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Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version)
by Peter Olsson 21 Dec '11

21 Dec '11

Did you try to remove all tables, so FS can create everything from scratch? /Peter ----- Reply message ----- Från: "king2kin" <xing2kin(a)yahoo.com> Datum: mån, dec 19, 2011 07:11 Rubrik: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> No, this occurs on the latest GIT version of freeswitch that I downloaded on Dec. 18, 2011 by command: { git clone git://git.freeswitch.org/freeswitch.git } The early GIT version (e.g. 2011-08-31) doesn't have such problems. The latest version can run on windows 2003, but keep printing out error messages in freeswitch.log and console, for example: [DEBUG] switch_core_sqldb.c:890 SQL ERR [no such table: sip_registrations] [delete from sip_registrations where (contact like '%TCP%' or status like '%TCP%' or status like '%TLS%') and hostname='W2k3T602' and network_ip like '%' and network_port like '%' and sip_username like '%' and mwi_user like '%' and mwi_host like '%' and orig_server_host like '%' and orig_hostname like '%'] Auto Generating Table! [DEBUG] switch_core_sqldb.c:897 SQL ERR [no such table: sip_subscriptions] [DEBUG] switch_core_sqldb.c:897 SQL ERR [no such table: sip_presence] [DEBUG] switch_core_sqldb.c:890 SQL ERR [no such table: sip_dialogs] ...... also, while running the new version of freeswitch, the following error messages keep coming up in freeswitch.log: { 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it } From: Brian West <brian(a)freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Sent: Monday, December 19, 2011 9:24 AM Subject: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) You're using a very old version of FreeSWITCH... why ? /b On Dec 18, 2011, at 7:11 PM, king2kin wrote: No, these tables were not created automatically by freeswitch on windows 2003 while MySQL is running. During the past three months, I had been playing around the GIT version of 2011-08-31, whose log file [freeswitch.log] didn't complain those database tables that I listed in my previous email. It seems to me that these db tables come up only in more recent GIT version. From: Jeff Lenk <jeff(a)jefflenk.com<mailto:jeff@jefflenk.com>> To: freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Sent: Monday, December 19, 2011 1:30 AM Subject: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) All the required tables should be created on first run. Same as always. Is that not happening? -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian(a)freeswitch.org<mailto:brian@freeswitch.org> http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com<http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://wiki.freeswitch.org<http://wiki.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@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://www.freeswitch.org/> !DSPAM:4eeed36a32763536013952!

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Linksys PAP2-T and echo
by Boris Kovalenko 21 Dec '11

21 Dec '11

Hello! I'm wondering if it is possible to solve echo with Linksys PAP2-T devices. Does anybody here uses it? I read many articles and still can't understand how to eleminate echo :( I read http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found that Asterisk has its own EC. Does Freeswitch has it too? I found this article http://docs.freeswitch.org/echo_can_page.html but can't understand - has Freeswitch EC or not? If yes - how to turn it on and off? -- Regards, Boris

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Re: [Freeswitch-users] G723 usage - need something special for transcoding?
by ib-freeswitch＠bzsolutions.it 21 Dec '11

