parameter. > I think postd stands for "post dial" and is a standard parameter that can > be > added to the destination address. > RFC 3261, section 19.1.6 has the following example: > Thus, tel:+358-555-1234567;postd=pp22 becomes > sip:+358-555-1234567;postd=pp22 at foo.com;user=phone > > There should be a way to add a postd string to the new destination address. > So you can put the "user id" into the postd string. > > When the divert/transfer occurs, the new system will get the call as if it > was an originated call. > Looking at the destination string, the presence or absence of postd can > determine how the > call should be handled. If a postd is provided, the value can be used as > the > "user id". > > Note: I was looking at conferencing so planned to pass the conference > information this way. > > Another note: I don't think that "user=phone" is necessary. The systems > seem > to be designed to > work with it provided or not provided. > > If/When I find time, I will code this up as a JavaScript example and add it > to the Wiki. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Two-FreeSWITCH-Box-IVR-Call-Transfer-tp5265404p5265595.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/f65c4276/attachment.html From jan.berger at video24.no Fri Jul 9 04:42:18 2010 From: jan.berger at video24.no (Jan Berger) Date: Fri, 9 Jul 2010 13:42:18 +0200 Subject: [Freeswitch-users] originate call hangup signal In-Reply-To: References: <2B4FFD98-7682-44AB-ADB8-9B9700B5AC32@freeswitch.org> Message-ID: <06271C2A41CE49C3901D0B611D63B56A@dell9400> Have you checked your NAI setting? This is usually the main reason why outboand calls trouble. And the native and Libpri might have a different default settings. If not get a trace with the native and one with libpri . I need to see HDLC either in raw form or decoded on layer 2. Jan _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 8. juli 2010 18:58 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] originate call hangup signal Interesting. I've had more success with the libpri stack than the native stack because the native stack is definitely a work in progress. I assisted Mike J in getting 5ESS support in the native stack, but we definitely don't have explicit 4ESS support. I think you are better off using the libpri method and debugging it than trying to get the native PRI stack to work on a 4ESS connection. Hop on #openzap on irc.freenode.net and ask for some assistance there. We have a few guys who are familiar with PRI and libpri, etc. who might be able to help you figure out what's going on. -MC On Thu, Jul 8, 2010 at 1:01 AM, Tony Tin wrote: It's Digium TE220. I'm using the OpenZAP native stack, because I can not get the outbound call work with libpri compatibility stack. Attached is the freeswitch.log. I'm not sure whether it includes the d-chan trace, though I already enabled the "q931_dump". I originated a call to my mobile on freeswitch console with command "originate OpenZAP/2/A/98855404 6899", I answered the call then hung up, after around 30 seconds, I saw there is terminator event on the console and the call hangup. Thanks Regards, Tony On Thu, Jul 8, 2010 at 12:42 PM, Michael S Collins wrote: Okay, next question: which PRI are you using? Is it Digium-based or Sangoma hardware? If the former then use the libpri method; the latter use freetdm. I think they're both covered on the wiki. You need to get an ISDN trace on the d-chan to see what is actually being sent to/from telco. -MC Sent from my iPhone On Jul 7, 2010, at 7:57 PM, Tony Tin wrote: Thanks for your help. It's a 4ESS IDSN and the carrier does provide disconnect supervision, is there any way to bypass it ? Regards, Tony On Thu, Jul 8, 2010 at 7:14 AM, Michael Collins < msc at freeswitch.org> wrote: It depends on where the "hangup signal" comes from. Is this an analog line? If so, does the carrier provide disconnect supervision? It's entirely possible that the other end isn't doing a good job of telling FreeSWITCH that the call is over. -MC On Tue, Jul 6, 2010 at 10:35 PM, Tony Tin < tony.tin at noahmedia.com.hk> wrote: Hi, When I use OpenZAP channel to originate a call, after the called party hangup the phone. It takes freeswitch around 40 seconds to catch the hangup signal and stop the dial plan. I'm wondering whether there is a way to shorten that duration. Thanks. Regards, Tony _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/2d53c192/attachment-0001.html From jan.berger at video24.no Fri Jul 9 05:01:14 2010 From: jan.berger at video24.no (Jan Berger) Date: Fri, 9 Jul 2010 14:01:14 +0200 Subject: [Freeswitch-users] Native stacks Message-ID: <5195B7332FD141F8950E89C65F3BE491@dell9400> Hi, Creating a new thread. If it is any interest for a native stack in FreeSWITCH we could address it later. It lacks the state-engine. Actually, last time I asked I was told someone in Germany was working on this? But, as I looked at it recently I concluded that with Sangoma supporting their own boards, and Digium maintaining libpri for theirs - we basically have covered a majority of the E1/T1 boards that is used in here. Correct me if I am wrong. Jan Interesting. I've had more success with the libpri stack than the native stack because the native stack is definitely a work in progress. I assisted Mike J in getting 5ESS support in the native stack, but we definitely don't have explicit 4ESS support. I think you are better off using the libpri method and debugging it than trying to get the native PRI stack to work on a 4ESS connection -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/a4774244/attachment.html From helmut.kuper at ewetel.de Fri Jul 9 05:36:52 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 09 Jul 2010 14:36:52 +0200 Subject: [Freeswitch-users] Native stacks In-Reply-To: <5195B7332FD141F8950E89C65F3BE491@dell9400> References: <5195B7332FD141F8950E89C65F3BE491@dell9400> Message-ID: <4C3717E4.602@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Jan, yes, Stefan Knoblich (stkn at freeswitch.org) is working or has worked on a new ISDN stack supporting timers for the call states. I have and using a openzap Q931StateTE stack which is basing on his early work. A added and fixed some stuff on it, so that it works quite good for basic calls on a Avaya-PRI as well as on a EWSD-PRI. The changes I did I have sent to Stefan, but got no new informations from him since a few months. Maybe he is busy, maybe things are passed to Sangoma, maybe there is no big interest in a native stack, dunno... At least I have a need for a reliable slim isdn stack supporting Q931 and if possible QSIG. It looks that libpri is the better choice as a future safe stack. But I've never tested it with FS. regards from sunny and hot germany Helmut On 09.07.2010 14:01, Jan Berger wrote: > Hi, > > > > Creating a new thread. > > > > If it is any interest for a native stack in FreeSWITCH we could address it > later. It lacks the state-engine. Actually, last time I asked I was told > someone in Germany was working on this? > > > > But, as I looked at it recently I concluded that with Sangoma supporting > their own boards, and Digium maintaining libpri for theirs - we basically > have covered a majority of the E1/T1 boards that is used in here. Correct me > if I am wrong. > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMNxfk4tZeNddg3dwRAqMxAJ0Vh5NnUY3HBEexz7ps8zR3GfGa6ACfTdhk Z+nSJ/7WC12tJyG/F3UxIEY= =5nmh -----END PGP SIGNATURE----- From jan.berger at video24.no Fri Jul 9 06:16:47 2010 From: jan.berger at video24.no (Jan Berger) Date: Fri, 9 Jul 2010 15:16:47 +0200 Subject: [Freeswitch-users] Native stacks In-Reply-To: <4C3717E4.602@ewetel.de> References: <5195B7332FD141F8950E89C65F3BE491@dell9400> <4C3717E4.602@ewetel.de> Message-ID: <70AA47CA5E7B4209A6ECA97A7F55CA23@dell9400> What E1/T1 hardware are you using? Libpri is maintained by a larger group. They had started to get QSIG last time I checked - they also have basic SS7 for simple ISUP calls in libss7. The stack itself is a maintenance problem, but Digium has the strength to maintain it - and in the end that is all that matters unless you want to maintain things yourself. Jan -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Helmut Kuper Sent: 9. juli 2010 14:37 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Native stacks -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Jan, yes, Stefan Knoblich (stkn at freeswitch.org) is working or has worked on a new ISDN stack supporting timers for the call states. I have and using a openzap Q931StateTE stack which is basing on his early work. A added and fixed some stuff on it, so that it works quite good for basic calls on a Avaya-PRI as well as on a EWSD-PRI. The changes I did I have sent to Stefan, but got no new informations from him since a few months. Maybe he is busy, maybe things are passed to Sangoma, maybe there is no big interest in a native stack, dunno... At least I have a need for a reliable slim isdn stack supporting Q931 and if possible QSIG. It looks that libpri is the better choice as a future safe stack. But I've never tested it with FS. regards from sunny and hot germany Helmut On 09.07.2010 14:01, Jan Berger wrote: > Hi, > > > > Creating a new thread. > > > > If it is any interest for a native stack in FreeSWITCH we could address it > later. It lacks the state-engine. Actually, last time I asked I was told > someone in Germany was working on this? > > > > But, as I looked at it recently I concluded that with Sangoma supporting > their own boards, and Digium maintaining libpri for theirs - we basically > have covered a majority of the E1/T1 boards that is used in here. Correct me > if I am wrong. > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMNxfk4tZeNddg3dwRAqMxAJ0Vh5NnUY3HBEexz7ps8zR3GfGa6ACfTdhk Z+nSJ/7WC12tJyG/F3UxIEY= =5nmh -----END PGP SIGNATURE----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From too.goa at gmail.com Thu Jul 8 23:12:28 2010 From: too.goa at gmail.com (Goa) Date: Fri, 09 Jul 2010 12:12:28 +0600 Subject: [Freeswitch-users] Error Loading module mod_spidermonkey.so Message-ID: <4C36BDCC.3020309@gmail.com> Hello! I'm trying to setup FS on FreeBSD 8.0-RELEASE i386 from http://files.freeswitch.org/freeswitch-1.0.6.tar.gz but there is an error in freeswitch.log 2010-07-09 05:36:45.246801 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/lib/libjs.so.1: Undefined symbol "PR_LocalTimeParameters"** Also I see the same situation http://jira.freeswitch.org/browse/FSBUILD-41;jsessionid=822BB66E72FFF9D2C18888FC208D9189?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel but http://files.freeswitch.org/freeswitch-1.0.6.tar.gz was created later, than that case. Thank you. Goa From too.goa at gmail.com Fri Jul 9 04:38:27 2010 From: too.goa at gmail.com (Goa) Date: Fri, 09 Jul 2010 17:38:27 +0600 Subject: [Freeswitch-users] Error making mod_conference Message-ID: <4C370A33.10204@gmail.com> Hello! I'm trying to setup FS on FreeBSD 8.0-RELEASE i386. Now from snapshot from http://files.freeswitch.org/freeswitch-snapshot.tar.gz #gmake install, but errors http://pastebin.freeswitch.org/13418 Thank you. Goa From brian at freeswitch.org Fri Jul 9 06:28:44 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 9 Jul 2010 08:28:44 -0500 Subject: [Freeswitch-users] Two FreeSWITCH Box IVR Call Transfer In-Reply-To: References: <1278513656974-5265595.post@n2.nabble.com> Message-ID: You could also encrypt it then decrypt it on the far side. /b On Jul 9, 2010, at 6:35 AM, Jigar Thakkar wrote: > Thanks Brian, It worked. > > @David, I like to explore the way you suggested. But you may have a look at what Brian told. > > Is it secure to transfer data in header ? > If not then what can be the way to make it secure? > > Thanks & Regards, > > Jigar. From brian at freeswitch.org Fri Jul 9 06:32:41 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 9 Jul 2010 08:32:41 -0500 Subject: [Freeswitch-users] Error Loading module mod_spidermonkey.so In-Reply-To: <4C36BDCC.3020309@gmail.com> References: <4C36BDCC.3020309@gmail.com> Message-ID: <2979973A-00C7-4E58-8C96-422F8225B157@freeswitch.org> Why don't you get the source code from git? /b On Jul 9, 2010, at 1:12 AM, Goa wrote: > Hello! > I'm trying to setup FS on FreeBSD 8.0-RELEASE i386 > from http://files.freeswitch.org/freeswitch-1.0.6.tar.gz > > but there is an error in freeswitch.log > > 2010-07-09 05:36:45.246801 [CRIT] switch_loadable_module.c:882 Error > Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so > **/usr/local/freeswitch/lib/libjs.so.1: Undefined symbol > "PR_LocalTimeParameters"** > > Also I see the same situation > http://jira.freeswitch.org/browse/FSBUILD-41;jsessionid=822BB66E72FFF9D2C18888FC208D9189?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel > > but http://files.freeswitch.org/freeswitch-1.0.6.tar.gz was created > later, than that case. > > Thank you. > Goa From brian at freeswitch.org Fri Jul 9 06:32:47 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 9 Jul 2010 08:32:47 -0500 Subject: [Freeswitch-users] Error making mod_conference In-Reply-To: <4C370A33.10204@gmail.com> References: <4C370A33.10204@gmail.com> Message-ID: <875F2CBE-6E62-480D-AAEE-72B793F6773D@freeswitch.org> Why don't you get the source code from git? /b On Jul 9, 2010, at 6:38 AM, Goa wrote: > Hello! > I'm trying to setup FS on FreeBSD 8.0-RELEASE i386. > Now from snapshot from > http://files.freeswitch.org/freeswitch-snapshot.tar.gz > > #gmake install, but errors > http://pastebin.freeswitch.org/13418 > > Thank you. > Goa From brian at freeswitch.org Fri Jul 9 06:33:46 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 9 Jul 2010 08:33:46 -0500 Subject: [Freeswitch-users] Contact Header Modification In-Reply-To: References: Message-ID: <24D2479A-646F-47B6-BCCA-87915D5BF38F@freeswitch.org> Any provider, hardware, software that requires specific things in the contact header is broken. Its your contact header not theirs to dictate. /b On Jul 9, 2010, at 4:39 AM, David Ponzone wrote: > Perhaps what you're looking for is the following gateway param: > > > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 From brian at freeswitch.org Fri Jul 9 06:35:12 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 9 Jul 2010 08:35:12 -0500 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: <201007071616.41535.sos@sokhapkin.dyndns.org> Message-ID: <1333C24C-C580-48A2-92B3-92B8ABAA1E0A@freeswitch.org> You also need to set ext-sip-ip and ext-rtp-ip also can you get me a pcap? I know it works because I have personally done this on EC2 along with many others. Its as simple as understanding NAT and FreeSWITCH. /b On Jul 9, 2010, at 2:57 AM, Vitalii Colosov wrote: > You should have run another ngrep to see if media traffic is OK between FS and siptraffic: From david.ponzone at gmail.com Fri Jul 9 06:42:27 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Fri, 9 Jul 2010 15:42:27 +0200 Subject: [Freeswitch-users] Contact Header Modification In-Reply-To: <24D2479A-646F-47B6-BCCA-87915D5BF38F@freeswitch.org> References: <24D2479A-646F-47B6-BCCA-87915D5BF38F@freeswitch.org> Message-ID: <78563FB1-EFCA-4F2F-B2B0-2E3C68C971B6@gmail.com> Brian, you know that SIP is all broken anyway :) David Ponzone Direction Technique email: david.ponzone at 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 pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 09/07/2010 ? 