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How to setup a span as PRI_NET
by lakshmanan ganapathy 17 Apr '10

17 Apr '10

Hi all, In my office we have a Hard PBX, with some 4 extensions. We also have sangoma A102 card. >From the Hard PBX, if 0 is pressed, it is setup in a way that it will go to outside world. I've connected that line to span2 of the card. The span2 in the A102 card, is configured as PRI_NET. wanrouter status shows connected for the span2. But if I dial from the extension, I got the following in the sangoma_dchan log. 2010-04-03 12:49:49 INCOMING [ 00 01 54 50 08 02 01 64 05 04 03 80 90 a3 6c 0c 01 81 34 34 33 39 31 31 34 36 30 30 7d 02 91 81 ] Call Ref:0164 Type:Setup (0x5) Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64 Kbit/s(16) L1Prot:G.711 A-Law(3) Calling Party Number:4439114600(l:10) plan:isdn(1) type:unknown(0)scr:user, passed(1) pres:allowed(0) High-Layer Compatibility:Undecodedhex [ 7d 02 91 81 ] 2010-04-03 12:49:49 OUTGOING [ 02 01 50 56 08 02 81 64 5a 08 02 82 a2 ] Call Ref:0164 Type:Release Compl (0x5a) Cause:coding:ITU-T(0) location:Public network, local user(2) val:No Circuit/Channel Available(34) The call in not reaching freeswitch ( I enabled debug log. But nothing is printing in it ). Can someone suggest how to make this work. Please ask me if you need more information?, since I don't know what to give now.

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Wiki info: fs_ivrd
by Michael Collins 17 Apr '10

17 Apr '10

Gang, I just finished some awesome documentation on using fs_ivrd and the Perl ESL::IVR module: http://wiki.freeswitch.org/wiki/Ivrd If anyone else is using fs_ivrd and has some examples then by all means please add to this page. Also, feel free to check the page's accuracy. Thanks! -MC

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performance comparison between centos and freebsd
by Woody Dickson 17 Apr '10

17 Apr '10

Hi, I have tested with running freeswitch1.0.5 on freebsd and centos 5.4. The expectation is that freeswitch 1.0.5 acting as media proxy would perform better in freebsd, but I found that freebsd can only sustain half of the total concurrent calls as in centos 5.4 (120 vs 60). The test is run on both ATOM CPU and VIA c7 and the result is relatively the same. Does anyone know why? Is this some sort of setting issues in freebsd kernel? I have tried with pure freebsd and pfsense and the result is the same. Woody

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Dial Group & Handy-Tone 386
by Garrison Hoffman 16 Apr '10

16 Apr '10

I'm trying to set up a dial group using an IP phone (Grandstream GXP 2020) and a phone via Handy-Tone 386, but there seems to be an issue with the Handy-tone. Routing a DID directly to either extension works without a hitch, and both phones ring using the following dial group: <extension name="group_dial_test"> <condition field="destination_number" expression="^2000$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="group/test@${domain_name}"/> </condition> </extension> Incidentally, I've also seen the following syntax, is there a difference? <action application="bridge" data="${group_call(test@${domain_name}+A)}"/> The only problem is that when the call is answered by the phone connected to the Handy-Tone FXS port, neither end has audio. The GXP 2020 works in all cases.

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DTMF issues
by Dan Lane 15 Apr '10

15 Apr '10

I've got an issue where DTMF has stopped working with outbound calls (DTMF works on inbound calls from them) via one of my service providers (still works fine with several other providers) so I assumed it was a change the termination provider had made and duly reported it to them. Their response was: "We have analysed rtp from xlite to freeswitch to us and xlite to us 1. volume is being made much quieter by freeswitch, i.e. standard dtmf volume 10 is being re-written to 7 2. dtmf is shorter, i.e messages from phones are being re-written to a really short version 3. timestamp discontinuity Audio to RTP conversion point. timestamps are indicating missing packets received via freeswitch" The service provider is running a mixture of Cisco AS 5300's and 5350xm's, I don't think it's a bug in Freeswitch (I have a number of Freeswitch boxes that are all affected running different versions from 1.0.4pre7 to r17188) as DTMF was working once upon a time... can anyone think of anything obvious that either I or the service provider could be missing? Thanks.