21 Dec '11

Usually G729 it's the standard, if you phone support it, you don't need to do transcoding, test it in dialplan: <condition field="destination_number" expression="^5([0-9#]+)$"> <action application="set" data="custom_leg=ALEG"/> <action application="set" data="disable-transcoding=true"/> <action application="set" data="late-negotiation=true"/> <action application="set" data="inherit_codec=true"/> .... bridge bla bla bla.... http://wiki.freeswitch.org/wiki/Variable_inherit_codec Set on the phone only the codecs supperted by providers and it will work... An usefull tools to debug what happen it's SIPGREP You can find it in __WHERE_YOU_HAVE_FS_SOURCES/freeswitch/scripts/trace/sipgrep ----- Original Message ----- From: "Avi Marcus" <avi(a)avimarcus.net> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Wednesday, December 21, 2011 3:06:37 PM GMT +01:00 Amsterdam / Berlin / Bern / Rome / Stockholm / Vienna Subject: Re: [Freeswitch-users] G723 usage - need something special for transcoding? Ah, well, that would explain it then.. are there any other common open licenses other than pcmu/pcma for PSTN interconnects? e.g. they don't offer speex.. heck, the underlying carrier doesn't have g711. -Avi Marcus On Wed, Dec 21, 2011 at 3:54 PM, < ib-freeswitch(a)bzsolutions.it > wrote: Hi, freeswitch supports G.723 only in pass through, if you want to transcode, there is a sangoma card that can do it, commercial license is avaiable only for G.729. G723 / G729 / AMR are patented, and you can't use use the algorithm without permission of patent holders. If ALEG and BLEG support G.723 you can change codec negotiation mode in freeswitch. You can found usefull info here on WiKi: http://wiki.freeswitch.org/wiki/Codec_negotiation ----- Original Message ----- From: "Avi Marcus" < avi(a)avimarcus.net > To: "FreeSWITCH Users Help" < FreeSWITCH-users(a)lists.freeswitch.org > Sent: Wednesday, December 21, 2011 10:41:19 AM GMT +01:00 Amsterdam / Berlin / Bern / Rome / Stockholm / Vienna Subject: [Freeswitch-users] G723 usage - need something special for transcoding? Is g723 supposed to work out of the box? I added G723 to the codec negotiation.. and I see it chose it on leg B: Set Codec sofia/external/551137119355 G723/8000 30 ms 240 samples 6300 bits Leg A is... Set Codec sofia/internal/ 1000(a)sip.domain.com PCMU/8000 20 ms 160 samples 64000 bits Leg A can't hear leg B... I don't know about leg B. Is the ptimes an issue..? Shouldn't FS just transcode the call? -Avi Marcus _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Re: [Freeswitch-users] G723 usage - need something special for transcoding?
by ib-freeswitch＠bzsolutions.it 21 Dec '11

21 Dec '11

Hi, freeswitch supports G.723 only in pass through, if you want to transcode, there is a sangoma card that can do it, commercial license is avaiable only for G.729. G723 / G729 / AMR are patented, and you can't use use the algorithm without permission of patent holders. If ALEG and BLEG support G.723 you can change codec negotiation mode in freeswitch. You can found usefull info here on WiKi: http://wiki.freeswitch.org/wiki/Codec_negotiation ----- Original Message ----- From: "Avi Marcus" <avi(a)avimarcus.net> To: "FreeSWITCH Users Help" <FreeSWITCH-users(a)lists.freeswitch.org> Sent: Wednesday, December 21, 2011 10:41:19 AM GMT +01:00 Amsterdam / Berlin / Bern / Rome / Stockholm / Vienna Subject: [Freeswitch-users] G723 usage - need something special for transcoding? Is g723 supposed to work out of the box? I added G723 to the codec negotiation.. and I see it chose it on leg B: Set Codec sofia/external/551137119355 G723/8000 30 ms 240 samples 6300 bits Leg A is... Set Codec sofia/internal/ 1000(a)sip.domain.com PCMU/8000 20 ms 160 samples 64000 bits Leg A can't hear leg B... I don't know about leg B. Is the ptimes an issue..? Shouldn't FS just transcode the call? -Avi Marcus _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Freeswitch inside-out
by Valter Nogueira 21 Dec '11

21 Dec '11

I have pointed out my emacs+cscope to freeswtich source code and I am trying to figure out how does it work? So far, I figure out that switch_loadable_module_init creates one thread per runtime module and that task_thread_loop plays an important role in running freeswitch. Is there any doc about freeswitch internals, since I yet want to discover how channels are initiated and bridged together (and how the "talk" flows between them) thanks, Valter * *

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G723 usage - need something special for transcoding?
by Avi Marcus 21 Dec '11

21 Dec '11

Is g723 supposed to work out of the box? I added G723 to the codec negotiation.. and I see it chose it on leg B: Set Codec sofia/external/551137119355 G723/8000 30 ms 240 samples 6300 bits Leg A is... Set Codec sofia/internal/1000(a)sip.domain.com PCMU/8000 20 ms 160 samples 64000 bits Leg A can't hear leg B... I don't know about leg B. Is the ptimes an issue..? Shouldn't FS just transcode the call? -Avi Marcus

