August 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Polycom firmware and TLS
by Kristian Kielhofner 23 Aug '12

23 Aug '12

All, Which versions of Polycom firmware have you had the most success using TLS with? Thanks! -- Kristian Kielhofner

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T.38 three re-invites then bye
by Leon de Rooij 23 Aug '12

23 Aug '12

Hi all, I've been having a problem with a new provider with sending them faxes over T.38. I don't think FreeSWITCH is to blame at all, but maybe FS can fix my issue, or someone here can help me find the cause of the problem ? The call flow is as follows: CPE 1.1.1.1 | V 2.2.2.2 FreeSWITCH 3.3.3.3 | V 4.4.4.4 Provider In FreeSWITCH, I've set t38-passthru to true on the profile where the CPE INVITEs to. Further, I'm doing late negotiation (while offering the a-leg codecs to the b-leg and the other way around). RTP is flowing through FS. In short, this is happening: CPE -> FS -> Provider INVITE (with audio PCMA, PCMU codecs) Provider -> FS -> CPE 200 OK (with audio PCMA) CPE -> FS -> Provider reINVITE (with T.38 in SDP) Provider -> FS -> CPE 200 OK (with T.38 in SDP - although with different error correction, is this a problem?) Provider -> FS -> CPE reINVITE (again with T.38 in SDP) CPE -> FS -> Provider 200 OK (with T.38 in SDP) CPE -> FS -> Provider reINVITE (with T.38 in SDP) Provider -> FS -> CPE BYE (of course, or it'll loop forever) Does a setting in FreeSWITCH exist so that it can change the error correction (because both endpoints seem to want a different one ?) Could that be the cause of the problem ? Hope anyone can help. Thanks, Leon Here's the trace - I left out the tryings, acks, removed some unnecessary sip headers to make things a bit less verbose: U 2012/08/23 15:34:16.334938 3.3.3.3:5060 -> 4.4.4.4:5060 INVITE sip:+888@4.4.4.4 SIP/2.0. From: "+777" <sip:+777@provider.voip.tel>;tag=70aprjj2Z1QKK. To: <sip:+888@4.4.4.4>. Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa. CSeq: 32528364 INVITE. Contact: <sip:gw+provider-voip-tel@3.3.3.3:5060;transport=udp;gw=provider-voip-tel>. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 253. X-Designated-Service: fax/t38. . v=0. o=FreeSWITCH 1345708964 1345708965 IN IP4 3.3.3.3. s=FreeSWITCH. c=IN IP4 3.3.3.3. t=0 0. m=audio 19892 RTP/AVP 8 0 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:30. U 2012/08/23 15:34:18.192303 4.4.4.4:5060 -> 3.3.3.3:5060 SIP/2.0 180 Ringing. From: "+777" <sip:+777@provider.voip.tel>;tag=70aprjj2Z1QKK. To: <sip:+888@4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad. Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa. CSeq: 32528364 INVITE. Server: DC-SIP/2.0. Content-Length: 0. Contact: <sip:+888@4.4.4.4:5060;transport=udp>. . U 2012/08/23 15:34:23.904745 4.4.4.4:5060 -> 3.3.3.3:5060 SIP/2.0 200 OK. From: "+777" <sip:+777@provider.voip.tel>;tag=70aprjj2Z1QKK. To: <sip:+888@4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad. Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa. CSeq: 32528364 INVITE. Server: DC-SIP/2.0. Contact: <sip:+888@4.4.4.4:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 229. . v=0. o=- 3554717663 3554717663 IN IP4 4.4.4.4. s=-. c=IN IP4 4.4.4.4. t=0 0. m=audio 22054 RTP/AVP 8 101. a=ptime:30. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=silenceSupp:off - - - -. U 2012/08/23 15:34:23.996849 3.3.3.3:5060 -> 4.4.4.4:5060 INVITE sip:+888@4.4.4.4:5060;transport=udp SIP/2.0. From: "+777" <sip:+777@provider.voip.tel>;tag=70aprjj2Z1QKK. To: <sip:+888@4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad. Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa. CSeq: 32528365 INVITE. Contact: <sip:gw+provider-voip-tel@3.3.3.3:5060;transport=udp;gw=provider-voip-tel>. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 339. X-Designated-Service: fax/t38. . v=0. o=FreeSWITCH 1345708964 1345708966 IN IP4 3.3.3.3. s=FreeSWITCH. c=IN IP4 3.3.3.3. t=0 0. m=image 19892 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:14400. a=T38FaxTranscodingMMR. a=T38FaxTranscodingJBIG. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:500. a=T38FaxMaxDatagram:512. a=T38FaxUdpEC:t38UDPFEC. U 2012/08/23 15:34:24.093598 4.4.4.4:5060 -> 3.3.3.3:5060 SIP/2.0 200 OK. From: "+777" <sip:+777@provider.voip.tel>;tag=70aprjj2Z1QKK. To: <sip:+888@4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad. Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa. CSeq: 32528365 INVITE. Server: DC-SIP/2.0. Contact: <sip:+888@4.4.4.4:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 272. . v=0. o=- 3554717663 3554717664 IN IP4 4.4.4.4. s=-. c=IN IP4 4.4.4.4. t=0 0. m=image 22054 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:14400. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:72. a=T38FaxMaxDatagram:316. a=T38FaxUdpEC:t38UDPRedundancy. U 2012/08/23 15:34:33.885255 4.4.4.4:5060 -> 3.3.3.3:5060 INVITE sip:gw+provider-voip-tel@3.3.3.3:5060;transport=udp;gw=provider-voip-tel SIP/2.0. Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa. From: <sip:+888@4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad. To: "+777" <sip:+777@provider.voip.tel>;tag=70aprjj2Z1QKK. CSeq: 693734809 INVITE. Contact: <sip:+888@4.4.4.4:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 272. . v=0. o=- 3554717663 3554717665 IN IP4 4.4.4.4. s=-. c=IN IP4 4.4.4.4. t=0 0. m=image 22054 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:14400. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:72. a=T38FaxMaxDatagram:316. a=T38FaxUdpEC:t38UDPRedundancy. U 2012/08/23 15:34:33.939092 3.3.3.3:5060 -> 4.4.4.4:5060 SIP/2.0 200 OK. From: <sip:+888@4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad. To: "+777" <sip:+777@provider.voip.tel>;tag=70aprjj2Z1QKK. Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa. CSeq: 693734809 INVITE. Contact: <sip:gw+provider-voip-tel@3.3.3.3:5060;transport=udp;gw=provider-voip-tel>. Accept: application/sdp. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 345. . v=0. o=FreeSWITCH 1345708964 1345708967 IN IP4 3.3.3.3. s=FreeSWITCH. c=IN IP4 3.3.3.3. t=0 0. m=image 19892 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:14400. a=T38FaxTranscodingMMR. a=T38FaxTranscodingJBIG. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:72. a=T38FaxMaxDatagram:512. a=T38FaxUdpEC:t38UDPRedundancy. U 2012/08/23 15:34:33.950201 3.3.3.3:5060 -> 4.4.4.4:5060 INVITE sip:+888@4.4.4.4:5060;transport=udp SIP/2.0. From: "+777" <sip:+777@provider.voip.tel>;tag=70aprjj2Z1QKK. To: <sip:+888@4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad. Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa. CSeq: 32528366 INVITE. Contact: <sip:gw+provider-voip-tel@3.3.3.3:5060;transport=udp;gw=provider-voip-tel>. User-Agent: c4-provider-ims-intx. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 338. X-Designated-Service: fax/t38. . v=0. o=FreeSWITCH 1345708964 1345708968 IN IP4 3.3.3.3. s=FreeSWITCH. c=IN IP4 3.3.3.3. t=0 0. m=image 19892 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:14400. a=T38FaxTranscodingMMR. a=T38FaxTranscodingJBIG. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:72. a=T38FaxMaxDatagram:512. a=T38FaxUdpEC:t38UDPFEC. U 2012/08/23 15:34:34.043165 4.4.4.4:5060 -> 3.3.3.3:5060 BYE sip:gw+provider-voip-tel@3.3.3.3:5060;transport=udp;gw=provider-voip-tel SIP/2.0. Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa. From: <sip:+888@4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad. To: "+777" <sip:+777@provider.voip.tel>;tag=70aprjj2Z1QKK. CSeq: 693734810 BYE. Content-Length: 0. Reason: Q.850; cause=41;text="Temporary failure". .