15:33, Brian West a ?crit : > Any provider, hardware, software that requires specific things in > the contact header is broken. Its your contact header not theirs to > dictate. > > /b > > On Jul 9, 2010, at 4:39 AM, David Ponzone wrote: > >> Perhaps what you're looking for is the following gateway param: >> >> >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/7c97f492/attachment.html From phone.bytes at gmail.com Fri Jul 9 06:46:21 2010 From: phone.bytes at gmail.com (Phone) Date: Fri, 09 Jul 2010 07:46:21 -0600 Subject: [Freeswitch-users] Native stacks In-Reply-To: <70AA47CA5E7B4209A6ECA97A7F55CA23@dell9400> References: <5195B7332FD141F8950E89C65F3BE491@dell9400> <4C3717E4.602@ewetel.de> <70AA47CA5E7B4209A6ECA97A7F55CA23@dell9400> Message-ID: <4C37282D.2070602@gmail.com> I am interested in this discussion of the PRI stack. We are currently using libpri with a Samgoma B601DE. Things are working fine, except for the fact that we are not receiving Caller_ID "Name" on incoming calls. In examining traces, it appears that the name comes along milliseconds later in a separate facility message, and FS is not picking this up? I am curious if others using libpri with FS and Sangoma are getting Caller-ID "Name" delivery. Not sure if we are not configured quite right, or if there is some other libpri issue with FS. Jan Berger wrote: > What E1/T1 hardware are you using? > > Libpri is maintained by a larger group. They had started to get QSIG last > time I checked - they also have basic SS7 for simple ISUP calls in libss7. > The stack itself is a maintenance problem, but Digium has the strength to > maintain it - and in the end that is all that matters unless you want to > maintain things yourself. > > Jan > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Helmut > Kuper > Sent: 9. juli 2010 14:37 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Native stacks > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi Jan, > > yes, Stefan Knoblich (stkn at freeswitch.org) is working or has worked on a > new ISDN stack supporting timers for the call states. > > I have and using a openzap Q931StateTE stack which is basing on his > early work. A added and fixed some stuff on it, so that it works quite > good for basic calls on a Avaya-PRI as well as on a EWSD-PRI. > > The changes I did I have sent to Stefan, but got no new informations > from him since a few months. > > Maybe he is busy, maybe things are passed to Sangoma, maybe there is no > big interest in a native stack, dunno... > > > At least I have a need for a reliable slim isdn stack supporting Q931 > and if possible QSIG. > > It looks that libpri is the better choice as a future safe stack. But > I've never tested it with FS. > > > regards from sunny and hot germany > > Helmut > > > > On 09.07.2010 14:01, Jan Berger wrote: > >> Hi, >> >> >> >> Creating a new thread. >> >> >> >> If it is any interest for a native stack in FreeSWITCH we could address it >> later. It lacks the state-engine. Actually, last time I asked I was told >> someone in Germany was working on this? >> >> >> >> But, as I looked at it recently I concluded that with Sangoma supporting >> their own boards, and Digium maintaining libpri for theirs - we basically >> have covered a majority of the E1/T1 boards that is used in here. Correct >> > me > >> if I am wrong. >> >> > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFMNxfk4tZeNddg3dwRAqMxAJ0Vh5NnUY3HBEexz7ps8zR3GfGa6ACfTdhk > Z+nSJ/7WC12tJyG/F3UxIEY= > =5nmh > -----END PGP SIGNATURE----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Fri Jul 9 06:48:24 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 9 Jul 2010 08:48:24 -0500 Subject: [Freeswitch-users] Contact Header Modification In-Reply-To: <78563FB1-EFCA-4F2F-B2B0-2E3C68C971B6@gmail.com> References: <24D2479A-646F-47B6-BCCA-87915D5BF38F@freeswitch.org> <78563FB1-EFCA-4F2F-B2B0-2E3C68C971B6@gmail.com> Message-ID: Sip in general isn't that badly broken... its humans that misinterpret the spec in their own quirky ways that makes sip suck at times... if the RFC had come with a working implementation to test against this would have never happened. /b On Jul 9, 2010, at 8:42 AM, David Ponzone wrote: > Brian, > > you know that SIP is all broken anyway :) > From brian at freeswitch.org Fri Jul 9 06:49:02 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 9 Jul 2010 08:49:02 -0500 Subject: [Freeswitch-users] Native stacks In-Reply-To: <4C37282D.2070602@gmail.com> References: <5195B7332FD141F8950E89C65F3BE491@dell9400> <4C3717E4.602@ewetel.de> <70AA47CA5E7B4209A6ECA97A7F55CA23@dell9400> <4C37282D.2070602@gmail.com> Message-ID: <3C444E43-479A-4626-B26C-E25B2770A3A6@freeswitch.org> because you need to SLEEP a few seconds before you answer just like you do in Asterisk. /b On Jul 9, 2010, at 8:46 AM, Phone wrote: > I am interested in this discussion of the PRI stack. We are currently > using libpri with a Samgoma B601DE. Things are working fine, except for > the fact that we are not receiving Caller_ID "Name" on incoming calls. > In examining traces, it appears that the name comes along milliseconds > later in a separate facility message, and FS is not picking this up? > > I am curious if others using libpri with FS and Sangoma are getting > Caller-ID "Name" delivery. Not sure if we are not configured quite > right, or if there is some other libpri issue with FS. From jan.berger at video24.no Fri Jul 9 06:59:59 2010 From: jan.berger at video24.no (Jan Berger) Date: Fri, 9 Jul 2010 15:59:59 +0200 Subject: [Freeswitch-users] Native stacks In-Reply-To: <4C37282D.2070602@gmail.com> References: <5195B7332FD141F8950E89C65F3BE491@dell9400> <4C3717E4.602@ewetel.de> <70AA47CA5E7B4209A6ECA97A7F55CA23@dell9400> <4C37282D.2070602@gmail.com> Message-ID: Why are you using libpri with Sangoma boards? I am not that familiar with Sangoma boards myself, but looking at FreeTDM I would expect this to be the most optional path for Sangoma hardware? Jan -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phone Sent: 9. juli 2010 15:46 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Native stacks I am interested in this discussion of the PRI stack. We are currently using libpri with a Samgoma B601DE. Things are working fine, except for the fact that we are not receiving Caller_ID "Name" on incoming calls. In examining traces, it appears that the name comes along milliseconds later in a separate facility message, and FS is not picking this up? I am curious if others using libpri with FS and Sangoma are getting Caller-ID "Name" delivery. Not sure if we are not configured quite right, or if there is some other libpri issue with FS. Jan Berger wrote: > What E1/T1 hardware are you using? > > Libpri is maintained by a larger group. They had started to get QSIG last > time I checked - they also have basic SS7 for simple ISUP calls in libss7. > The stack itself is a maintenance problem, but Digium has the strength to > maintain it - and in the end that is all that matters unless you want to > maintain things yourself. > > Jan > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Helmut > Kuper > Sent: 9. juli 2010 14:37 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Native stacks > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi Jan, > > yes, Stefan Knoblich (stkn at freeswitch.org) is working or has worked on a > new ISDN stack supporting timers for the call states. > > I have and using a openzap Q931StateTE stack which is basing on his > early work. A added and fixed some stuff on it, so that it works quite > good for basic calls on a Avaya-PRI as well as on a EWSD-PRI. > > The changes I did I have sent to Stefan, but got no new informations > from him since a few months. > > Maybe he is busy, maybe things are passed to Sangoma, maybe there is no > big interest in a native stack, dunno... > > > At least I have a need for a reliable slim isdn stack supporting Q931 > and if possible QSIG. > > It looks that libpri is the better choice as a future safe stack. But > I've never tested it with FS. > > > regards from sunny and hot germany > > Helmut > > > > On 09.07.2010 14:01, Jan Berger wrote: > >> Hi, >> >> >> >> Creating a new thread. >> >> >> >> If it is any interest for a native stack in FreeSWITCH we could address it >> later. It lacks the state-engine. Actually, last time I asked I was told >> someone in Germany was working on this? >> >> >> >> But, as I looked at it recently I concluded that with Sangoma supporting >> their own boards, and Digium maintaining libpri for theirs - we basically >> have covered a majority of the E1/T1 boards that is used in here. Correct >> > me > >> if I am wrong. >> >> > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFMNxfk4tZeNddg3dwRAqMxAJ0Vh5NnUY3HBEexz7ps8zR3GfGa6ACfTdhk > Z+nSJ/7WC12tJyG/F3UxIEY= > =5nmh > -----END PGP SIGNATURE----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From phone.bytes at gmail.com Fri Jul 9 07:00:52 2010 From: phone.bytes at gmail.com (Phone) Date: Fri, 09 Jul 2010 08:00:52 -0600 Subject: [Freeswitch-users] Native stacks In-Reply-To: <3C444E43-479A-4626-B26C-E25B2770A3A6@freeswitch.org> References: <5195B7332FD141F8950E89C65F3BE491@dell9400> <4C3717E4.602@ewetel.de> <70AA47CA5E7B4209A6ECA97A7F55CA23@dell9400> <4C37282D.2070602@gmail.com> <3C444E43-479A-4626-B26C-E25B2770A3A6@freeswitch.org> Message-ID: <4C372B94.2040605@gmail.com> As I recall, we already tried that...but we will pursue that solution further. Thanks! Brian West wrote: > because you need to SLEEP a few seconds before you answer just like you do in Asterisk. > > /b > > On Jul 9, 2010, at 8:46 AM, Phone wrote: > > >> I am interested in this discussion of the PRI stack. We are currently >> using libpri with a Samgoma B601DE. Things are working fine, except for >> the fact that we are not receiving Caller_ID "Name" on incoming calls. >> In examining traces, it appears that the name comes along milliseconds >> later in a separate facility message, and FS is not picking this up? >> >> I am curious if others using libpri with FS and Sangoma are getting >> Caller-ID "Name" delivery. Not sure if we are not configured quite >> right, or if there is some other libpri issue with FS. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From chris.chen2004 at gmail.com Fri Jul 9 07:07:35 2010 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 9 Jul 2010 10:07:35 -0400 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: <1333C24C-C580-48A2-92B3-92B8ABAA1E0A@freeswitch.org> References: <201007071616.41535.sos@sokhapkin.dyndns.org> <1333C24C-C580-48A2-92B3-92B8ABAA1E0A@freeswitch.org> Message-ID: I second that, I have one FS running on EC2 instance for production, so far no complaints yet. You do need good understanding of both NAT and FreeSWITCH. Chris On Fri, Jul 9, 2010 at 9:35 AM, Brian West wrote: > You also need to set ext-sip-ip and ext-rtp-ip also can you get me a pcap? > I know it works because I have personally done this on EC2 along with many > others. > > Its as simple as understanding NAT and FreeSWITCH. > > /b > > On Jul 9, 2010, at 2:57 AM, Vitalii Colosov wrote: > > > You should have run another ngrep to see if media traffic is OK between > FS and siptraffic: > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/236eb031/attachment.html From phone.bytes at gmail.com Fri Jul 9 07:10:55 2010 From: phone.bytes at gmail.com (Phone) Date: Fri, 09 Jul 2010 08:10:55 -0600 Subject: [Freeswitch-users] Native stacks In-Reply-To: References: <5195B7332FD141F8950E89C65F3BE491@dell9400> <4C3717E4.602@ewetel.de> <70AA47CA5E7B4209A6ECA97A7F55CA23@dell9400> <4C37282D.2070602@gmail.com> Message-ID: <4C372DEF.4090402@gmail.com> Jan Berger wrote: > Why are you using libpri with Sangoma boards? I am not that familiar with > Sangoma boards myself, but looking at FreeTDM I would expect this to be the > most optional path for Sangoma hardware? > > Jan > > > We were hoping to be able to get 2B Channel Transfer working, and understood that using libpri would allow this until the feature is added into the Sangoma Stack. There may have also been an issue with the card we are using, as it was new... not sure about that though. So far, no success with 2B Channel Transfer. Probably we should revisit FreeTMD, as this was a few months ago. From paul.gore.j at gmail.com Fri Jul 9 07:20:17 2010 From: paul.gore.j at gmail.com (paul gore) Date: Fri, 9 Jul 2010 10:20:17 -0400 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: <201007071616.41535.sos@sokhapkin.dyndns.org> Message-ID: As I mentioned in my first post I did that, I did everything as per ec2 wiki, incuding setting ext-ip in profiles. I have everything working fine, except no audio via siptraffic.com. I will do pcap and rtp trace via ngrep over weekend and update. On 7/9/10, k xd wrote: > I ever met same issue in EC2. > > Modify the sip_profile configuration file like "internal.xml" > Replace the below item with actual ip address: > > > Thanks, > Will > > On Fri, Jul 9, 2010 at 7:31 AM, paul gore wrote: > >> I got ngrep trace for port 5060 while making a call to a US number via >> siptraffic.com from X-Lite phone. I canceled the call after ~ 20 sec., I >> heard no audio not even ringing. >> Is there anything in this trace which can help identify the problem? >> >> 10.194.206.102:5060 - is my local EC2 IP >> 184.72.206.204:5060 - is my public EC2 IP >> 77.72.169.128:5060 - siptraffic.com proxy IP >> >> Thanks! >> >> >> >> 67.33.160.119:18294 -> 10.194.206.102:5060 >> INVITE >> sip:45517705678570 at myserver.comSIP/2.0..Via: >> SIP/2.0/UDP 192.168.0.8:29486 >> ;branch=z9hG4bK-d87543-f >> 524431af92cef56-1--d87543-;rport..Max-Forwards: 70..Contact: < >> sip:4000002 at 67.33.160.119:18027>..To: "45517 >> >> 709248570">..From: >> "4000002" >> >;tag=5f1ec15f..Call- >> ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Allow: >> INVITE, ACK, CANCEL, OPTIONS, BYE >> , REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: >> application/sdp..Proxy-Authorization: Digest user >> name="4000002",realm="myserver.com >> ",nonce="cf9019cc-f44a-4568-97d1-e9883fb1821f",uri="sip:45517705678570 at v >> ersafon.com >> ",response="57da72527524e0e065c9a3221bfadd38",cnonce="140f655ff3427f6ba3767ab7040231f3",nc=0000 >> 0001,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp >> 41150..Content-Length: 417....v=0..o=- >> 8 2 IN IP4 192.168.0.8..s=CounterPath X-Lite 3.0..c=IN IP4 >> 192.168.0.8..t=0 0..m=audio 46298 RTP/AVP 107 >> 119 100 106 0 105 98 8 101..a=alt:1 1 : tomYv1/D Yont/s+3 192.168.0.8 >> 46298..a=fmtp:101 0-15..a=rtpmap:107 >> BV32/16000..a=rtpmap:119 BV32-FEC/16000..a=rtpmap:100 >> SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap >> :105 SPEEX-FEC/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:101 >> telephone-event/8000..a=sendrecv.. >> # >> U 10.194.206.102:5060 -> 67.33.160.119:18294 >> SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.0.8:29486 >> ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;r >> port=18294;received=67.33.160.119..From: "4000002" < >> sip:4000002 at myserver.com >;tag=5f1ec15f..To: >> "455177092 >> 48570" >> >..Call-ID: >> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 I >> NVITE..User-Agent: myserver..Content-Length: 0.... >> # >> U 10.194.206.102:5080 -> 77.72.169.128:5060 >> INVITE >> sip:0017705678570 at sip.siptraffic.comSIP/2.0..Via: >> SIP/2.0/UDP 184.72.206.204:5080 >> ;rport;branch=z9h >> G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < >> sip:0014444295793 at 184.72.206.204 >> >;tag=18853e82KDe7j. >> .To: >> >..Call-ID: >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 >> 2 INVITE..Contact: > ;transport=udp;gw=voicetrading.com>..User-A >> gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, >> UPDATE, INFO, REGISTER, REFER, NOTIFY.. >> Supported: timer, precondition, path, replaces..Allow-Events: talk, >> refer..Content-Type: application/sdp.. >> Content-Disposition: session..Content-Length: 295..X-FS-Support: >> update_display..Remote-Party-ID: "4000002 >> " > >;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH >> 1278518039 >> 1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 >> 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8 >> 3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 >> GSM/8000..a=rtpmap:101 telephone-event/80 >> 00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20.. >> # >> U 77.72.169.128:5060 -> 10.194.206.102:5080 >> SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 >> ;rport;branch=z9hG4bKBU626KBp16t5Q..From >> : "4000002" >> >;tag=18853e82KDe7j..To: >> >> m>;tag=20113ac4c230cd6412168..Contact: >> sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c >> 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip >> Registrar/Proxy Server)..Allow: ACK,BYE,C >> ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: >> application/sdp..Content-Length: 198....v=0..o=C >> ARRIER 1278549617 1278549617 IN IP4 77.72.168.40..s=SIP Call..c=IN IP4 >> 77.72.168.40..t=0 0..m=audio 57672 >> RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 >> telephone-event/8000..a=ptime:20.. >> # >> U 10.194.206.102:5060 -> 67.33.160.119:18294 >> SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.8:29486 >> ;branch=z9hG4bK-d87543-f524431af92cef56-1- >> -d87543-;rport=18294;received=67.33.160.119..From: "4000002" < >> sip:4000002 at myserver.com >;tag=5f1ec15f..To: >> "45517705678570" >> >;tag=BXgB1FZBUZ3Da..Call-ID: >> ZDAzODE0Y2JkZjYzODE5NmVmNjk >> zMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Contact: >> ..User-A >> gent: myserver..Accept: application/sdp..Allow: INVITE, ACK, BYE, >> CANCEL, >> OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, >> precondition, path, replaces..Allow-Events: >> talk, presence, dialog, line-seize, call-info, sla, >> include-session-description, presence.winfo, message- >> summary, refer..Content-Type: application/sdp..Content-Disposition: >> session..Content-Length: 251..Remote-P >> arty-ID: "45517705678570" >> >> >;party=calling;privacy=off;screen=no....v=0.. >> o=FreeSWITCH 1278530815 1278530816 IN IP4 >> 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m= >> audio 18788 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 >> telephone-event/8000..a=fmtp:101 0-16..a=sil >> enceSupp:off - - - -..a=ptime:20.. >> # >> >> >> >> U 77.72.169.128:5060 -> 10.194.206.102:5080 >> SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 >> ;rport;branch=z9hG4bKBU626KBp16t5Q..From >> : "4000002" >> >;tag=18853e82KDe7j..To: >> >> m>;tag=20113ac4c230cd6412168..Contact: >> sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c >> 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip >> Registrar/Proxy Server)..Allow: ACK,BYE,C >> ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: >> application/sdp..Content-Length: 204....v=0..o=C >> ARRIER 1278549619 1278549619 IN IP4 208.167.230.118..s=SIP Call..c=IN >> IP4 >> 208.167.230.118..t=0 0..m=audio >> 57786 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 >> telephone-event/8000..a=ptime:20.. >> # >> >> U 67.33.160.119:18294 -> 10.194.206.102:5060 >> .... >> # >> U 67.33.160.119:18294 -> 10.194.206.102:5060 >> CANCEL >> sip:45517705678570 at myserver.comSIP/2.0..Via: >> SIP/2.0/UDP 192.168.0.8:29486 >> ;branch=z9hG4bK-d87543-f >> 524431af92cef56-1--d87543-;rport..To: "45517705678570"< >> sip:45517705678570 at myserver.com >> >..From: >> "4000002"> ip:4000002 at myserver.com >> >;tag=5f1ec15f..Call-ID: >> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 CANC >> EL..Proxy-Authorization: Digest username="4000002",realm="myserver.com >> ",nonce="cf9019cc-f44a-4568-97d1-e98 >> >> 83fb1821f",uri="sip:45517705678570 at myserver.com >> ",response="46c7e289f7490c807565c561699b03d6",cnonce="a226c >> >> 55446c605ee229f045602b29135",nc=00000002,qop=auth,algorithm=MD5..User-Agent: >> X-Lite release 1011s stamp 41 >> 150..Content-Length: 0.... >> # >> U 10.194.206.102:5060 -> 67.33.160.119:18294 >> SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.8:29486 >> ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;rport >> =18294;received=67.33.160.119..From: "4000002" >> >;tag=5f1ec15f..To: >> "4551770567857 >> 0" > >;tag=BXgB1FZBUZ3Da..Call-ID: >> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkY >> mY...CSeq: 2 CANCEL..Content-Length: 0.... >> # >> U 10.194.206.102:5060 -> 67.33.160.119:18294 >> SIP/2.0 487 Request Terminated..Via: SIP/2.0/UDP 192.168.0.8:29486 >> ;branch=z9hG4bK-d87543-f524431af92cef56- >> 1--d87543-;rport=18294;received=67.33.160.119..From: "4000002" < >> sip:4000002 at myserver.com >;tag=5f1ec15f..To >> : "45517705678570" >> >;tag=BXgB1FZBUZ3Da..Call-ID: >> ZDAzODE0Y2JkZjYzODE5NmVmN >> jkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..User-Agent: myserver..Allow: >> INVITE, >> ACK, BYE, CANCEL, OPTIONS, MESSA >> GE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, >> SUBSCRIBE..Supported: >> timer, precondition, path, repla >> ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, >> include-session-description, presen >> ce.winfo, message-summary, refer..Content-Length: 0.... >> # >> U 10.194.206.102:5080 -> 77.72.169.128:5060 >> CANCEL >> sip:0017705678570 at sip.siptraffic.comSIP/2.0..Via: >> SIP/2.0/UDP 184.72.206.204:5080 >> ;rport;branch=z9h >> G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < >> sip:0014444295793 at 184.72.206.204 >> >;tag=18853e82KDe7j. >> .To: >> >..Call-ID: >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 >> 2 CANCEL..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length: >> 0.... >> # >> >> U 67.33.160.119:18294 -> 10.194.206.102:5060 >> ACK sip:45517705678570 at myserver.com >> SIP/2.0..Via: SIP/2.0/UDP >> 192.168.0.8:29486 >> ;branch=z9hG4bK-d87543-f524 >> 431af92cef56-1--d87543-;rport..To: "45517705678570" < >> sip:45517705678570 at myserver.com >> >;tag=BXgB1FZBUZ3Da..F >> rom: "4000002"> >;tag=5f1ec15f..Call-ID: >> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYm >> Y...CSeq: 2 ACK..Content-Length: 0.... >> # >> >> U 77.72.169.128:5060 -> 10.194.206.102:5080 >> SIP/2.0 200 Ok..Via: SIP/2.0/UDP >> 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From: >> "4000002" > ip:0014444295793 at 184.72.206.204 >> >;tag=18853e82KDe7j..To: >> >..Contact: >> s >> ip:0017705678570 at 77.72.169.128:5060..Call-ID: >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 CANCEL >> ..Server: (Very nice Sip Registrar/Proxy Server)..Allow: >> ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSA >> GE..Content-Length: 0.... >> # >> U 77.72.169.128:5060 -> 10.194.206.102:5080 >> SIP/2.0 487 Request terminated..Via: SIP/2.0/UDP 184.72.206.204:5080 >> ;rport;branch=z9hG4bKBU626KBp16t5Q..Fr >> om: "4000002" >> >;tag=18853e82KDe7j..To: >> > com>..Contact: sip:0017705678570 at 77.72.169.128:5060..Call-ID: >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: >> 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow: >> ACK,BYE,CANCEL,INVITE,REGISTER,OP >> TIONS,INFO,MESSAGE..Content-Length: 0.... >> # >> U 10.194.206.102:5080 -> 77.72.169.128:5060 >> ACK >> sip:0017705678570 at sip.siptraffic.comSIP/2.0..Via: >> SIP/2.0/UDP 184.72.206.204:5080 >> ;rport;branch=z9hG4b >> KBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < >> sip:0014444295793 at 184.72.206.204 >> >;tag=18853e82KDe7j..To >> : >> >..Call-ID: >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 A >> CK..Content-Length: 0.... >> # >> >> >> >> >> >> >> >> >> On Wed, Jul 7, 2010 at 5:50 PM, paul gore wrote: >> >>> Seems like siptraffic uses 6 ip addresses for media, can that be the >>> problem? Is there any setting in a gateway config xml which helps with >>> that? >>> I will do ngrep thing and update. >>> >>> On 7/7/10, paul gore wrote: >>> > This provider does work on another box which is not natted as ec2. >>> > Most puzzling here though is why call originaion via api even not >>> > going via siptraffic still gets no audio. >>> > >>> > On 7/7/10, Tony Graziano wrote: >>> >> You should try from a standalone or local installation to ensure it >>> works >>> >> with this provider and your account before you attempt to run it on >>> >> ec2 >>> >> (imo). >>> >> >>> >> On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin >>> >> wrote: >>> >> >>> >>> What "doesn't work" means? It could be (and most likely is not) >>> >>> FS-related >>> >>> problem >>> >>> >>> >>> On Wednesday 07 July 2010, Madovsky wrote: >>> >>> > I had same problem from this provider without to explain why. >>> >>> > One day it works, another day it doesn't, their support is crap... >>> >>> > >>> >>> > ----- Original Message ----- >>> >>> > From: Anthony Minessale >>> >>> > To: freeswitch-users at lists.freeswitch.org >>> >>> > Sent: Wednesday, July 07, 2010 2:37 PM >>> >>> > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on >>> >>> > outgoing >>> >>> > calls >>> >>> > >>> >>> > >>> >>> > not really, not with so little information. >>> >>> > >>> >>> > >>> >>> > >>> >>> > On Wed, Jul 7, 2010 at 1:30 PM, paul gore >>> >>> wrote: >>> >>> > >>> >>> > Firewall is configured according to the wiki, I also tried to >>> open >>> >>> all >>> >>> > udp ports, issue persists. >>> >>> > Actually the problem became more complex - outgoing calls don't >>> >>> > work >>> >>> > with one particular termination provider, siptraffic.com , any >>> >>> > ideas >>> >>> > why? >>> >>> > Outgoing calls also don't work when originating a call via js >>> >>> > script >>> >>> > or via FS api. Any clues on that one? >>> >>> > >>> >>> > On 7/6/10, paul gore wrote: >>> >>> > > Hi there, >>> >>> > > I am experimenting with FS on EC2, I like results, but stuck >>> on >>> >>> weird >>> >>> > > audio issue - I followed FreeSwitch EC2 wiki article and >>> >>> > modified >>> >>> > > internal profile >>> >>> > > and vars.xml accordingly, but unfortunately still cannot get >>> it >>> >>> > > working. Incoming and outgoing calls made using a SIP phone >>> >>> > to >>> >>> > FS >>> >>> > > extensions work just fine. As well as calls to FS from PSTN. >>> But >>> >>> > > calls to PSTN via gateways result in no audio at all, no >>> >>> > ring, >>> >>> > > nothing, SIP signaling goes through OK. Sofia status profile >>> >>> > shows >>> >>> > > correct values for Ext-RTP-IP for both profiles - >>> >>> > > my static public IP, RTP-IP shows local IP. >>> >>> > > Any thoughts on that? Anybody can share working profile >>> >>> configuration >>> >>> > > may be? >>> >>> > > Please help, I really need to get this going. >>> >>> > > >>> >>> > > Thanks. >>> >>> >>> > >>> >>> > _______________________________________________ >>> >>> > FreeSWITCH-users mailing list >>> >>> > FreeSWITCH-users at 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 http://www.freeswitch.org/ >>> >>> > ClueCon http://www.cluecon.com/ >>> >>> > Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> > >>> >>> > AIM: anthm >>> >>> > >>> >>> > MSN:anthony_minessale at hotmail.com >>> >>> > >>> >>> > >>> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> >>> > >>> >>> > IRC: irc.freenode.net #freeswitch >>> >>> > >>> >>> > FreeSWITCH Developer Conference >>> >>> > >>> >>> > sip:888 at conference.freeswitch.org >>> >>> > >>> >>> > >>> >>> > googletalk:conf+888 at conference.freeswitch.org >>> >>> > >>> >>> > pstn:+19193869900 >>> >>> > >>> >>> > >>> >>> > >>> >>> > >>> >>> >>> --------------------------------------------------------------------------- >>> >>> > --- >>> >>> >>> > >>> >>> > >>> >>> > _______________________________________________ >>> >>> > FreeSWITCH-users mailing list >>> >>> > FreeSWITCH-users at 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 at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >>> >> >>> >> >>> >> >>> >> -- >>> >> ====================== >>> >> Tony Graziano, Manager >>> >> Telephone: 434.984.8430 >>> >> sip: tgraziano at voice.myitdepartment.net >>> >> Fax: 434.984.8431 >>> >> >>> >> Email: tgraziano at myitdepartment.net >>> >> >>> >> LAN/Telephony/Security and Control Systems Helpdesk: >>> >> Telephone: 434.984.8426 >>> >> sip: helpdesk at voice.myitdepartment.net >>> >> Fax: 434.984.8427 >>> >> >>> >> Helpdesk Contract Customers: >>> >> http://www.myitdepartment.net/gethelp/ >>> >> >>> >> Why do mathematicians always confuse Halloween and Christmas? >>> >> Because 31 Oct = 25 Dec. >>> >> >>> > >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From anthony.minessale at gmail.com Fri Jul 9 07:57:14 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 9 Jul 2010 09:57:14 -0500 Subject: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls In-Reply-To: References: <201007071616.41535.sos@sokhapkin.dyndns.org> Message-ID: in the future post logs to pastebin, notice how big and ugly this thread is now quoting your log and toting it around on each reply. Also consider coming to IRC for real-time interaction and post the conclusion here once you figure it out, On Fri, Jul 9, 2010 at 9:20 AM, paul gore wrote: > As I mentioned in my first post I did that, I did everything as per > ec2 wiki, incuding setting ext-ip in profiles. > I have everything working fine, except no audio via siptraffic.com. > I will do pcap and rtp trace via ngrep over weekend and update. > > On 7/9/10, k xd wrote: > > I ever met same issue in EC2. > > > > Modify the sip_profile configuration file like "internal.xml" > > Replace the below item with actual ip address: > > > > > > Thanks, > > Will > > > > On Fri, Jul 9, 2010 at 7:31 AM, paul gore wrote: > > > >> I got ngrep trace for port 5060 while making a call to a US number via > >> siptraffic.com from X-Lite phone. I canceled the call after ~ 20 sec., > I > >> heard no audio not even ringing. > >> Is there anything in this trace which can help identify the problem? > >> > >> 10.194.206.102:5060 - is my local EC2 IP > >> 184.72.206.204:5060 - is my public EC2 IP > >> 77.72.169.128:5060 - siptraffic.com proxy IP > >> > >> Thanks! > >> > >> > >> > >> 67.33.160.119:18294 -> 10.194.206.102:5060 > >> INVITE > >> sip:45517705678570 at myserver.com < > sip%3A45517705678570 at myserver.com > >SIP/2.0..Via: > >> SIP/2.0/UDP 192.168.0.8:29486 > >> ;branch=z9hG4bK-d87543-f > >> 524431af92cef56-1--d87543-;rport..Max-Forwards: 70..Contact: < > >> sip:4000002 at 67.33.160.119:18027>..To: "45517 > >> > >> 709248570" > > >>..From: > >> "4000002" < > sip%3A4000002 at myserver.com > > >> >;tag=5f1ec15f..Call- > >> ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 > INVITE..Allow: > >> INVITE, ACK, CANCEL, OPTIONS, BYE > >> , REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: > >> application/sdp..Proxy-Authorization: Digest user > >> name="4000002",realm="myserver.com > >> ",nonce="cf9019cc-f44a-4568-97d1-e9883fb1821f",uri="sip:45517705678570 at v > >> ersafon.com > >> > ",response="57da72527524e0e065c9a3221bfadd38",cnonce="140f655ff3427f6ba3767ab7040231f3",nc=0000 > >> 0001,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp > >> 41150..Content-Length: 417....v=0..o=- > >> 8 2 IN IP4 192.168.0.8..s=CounterPath X-Lite 3.0..c=IN IP4 > >> 192.168.0.8..t=0 0..m=audio 46298 RTP/AVP 107 > >> 119 100 106 0 105 98 8 101..a=alt:1 1 : tomYv1/D Yont/s+3 192.168.0.8 > >> 46298..a=fmtp:101 0-15..a=rtpmap:107 > >> BV32/16000..a=rtpmap:119 BV32-FEC/16000..a=rtpmap:100 > >> SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap > >> :105 SPEEX-FEC/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:101 > >> telephone-event/8000..a=sendrecv.. > >> # > >> U 10.194.206.102:5060 -> 67.33.160.119:18294 > >> SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.0.8:29486 > >> ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;r > >> port=18294;received=67.33.160.119..From: "4000002" < > >> sip:4000002 