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Call/Transaction Does Not Exist
by Joe Flemmings 21 Dec '11

21 Dec '11

When i call from one extension and hangup, the other call does not hangup and continues ringing. Please see log bellow. Any ideas? U 10.10.78.24:55479 -> 10.10.78.36:5060 INVITE sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213@10.10.78.36;user=phone>..CSeq: 1 INVITE..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: < sip:pbxjker_2564@10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MES SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: en..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=- 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone..c=IN IP4 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8 18 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000.. # U 10.10.78.36:5060 -> 10.10.78.24:52732 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One" <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: < sip:213@10.10.78.36;user=phone> ..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 1 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 -0500..Content-Length: 0.... # U 10.10.78.36:5060 -> 10.10.78.24:52732 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213@6 6.113.78.36;user=phone>;tag=71XvH76tyavaF..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 1 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 -0500..Accept: appl ication/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, prese nce, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Proxy-Authenticate: Digest realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac ", algorithm=MD5, qop="auth"..Content-Length: 0.... # U 10.10.78.24:55479 -> 10.10.78.36:5060 ACK sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <si p:213@10.10.78.36;user=phone>;tag=71XvH76tyavaF..CSeq: 1 ACK..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: < sip:pbxjker_2564@10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL, OPTION S, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: en..Max-Forwards: 70..Content-Length: 0.... # U 10.10.78.24:55479 -> 10.10.78.36:5060 INVITE sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213@10.10.78.36;user=phone>..CSeq: 2 INVITE..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: < sip:pbxjker_2564@10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MES SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: en..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Proxy-Authoriz ation: Digest username="pbxjker_2564", realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000001, uri="sip:213@10.10.78.36:5060 ;user=phone", response="82a75edb7ba97bfe7dea585684701d0c", algorithm=MD5..Max-Forwards: 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=- 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone ..c=IN IP4 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8 18 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000.. # U 10.10.78.36:5060 -> 10.10.78.24:52732 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: < sip:213@10.10.78.36;user=phone> ..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 -0500..Content-Length: 0.... # U 10.10.78.36:5060 -> 10.10.78.24:52783 INVITE sip:pbxjker_213@10.10.78.24:52783 SIP/2.0..Via: SIP/2.0/UDP 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..Max-Forwards: 69..From: "Polycom One" <sip:pbxjker_2564@10.10.78.36>;tag=9KgeNX81 rv8Fp..To: <sip:pbxjker_213@10.10.78.24:52783>..Call-ID: 4cedf474-6ee4-122f-0f99-00093d12d5dd..CSeq: 18845076 INVITE..Contact: < sip:mod_sofia@10.10.78.36:5060>..User-Agent: FreeSWITCH-mod_sofia/1.0. head-git-1086cba 2011-05-23 22-51-43 -0500..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: 337..X-FS-Support: update_display..Remote-Party-ID: "Polycom One" <sip:pbxjker_2564@10.10.78.36>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1318333156 1318333157 IN IP4 10.10.78.36..s=FreeSWITCH..c=IN IP4 10.10.78.36..t=0 0..m=audio 29124 RTP/AVP 9 98 99 0 8 3 18 101 13..a=rtpmap:98 G7221/32000..a=fmtp:98 bitrate=48000..a=rtpmap:99 G7221/16000..a=fmtp:99 bitrate=320 00..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ptime:20.. # U 10.10.78.24:55516 -> 10.10.78.36:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two" <sip:pbxjker_213@99.65.2 17.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID: 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact: < sip:pbxjker_213@10.10.78.24:52783>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/3.2 .2.0477..Accept-Language: en..Content-Length: 0.... # U 10.10.78.24:55516 -> 10.10.78.36:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two" <sip:pbxjker_213@99.65. 217.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID: 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact: < sip:pbxjker_213@10.10.78.24:52783>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/3. 2.2.0477..Allow-Events: talk,hold,conference..Accept-Language: en..Content-Length: 0.... # U 10.10.78.36:5060 -> 10.10.78.24:52732 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: < sip:213@10.10.78.36;user=phone >;tag=8aQNK2QyUKjXa..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 INVITE..Contact: <sip:213@10.10.78.36:5060;transport=udp>..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-2 3 22-51-43 -0500..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..A llow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Content-Length: 0..Remote-Party-ID: "Outbound Call" <pbxjker _213>;party=calling;privacy=off;screen=no.... # U 10.10.78.24:55479 -> 10.10.78.36:5060 CANCEL sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2 13(a)10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: < sip:pbxjker_2564@10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCR IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest username="pbxjker_2564", realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-462 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002, uri="sip:213@10.10.78.36:5060;user=phone", response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards: 70..Content-Length: 0.. .. # U 10.10.78.36:5060 -> 10.10.78.24:5060 SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213@66.11 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 CANCEL..Content-Length: 0.... # U 10.10.78.24:55479 -> 10.10.78.36:5060 CANCEL sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2 13(a)10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: < sip:pbxjker_2564@10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCR IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest username="pbxjker_2564", realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-462 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002, uri="sip:213@10.10.78.36:5060;user=phone", response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards: 70..Content-Length: 0.. .. # U 10.10.78.36:5060 -> 10.10.78.24:5060 SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213@66.11 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 CANCEL..Content-Length: 0.... # U 10.10.78.24:55479 -> 10.10.78.36:5060 CANCEL sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2 13(a)10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: < sip:pbxjker_2564@10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCR IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest username="pbxjker_2564", realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-462 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002, uri="sip:213@10.10.78.36:5060;user=phone", response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards: 70..Content-Length: 0.. .. # U 10.10.78.36:5060 -> 10.10.78.24:5060 SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213@66.11 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 CANCEL..Content-Length: 0.... #