2 1

Gsmopen : no audio in huawei e1550
by Sameer Khan 23 Aug '12

23 Aug '12

Hi, I can't get audio working with e1550 I have this config : <interface id="1" name="gsm01"> <param name="controldevice_name" value="/dev/ttyUSB0"/> <param name="controldevice_audio_name" value="/dev/ttyUSB3"/> I tried swaping 0 and 3 but having same results I am originating a call from softphone and answering it on cell phone but softphone doesn't get notified of being answered and keeps ringing Please help Thanks

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Re: [Freeswitch-users] mod_gsmopen serial init failed
by Mi Ke 23 Aug '12

23 Aug '12

It appears that selinux policies prevented freeswitch user from accessing /dev/ttyUSB* [root@freeswitch ~]# ls -l /dev/ttyUSB* crw-rw----. 1 root dialout 188, 0 Aug 21 10:57 /dev/ttyUSB0 crw-rw----. 1 root dialout 188, 1 Aug 21 10:48 /dev/ttyUSB1 crw-rw----. 1 root dialout 188, 2 Aug 21 10:48 /dev/ttyUSB2 so adding freeswitch user to dialout group by running "usermod -a -G dialout freeswitch" worked for me. Have a nice day Mike ----- Original Message ----- From: Mi Ke Sent: 08/21/12 12:58 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] mod_gsmopen serial init failed Hi All, I have problems getting mod_gsmopen to see my E1550. Both minicom and wvdial see the dongle on /dev/ttyUSB0: ttyUSB0<*1>: ATQ0 V1 E1 -- OK ttyUSB0<*1>: ATQ0 V1 E1 Z -- OK ttyUSB0<*1>: ATQ0 V1 E1 S0=0 -- OK ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 -- OK ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 -- OK ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0 -- OK ttyUSB0<*1>: Modem Identifier: ATI -- Manufacturer: huawei ttyUSB0<*1>: Speed 9600: AT -- OK ttyUSB0<*1>: Max speed is 9600; that should be safe. ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0 -- OK Welcome to minicom 2.3 OPTIONS: I18n Compiled on Aug 19 2010, 05:50:19. Port /dev/ttyUSB0 Press CTRL-A Z for help on special keys AT S7=45 S0=0 L1 V1 X4 &c1 E1 Q0 OK at+cpin? +CPIN: READY OK but mod_gsmopen does not want to see it: 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1203 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1203 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 0 èéòàù 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1204 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1204 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 1 ç°§^£ 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1205 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1205 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 2 новости 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1206 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1206 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 3 ﺎﻠﺠﻤﻋﺓ 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1207 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1207 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 4 ראת 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1208 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1208 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 5 לק 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1209 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1209 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 6 人大 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1226 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1226 ][none ][-1,-1,-1] globals.debug=0 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1228 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1228 ][none ][-1,-1,-1] globals.debug=8 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1234 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1234 ][none ][-1,-1,-1] globals.dialplan=XML 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1240 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1240 ][none ][-1,-1,-1] globals.context=external 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1231 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1231 ][none ][-1,-1,-1] globals.hold_music= 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1237 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1237 ][none ][-1,-1,-1] globals.destination=5000 2012-08-21 09:18:18.973501 [WARNING] mod_gsmopen.cpp:1653 rev ccae5cd|07bc7ba[(nil)|37 ][WARNINGA 1653 ][gsm00 ][-1, 0, 0] STARTING interface_id=1 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1654 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1654 ][gsm00 ][-1, 0, 0] id=1 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1655 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1655 ][gsm00 ][-1, 0, 0] name=gsm00 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1656 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1656 ][gsm00 ][-1, 0, 0] hold-music= 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1657 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1657 ][gsm00 ][-1, 0, 0] context=default 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1658 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1658 ][gsm00 ][-1, 0, 0] dialplan=XML 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1659 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1659 ][gsm00 ][-1, 0, 0] destination=5000 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1660 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1660 ][gsm00 ][-1, 0, 0] controldevice_name=/dev/ttyUSB0 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1661 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1661 ][gsm00 ][-1, 0, 0] controldevice_audio_name=/dev/ttyUSB2 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1663 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1663 ][gsm00 ][-1, 0, 0] gsmopen_serial_sync_period=300 2012-08-21 09:18:18.973501 [ERR] gsmopen_protocol.cpp:122 rev ccae5cd|07bc7ba[(nil)|37 ][ERRORA 122 ][gsm00 ][-1, 0, 0] port /dev/ttyUSB0, NOT open 2012-08-21 09:18:18.973501 [ERR] mod_gsmopen.cpp:1670 rev ccae5cd|07bc7ba[(nil)|37 ][ERRORA 1670 ][gsm00 ][-1, 0, 0] gsmopen_serial_init failed 2012-08-21 09:18:18.973501 [ERR] mod_gsmopen.cpp:1671 rev ccae5cd|07bc7ba[(nil)|37 ][ERRORA 1671 ][gsm00 ][-1, 0, 0] STARTING interface_id=1 FAILED 2012-08-21 09:18:18.973501 [ERR] mod_gsmopen.cpp:2928 rev ccae5cd|07bc7ba[(nil)|37 ][ERRORA 2928 ][gsm00 ][-1, 0, 0] ALARM on interface gsm00: 2012-08-21 09:18:18.973501 [CONSOLE] switch_loadable_module.c:1328 Successfully Loaded [mod_gsmopen] 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'gsmopen' 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsm' 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen' 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen_boost_audio' 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen_dump' 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen_sendsms' 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'sms' Dongle is voice enabled, PIN request is disabled...What do I do wrong ? Cheers / Mike