at myserver.com < > sip%3A4000002 at myserver.com > >>;tag=5f1ec15f..To: > >> "455177092 > >> 48570" > >> < > sip%3A45517705678570 at myserver.com > >>..Call-ID: > >> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 I > >> NVITE..User-Agent: myserver..Content-Length: 0.... > >> # > >> U 10.194.206.102:5080 -> 77.72.169.128:5060 > >> INVITE > >> sip:0017705678570 at sip.siptraffic.com > > >SIP/2.0..Via: > >> SIP/2.0/UDP 184.72.206.204:5080 > >> ;rport;branch=z9h > >> G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < > >> sip:0014444295793 at 184.72.206.204 < > sip%3A0014444295793 at 184.72.206.204 > > >> >;tag=18853e82KDe7j. > >> .To: > >> > > >>..Call-ID: > >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 > >> 2 INVITE..Contact: >> ;transport=udp;gw=voicetrading.com>..User-A > >> gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, > >> UPDATE, INFO, REGISTER, REFER, NOTIFY.. > >> Supported: timer, precondition, path, replaces..Allow-Events: talk, > >> refer..Content-Type: application/sdp.. > >> Content-Disposition: session..Content-Length: 295..X-FS-Support: > >> update_display..Remote-Party-ID: "4000002 > >> " > >> > >>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH > >> 1278518039 > >> 1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4 > >> 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8 > >> 3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 > >> GSM/8000..a=rtpmap:101 telephone-event/80 > >> 00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20.. > >> # > >> U 77.72.169.128:5060 -> 10.194.206.102:5080 > >> SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 > >> ;rport;branch=z9hG4bKBU626KBp16t5Q..From > >> : "4000002" > >> < > sip%3A0014444295793 at 184.72.206.204 > >>;tag=18853e82KDe7j..To: > >> > > > > >> m>;tag=20113ac4c230cd6412168..Contact: > >> sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c > >> 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip > >> Registrar/Proxy Server)..Allow: ACK,BYE,C > >> ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: > >> application/sdp..Content-Length: 198....v=0..o=C > >> ARRIER 1278549617 1278549617 IN IP4 77.72.168.40..s=SIP Call..c=IN IP4 > >> 77.72.168.40..t=0 0..m=audio 57672 > >> RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 > >> telephone-event/8000..a=ptime:20.. > >> # > >> U 10.194.206.102:5060 -> 67.33.160.119:18294 > >> SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.8:29486 > >> ;branch=z9hG4bK-d87543-f524431af92cef56-1- > >> -d87543-;rport=18294;received=67.33.160.119..From: "4000002" < > >> sip:4000002 at myserver.com < > sip%3A4000002 at myserver.com > >>;tag=5f1ec15f..To: > >> "45517705678570" > >> < > sip%3A45517705678570 at myserver.com > >>;tag=BXgB1FZBUZ3Da..Call-ID: > >> ZDAzODE0Y2JkZjYzODE5NmVmNjk > >> zMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Contact: > >> ..User-A > >> gent: myserver..Accept: application/sdp..Allow: INVITE, ACK, BYE, > >> CANCEL, > >> OPTIONS, MESSAGE, UPDATE, INFO, > >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, > >> precondition, path, replaces..Allow-Events: > >> talk, presence, dialog, line-seize, call-info, sla, > >> include-session-description, presence.winfo, message- > >> summary, refer..Content-Type: application/sdp..Content-Disposition: > >> session..Content-Length: 251..Remote-P > >> arty-ID: "45517705678570" > >> > > > > >> >;party=calling;privacy=off;screen=no....v=0.. > >> o=FreeSWITCH 1278530815 1278530816 IN IP4 > >> 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m= > >> audio 18788 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 > >> telephone-event/8000..a=fmtp:101 0-16..a=sil > >> enceSupp:off - - - -..a=ptime:20.. > >> # > >> > >> > >> > >> U 77.72.169.128:5060 -> 10.194.206.102:5080 > >> SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080 > >> ;rport;branch=z9hG4bKBU626KBp16t5Q..From > >> : "4000002" > >> < > sip%3A0014444295793 at 184.72.206.204 > >>;tag=18853e82KDe7j..To: > >> > > > > >> m>;tag=20113ac4c230cd6412168..Contact: > >> sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c > >> 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip > >> Registrar/Proxy Server)..Allow: ACK,BYE,C > >> ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type: > >> application/sdp..Content-Length: 204....v=0..o=C > >> ARRIER 1278549619 1278549619 IN IP4 208.167.230.118..s=SIP Call..c=IN > >> IP4 > >> 208.167.230.118..t=0 0..m=audio > >> 57786 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 > >> telephone-event/8000..a=ptime:20.. > >> # > >> > >> U 67.33.160.119:18294 -> 10.194.206.102:5060 > >> .... > >> # > >> U 67.33.160.119:18294 -> 10.194.206.102:5060 > >> CANCEL > >> sip:45517705678570 at myserver.com < > sip%3A45517705678570 at myserver.com > >SIP/2.0..Via: > >> SIP/2.0/UDP 192.168.0.8:29486 > >> ;branch=z9hG4bK-d87543-f > >> 524431af92cef56-1--d87543-;rport..To: "45517705678570"< > >> sip:45517705678570 at myserver.com > >> > >>..From: > >> "4000002" >> ip:4000002 at myserver.com > >> > >>;tag=5f1ec15f..Call-ID: > >> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 CANC > >> EL..Proxy-Authorization: Digest username="4000002",realm=" > myserver.com > >> ",nonce="cf9019cc-f44a-4568-97d1-e98 > >> > >> 83fb1821f",uri="sip:45517705678570 at myserver.com > > > >> ",response="46c7e289f7490c807565c561699b03d6",cnonce="a226c > >> > >> > 55446c605ee229f045602b29135",nc=00000002,qop=auth,algorithm=MD5..User-Agent: > >> X-Lite release 1011s stamp 41 > >> 150..Content-Length: 0.... > >> # > >> U 10.194.206.102:5060 -> 67.33.160.119:18294 > >> SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.8:29486 > >> ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;rport > >> =18294;received=67.33.160.119..From: "4000002" > >> < > sip%3A4000002 at myserver.com > >>;tag=5f1ec15f..To: > >> "4551770567857 > >> 0" > >> > >>;tag=BXgB1FZBUZ3Da..Call-ID: > >> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkY > >> mY...CSeq: 2 CANCEL..Content-Length: 0.... > >> # > >> U 10.194.206.102:5060 -> 67.33.160.119:18294 > >> SIP/2.0 487 Request Terminated..Via: SIP/2.0/UDP 192.168.0.8:29486 > >> ;branch=z9hG4bK-d87543-f524431af92cef56- > >> 1--d87543-;rport=18294;received=67.33.160.119..From: "4000002" < > >> sip:4000002 at myserver.com < > sip%3A4000002 at myserver.com > >>;tag=5f1ec15f..To > >> : "45517705678570" > >> < > sip%3A45517705678570 at myserver.com > >>;tag=BXgB1FZBUZ3Da..Call-ID: > >> ZDAzODE0Y2JkZjYzODE5NmVmN > >> jkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..User-Agent: myserver..Allow: > >> INVITE, > >> ACK, BYE, CANCEL, OPTIONS, MESSA > >> GE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, > >> SUBSCRIBE..Supported: > >> timer, precondition, path, repla > >> ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, > >> include-session-description, presen > >> ce.winfo, message-summary, refer..Content-Length: 0.... > >> # > >> U 10.194.206.102:5080 -> 77.72.169.128:5060 > >> CANCEL > >> sip:0017705678570 at sip.siptraffic.com > > >SIP/2.0..Via: > >> SIP/2.0/UDP 184.72.206.204:5080 > >> ;rport;branch=z9h > >> G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < > >> sip:0014444295793 at 184.72.206.204 < > sip%3A0014444295793 at 184.72.206.204 > > >> >;tag=18853e82KDe7j. > >> .To: > >> > > >>..Call-ID: > >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647 > >> 2 CANCEL..Reason: > Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length: > >> 0.... > >> # > >> > >> U 67.33.160.119:18294 -> 10.194.206.102:5060 > >> ACK sip:45517705678570 at myserver.com > >> >SIP/2.0..Via: > SIP/2.0/UDP > >> 192.168.0.8:29486 > >> ;branch=z9hG4bK-d87543-f524 > >> 431af92cef56-1--d87543-;rport..To: "45517705678570" < > >> sip:45517705678570 at myserver.com < > sip%3A45517705678570 at myserver.com > > >> >;tag=BXgB1FZBUZ3Da..F > >> rom: "4000002" > >> > >>;tag=5f1ec15f..Call-ID: > >> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYm > >> Y...CSeq: 2 ACK..Content-Length: 0.... > >> # > >> > >> U 77.72.169.128:5060 -> 10.194.206.102:5080 > >> SIP/2.0 200 Ok..Via: SIP/2.0/UDP > >> 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From: > >> "4000002" >> ip:0014444295793 at 184.72.206.204 > >> > >>;tag=18853e82KDe7j..To: > >> > > >>..Contact: > >> s > >> ip:0017705678570 at 77.72.169.128:5060..Call-ID: > >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 CANCEL > >> ..Server: (Very nice Sip Registrar/Proxy Server)..Allow: > >> ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSA > >> GE..Content-Length: 0.... > >> # > >> U 77.72.169.128:5060 -> 10.194.206.102:5080 > >> SIP/2.0 487 Request terminated..Via: SIP/2.0/UDP 184.72.206.204:5080 > >> ;rport;branch=z9hG4bKBU626KBp16t5Q..Fr > >> om: "4000002" > >> < > sip%3A0014444295793 at 184.72.206.204 > >>;tag=18853e82KDe7j..To: > >> >> com>..Contact: sip:0017705678570 at 77.72.169.128:5060..Call-ID: > >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: > >> 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy > Server)..Allow: > >> ACK,BYE,CANCEL,INVITE,REGISTER,OP > >> TIONS,INFO,MESSAGE..Content-Length: 0.... > >> # > >> U 10.194.206.102:5080 -> 77.72.169.128:5060 > >> ACK > >> sip:0017705678570 at sip.siptraffic.com > > >SIP/2.0..Via: > >> SIP/2.0/UDP 184.72.206.204:5080 > >> ;rport;branch=z9hG4b > >> KBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" < > >> sip:0014444295793 at 184.72.206.204 < > sip%3A0014444295793 at 184.72.206.204 > > >> >;tag=18853e82KDe7j..To > >> : > >> > > >>..Call-ID: > >> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 A > >> CK..Content-Length: 0.... > >> # > >> > >> > >> > >> > >> > >> > >> > >> > >> On Wed, Jul 7, 2010 at 5:50 PM, paul gore > wrote: > >> > >>> Seems like siptraffic uses 6 ip addresses for media, can that be the > >>> problem? Is there any setting in a gateway config xml which helps with > >>> that? > >>> I will do ngrep thing and update. > >>> > >>> On 7/7/10, paul gore wrote: > >>> > This provider does work on another box which is not natted as ec2. > >>> > Most puzzling here though is why call originaion via api even not > >>> > going via siptraffic still gets no audio. > >>> > > >>> > On 7/7/10, Tony Graziano wrote: > >>> >> You should try from a standalone or local installation to ensure it > >>> works > >>> >> with this provider and your account before you attempt to run it on > >>> >> ec2 > >>> >> (imo). > >>> >> > >>> >> On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin > >>> >> wrote: > >>> >> > >>> >>> What "doesn't work" means? It could be (and most likely is not) > >>> >>> FS-related > >>> >>> problem > >>> >>> > >>> >>> On Wednesday 07 July 2010, Madovsky wrote: > >>> >>> > I had same problem from this provider without to explain why. > >>> >>> > One day it works, another day it doesn't, their support is > crap... > >>> >>> > > >>> >>> > ----- Original Message ----- > >>> >>> > From: Anthony Minessale > >>> >>> > To: freeswitch-users at lists.freeswitch.org > >>> >>> > Sent: Wednesday, July 07, 2010 2:37 PM > >>> >>> > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on > >>> >>> > outgoing > >>> >>> > calls > >>> >>> > > >>> >>> > > >>> >>> > not really, not with so little information. > >>> >>> > > >>> >>> > > >>> >>> > > >>> >>> > On Wed, Jul 7, 2010 at 1:30 PM, paul gore < > paul.gore.j at gmail.com> > >>> >>> wrote: > >>> >>> > > >>> >>> > Firewall is configured according to the wiki, I also tried to > >>> open > >>> >>> all > >>> >>> > udp ports, issue persists. > >>> >>> > Actually the problem became more complex - outgoing calls > don't > >>> >>> > work > >>> >>> > with one particular termination provider, siptraffic.com , > any > >>> >>> > ideas > >>> >>> > why? > >>> >>> > Outgoing calls also don't work when originating a call via js > >>> >>> > script > >>> >>> > or via FS api. Any clues on that one? > >>> >>> > > >>> >>> > On 7/6/10, paul gore wrote: > >>> >>> > > Hi there, > >>> >>> > > I am experimenting with FS on EC2, I like results, but > stuck > >>> on > >>> >>> weird > >>> >>> > > audio issue - I followed FreeSwitch EC2 wiki article and > >>> >>> > modified > >>> >>> > > internal profile > >>> >>> > > and vars.xml accordingly, but unfortunately still cannot > get > >>> it > >>> >>> > > working. Incoming and outgoing calls made using a SIP phone > >>> >>> > to > >>> >>> > FS > >>> >>> > > extensions work just fine. As well as calls to FS from > PSTN. > >>> But > >>> >>> > > calls to PSTN via gateways result in no audio at all, no > >>> >>> > ring, > >>> >>> > > nothing, SIP signaling goes through OK. Sofia status > profile > >>> >>> > shows > >>> >>> > > correct values for Ext-RTP-IP for both profiles - > >>> >>> > > my static public IP, RTP-IP shows local IP. > >>> >>> > > Any thoughts on that? Anybody can share working profile > >>> >>> configuration > >>> >>> > > may be? > >>> >>> > > Please help, I really need to get this going. > >>> >>> > > > >>> >>> > > Thanks. > >>> > >>> >>> > > >>> >>> > _______________________________________________ > >>> >>> > FreeSWITCH-users mailing list > >>> >>> > FreeSWITCH-users at 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 http://www.freeswitch.org/ > >>> >>> > ClueCon http://www.cluecon.com/ > >>> >>> > Twitter: http://twitter.com/FreeSWITCH_wire > >>> >>> > > >>> >>> > AIM: anthm > >>> >>> > > >>> >>> > MSN:anthony_minessale at hotmail.com > > > > >>> > > > > >>> > > >>> >>> > > >>> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >>> > > > > >>> > > >>> >>> > IRC: irc.freenode.net #freeswitch > >>> >>> > > >>> >>> > FreeSWITCH Developer Conference > >>> >>> > > >>> >>> > sip:888 at conference.freeswitch.org > > > > >>> > > > > >>> > > >>> >>> > > >>> >>> > googletalk:conf+888 at conference.freeswitch.org > > > > >>> > > > > >>> > > >>> >>> > pstn:+19193869900 > >>> >>> > > >>> >>> > > >>> >>> > > >>> >>> > > >>> >>> > >>> > --------------------------------------------------------------------------- > >>> >>> > --- > >>> > >>> >>> > > >>> >>> > > >>> >>> > _______________________________________________ > >>> >>> > FreeSWITCH-users mailing list > >>> >>> > FreeSWITCH-users at 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 at lists.freeswitch.org > >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> >>> UNSUBSCRIBE: > >>> http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> >>> http://www.freeswitch.org > >>> >>> > >>> >> > >>> >> > >>> >> > >>> >> -- > >>> >> ====================== > >>> >> Tony Graziano, Manager > >>> >> Telephone: 434.984.8430 > >>> >> sip: tgraziano at voice.myitdepartment.net > >>> >> Fax: 434.984.8431 > >>> >> > >>> >> Email: tgraziano at myitdepartment.net > >>> >> > >>> >> LAN/Telephony/Security and Control Systems Helpdesk: > >>> >> Telephone: 434.984.8426 > >>> >> sip: helpdesk at voice.myitdepartment.net > >>> >> Fax: 434.984.8427 > >>> >> > >>> >> Helpdesk Contract Customers: > >>> >> http://www.myitdepartment.net/gethelp/ > >>> >> > >>> >> Why do mathematicians always confuse Halloween and Christmas? > >>> >> Because 31 Oct = 25 Dec. > >>> >> > >>> > > >>> > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/6bead1df/attachment-0001.html From helmut.kuper at ewetel.de Fri Jul 9 08:03:28 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 09 Jul 2010 17:03:28 +0200 Subject: [Freeswitch-users] Native stacks In-Reply-To: <70AA47CA5E7B4209A6ECA97A7F55CA23@dell9400> References: <5195B7332FD141F8950E89C65F3BE491@dell9400> <4C3717E4.602@ewetel.de> <70AA47CA5E7B4209A6ECA97A7F55CA23@dell9400> Message-ID: <4C373A40.1020207@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Jan, sounds great to me SS7 and QSIG ... would be too nice. I use Sangoma A101 and A104d. regards On 09.07.2010 15:16, Jan Berger wrote: > What