- FreeSWITCH-users - FreeSWITCH Mailing Lists

will a Faster Spidermonkey help FreeSwitch ?
by jay binks 24 Aug '08

24 Aug '08

my question is... will these sort of optimizations help when used with FreeSwitch ? http://arstechnica.com/news.ars/post/20080822-firefox-to-get-massive-javasc… http://blog.mozilla.com/schrep/2008/08/22/what-can-you-do-when-your-browser… would be interesting to see how this compares to the likes of Lua etc.. -- Sincerely Jay Binks

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Registration issue with Polycom
by Craig Guy 24 Aug '08

24 Aug '08

Hi, I've got an issue trying to register a Polycom IP320 against FS 1.0.1, as below the Polycom is coming from and registering with a contact port 5076, however FS is sending the 401 to port 5060. Any ideas? I have a Snom registering successfully as 1000 from behind a different NAT. Any ideas? # U 203.23.223.109:5076 -> 4.53.88.29:5060 REGISTER sip:4.53.88.29 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK7db8749a8ECDE4D3. From: "Craig Guy" <sip:1001@4.53.88.29>;tag=914B0517-5EB6FA00. To: <sip:1001@4.53.88.29>. CSeq: 1 REGISTER. Call-ID: 7ef2a3fc-a794db15-262aa31e(a)192.168.1.1. Contact: <sip:1001@203.23.223.109:5076>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER". User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049. Max-Forwards: 70. Expires: 3600. Content-Length: 0. . # U 4.53.88.29:5060 -> 203.23.223.109:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bK7db8749a8ECDE4D3;received=203.23.223.109. From: "Craig Guy" <sip:1001@4.53.88.29>;tag=914B0517-5EB6FA00. To: <sip:1001@4.53.88.29>;tag=7U95e600jSH5e. Call-ID: 7ef2a3fc-a794db15-262aa31e(a)192.168.1.1. CSeq: 1 REGISTER. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: 100rel, timer, precondition, path, replaces. WWW-Authenticate: Digest realm="4.53.88.29", nonce="b59a58b1-bdc6-4487-b346-12838168039d", algorithm=MD5, qop="auth". Content-Length: 0. .

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bypass media
by Juan Backson 23 Aug '08

23 Aug '08

Hi, Is there anyway that I can tell if one of either leg A or leg B is a client behind NAT? I would like to find a way to do that in order to dynamically determine whether bypass_media or proxy_media should be used. Thanks, JB

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FreeSwitch on openSUSE 11.0
by Robert Clayton 23 Aug '08

23 Aug '08

All, Is there any documentation on the steps for installing FreeSwitch on openSUSE 11.0. I tried to install from the Buils Service on a new OS installation but it fails. Bob

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T1 RBS Support Revisited
by John Wehle 22 Aug '08

22 Aug '08

> Is there support in FreeSWITCH for MF? To answer my own question ... Perhaps I can add a ZAP_COMMAND_SEND_MF case to zap_channel_command patterned after ZAP_COMMAND_SEND_DTMF and just use teletone_set_tone to set the proper frequencies??? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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SBC Abilities
by eric c 22 Aug '08

22 Aug '08

Hello, Looking at the website shows that FS can act as a SBC but i haven't found any page on the wiki that goes over the ability that an SBC can handle when it comes to DoS like behavior such as to rate limit per IP or automatic blacklist based on too many registers/request storms. Is this possible with FS at the moment? thanks, eric clope

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Sending Interrupt signal during MOH
by Juan Backson 22 Aug '08

22 Aug '08

Hi, Does anyone know how to how to insert an interrupt message ( ex. Please wait and our agent will be with you momentarily) every 15 seconds when the MOH is being played? Any suggestion or idea how to work around this requirement will be helpful. Thanks, JB

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multiple sip profile , centrex like function
by Lee JJ 22 Aug '08

22 Aug '08

Hello Brian : I finally work something out ! # the dialplan <extension name="centrex testing"> <condition field="destination_number" expression="^(1[0-9][0-9][0-9])$"> <action application="set" data="dialed_ext=$1"/> <action application="export" data="dialed_ext=$1"/> </condition> <condition field="${sip_from_host}" expression="^${sip_to_host}$"> <action application="bridge" data="sofia/${sofia_profile_name}/${dialed_ext}" /> <anti-action application="bridge" data="sofia/${sip_to_host}/${dialed_ext}" /> </condition> </extension> # the same EXT 1528@inter2 bridged with 1528@internal 2008-08-22 15:58:06 [NOTICE] sofia.c:2491 sofia_handle_sip_i_state() Channel [sofia/inter2/1528] has been answered 2008-08-22 15:58:06 [DEBUG] mod_sofia.c:430 sofia_answer_channel() Local SDP sofia/internal/1528(a)210.243.126.69: 2008-08-22 15:58:06 [NOTICE] switch_ivr_originate.c:1298 switch_ivr_originate() Channel [sofia/internal/1528(a)210.243.126.69] has been answered But there is one issue 2008-08-22 15:06:11 [WARNING] sofia.c:3130 sofia_handle_sip_i_invite() IP 219.81.16.134 Rejected by acl domains I have to disable the acl in profiles  otherwise it won't allow me to call "sip:1528@internal" from "sip:1528@inter2" ( PLS explain the security issue ) # the following diaplan tested , really nothing to do with ENUM very good DNS working <extension name="apnic Testing" continue="true"> <condition field="destination_number" expression="^(info|billing)"> <action application="bridge" data="sofia/default/$1@${sip_to_host}" /> </condition> </extension> 1 0.000000 210.243.126.69 -> 203.119.0.97 SIP/SDP Request: INVITE sip:billing@voip.apnic.net, with session description 2 0.246246 203.119.0.97 -> 210.243.126.69 SIP Status: 100 Trying 3 0.659271 203.119.0.97 -> 210.243.126.69 SIP Status: 180 Ringing 4 2.589506 210.243.126.69 -> 203.119.0.97 SIP Request: CANCEL sip:billing@voip.apnic.net Also been through these points. http://lists.freeswitch.org/pipermail/freeswitch-users/2008-August/005531.h…

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Custom MOH
by mark morreny 22 Aug '08

22 Aug '08

Dear Freeswitch friends, I am working on making my own custom MOH specific to our company's need. I tried recording the content into a normal .wav file and then played it by setting ringback=<CUSTOMER MOH WAV PATH>. What I get is only a long series of beep sound instead of the recorded content. What did I do wrong? Is there any special format that I should use when creating a custom MOH? If I want to convert a mp3 file into MOH, how would I do that? Thanks, Mark

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How to originate a session from outside
by Adeel Ansari 22 Aug '08

22 Aug '08

Hi All, I had success in bridging a session, using my Java Program as a hook, and manage to connect one mobile phone to another. It goes like below. === --> originate sofia/gateway/gizmo2/01160176xxxxxx &java(net.digitalapps.freeswitch.CallTest) === In this case, freeswitch call my Java program. Now, My intension is to not use console at all, instead use my java program to originate a session in freeswitch and then bridge the session. So, call freeswitch from my program, not the other way around. Is it possible? Need suggestions in any case. I mean if I need to come up with my own client. Actually, I was looking JAIN-SIP to achieve this. But then thought of asking your suggestions. With JAIN I might go like this, place a call, hit the dial plan invoke my Java Hook Program and bridge the call. But I am afraid that it might connect my client with the mobile phone, not one phone to another. Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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