2 1

Wrong energy level announcement in conference
by Emrah 22 Aug '12

22 Aug '12

Hello, My conference room is set with default energy-level = 300 The first time I use the 7 key, I hear "Zero", while in reality, I went from 300 to 100. Where can we configure the adjustment levels and matching audio prompts? So to have zero for 0, one for 100, two for 200, and have each 7 or 9 key press change the value by 100? Best to all, Emrah

1 0

FreeSWITCH Community Conference Call Today (Important Information - Please Read)
by Michael Collins 22 Aug '12

22 Aug '12

Hello all! The conference will start in less than an hour. Here's our agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22 We've made a change to the system. Everyone will come in muted. When you press 0 it will send a message into the #FreeSWITCH IRC channel - "user has a question". One of the conference moderators will unmute you so that you can ask your question. We are hoping this will make the conference more enjoyable for everyone. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

1 0

Routing inbound calls to extentions with did intact
by Ian Halliday 22 Aug '12

22 Aug '12

Hello All, I use freeswitch to route inbound calls to some of the PBX's we host, and I use the following method (Below) to do so. It's been working great (20,000 calls in the first month without a problem), but I wanted to get a second opinion to see if there was a better way to do such a thing (keeping the DID intact). Any help would be appreciated. (I used an example DID and extention) <extension name="setdid"> <condition field="destination_number" expression="^setdid$"> <action application="set" data="dialed_ext=${did}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="disable_q850_reason=true"/> <action application="bridge" data="sofia/internal/${did}${regex(${sofia_contact(${user})}|^[^\(a)]+(.*)|%1)}"/> </condition> </extension> <extension name="13125551234"> <condition field="destination_number" expression="^13125551234$"> <action application="set" data="user=user1002"/> <action application="set" data="did=${destination_number}"/> <action application="execute_extension" data="setdid"/> </condition> </extension> Thanks!