E1/T1 hardware are you using? > > Libpri is maintained by a larger group. They had started to get QSIG last > time I checked - they also have basic SS7 for simple ISUP calls in libss7. > The stack itself is a maintenance problem, but Digium has the strength to > maintain it - and in the end that is all that matters unless you want to > maintain things yourself. > > Jan -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMNzo/4tZeNddg3dwRAnS2AJkBBNLQMY56ofyoLdlbbLvyJzoQfgCdGO1j 4wRzrKPmE0cud5Kg5SR/GPI= =yZ8N -----END PGP SIGNATURE----- From anthony.minessale at gmail.com Fri Jul 9 08:18:23 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 9 Jul 2010 10:18:23 -0500 Subject: [Freeswitch-users] Native stacks In-Reply-To: <4C373A40.1020207@ewetel.de> References: <5195B7332FD141F8950E89C65F3BE491@dell9400> <4C3717E4.602@ewetel.de> <70AA47CA5E7B4209A6ECA97A7F55CA23@dell9400> <4C373A40.1020207@ewetel.de> Message-ID: I want the native stack to succeed because our goal was to have an open source BSD licensed free ISDN stack so we can put an end to people selling it for ungodly fees. The libpri is fine to use if a user assembles it himself but ultimately it's GPL and there is a grey-area license conflict with using with FS. Technically OpenZAP/FreeTDM is BSD and compat with libpri but according to the greedy GPL, when you load it the GPL infects the whole code and makes it also GPL. FS is MPL and is happily compat with OpenZAP/FreeTDM but there is a philosophical debate as to if the BSD lib in the middle that completely abstracts the 2 entities, protects FS from the GPL FreeTDM is still OpenZAP, just with another name. Sangoma is only working on their own modules for FreeTDM and the API as a whole still supports everything OpenZAP does. I think stkn will merge his stack back into FreeTDM and it can continue to be developed for those who don't have Sangoma cards. On Fri, Jul 9, 2010 at 10:03 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi Jan, > > sounds great to me SS7 and QSIG ... would be too nice. > > I use Sangoma A101 and A104d. > > regards > > On 09.07.2010 15:16, Jan Berger wrote: > > What E1/T1 hardware are you using? > > > > Libpri is maintained by a larger group. They had started to get QSIG last > > time I checked - they also have basic SS7 for simple ISUP calls in > libss7. > > The stack itself is a maintenance problem, but Digium has the strength to > > maintain it - and in the end that is all that matters unless you want to > > maintain things yourself. > > > > Jan > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFMNzo/4tZeNddg3dwRAnS2AJkBBNLQMY56ofyoLdlbbLvyJzoQfgCdGO1j > 4wRzrKPmE0cud5Kg5SR/GPI= > =yZ8N > -----END PGP SIGNATURE----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/3038e35e/attachment.html From anthony.minessale at gmail.com Fri Jul 9 08:31:42 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 9 Jul 2010 10:31:42 -0500 Subject: [Freeswitch-users] SIP2VXML In-Reply-To: <908418189EF44BB98023F68CD865A864@dell9400> References: <39ED5E8D0A5D42BCA019A827DE9F7A4C@dell9400> <908418189EF44BB98023F68CD865A864@dell9400> Message-ID: I think the problem is that too many people treat VXML like it's a gold mine or somehow valuable. That is a no no for a standard. So when it was first invented, people tried to sell the editors for money, the only implementation were commercial and there was no real acceptance to it. The first open source VXML stack was dependency-ridden and archaic to even build for many years. The value could be realized by providing hosting services for it, or developing it for customers, not by making them pay money to even use the editor. I believe a standardized way to serialize an IVR and execute it from a neutral server is a good idea but I am a bit sketchy on whether or not VXML is right. Imagine if email was a secret protocol and you had to pay money for your email composer...Oh yeah, that happened and everyone laughed in their faces and now it's free. When I once tried to implement VXML, I started trying to use openVXI, while hunting down the dependancies, I learned about spidermonkey and started to wonder why to bother rendering the XML into a script and running it when you could just write the whole IVR in JS. So I made res_js for Asterisk. When I started FS, I also made a mod_spidermonkey as we can all see. I also use XML in FS so I don't have a vendetta against XML, (I do get annoyed by those who over do it with XML). I just get an uneasy feeling about it and there is something to be said since we have 100% of the ingredients for VXML in FS and never implemented it. Like I said I am open to the idea of supporting it, to give it a fighting chance to get adopted. But teenagers are harder to get adopted as Andy pointed out =p On Fri, Jul 9, 2010 at 2:11 AM, Jan Berger wrote: > So why does it "suck"? > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andy > Spitzer > Sent: 9. juli 2010 06:27 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] SIP2VXML > > Woof! > > On Mon, 05 Jul 2010 18:58:05 -0400, Anthony Minessale > wrote: > > > It's almost like people think with all the stuff FS can do out of the > > box, supporting this fly-by-night phenomenon called VXML that has failed > > to gain traction after 5 years is somehow too valuable to contribute lol > > VoiceXML started 11 years ago, in 1999. > > And, still, "VoiceXML sucks" (my signature phrase for all posts about > VoiceXML. Google it.) > > --Woof! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/3197fa14/attachment.html From dswardstrom at remotelink.com Fri Jul 9 09:32:08 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Fri, 9 Jul 2010 09:32:08 -0700 (PDT) Subject: [Freeswitch-users] Two FreeSWITCH Box IVR Call Transfer In-Reply-To: References: <1278513656974-5265595.post@n2.nabble.com> Message-ID: <1278693128660-5275046.post@n2.nabble.com> If you are using SIP, then anything in the Headers in the SIP message are subject to being examined by some 3rd party. They are not secure. They could also be spoofed or intercepted and modified (man in the middle). You can use SIPS, that is why it was designed. Don't know what kind of issue this would raise with carriers. Another possibility since you control the original destination and the actual destination system. Use a common database, put a record into the DB that the call is being transferred and use as the postd number the Key. Probably should put From (ANI), To (DNIS), user info, and other information into the record. On the receiving side, look up the entry/record, verify what can be verified, the use the information in the record to determine the user. This is similar to the sophisticated capabilities used in some ACD installations. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Two-FreeSWITCH-Box-IVR-Call-Transfer-tp5265404p5275046.html Sent from the freeswitch-users mailing list archive at Nabble.com. From nick at hkcradio.com Fri Jul 9 09:31:17 2010 From: nick at hkcradio.com (Nick G) Date: Fri, 09 Jul 2010 12:31:17 -0400 Subject: [Freeswitch-users] Freeswitch+Siemens A580IP Message-ID: <4C374ED5.3040407@hkcradio.com> Greetings. I have a bit of a problem with placing outgoing calls with my Siemens on my freeswitch PBX, which seems to have to do with the way it handles sending a ring signal to someone on the call. Instead of the method used by other PBX's, it seems to send audio back, rather than being content to let my sip device take care of sending me the ring signal. Consequently, it has a hard time negotiating media when someone actually picks up the call. Can I fix this? If so, how? Thanks, Nick From luixsansan at hotmail.com Fri Jul 9 09:46:16 2010 From: luixsansan at hotmail.com (luixsansan at hotmail.com) Date: Fri, 9 Jul 2010 18:46:16 +0200 Subject: [Freeswitch-users] dtmf in Skype In-Reply-To: References: Message-ID: Hello Giovanni, I have made more tests and what I found is that when I made the call from skypeClient to skypeClient I can not open a keyboard to send dtmf signals ( I have not found any option in the configuration that allows a keyboard), it seems that other version of skype is needed for this to work. Do you know wich? My error was that once the call was established I opened a keyboard that is used to make calls to the PSTN. Then I need a Dialplan to send an incoming call to the extension where skype is receiving calls. Is this correct? Kind regards. Luis. -------------------------------------------------- From: "Giovanni Maruzzelli" Sent: Friday, July 09, 2010 1:22 PM To: Subject: Re: [Freeswitch-users] dtmf in Skype > On Fri, Jul 9, 2010 at 12:47 PM, wrote: > > config file is ok >> >> then, when I call from another Skype Client in other computer to user1 >> extension 5000 answers and I listen to the IVR demo, I pulse an extension >> or >> a number of the menu options but nothing happens, I keep listening the >> menu >> until FS hangs up for timeout. > > this is very strange, because the skype clients sends the dtmf as out > of band signaling, so it is always seen by skypopen (eg: no fiddling > with audio dtmf recognition). > > can you please open a Jira issue? > > and please do this: > > 1) if you have put mod_skypopen in modules.conf.xml, please comment it > out (let's not start it when we don't see it ;) ) > 2) shutdown and restart FS > 3) from its command line: > 4) "fsctl loglevel 7" > 5) "console loglevel 7" > 6) "load mod_skypopen" > 7) receive the call and press some dtmf in the remote skype client keypad > > and attach to the Jira all the output, since the beginning (attach as > a file, would be better than paste as comment. > > >> -------------------------------------------------- >> From: "Giovanni Maruzzelli" >> Sent: Thursday, July 08, 2010 9:47 PM >> To: >> Subject: Re: [Freeswitch-users] dtmf in Skype >> >>> that is not true ;) Is very clear, indeed (I wrote it ;) ) >>> >>> look for dtmf in all the wiki page, you'll find some parameter to be >>> used. >>> >>> If it's still unclear, please rewrite here, and describe fully what >>> you'r doing and what is the problem >>> >>> -giovanni >>> >>> On Thu, Jul 8, 2010 at 9:03 PM, wrote: >>>> Hello, >>>> >>>> I have FS ver. 1.0.6, Windows Vista Ultimate and Skype 4.2.0.169 for >>>> Windows. (user1) >>>> >>>> On a laptop I have Windows Vista Home and Skype 4.1.0.179 (user2) >>>> >>>> When I make a call from user2 to user1 I listen the IVR menu and see >>>> the >>>> activity on FS console. The problem is that when I click on the skype's >>>> phone keyboard I can not listen to any DTMF signal and user1 is not >>>> receiving DTMF signals ( the IVR menu is not interrupted). >>>> >>>> Is there a configuration change I have to do in order to make skype >>>> send/receive dtmf signals or do I need another type of skype client? >>>> >>>> The info in the document for mod_skypopen is not very clear. >>>> >>>> >>>> Thank you. >>>> >>>> Luis. >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From rupa at rupa.com Fri Jul 9 09:55:40 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 9 Jul 2010 11:55:40 -0500 Subject: [Freeswitch-users] Freeswitch+Siemens A580IP In-Reply-To: <4C374ED5.3040407@hkcradio.com> References: <4C374ED5.3040407@hkcradio.com> Message-ID: They have a problem with g722 and early media. Either turn off g722 on the phone or actually answer the call prior to bridging. I have a ticket open with Siemens on this with traces but have heard nothing recently from them. I have the following in my dialplan towards the top to unconditionally "answer" if the endpoint is a A580ip: On Fri, Jul 9, 2010 at 11:31 AM, Nick G wrote: > Greetings. > I have a bit of a problem with placing outgoing calls with my > Siemens on my freeswitch PBX, which seems to have to do with the way it > handles sending a ring signal to someone on the call. Instead of the > method used by other PBX's, it seems to send audio back, rather than > being content to let my sip device take care of sending me the ring > signal. Consequently, it has a hard time negotiating media when someone > actually picks up the call. Can I fix this? If so, how? > Thanks, > Nick > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/23491e67/attachment.html From nick at hkcradio.com Fri Jul 9 10:36:43 2010 From: nick at hkcradio.com (Nick G) Date: Fri, 09 Jul 2010 13:36:43 -0400 Subject: [Freeswitch-users] Freeswitch+Siemens A580IP In-Reply-To: References: <4C374ED5.3040407@hkcradio.com> Message-ID: <4C375E2B.8030301@hkcradio.com> Thanks so much. Which file should this be in? Sorry, I'm a FusionPBX using n00b. Thanks, Nick On 7/9/2010 12:55 PM, Rupa Schomaker wrote: > They have a problem with g722 and early media. Either turn off g722 > on the phone or actually answer the call prior to bridging. I have a > ticket open with Siemens on this with traces but have heard nothing > recently from them. > > I have the following in my dialplan towards the top to unconditionally > "answer" if the endpoint is a A580ip: > > > > > > > > > > On Fri, Jul 9, 2010 at 11:31 AM, Nick G > wrote: > > Greetings. > I have a bit of a problem with placing outgoing calls with my > Siemens on my freeswitch PBX, which seems to have to do with the > way it > handles sending a ring signal to someone on the call. Instead of the > method used by other PBX's, it seems to send audio back, rather than > being content to let my sip device take care of sending me the ring > signal. Consequently, it has a hard time negotiating media when > someone > actually picks up the call. Can I fix this? If so, how? > Thanks, > Nick > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/e52a1d9d/attachment.html From gmaruzz at celliax.org Fri Jul 9 10:57:49 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 9 Jul 2010 19:57:49 +0200 Subject: [Freeswitch-users] dtmf in Skype In-Reply-To: References: Message-ID: On Fri, Jul 9, 2010 at 6:46 PM, wrote: > I have made more tests and what I found is that when I made the call from > skypeClient to skypeClient I can not open a keyboard to send dtmf signals > ( I have not found any option in the configuration that allows a keyboard), > it seems that other version of skype is needed for this to work. Do you know > wich? I don't understand this. We all are using the same skypeclient in windows, and you have a little arrow besides the horizontal volume slider. You click on that arrow after having a live call connected, and a dtmf keyboard appear. Why don't you have looked in the skypeclient help or googled for this? :) The extension where an incoming call will go is defined in the skypopen.conf.xml file Hope this helps, -giovanni > Kind regards. > > Luis. > > > -------------------------------------------------- > From: "Giovanni Maruzzelli" > Sent: Friday, July 09, 2010 1:22 PM > To: > Subject: Re: [Freeswitch-users] dtmf in Skype > >> On Fri, Jul 9, 2010 at 12:47 PM, ? wrote: >> >> config file is ok >>> >>> then, when I call from another Skype Client in other computer to user1 >>> extension 5000 answers and I listen to the IVR demo, I pulse an extension >>> or >>> a number of the menu options but nothing happens, I keep listening the >>> menu >>> until FS hangs up for timeout. >> >> this is very strange, because the skype clients sends the dtmf as out >> of band signaling, so it is always seen by skypopen (eg: no fiddling >> with audio dtmf recognition). >> >> can you please open a Jira issue? >> >> and please do this: >> >> 1) if you have put mod_skypopen in modules.conf.xml, please comment it >> out (let's not start it when we don't see it ;) ) >> 2) shutdown and restart FS >> 3) from its command line: >> 4) "fsctl loglevel 7" >> 5) "console loglevel 7" >> 6) "load mod_skypopen" >> 7) receive the call and press some dtmf in the remote skype client keypad >> >> and attach to the Jira all the output, since the beginning (attach as >> a file, would be better than paste as comment. >> >> >>> -------------------------------------------------- >>> From: "Giovanni Maruzzelli" >>> Sent: Thursday, July 08, 2010 9:47 PM >>> To: >>> Subject: Re: [Freeswitch-users] dtmf in Skype >>> >>>> that is not true ;) Is very clear, indeed (I wrote it ;) ) >>>> >>>> look for dtmf in all the wiki page, you'll find some parameter to be >>>> used. >>>> >>>> If it's still unclear, please rewrite here, and describe fully what >>>> you'r doing and what is the problem >>>> >>>> -giovanni >>>> >>>> On Thu, Jul 8, 2010 at 9:03 PM, ? wrote: >>>>> Hello, >>>>> >>>>> I have FS ver. 1.0.6, Windows Vista Ultimate and Skype 4.2.0.169 for >>>>> Windows. (user1) >>>>> >>>>> On a laptop I have Windows Vista Home and Skype 4.1.0.179 (user2) >>>>> >>>>> When I make a call from user2 to user1 I listen the IVR menu and see >>>>> the >>>>> activity on FS console. The problem is that when I click on the skype's >>>>> phone keyboard I can not listen to any DTMF signal and user1 is not >>>>> receiving DTMF signals ( the IVR menu is not interrupted). >>>>> >>>>> Is there a configuration change I have to do in order to make skype >>>>> send/receive dtmf signals or do I need another type of skype client? >>>>> >>>>> The info in the document for mod_skypopen is not very clear. >>>>> >>>>> >>>>> Thank you. >>>>> >>>>> Luis. >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>> -- >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> Cell : +39-347-2665618 >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From msc at freeswitch.org Fri Jul 9 10:58:16 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 9 Jul 2010 10:58:16 -0700 Subject: [Freeswitch-users] Freeswitch+Siemens A580IP In-Reply-To: <4C375E2B.8030301@hkcradio.com> References: <4C374ED5.3040407@hkcradio.com> <4C375E2B.8030301@hkcradio.com> Message-ID: On Fri, Jul 9, 2010 at 10:36 AM, Nick G wrote: > Thanks so much. Which file should this be in? Sorry, I'm a FusionPBX > using n00b. > Any time you see a line like this: Then we're talking about a dialplan file. I don't know the nomenclature for FuPBX but in the vanilla FS install the first place to look would be conf/dialplan/default.xml... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/035331e0/attachment.html From too.goa at gmail.com Fri Jul 9 11:28:48 2010 From: too.goa at gmail.com (Goa) Date: Sat, 10 Jul 2010 00:28:48 +0600 Subject: [Freeswitch-users] Making mod_conference Message-ID: <4C376A60.9010101@gmail.com> Hello! I'm trying to setup FS on FreeBSD 8.0-RELEASE i386. Now from snapshot from http://files.freeswitch.org/freeswitch-snapshot.tar.gz #gmake install, but errors http://pastebin.freeswitch.org/13418 Thank you. Goa From brian at freeswitch.org Fri Jul 9 11:37:54 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 9 Jul 2010 13:37:54 -0500 Subject: [Freeswitch-users] Making mod_conference In-Reply-To: <4C376A60.9010101@gmail.com> References: <4C376A60.9010101@gmail.com> Message-ID: <295B4C5A-4F16-4664-9CA5-2BB7E08E80C6@freeswitch.org> I have already asked if you can pull from GIT. /b On Jul 9, 2010, at 1:28 PM, Goa wrote: > Hello! > I'm trying to setup FS on FreeBSD 8.0-RELEASE i386. > Now from snapshot from > http://files.freeswitch.org/freeswitch-snapshot.tar.gz > > #gmake install, but errors > http://pastebin.freeswitch.org/13418 > > Thank you. > Goa From msc at freeswitch.org Fri Jul 9 11:55:39 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 9 Jul 2010 11:55:39 -0700 Subject: [Freeswitch-users] Making mod_conference In-Reply-To: <4C376A60.9010101@gmail.com> References: <4C376A60.9010101@gmail.com> Message-ID: http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code or http://wiki.freeswitch.org/wiki/Git_Install -MC On Fri, Jul 9, 2010 at 11:28 AM, Goa wrote: > Hello! > I'm trying to setup FS on FreeBSD 8.0-RELEASE i386. > Now from snapshot from > http://files.freeswitch.org/freeswitch-snapshot.tar.gz > > #gmake install, but errors > http://pastebin.freeswitch.org/13418 > > Thank you. > Goa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/cf0150b3/attachment-0001.html From engineerzuhairraza at gmail.com Fri Jul 9 12:35:52 2010 From: engineerzuhairraza at gmail.com (Zuhair Raza) Date: Sat, 10 Jul 2010 00:35:52 +0500 Subject: [Freeswitch-users] Hi guyz Message-ID: Hi.. What is the alternate to agi, as in asterisk, in freeswitch ? -- Regards, Zuhair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100710/c67c3231/attachment.html From sos at sokhapkin.dyndns.org Fri Jul 9 12:45:07 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 9 Jul 2010 15:45:07 -0400 Subject: [Freeswitch-users] Hi guyz In-Reply-To: References: Message-ID: <201007091545.07262.sos@sokhapkin.dyndns.org> http://wiki.freeswitch.org/wiki/Languages_for_Call_Control On Friday 09 July 2010, Zuhair Raza wrote: > Hi.. > What is the alternate to agi, as in asterisk, in freeswitch ? > From chavpaskov at shaw.ca Fri Jul 9 13:02:55 2010 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Fri, 09 Jul 2010 13:02:55 -0700 Subject: [Freeswitch-users] Fwd: Mod_lcr lcr_user_rate suggestion Message-ID: An embedded message was scrubbed... From: Tchavdar Paskov Subject: Mod_lcr lcr_user_rate suggestion Date: Fri, 09 Jul 2010 11:06:40 -0700 Size: 2735 Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/413878a3/attachment.mht From saeedahmad1981 at gmail.com Fri Jul 9 15:12:12 2010 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Sat, 10 Jul 2010 00:12:12 +0200 Subject: [Freeswitch-users] Freeswitch - wholesale setup In-Reply-To: <79C486FA6812401A81187C7C40D611E3@MOBILEE1705> References: <07d601cb1eb1$64f05db0$2ed11910$@com> <79C486FA6812401A81187C7C40D611E3@MOBILEE1705> Message-ID: Hi, Definitely all of your questions can't be covered but, if you some programming knowledge then look for: - mod_xml_curl for doing ip based auth credit checks basically all of your routing needs there are good examples on wiki - mod_xml_cdr to generate the xml based cdr which you can load into database, search on list for more details. - SAT On Thu, Jul 8, 2010 at 6:04 PM, Madovsky wrote: > dear Eric, > > you need to learn from WIKI or pay a consultant. > > Regards > > Franck > > ----- Original Message ----- > *From:* Peder > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Thursday, July 08, 2010 11:22 AM > *Subject:* Re: [Freeswitch-users] Freeswitch - wholesale setup > > Um, so you want to setup to be a wholesale SIP provider, but everything > on the wiki is way too technical for you? Your best bet is to email > consulting at freeswitch.org and pay for help. If you ask a specific > question you will get answers, but essentially saying ?I don?t know > anything, please set it all up for me? is going to result in no help at all. > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Voip Newbie > *Sent:* Thursday, July 08, 2010 9:10 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Freeswitch - wholesale setup > > > > Dear List, > > I am new to voip and to freeswitch and I am trying to learn how to setup a > wholesale scenario using FreeSwitch. I have very little experience in > running wholesale voip business on voipswitch. > > I know all or most of the answers to my questions are already either in > wiki or in list archive, and I am really sorry to email the list asking for > help but with the little knowledge I have its way over my head and I failed > to compile the information in right way to achieve any result, please > forgive me if these questions bother any of you at all. > > Here is what I wish to achieve with FreeSwitch: > > 1) Setup username/password less accounts for customers with IP > authentication. One customer can have multiple IPs. Customer can send > traffic using SIP or H323 protocol. A prefix will be assigned to customer > for sending traffic. Eg. 1234 + Country code + Area code + Number. > 2) Customer can be post paid or per paid, so need to disable customer's > ability to call when assigned credit limit is reached. > 3) Setup providers (Gateways) which do not provide username/password for > authentication and do not require FreeSwitch to register with them, > FreeSwitch IP will be allowed to send traffic directly with a 3-4 digit > prefix. Provider can be on H323 or SIP either and can have multiple IPs (1 > primary and other for fail-over) > 4) Customer and providers need access to CDR. So we need to configure > Freeswitch the way that it can store CDRs in MySQL database, that database > can be accessed by a web application to show CDR on web. > > *Hardware and OS info:* > CentOS 5.4 (Linux 2.6.18-164.15.1.el5PAE on i686) > Intel(R) Xeon(R) CPU E5420 @ 2.50GHz, 4 cores > 4 GB RAM. > 1 GIGABIT NIC with a Public IP address > > *Progress so far:* > 1) FreeSwitch installed with installation procedure at FreeSwitch Wiki ( > http://wiki.freeswitch.org/wiki/Installation_Guide#Download_Source_Tarball > ) > 2) Registered and called already created extensions (1001,1002) from > x-lite, called echo extensions and everything worked fine. > > *Questions:* > 1) I can see there are 2 H323 mods available for FreeSwitch, which one is > better to use in production. > a) http://wiki.freeswitch.org/wiki/FreeSwitch_H323 > b) http://wiki.freeswitch.org/wiki/Mod_h323 > 2) How and where (location in freeswitch conf) can I create customers, add > IP addresses to authorize without username and password, assign a prefix to > the customer? An example would be nice. > 3) How and where (location in freeswitch conf) I can create gateways. An > example would be nice. > 4) Where to create dialplan to route customer calls to provider. An example > would be nice. > 5) How to manage credit assigned to customers, and how to bill the calls? > 6) How to configure Freeswitch so that it can dump CDRs to a mysql > database. > 7) According to FreeSwitch feature list, it supports g723 and g729 in > pass-througe mode, so this means if both customer and providers support g729 > & g723, calls will pass? > 8) Is there any limit on g729 and g723 calls in pass-through mood? > 9) Approx how many concurrent calls can FreeSwitch support in H323 and in > SIP based on the hardware info given above. > > Any pointers, help, links, examples will be highly appreciated. > > Thanks in advance. > > > > -Eric > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100710/9e3390dd/attachment.html From msc at freeswitch.org Fri Jul 9 15:26:38 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 9 Jul 2010 15:26:38 -0700 Subject: [Freeswitch-users] Hi guyz In-Reply-To: References: Message-ID: On Fri, Jul 9, 2010 at 12:35 PM, Zuhair Raza wrote: > Hi.. > What is the alternate to agi, as in asterisk, in freeswitch ? > > You might also appreciate this page: http://wiki.freeswitch.org/wiki/Rosetta_stone -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/b1adef1a/attachment.html From msc at freeswitch.org Fri Jul 9 15:27:29 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 9 Jul 2010 15:27:29 -0700 Subject: [Freeswitch-users] Freeswitch - wholesale setup In-Reply-To: References: <07d601cb1eb1$64f05db0$2ed11910$@com> <79C486FA6812401A81187C7C40D611E3@MOBILEE1705> Message-ID: Don't forget to buy the FreeSWITCH book! https://www.packtpub.com/freeswitch-1-0-5-build-robust-high-performance-telephony-systems/book -MC On Fri, Jul 9, 2010 at 3:12 PM, Saeed Ahmed wrote: > Hi, > > Definitely all of your questions can't be covered but, if you some > programming knowledge then look for: > > - mod_xml_curl > for doing ip based auth > credit checks > basically all of your routing needs > there are good examples on wiki > > - mod_xml_cdr > to generate the xml based cdr which you can load into database, search on > list for more details. > > - SAT > > On Thu, Jul 8, 2010 at 6:04 PM, Madovsky wrote: > >> dear Eric, >> >> you need to learn from WIKI or pay a consultant. >> >> Regards >> >> Franck >> >> ----- Original Message ----- >> *From:* Peder >> *To:* freeswitch-users at lists.freeswitch.org >> *Sent:* Thursday, July 08, 2010 11:22 AM >> *Subject:* Re: [Freeswitch-users] Freeswitch - wholesale setup >> >> Um, so you want to setup to be a wholesale SIP provider, but everything >> on the wiki is way too technical for you? Your best bet is to email >> consulting at freeswitch.org and pay for help. If you ask a specific >> question you will get answers, but essentially saying ?I don?t know >> anything, please set it all up for me? is going to result in no help at all. >> >> >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Voip Newbie >> *Sent:* Thursday, July 08, 2010 9:10 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] Freeswitch - wholesale setup >> >> >> >> Dear List, >> >> I am new to voip and to freeswitch and I am trying to learn how to setup a >> wholesale scenario using FreeSwitch. I have very little experience in >> running wholesale voip business on voipswitch. >> >> I know all or most of the answers to my questions are already either in >> wiki or in list archive, and I am really sorry to email the list asking for >> help but with the little knowledge I have its way over my head and I failed >> to compile the information in right way to achieve any result, please >> forgive me if these questions bother any of you at all. >> >> Here is what I wish to achieve with FreeSwitch: >> >> 1) Setup username/password less accounts for customers with IP >> authentication. One customer can have multiple IPs. Customer can send >> traffic using SIP or H323 protocol. A prefix will be assigned to customer >> for sending traffic. Eg. 1234 + Country code + Area code + Number. >> 2) Customer can be post paid or per paid, so need to disable customer's >> ability to call when assigned credit limit is reached. >> 3) Setup providers (Gateways) which do not provide username/password for >> authentication and do not require FreeSwitch to register with them, >> FreeSwitch IP will be allowed to send traffic directly with a 3-4 digit >> prefix. Provider can be on H323 or SIP either and can have multiple IPs (1 >> primary and other for fail-over) >> 4) Customer and providers need access to CDR. So we need to configure >> Freeswitch the way that it can store CDRs in MySQL database, that database >> can be accessed by a web application to show CDR on web. >> >> *Hardware and OS info:* >> CentOS 5.4 (Linux 2.6.18-164.15.1.el5PAE on i686) >> Intel(R) Xeon(R) CPU E5420 @ 2.50GHz, 4 cores >> 4 GB RAM. >> 1 GIGABIT NIC with a Public IP address >> >> *Progress so far:* >> 1) FreeSwitch installed with installation procedure at FreeSwitch Wiki ( >> http://wiki.freeswitch.org/wiki/Installation_Guide#Download_Source_Tarball >> ) >> 2) Registered and called already created extensions (1001,1002) from >> x-lite, called echo extensions and everything worked fine. >> >> *Questions:* >> 1) I can see there are 2 H323 mods available for FreeSwitch, which one is >> better to use in production. >> a) http://wiki.freeswitch.org/wiki/FreeSwitch_H323 >> b) http://wiki.freeswitch.org/wiki/Mod_h323 >> 2) How and where (location in freeswitch conf) can I create customers, add >> IP addresses to authorize without username and password, assign a prefix to >> the customer? An example would be nice. >> 3) How and where (location in freeswitch conf) I can create gateways. An >> example would be nice. >> 4) Where to create dialplan to route customer calls to provider. An >> example would be nice. >> 5) How to manage credit assigned to customers, and how to bill the calls? >> 6) How to configure Freeswitch so that it can dump CDRs to a mysql >> database. >> 7) According to FreeSwitch feature list, it supports g723 and g729 in >> pass-througe mode, so this means if both customer and providers support g729 >> & g723, calls will pass? >> 8) Is there any limit on g729 and g723 calls in pass-through mood? >> 9) Approx how many concurrent calls can FreeSwitch support in H323 and in >> SIP based on the hardware info given above. >> >> Any pointers, help, links, examples will be highly appreciated. >> >> Thanks in advance. >> >> >> >> -Eric >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/354e31be/attachment-0001.html From paul.gore.j at gmail.com Fri Jul 9 16:15:01 2010 From: paul.gore.j at gmail.com (paul gore) Date: Fri, 9 Jul 2010 19:15:01 -0400 Subject: [Freeswitch-users] High cpu usage on EC2 Centos 5.4 Message-ID: Hello there, Seems like I hit another problem running FS 1.0.6 on EC2 - CPU usage by FS alone hits 65% for just one call! Here's output from top command: 26348 root 15 0 418m 42m 8024 S 65.8 0.6 808:37.18 freeswitch We use Rightscale image Centos 5.4 EBS and m1.large instance type. What can be possible reasons for that? I have never seen FS using more than 10% CPU for 50 calls on a regular box under Centos 5.2, we have version FS 1.0.5 there. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/efbb36cb/attachment.html From infos at madovsky.org Fri Jul 9 16:37:09 2010 From: infos at madovsky.org (Madovsky) Date: Fri, 9 Jul 2010 19:37:09 -0400 Subject: [Freeswitch-users] Freeswitch - wholesale setup References: <07d601cb1eb1$64f05db0$2ed11910$@com><79C486FA6812401A81187C7C40D611E3@MOBILEE1705> Message-ID: I don't have a car but be sure I will have my FS book ! :D ----- Original Message ----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Friday, July 09, 2010 6:27 PM Subject: Re: [Freeswitch-users] Freeswitch - wholesale setup Don't forget to buy the FreeSWITCH book! https://www.packtpub.com/freeswitch-1-0-5-build-robust-high-performance-telephony-systems/book -MC On Fri, Jul 9, 2010 at 3:12 PM, Saeed Ahmed wrote: Hi, Definitely all of your questions can't be covered but, if you some programming knowledge then look for: - mod_xml_curl for doing ip based auth credit checks basically all of your routing needs there are good examples on wiki - mod_xml_cdr to generate the xml based cdr which you can load into database, search on list for more details. - SAT On Thu, Jul 8, 2010 at 6:04 PM, Madovsky wrote: dear Eric, you need to learn from WIKI or pay a consultant. Regards Franck ----- Original Message ----- From: Peder To: freeswitch-users at lists.freeswitch.org Sent: Thursday, July 08, 2010 11:22 AM Subject: Re: [Freeswitch-users] Freeswitch - wholesale setup Um, so you want to setup to be a wholesale SIP provider, but everything on the wiki is way too technical for you? Your best bet is to email consulting at freeswitch.org and pay for help. If you ask a specific question you will get answers, but essentially saying ?I don?t know anything, please set it all up for me? is going to result in no help at all. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Voip Newbie Sent: Thursday, July 08, 2010 9:10 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch - wholesale setup Dear List, I am new to voip and to freeswitch and I am trying to learn how to setup a wholesale scenario using FreeSwitch. I have very little experience in running wholesale voip business on voipswitch. I know all or most of the answers to my questions are already either in wiki or in list archive, and I am really sorry to email the list asking for help but with the little knowledge I have its way over my head and I failed to compile the information in right way to achieve any result, please forgive me if these questions bother any of you at all. Here is what I wish to achieve with FreeSwitch: 1) Setup username/password less accounts for customers with IP authentication. One customer can have multiple IPs. Customer can send traffic using SIP or H323 protocol. A prefix will be assigned to customer for sending traffic. Eg. 1234 + Country code + Area code + Number. 2) Customer can be post paid or per paid, so need to disable customer's ability to call when assigned credit limit is reached. 3) Setup providers (Gateways) which do not provide username/password for authentication and do not require FreeSwitch to register with them, FreeSwitch IP will be allowed to send traffic directly with a 3-4 digit prefix. Provider can be on H323 or SIP either and can have multiple IPs (1 primary and other for fail-over) 4) Customer and providers need access to CDR. So we need to configure Freeswitch the way that it can store CDRs in MySQL database, that database can be accessed by a web application to show CDR on web. Hardware and OS info: CentOS 5.4 (Linux 2.6.18-164.15.1.el5PAE on i686) Intel(R) Xeon(R) CPU E5420 @ 2.50GHz, 4 cores 4 GB RAM. 1 GIGABIT NIC with a Public IP address Progress so far: 1) FreeSwitch installed with installation procedure at FreeSwitch Wiki (http://wiki.freeswitch.org/wiki/Installation_Guide#Download_Source_Tarball) 2) Registered and called already created extensions (1001,1002) from x-lite, called echo extensions and everything worked fine. Questions: 1) I can see there are 2 H323 mods available for FreeSwitch, which one is better to use in production. a) http://wiki.freeswitch.org/wiki/FreeSwitch_H323 b) http://wiki.freeswitch.org/wiki/Mod_h323 2) How and where (location in freeswitch conf) can I create customers, add IP addresses to authorize without username and password, assign a prefix to the customer? An example would be nice. 3) How and where (location in freeswitch conf) I can create gateways. An example would be nice. 4) Where to create dialplan to route customer calls to provider. An example would be nice. 5) How to manage credit assigned to customers, and how to bill the calls? 6) How to configure Freeswitch so that it can dump CDRs to a mysql database. 7) According to FreeSwitch feature list, it supports g723 and g729 in pass-througe mode, so this means if both customer and providers support g729 & g723, calls will pass? 8) Is there any limit on g729 and g723 calls in pass-through mood? 9) Approx how many concurrent calls can FreeSwitch support in H323 and in SIP based on the hardware info given above. Any pointers, help, links, examples will be highly appreciated. Thanks in advance. -Eric ------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/f6d667e4/attachment.html From sos at sokhapkin.dyndns.org Fri Jul 9 16:53:23 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 9 Jul 2010 19:53:23 -0400 Subject: [Freeswitch-users] mod_nibblebill and mod_xml_cdr problem Message-ID: <201007091953.24035.sos@sokhapkin.dyndns.org> When customer's balance falls below nobal_amt and mod_nibblebill transfers the call to nobal_action (which just does hangup), mod_xml_cdr doesn't send all leg a channel variables to the url (). More precisely, it sends only channel variables set by FS, but no variables set by dialplan on leg a. I suspect it sends only leg b variables. Am I right? From brian at freeswitch.org Fri Jul 9 17:03:56 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 9 Jul 2010 19:03:56 -0500 Subject: [Freeswitch-users] mod_nibblebill and mod_xml_cdr problem In-Reply-To: <201007091953.24035.sos@sokhapkin.dyndns.org> References: <201007091953.24035.sos@sokhapkin.dyndns.org> Message-ID: Highly doubt it... if the variable was set in the dialplan its the same as if FS set them its the exact same API... so the absence of the variable leads me to think you never actually set them to what you thought you set them.... or they are only on the b-leg. /b On Jul 9, 2010, at 6:53 PM, Sergey Okhapkin wrote: > When customer's balance falls below nobal_amt and mod_nibblebill transfers the > call to nobal_action (which just does hangup), mod_xml_cdr doesn't send all > leg a channel variables to the url (). > More precisely, it sends only channel variables set by FS, but no variables > set by dialplan on leg a. I suspect it sends only leg b variables. Am I right? From sos at sokhapkin.dyndns.org Fri Jul 9 17:13:54 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 9 Jul 2010 20:13:54 -0400 Subject: [Freeswitch-users] mod_nibblebill and mod_xml_cdr problem In-Reply-To: References: <201007091953.24035.sos@sokhapkin.dyndns.org> Message-ID: <201007092013.54385.sos@sokhapkin.dyndns.org> All variables set by dialplan are sent to xml_cdr url if call was terminated normally (either party did hang up). But no dialplan variables are in xml_cdr xml when call was terminated by mod_nibblebill. I can send you xml examples. On Friday 09 July 2010, Brian West wrote: > Highly doubt it... if the variable was set in the dialplan its the same as > if FS set them its the exact same API... so the absence of the variable > leads me to think you never actually set them to what you thought you set > them.... or they are only on the b-leg. > > /b > > On Jul 9, 2010, at 6:53 PM, Sergey Okhapkin wrote: > > When customer's balance falls below nobal_amt and mod_nibblebill > > transfers the call to nobal_action (which just does hangup), mod_xml_cdr > > doesn't send all leg a channel variables to the url ( > name="log-b-leg" value="false"/>). More precisely, it sends only channel > > variables set by FS, but no variables set by dialplan on leg a. I suspect > > it sends only leg b variables. Am I right? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From infos at madovsky.org Fri Jul 9 17:20:03 2010 From: infos at madovsky.org (Madovsky) Date: Fri, 9 Jul 2010 20:20:03 -0400 Subject: [Freeswitch-users] mod_nibblebill and mod_xml_cdr problem References: <201007091953.24035.sos@sokhapkin.dyndns.org> <201007092013.54385.sos@sokhapkin.dyndns.org> Message-ID: <868D866334F6496E969AA6D437F0F59B@MOBILEE1705> yes examples would be nice... ----- Original Message ----- From: "Sergey Okhapkin" To: Sent: Friday, July 09, 2010 8:13 PM Subject: Re: [Freeswitch-users] mod_nibblebill and mod_xml_cdr problem > All variables set by dialplan are sent to xml_cdr url if call was > terminated > normally (either party did hang up). But no dialplan variables are in > xml_cdr > xml when call was terminated by mod_nibblebill. I can send you xml > examples. > > On Friday 09 July 2010, Brian West wrote: >> Highly doubt it... if the variable was set in the dialplan its the same >> as >> if FS set them its the exact same API... so the absence of the variable >> leads me to think you never actually set them to what you thought you >> set >> them.... or they are only on the b-leg. >> >> /b >> >> On Jul 9, 2010, at 6:53 PM, Sergey Okhapkin wrote: >> > When customer's balance falls below nobal_amt and mod_nibblebill >> > transfers the call to nobal_action (which just does hangup), >> > mod_xml_cdr >> > doesn't send all leg a channel variables to the url (> > name="log-b-leg" value="false"/>). More precisely, it sends only >> > channel >> > variables set by FS, but no variables set by dialplan on leg a. I >> > suspect >> > it sends only leg b variables. Am I right? >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sos at sokhapkin.dyndns.org Fri Jul 9 17:41:13 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 9 Jul 2010 20:41:13 -0400 Subject: [Freeswitch-users] mod_nibblebill and mod_xml_cdr problem In-Reply-To: <201007092013.54385.sos@sokhapkin.dyndns.org> References: <201007091953.24035.sos@sokhapkin.dyndns.org> <201007092013.54385.sos@sokhapkin.dyndns.org> Message-ID: <201007092041.13464.sos@sokhapkin.dyndns.org> More info - if I execute a lua script from mod_cdr_csv, all channel variables set by dialplan are accessible from the script (even when the call was hung up by mod_nibblebill) and the script logs CDR to DB right. But no variables are sent by mod_xml_cdr. Attached is an example of mod_xml_cdr, before the call was setup, dialplan sets variables "accountcode", "A2B_tp_trunk" and "A2B_rateinitial". The variables are not in the XML file. Yes, I use a2billing GUI and DB with FreeSWITCH:-) On Friday 09 July 2010, Sergey Okhapkin wrote: > All variables set by dialplan are sent to xml_cdr url if call was > terminated normally (either party did hang up). But no dialplan variables > are in xml_cdr xml when call was terminated by mod_nibblebill. I can send > you xml examples. > > On Friday 09 July 2010, Brian West wrote: > > Highly doubt it... if the variable was set in the dialplan its the same > > as if FS set them its the exact same API... so the absence of the > > variable leads me to think you never actually set them to what you > > thought you set them.... or they are only on the b-leg. > > > > /b > > > > On Jul 9, 2010, at 6:53 PM, Sergey Okhapkin wrote: > > > When customer's balance falls below nobal_amt and mod_nibblebill > > > transfers the call to nobal_action (which just does hangup), > > > mod_xml_cdr doesn't send all leg a channel variables to the url ( > > name="log-b-leg" value="false"/>). More precisely, it sends only > > > channel variables set by FS, but no variables set by dialplan on leg a. > > > I suspect it sends only leg b variables. Am I right? > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- A non-text attachment was scrubbed... Name: a_88be0e3a-38d2-4384-8ece-d4aeabf98cac.cdr.xml Type: application/xml Size: 15003 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100709/edafaf49/attachment.rdf From sos at sokhapkin.dyndns.org Fri Jul 9 17:51:13 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 9 Jul 2010 20:51:13 -0400 Subject: [Freeswitch-users] mod_nibblebill and mod_xml_cdr problem In-Reply-To: <201007092041.13464.sos@sokhapkin.dyndns.org> References: <201007091953.24035.sos@sokhapkin.dyndns.org> <201007092013.54385.sos@sokhapkin.dyndns.org> <201007092041.13464.sos@sokhapkin.dyndns.org> Message-ID: <201007092051.13853.sos@sokhapkin.dyndns.org> BTW, search in the XML file for "variable_" returns nothing. No any variable is logged. On Friday 09 July 2010, Sergey Okhapkin wrote: > More info - if I execute a lua script from mod_cdr_csv, all channel > variables set by dialplan are accessible from the script (even when the > call was hung up by mod_nibblebill) and the script logs CDR to DB right. > But no variables are sent by mod_xml_cdr. > > Attached is an example of mod_xml_cdr, before the call was setup, dialplan > sets variables "accountcode", "A2B_tp_trunk" and "A2B_rateinitial". The > variables are not in the XML file. > > Yes, I use a2billing GUI and DB with FreeSWITCH:-) > > On Friday 09 July 2010, Sergey Okhapkin wrote: > > All variables set by dialplan are sent to xml_cdr url if call was > > terminated normally (either party did hang up). But no dialplan > > variables are in xml_cdr xml when call was terminated by mod_nibblebill. > > I can send you xml examples. > > > > On Friday 09 July 2010, Brian West wrote: > > > Highly doubt it... if the variable was set in the dialplan its the same > > > as if FS set them its the exact same API... so the absence of the > > > variable leads me to think you never actually set them to what you > > > thought you set them.... or they are only on the b-leg. > > > > > > /b > > > > > > On Jul 9, 2010, at 6:53 PM, Sergey Okhapkin wrote: > > > > When customer's balance falls below nobal_amt and mod_nibblebill > > > > transfers the call to nobal_action (which just does hangup), > > > > mod_xml_cdr doesn't send all leg a channel variables to the url > > > > (). More precisely, it sends > > > > only channel variables set by FS, but no variables set by dialplan on > > > > leg a. I suspect it sends only leg b variables. Am I right? > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user > > >s http://www.freeswitch.