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Load balancing question
by Lon Baker 15 Apr '10

15 Apr '10

Hi gang, I am tasked with load balancing calls to a cluster of Freeswitch servers, all running different "applications". Some will be handling conferencing others regular call routing. We are using OpenSIPs to proxy to the clusters, but I'm trying to figure out a way to "know" which server is handling a particular conference for example. So we route all participants into the correct server and conference room. Any ideas on doing this? Thanks for any ideas. Lon

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Music On Hold for SNOM phones with freeswitch
by Christian van der Leeden 15 Apr '10

15 Apr '10

Hi, I'd like to configure freeswitch so it acts as music on hold server (replying to a SIP Invite with music) for a SNOM phone. The SNOM phone is registered on my Patton SmartNode. I've tried to point it to 9999(a)192.168.140.7 (ip of my server) but freeswitch responds with just installed freeswitch.org on my linux box. I'd like to use it play music on hold to a SNOM VoIP phone (it is registered on my patton SmartNode, not with the freeswitch server). I receive a 407 Proxy authentication required. This is on a out of the box "quick install" from the git trunk. What would I have to do, so freeswitch would answer to the invite? Christian P.S.: attached the session log INVITE sip:9999@192.168.140.7;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.140.42:1024;branch=z9hG4bK-d0e7ezmhzuzs;rport From: "Zentrale" <sip:central@patton.logicunited.com>;tag=d3h27cphyo To: <sip:9999@192.168.140.7;user=phone> Call-ID: 3c28f4234780-u9oqvzvds56e CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:central@192.168.140.42:1024>;reg-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom370/7.3.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 452 v=0 o=root 30256492 30256492 IN IP4 192.168.140.42 s=call c=IN IP4 192.168.140.42 t=0 0 m=audio 54834 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:TpHIcA93pqErUArxCuWRC1tcd7mp8LTs73clcwnE a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Received from udp:192.168.140.7:5060 at 15/4/2010 15:39:52:647 (362 bytes): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.140.42:1024;branch=z9hG4bK-d0e7ezmhzuzs;rport=1024 From: "Zentrale" <sip:central@patton.logicunited.com>;tag=d3h27cphyo To: <sip:9999@192.168.140.7;user=phone> Call-ID: 3c28f4234780-u9oqvzvds56e CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d6ee682 2010-04-13 13:38:47 -0700 Content-Length: 0 Received from udp:192.168.140.7:5060 at 15/4/2010 15:39:52:802 (853 bytes): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.140.42:1024;branch=z9hG4bK-d0e7ezmhzuzs;rport=1024 From: "Zentrale" <sip:central@patton.logicunited.com>;tag=d3h27cphyo To: <sip:9999@192.168.140.7;user=phone>;tag=4jg7ajj84BB7K Call-ID: 3c28f4234780-u9oqvzvds56e CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d6ee682 2010-04-13 13:38:47 -0700 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 Proxy-Authenticate: Digest realm="patton.logicunited.com", nonce="82f5fa24-ae3a-4def-a67f-b097f599bc34", algorithm=MD5, qop="auth" Content-Length: 0

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REMINDER: FreeSWITCH Users Group / Installfest in San Francisco next week!
by Darren Schreiber 15 Apr '10

15 Apr '10

Hi everyone, I posted about this a while ago but just wanted to remind those who may live in the Bay Area of an upcoming FreeSWITCH Users Group meetup and installfest. The location I've picked is easy to get to by car, train, Muni, walking or even on a Segway. It's Borders books, located right near the CalTrain station in San Francisco. We'll be meeting in the cafe. If you need help finding your way, just let me know, but I suspect Google Maps has got that covered for you. If you're interested, PLEASE BRING A LAPTOP WITH A LINUX VIRTUAL MACHINE RUNNING IF YOU WANT TO PARTICIPATE IN THE INSTALL-FEST. For more details, see the full listing: http://www.meetup.com/fsusers/calendar/13009468/ Please note that this isn't a FreeSWITCH sponsored event (or Bandwidth or anyone else), it's just a casual organized meetup to meet others and learn about FreeSWITCH. There is no fee, sales pitch or evil corporate agenda backing this. When: Sunday, April 25, 2010 4:00 PM Where: Borders Book Stores (in the Cafe) 200 King Street San Francisco, CA 94107 415-357-9931 Feel free to email me if you have questions at d(a)d-man.org I look forward to seeing all the locals ;-) Thanks much, Darren Schreiber

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[INFO] User-Agent in mod_sofia, and git-builds.
by Peter Olsson 15 Apr '10

15 Apr '10

I just wanted to share an experience I had with a git-build and Avaya Communications Manager PBX. It seems it can't handle the new User-Agent values correctly, and this started to occur after doing git builds. In the end I found that the ":" in the timestamp was what broke the INVITE. After overriding the user-agent in the sofia configuration it started to work again. Probably it's a Avaya problem - but I just wanted to inform the other people on the list about this. This is how my User-Agent looked like: FreeSWITCH-mod_sofia/1.0.head-git-d6ee682 2010-04-13 13:38:47 -0700 Regards, Peter Olsson

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MEDIA_TIMEOUT after long duration call
by Francisco Scaramanga 15 Apr '10

15 Apr '10

hi , I have a problem with long duration calls. After 11 hours and 35 minutes the call ends because of MEDIA_TIMEOUT. It seems to be reproduceable. Any idea? I use SIP phone SIEMENS Gigaset S685. 2010-04-12 19:00:51.421875 [DEBUG] sofia.c:3289 Channel sofia/internal/1010(a)192.168.1.234 entering state [completed][200] 2010-04-12 19:00:51.421875 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/1010(a)192.168.1.234 [BREAK] 2010-04-12 19:00:51.421875 [NOTICE] mod_dptools.c:649 Channel [sofia/internal/1010(a)192.168.1.234] has been answered 2010-04-12 19:00:51.421875 [DEBUG] switch_channel.c:182 sofia/internal/1010(a)192.168.1.234 receive message [AUDIO_SYNC] EXECUTE sofia/internal/1010(a)192.168.1.234 MyModul 2010-04-12 19:00:51.640625 [DEBUG] sofia.c:3289 Channel sofia/internal/1010(a)192.168.1.234 entering state [ready][200] 2010-04-13 06:35:53.406250 [NOTICE] mod_sofia.c:744 Hangup sofia/internal/1010(a)192.168.1.234 [CS_EXECUTE] [MEDIA_TIMEOUT] 2010-04-13 06:35:53.406250 [DEBUG] switch_channel.c:1683 Send signal sofia/internal/1010(a)192.168.1.234 [KILL] 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1010(a)192.168.1.234 [BREAK] 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1010(a)192.168.1.234) State EXECUTE going to sleep 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1010(a)192.168.1.234) Running State Change CS_HANGUP 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/1010(a)192.168.1.234) State HANGUP 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:338 Channel sofia/internal/1010(a)192.168.1.234 hanging up, cause: MEDIA_TIMEOUT 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:396 Sending BYE to sofia/internal/1010(a)192.168.1.234 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1010(a)192.168.1.234 Standard HANGUP, cause: MEDIA_TIMEOUT 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/1010(a)192.168.1.234) State HANGUP going to sleep 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:476 (sofia/internal/1010(a)192.168.1.234) State Change CS_HANGUP -> CS_REPORTING 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1010(a)192.168.1.234 [BREAK] 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1010(a)192.168.1.234) Running State Change CS_REPORTING 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/1010(a)192.168.1.234) State REPORTING 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1010(a)192.168.1.234 Standard REPORTING, cause: MEDIA_TIMEOUT 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/1010(a)192.168.1.234) State REPORTING going to sleep 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:411 (sofia/internal/1010(a)192.168.1.234) State Change CS_REPORTING -> CS_DESTROY 2010-04-13 06:35:53.406250 [DEBUG] switch_core_session.c:1068 Session 3 (sofia/internal/1010(a)192.168.1.234) Locked, Waiting on external entities 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1086 Session 3 (sofia/internal/1010(a)192.168.1.234) Ended 2010-04-13 06:35:53.406250 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/1010(a)192.168.1.234 [CS_DESTROY] 2010-04-13 06:35:53.406250 [DEBUG] switch_core_codec.c:122 Restore original codec. 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/1010(a)192.168.1.234) State DESTROY 2010-04-13 06:35:53.406250 [DEBUG] mod_sofia.c:255 sofia/internal/1010(a)192.168.1.234 SOFIA DESTROY 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1010(a)192.168.1.234 Standard DESTROY 2010-04-13 06:35:53.406250 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/1010(a)192.168.1.234) State DESTROY going to sleep _________________________________________________________________ http://redirect.gimas.net/?n=M1004xSkyDrive2 Ihre Daten brauchen Platz? SkyDrive gibt Ihnen 25 GB - gratis!

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