2 1

Build on Apple PPC?
by Lawrence Conroy 22 Aug '12

22 Aug '12

Hi there, I'm trying to build 1.21 stable from git (replacing my current 1.06 tarball build) on a TiBook (PPC G4) running Leopard (10.5.8). It's failing during configure -- "configure: error: no usable libodbc; please install unixodbc devel package or equivalent". I used the command line flag --disable-core-odbc-support, so don't understand ... why is configure looking for odbc at all? anyone have an idea why this flag is ignored? all the best, Lawrence

2 1

Wrong transport=tcp in SIP Contact field.
by Vladimir Ananiev 22 Aug '12

22 Aug '12

Hello all. I have a problem with some specific softphone connecting to the Freswitch system. This softphone connects with UDP protocol, but sends additional ;transport=tcp line in the end of Contact field, that makes freeswitch thinking that the transport is really TCP, not the UDP. And after that incoming calls are not working. Outgoing calls from that softphone are working just fine. Is there a way to force freeswitch to use UDP in this case ? I already have <param name="bind-params" value="transport=udp" /> in sip profile settings... See the packet dump from wireshark: User Datagram Protocol, Src Port: na-localise (5062), Dst Port: sip (5060) Source port: na-localise (5062) Destination port: sip (5060) Length: 793 Checksum: 0x34ff [validation disabled] Session Initiation Protocol Request-Line: REGISTER sip:sip.test.com SIP/2.0 Method: REGISTER Request-URI: sip:sip.test.com [Resent Packet: False] Message Header CSeq: 535 REGISTER Via: SIP/2.0/UDP 192.168.0.12:5062;branch=z9hG4bK2edb4d2b-d007-1910-9210-1cc1de60bc3c;rport User-Agent: TMSoftPhone [truncated] Authorization: Digest username="621", realm="sip.test.com", nonce="c4f876ae-eb58-11e1-8919-6df0d0a4f1d4", uri="sip:sip.test.com", algorithm=MD5, response="52625726279aaf004b61871164b61088", cnonce="0560cd01-d007-1910-91f From: <sip:621@sip.test.com>;tag=25ffcc01-d007-1910-91fb-1cc1de60bc3c Call-ID: 25ffcc01-d007-1910-91fa-1cc1de60bc3c To: <sip:621@sip.test.com> Contact: <sip:621@192.168.0.12:5062;transport=tcp> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 120 Content-Length: 0 Max-Forwards: 70 --- Ананьев Владимир Викторович Технический Директор ЗАО "СВЯЗЬ" Тел. +7 495 7238000, доб. 101 Факс.+7 495 7308111 E-mail: ananiev(a)svyaz.com

2 2

Re: [Freeswitch-users] FreeSWITCH Question
by Alex Lake 22 Aug '12

22 Aug '12

You want to do 90cps on a machine that has just 4G RAM?! On 21 Aug 2012, at 23:38, "Jason Caulfield" <jason.caulfield(a)intermetro.net> wrote: > It is not so much the cps but the concurrent calls. > > I am looking at maintaining a 30 cps with a 5 min. avg. call length which results in a sustained 9000 concurrent calls. > > This results in 85% mem. usage on a 4 GB machine, However CPU is at 25% (total) on a dual core with hyper threading. > > Current hardware does not support increase memory. > > I am striving for about 75% cpu (appox. 90 cps). > > Jason > > > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Avi Marcus > Sent: Tuesday, August 21, 2012 2:58 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] FreeSWITCH Question > > Do you have such a large cps/concurrent volume that you're actually seeing a performance hit? > > -Avi > > > On Wed, Aug 22, 2012 at 12:35 AM, Jason Caulfield <jason.caulfield(a)intermetro.net> wrote: > I am using FreeSWITCH in media bypass mode to decrease memory and cpu usage. > > I notice that mod_sofia creates threads for each call leg to maintain session context. > > Do you know of a way to configure FreeSWITCH to use a table to maintain the context when in media bypass mode to reduce the number of threads? > > I am hoping that this will speed things up by reducing thread context switching and reduce memory usage by decreasing memory allocation for each thread. > > Thanks for the help, > Jason > > _________________________________________________________________________ > 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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