3 6

quick question about eavesdrop
by Seven Du 21 Dec '11

21 Dec '11

Hi, first sorry I'm using an old version of FS FreeSWITCH Version 1.0.head (hacked-20110626T115851Z). Just a quick question I found that when I eavesdrop a bridged call between a sip channel and a skinny channel, if the eavesdrop applied to the sip uuid then everything is ok, but if applied to the skynny uuid, voice is stretched (speaks slower and slower), any hint on this? here are what I see in show channels. eavesdrop sip uuid uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid dc11d8a7-3a1c-42d6-a936-577438b44a8c,inbound,2011-12-21 22:14:41,1324476881,sofia/internal/6802@6802:5060,CS_EXECUTE,6802,6802,192.168.1.118,6807,bridge,skinny/internal/6807,XML,default,PCMU,8000,64000,PCMU,8000,64000,,ola,6802@6802,,ACTIVE,Outbound Call,internal/6807,SEND,dc11d8a7-3a1c-42d6-a936-577438b44a8c b07e40f4-1978-4142-b4fe-35e3ee00c8bd,outbound,2011-12-21 22:14:41,1324476881,SKINNY/internal/6807,CS_EXCHANGE_MEDIA,6802,6802,192.168.1.118,internal/6807,,,XML,default,PCMU,8000,64000,PCMU,8000,64000,,ola,,,ACTIVE,,,,dc11d8a7-3a1c-42d6-a936-577438b44a8c ef2aa92a-dad1-4738-99a2-0f3266b1f79f,outbound,2011-12-21 22:14:49,1324476889,sofia/internal/sip:6800@192.168.1.210:5062,CS_EXECUTE,OLA-QUEUE,00000000,,6800,eavesdrop,dc11d8a7-3a1c-42d6-a936-577438b44a8c,,default,L16,8000,128000,PCMU,8000,64000,,ola,6800@192.168.1.200,,ACTIVE,Outbound Call,6800,RECV,6a208fec-2297-407f-ba74-446122a91974 eavesdrop sccp uuid uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid 1f5118b4-2a9f-4682-adf6-1a378b381ee3,inbound,2011-12-21 22:13:19,1324476799,sofia/internal/6802@6802:5060,CS_EXECUTE,6802,6802,192.168.1.118,6807,bridge,skinny/internal/6807,XML,default,PCMU,8000,64000,PCMU,8000,64000,,ola,6802@6802,,ACTIVE,Outbound Call,internal/6807,SEND,1f5118b4-2a9f-4682-adf6-1a378b381ee3 7dfb8790-3d0d-4fbf-a716-7bf31285b5b0,outbound,2011-12-21 22:13:19,1324476799,SKINNY/internal/6807,CS_EXCHANGE_MEDIA,6802,6802,192.168.1.118,internal/6807,,,XML,default,PCMU,8000,64000,PCMU,8000,64000,,ola,,,ACTIVE,,,,1f5118b4-2a9f-4682-adf6-1a378b381ee3 df975188-fe2a-4817-b046-f911bf158447,outbound,2011-12-21 22:13:32,1324476812,sofia/internal/sip:6800@192.168.1.210:5062,CS_EXECUTE,OLA-QUEUE,00000000,,6800,eavesdrop,7dfb8790-3d0d-4fbf-a716-7bf31285b5b0,,default,L16,8000,128000,PCMU,8000,64000,,ola,6800@192.168.1.200,,ACTIVE,Outbound Call,6800,RECV,14fe5a58-75ea-416e-9f3d-30c57b7fa626 thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

1 0

How to invite an endpoint sip phone directly?
by envelopes envelopes 21 Dec '11

21 Dec '11

The phone is not registered in FS. Thanks

1 0

Guidelines for setting up contexts per user / gateway
by Nestor A Diaz 21 Dec '11

21 Dec '11

Hello Freeswitch Users List, I am trying to emulate asterisk behavior for user / gateway context under FreeSWITCH. For users inside the 'directory' directory I can set up different contexts using the 'user_context' variable, solved. But when i have to deal with external gateways sip_profiles seems to be the best place to put the configuration, and if i want to use just one sip profile due to firewall restricctions i am limited to just one context. According to FreeSWITCH FAQ: Q: How do I assign endpoints to contexts with different sets of extensions <http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_How_do_I_assign_endpoints…> Here are the different possible approaches: 1. use 1 profile per context you want to route to (each one needs a distinct ip:port) 2. use different domains in the registration data and use the auto context thing 3. send them all to a common context and execute_exten or transfer to somewhere else 4. send them to an IVR to decide where they go 5. use xml_curl to make it dish out a different dialplan based on who they are in the list of data you are fed I don't want to use 1 profile per context, so option 1 discarded. I don't want to use different domains in the registration data, since it have no semantic for a one company, so option 2 discarded, but for gateways it makes sense. I don't want to use an IVR for every user call, so option 4 discarded. So that gives me two choices, use xml_curl, which means i have to set up a webserver but that seems overkill, it seems to be an superset of option 3, so that left me with option 3 using just static XML files. I found this example on the dialplan XML: <extension name="Inbound_external"> <condition regex="any"> <regex field="${sip_from_host}" expression="domainA"/> <regex field="${sip_from_uri}" expression="1234567890@domainB"/> <regex field="${sip_from_uri}" expression="user@domainC"/> <regex field="caller_id_name" expression="^(John Smith)$"/> <regex field="caller_id_number" expression="^(55512341)|(55512342)|(55512343)$"/> <action application="set" data="domain_name=domainZ"/> <action application="transfer" data="${destination_number} XML domainZ"/> </condition> </extension> It seems to be the first place to start adapting to internal extension and transfering to the correct dialplan, anybody have a more refined example ? which variables can i trust / use in order to validate a call from a particular gateway and then set up a context transfer according to them. p.d. anybody knows about a good cdr web stats software for freeswitch ? i have used cdr-stats for asterisk in the past, newer version seems to have freeswitch support on the way, but stable ? any suggestions will be apreciated . Thank you !! -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-485-3020 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211@tiendalinux.com Email/MSN: nestor(a)tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia

1 0