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > From mrene_lists at avgs.ca Fri Jul 9 20:33:46 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 9 Jul 2010 23:33:46 -0400 Subject: [Freeswitch-users] Making mod_conference In-Reply-To: <295B4C5A-4F16-4664-9CA5-2BB7E08E80C6@freeswitch.org> References: <4C376A60.9010101@gmail.com> <295B4C5A-4F16-4664-9CA5-2BB7E08E80C6@freeswitch.org> Message-ID: <9313DF68-12FD-4054-BE44-490690090B20@avgs.ca> I second, thats fixed already. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-07-09, at 2:37 PM, Brian West wrote: > I have already asked if you can pull from GIT. > > /b > > On Jul 9, 2010, at 1:28 PM, Goa wrote: > >> Hello! >> I'm trying to setup FS on FreeBSD 8.0-RELEASE i386. >> Now from snapshot from >> http://files.freeswitch.org/freeswitch-snapshot.tar.gz >> >> #gmake install, but errors >> http://pastebin.freeswitch.org/13418 >> >> Thank you. >> Goa > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tony.tin at noahmedia.com.hk Sat Jul 10 01:38:47 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Sat, 10 Jul 2010 16:38:47 +0800 Subject: [Freeswitch-users] originate call hangup signal In-Reply-To: <06271C2A41CE49C3901D0B611D63B56A@dell9400> References: <2B4FFD98-7682-44AB-ADB8-9B9700B5AC32@freeswitch.org> <06271C2A41CE49C3901D0B611D63B56A@dell9400> Message-ID: Thanks for all the replies ! I've tried libpri compatible stack again. The outbound is working now after I removed the latest libpri package 1.4.11, fall back to 1.4.7 and recompiled everything again. Actually which version of DAHDI and libpri package do you recommend? Is there any easy way to bypass the "disconnect supervision" ? Thanks. Regards, Tony On Fri, Jul 9, 2010 at 7:42 PM, Jan Berger wrote: > Have you checked your NAI setting? This is usually the main reason why > outboand calls trouble. And the native and Libpri might have a different > default settings. If not get a trace with the native and one with libpri . I > need to see HDLC either in raw form or decoded on layer 2. > > > > Jan > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* 8. juli 2010 18:58 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] originate call hangup signal > > > > Interesting. I've had more success with the libpri stack than the native > stack because the native stack is definitely a work in progress. I assisted > Mike J in getting 5ESS support in the native stack, but we definitely don't > have explicit 4ESS support. I think you are better off using the libpri > method and debugging it than trying to get the native PRI stack to work on a > 4ESS connection. > > Hop on #openzap on irc.freenode.net and ask for some assistance there. We > have a few guys who are familiar with PRI and libpri, etc. who might be able > to help you figure out what's going on. > > -MC > > On Thu, Jul 8, 2010 at 1:01 AM, Tony Tin > wrote: > > It's Digium TE220. I'm using the OpenZAP native stack, because I can not > get the outbound call work with libpri compatibility stack. > > Attached is the freeswitch.log. I'm not sure whether it includes the d-chan > trace, though I already enabled the "q931_dump". > > I originated a call to my mobile on freeswitch console with command > "originate OpenZAP/2/A/98855404 6899", I answered the call then hung up, > after around 30 seconds, I saw there is terminator event on the console and > the call hangup. > > Thanks > > Regards, > Tony > > > > > On Thu, Jul 8, 2010 at 12:42 PM, Michael S Collins > wrote: > > Okay, next question: which PRI are you using? Is it Digium-based or Sangoma > hardware? If the former then use the libpri method; the latter use freetdm. > I think they're both covered on the wiki. You need to get an ISDN trace on > the d-chan to see what is actually being sent to/from telco. > > > > -MC > > Sent from my iPhone > > > On Jul 7, 2010, at 7:57 PM, Tony Tin wrote: > > Thanks for your help. > > It's a 4ESS IDSN and the carrier does provide disconnect supervision, is > there any way to bypass it ? > > Regards, > Tony > > On Thu, Jul 8, 2010 at 7:14 AM, Michael Collins < > msc at freeswitch.org> wrote: > > It depends on where the "hangup signal" comes from. Is this an analog line? > If so, does the carrier provide disconnect supervision? It's entirely > possible that the other end isn't doing a good job of telling FreeSWITCH > that the call is over. > > -MC > > On Tue, Jul 6, 2010 at 10:35 PM, Tony Tin < > tony.tin at noahmedia.com.hk> wrote: > > Hi, > > When I use OpenZAP channel to originate a call, after the called party > hangup the phone. It takes freeswitch around 40 seconds to catch the hangup > signal and stop the dial plan. I'm wondering whether there is a way to > shorten that duration. Thanks. > > Regards, > Tony > > > _______________________________________________ > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100710/8664908c/attachment.html From jan.berger at video24.no Sat Jul 10 04:39:44 2010 From: jan.berger at video24.no (Jan Berger) Date: Sat, 10 Jul 2010 13:39:44 +0200 Subject: [Freeswitch-users] SIP2VXML In-Reply-To: References: <39ED5E8D0A5D42BCA019A827DE9F7A4C@dell9400> <908418189EF44BB98023F68CD865A864@dell9400> Message-ID: I think it's more an issue that people need to get paid and to do that using FreeSWITCH you need to add services to it - have something to sell that is not free. Jan _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 9. juli 2010 17:32 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP2VXML I think the problem is that too many people treat VXML like it's a gold mine or somehow valuable. That is a no no for a standard. So when it was first invented, people tried to sell the editors for money, the only implementation were commercial and there was no real acceptance to it. The first open source VXML stack was dependency-ridden and archaic to even build for many years. The value could be realized by providing hosting services for it, or developing it for customers, not by making them pay money to even use the editor. I believe a standardized way to serialize an IVR and execute it from a neutral server is a good idea but I am a bit sketchy on whether or not VXML is right. Imagine if email was a secret protocol and you had to pay money for your email composer...Oh yeah, that happened and everyone laughed in their faces and now it's free. When I once tried to implement VXML, I started trying to use openVXI, while hunting down the dependancies, I learned about spidermonkey and started to wonder why to bother rendering the XML into a script and running it when you could just write the whole IVR in JS. So I made res_js for Asterisk. When I started FS, I also made a mod_spidermonkey as we can all see. I also use XML in FS so I don't have a vendetta against XML, (I do get annoyed by those who over do it with XML). I just get an uneasy feeling about it and there is something to be said since we have 100% of the ingredients for VXML in FS and never implemented it. Like I said I am open to the idea of supporting it, to give it a fighting chance to get adopted. But teenagers are harder to get adopted as Andy pointed out =p On Fri, Jul 9, 2010 at 2:11 AM, Jan Berger wrote: So why does it "suck"? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andy Spitzer Sent: 9. juli 2010 06:27 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP2VXML Woof! On Mon, 05 Jul 2010 18:58:05 -0400, Anthony Minessale wrote: > It's almost like people think with all the stuff FS can do out of the > box, supporting this fly-by-night phenomenon called VXML that has failed > to gain traction after 5 years is somehow too valuable to contribute lol VoiceXML started 11 years ago, in 1999. And, still, "VoiceXML sucks" (my signature phrase for all posts about VoiceXML. Google it.) --Woof! _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100710/080470f3/attachment-0001.html From jan.berger at video24.no Sat Jul 10 05:50:06 2010 From: jan.berger at video24.no (Jan Berger) Date: Sat, 10 Jul 2010 14:50:06 +0200 Subject: [Freeswitch-users] SIP2VXML In-Reply-To: References: <39ED5E8D0A5D42BCA019A827DE9F7A4C@dell9400> <908418189EF44BB98023F68CD865A864@dell9400> Message-ID: <9619C6C805584257984D578E4F0E677B@dell9400> I don't use OpenVXI for reasons like - dead project, performance, reliability, non-scaling architecture, wrong dependencies - and so on. We have a target of 20,000 sessions on my laptop alone - and I have already secured that and more. We can do more, but we start running out of memory. What I expect out of FreeSWITCH/CCXML/VXML is that we can scale up as much as FreeSWITCH can without CCXML/VXML being the show-stopper. Then OpenVXI came out it was adopted by tier1 companies that still use the stack today. Some people consider that a reference, but they fail to understand that those companies have the strength needed to pick up half-done-work and make it right. Though the ones I know of still struggle with performance issues. So when it was first invented, people tried to sell the editors for money, the only implementation were commercial and there was no real acceptance to it. The first open source VXML stack was dependency-ridden and archaic to even build for many years. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100710/b3ffb4eb/attachment.html From anthony.minessale at gmail.com Sat Jul 10 08:25:35 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 10 Jul 2010 10:25:35 -0500 Subject: [Freeswitch-users] SIP2VXML In-Reply-To: <9619C6C805584257984D578E4F0E677B@dell9400> References: <39ED5E8D0A5D42BCA019A827DE9F7A4C@dell9400> <908418189EF44BB98023F68CD865A864@dell9400> <9619C6C805584257984D578E4F0E677B@dell9400> Message-ID: Our xml parser is very low on memory because its a small barebones lib based on c pointer black magic that we came across. Scary to look at but fast and low in memory usage. Our onboard asr and tts is only as bad in resource consumption as the engine you choose, but we do have mrcp. The big one, as you have already identified, is the javascript lib. The one we have does not scale along with FS for sure. However, that is when using the js context to run the whole call. Maybe in the case of vxml it would be a different model? Lua has shown the best parallel scalibility of all the embedded languages because its small and runs a separate interpreter for each instance. If we found a js implementation that works in a similar way it might scale. On Jul 10, 2010 7:56 AM, "Jan Berger" wrote: I don?t use OpenVXI for reasons like ? dead project, performance, reliability, non-scaling architecture, wrong dependencies ? and so on. We have a target of 20,000 sessions on my laptop alone ? and I have already secured that and more. We can do more, but we start running out of memory. What I expect out of FreeSWITCH/CCXML/VXML is that we can scale up as much as FreeSWITCH can without CCXML/VXML being the show-stopper. Then OpenVXI came out it was adopted by tier1 companies that still use the stack today. Some people consider that a reference, but they fail to understand that those companies have the strength needed to pick up half-done-work and make it right. Though the ones I know of still struggle with performance issues. So when it was first invented, people tried to sell the editors for money, the only implementation... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100710/9fadde5e/attachment.html From steveu at coppice.org Sat Jul 10 09:40:49 2010 From: steveu at coppice.org (Steve Underwood) Date: Sun, 11 Jul 2010 00:40:49 +0800 Subject: [Freeswitch-users] SIP2VXML In-Reply-To: <9619C6C805584257984D578E4F0E677B@dell9400> References: <39ED5E8D0A5D42BCA019A827DE9F7A4C@dell9400> <908418189EF44BB98023F68CD865A864@dell9400> <9619C6C805584257984D578E4F0E677B@dell9400> Message-ID: <4C38A291.3030300@coppice.org> On 07/10/2010 08:50 PM, Jan Berger wrote: > > I don?t use OpenVXI for reasons like ? dead project, performance, > reliability, non-scaling architecture, wrong dependencies ? and so on. > We have a target of 20,000 sessions on my laptop alone ? and I have > already secured that and more. We can do more, but we start running > out of memory. What I expect out of FreeSWITCH/CCXML/VXML is that we > can scale up as much as FreeSWITCH can without CCXML/VXML being the > show-stopper. > The folk at Commetrex might disagree with your first 4 complaints. They claim huge performance, reliability and scaling improvements in the version of OpenVXI they are pushing forward. > > Then OpenVXI came out it was adopted by tier1 companies that still use > the stack today. Some people consider that a reference, but they fail > to understand that those companies have the strength needed to pick up > half-done-work and make it right. Though the ones I know of still > struggle with performance issues. > > > > So when it was first invented, people tried to sell the editors for > money, the only implementation were commercial and there was no real > acceptance to it. The first open source VXML stack was > dependency-ridden and archaic to even build for many years. > > > > Steve From jan.berger at video24.no Sat Jul 10 11:45:21 2010 From: jan.berger at video24.no (Jan Berger) Date: Sat, 10 Jul 2010 20:45:21 +0200 Subject: [Freeswitch-users] SIP2VXML In-Reply-To: <4C38A291.3030300@coppice.org> References: <39ED5E8D0A5D42BCA019A827DE9F7A4C@dell9400> <908418189EF44BB98023F68CD865A864@dell9400> <9619C6C805584257984D578E4F0E677B@dell9400> <4C38A291.3030300@coppice.org> Message-ID: <11C63002E6084B8E8C19137661C551A2@dell9400> Improvements yes, but that's their BladewareVXML. They have done a bit of work on that + the version of OpenVXI I talk about is the one from 2003 without the GPL license. Jan -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steve Underwood Sent: 10. juli 2010 18:41 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP2VXML On 07/10/2010 08:50 PM, Jan Berger wrote: > > I don't use OpenVXI for reasons like - dead project, performance, > reliability, non-scaling architecture, wrong dependencies - and so on. > We have a target of 20,000 sessions on my laptop alone - and I have > already secured that and more. We can do more, but we start running > out of memory. What I expect out of FreeSWITCH/CCXML/VXML is that we > can scale up as much as FreeSWITCH can without CCXML/VXML being the > show-stopper. > The folk at Commetrex might disagree with your first 4 complaints. They claim huge performance, reliability and scaling improvements in the version of OpenVXI they are pushing forward. > > Then OpenVXI came out it was adopted by tier1 companies that still use > the stack today. Some people consider that a reference, but they fail > to understand that those companies have the strength needed to pick up > half-done-work and make it right. Though the ones I know of still > struggle with performance issues. > > > > So when it was first invented, people tried to sell the editors for > money, the only implementation were commercial and there was no real > acceptance to it. The first open source VXML stack was > dependency-ridden and archaic to even build for many years. > > > > Steve _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sos at sokhapkin.dyndns.org Sat Jul 10 13:53:15 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sat, 10 Jul 2010 16:53:15 -0400 Subject: [Freeswitch-users] mod_nibblebill and mod_xml_cdr problem In-Reply-To: <201007092041.13464.sos@sokhapkin.dyndns.org> References: <201007091953.24035.sos@sokhapkin.dyndns.org> <201007092013.54385.sos@sokhapkin.dyndns.org> <201007092041.13464.sos@sokhapkin.dyndns.org> Message-ID: <201007101653.15812.sos@sokhapkin.dyndns.org> I replaced mod_xml_cdr with the following in mod_cdr_csv: