February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS on Public IP adrress
by lloyd thomas 18 Feb '11

18 Feb '11

I have put my FS box on a public IP address so that it can serve phones at remote sites. I have not set up any providers on it yet. I would like to make an internal call between two extensions, but I get the following error when I try to call freeswitch@internal> [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Content-Type] = [log/data] [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Content-Length] = [121] [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Level] = [7] [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Text-Channel] = [0] [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-File] = [sofia.c] [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Func] = [sofia_handle_sip_i_invite] [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [Log-Line] = [6081] [DEBUG] libs/esl/src/esl.c:985 esl_recv_event() RECV HEADER [User-Data] = [] [DEBUG] libs/esl/src/esl.c:1111 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: log/data Content-Length: 121 Log-Level: 7 Text-Channel: 0 Log-File: sofia.c Log-Func: sofia_handle_sip_i_invite Log-Line: 6081 User-Data: _undef_ 2011-02-18 17:41:40.638553 [DEBUG] sofia.c:6081 IP 188.221.84.33 Rejected by acl "domains". Falling back to Digest auth. 2011-02-18 17:41:40.638553 [DEBUG] sofia.c:6081 IP 188.221.84.33 Rejected by acl "domains". Falling back to Digest auth. I am using two polycom 550's to test this. Any Ideas? Thanks Lloydie T

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big audio delay in bridged conference
by George Niculae 18 Feb '11

18 Feb '11

Hi All, I have two conferences set in my dialplan - one is bridged and is using mod_loopback (as advised in http://jira.freeswitch.org/browse/FS-2741 ) the other is a normal one. Extensions in dialplan looks like: <extension name="3202"> <condition field="destination_number" expression="^202-conference$"> <action application="bridge" data="loopback/app=conference:202-conference@3202"/> </condition> </extension> <extension name="3203"> <condition field="destination_number" expression="^203-conference$"> <action application="conference" data="203-conference@3203"/> </condition> </extension> With the bridged conference I hit weird behaviour: there is a delay in audio that grows in time and can even touch 9 seconds in about 10 minutes. The other conference works just fine tested in same conditions as the first one. I have monitored FS console during both conferences but noticed nothing special. (running HEAD, commit 49a5effcdf2cea9e0ddcf146cf3fe85d1872e654) Any pointer greatly appreciated, Thanks, George

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CallManager trunk to FS
by Mitch Johnson 18 Feb '11

18 Feb '11

I'm having a few issues where the freeswitch 1.6 is not registering with the callmanager 8. [ERR] sofia_reg c:1765 CM8 Registration Failed with status DNS Error [503]. <include> <gateway name="CM8"> <param name="username" value="xxxx"/> <param name="password" value="xxxx"/> <param name="realm" value="CM8.localhost.local"/> </gateway> </include> I added the CM8.localhost.local to the hosts file as the CM is on the same network and doesn't need a DNS entry. Can I replace the value in the realm field with an IP Address? I did check the box MTP required on the callmanager. I've reloaded the xml file using the sofia profile external ... Thanks so much, Mitch

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Which CPU to select
by Vitalii Colosov 18 Feb '11

18 Feb '11

Hello list, Could you please suggest, based on your experience, which of the CPU will be enough to handle 1000 simultaneous calls using G711 codec on all end points? Is it really needed to buy Xeon with 4 cores? or Core 2 Duo or Quad will be perfectly enough? Assuming CPU frequency is ~2.5 Ghz The following page seems like not too much updated: http://wiki.freeswitch.org/wiki/Real-world_results Thank you, Vitalie

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ASR from Freeswitch to Loquendo - No Audio passed to MRCP Server
by Michel Habib 18 Feb '11

18 Feb '11

Dears, I am seeking expert advice on setting up Freeswitch to Loquendo MRCP Connection. I have freeswitch on one computer, configured for MRCP V1 - Loquendo Speech Suite 7. I have Loquendo set-up on the other computer - with MRCP V1 Profile. Using a simple lua script driving my extension, I can successfully:- - establish the connection to Loquendo Server. - define grammar - compile grammar successfully. - Begin Recognition Then it waits forever until i hangup the call. Looking at the Wireshark Traces, the RTP packets are transferred succesfully from softphone to freeswitch [i could hear my voice on the playback] But, when playing back the RTP packets between freeswitch and Loquendo, there is only deep silence [same thing with audio dump created by Loquendo]. Can you direct me where to troubleshoot this on the Freeswitch side? I will be more than happy to provide any logs you need to resolve this. Thank you, Michel.

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Re: [Freeswitch-users] dtmf via info
by Meftah Tayeb 18 Feb '11

18 Feb '11

ok, 1. why you need sip info DTMF? is not standard 2. you could try out <action application="set" data="dtmf-type=info"/>. if don't work, set it in the profile. 3. i strongly recomand you to use rfc2833 or inband dtmf. thank you Le 18/02/2011 09:57, Nikolay Kondratyev a écrit : > > Hi all, > > I’d like to send dtmf via info and I’d like to set it in the dial plan. > > Do I just need to set dtmf_type channel variable in the dial plan? > > I tried to set<action application="set" data="dtmf_type=info"/> in the dialplan, but I do not see Info messages…. > Should it work in this way at all? > Did I miss something? > Thanks in advance, > Nikolay. > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is > believed to be clean. > > > _______________________________________________ > 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 > -- Meftah Tayeb inum: +883510001288000 phone: +13477595883

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which language to implement dial plans in?
by Edward de Jong 18 Feb '11

18 Feb '11

I am new to freeswitch, and wondering which of the many languages to implement my rather complex dial plan in. I know i can't express my desires in the normal XML language, because I need looping and database access, so which language is the best tested and most reliable in terms of usage with FreeSwitch. My finalists so far are Lua, Python, and good old C. Haven't used python or lua yet, but I figure C will have a big performance advantage. but maybe not important considering how much greater the CPU load in the switch is devoted to handling audio streams... edj

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dtmf via info
by Nikolay Kondratyev 18 Feb '11

18 Feb '11

Hi all, I’d like to send dtmf via info and I’d like to set it in the dial plan. Do I just need to set dtmf_type channel variable in the dial plan? I tried to set <action application="set" data="dtmf_type=info"/> in the dialplan, but I do not see Info messages…. Should it work in this way at all? Did I miss something? Thanks in advance, Nikolay. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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Call Recording
by Jerry Richards 18 Feb '11

18 Feb '11

Does the *2 DTMF method still work for recording calls? I recall this working before without needing to make changes to the dialplan XML, but right now it does not appear to work on the Sep 20 Freeswitch trunk. Thanks, Jerry

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IVR application
by Jason b.a 18 Feb '11

18 Feb '11

Here is a call flow sample for playing a prompt, later on i need to make my IVR application virtual using cloud computing , in this example freeswitch can play the role of media server, for media processing , and the application server for signal processing and controlling the media server sure, is it right ! or should i use different media server such as SEMS ! sip Caller Application Server Media server -----------invite----------------> ----------------invite---------------------> <---------Trying----------------- <--------------200ok---------------------- <-------------200ok------------- --------------Ack---------------> -----------------Ack-----------------------> Caller in interactiong with media server ---------Info(play prompt X.wav)---------> <----------------Ok---------------------------

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