March 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Limit the size of queue in mod_fifo
by Marc de Corny 11 Mar '11

11 Mar '11

Hi All, does anyone know of a way of limiting the numbers of calls that are held in the mod_fifo queue so that we can deny more calls after having reacded that limit ? thanks Marc

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Conference silent issue for more than 2 callers
by kapil.rastogi 10 Mar '11

10 Mar '11

Hi, Login with 5 subscribers. 1st subscriber logs in, 2nd Subscriber log in and listen each other, 3rd subscribe and it is not listened and can’t listen others. 4th as 3rd and 5th as 3rd and 4th. Currently i m using freeswitch with diastar, that create a bridge between dialogic and freeswitch. Please help me to resolve the issue. Thanks ----- Regards, Kapil Rastogi Telemune Software Solutions P Ltd. kapil.rastogi(a)telemune.net +919013204760 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Conference-silent-issue-for-m… Sent from the freeswitch-users mailing list archive at Nabble.com.

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count
by Sam 10 Mar '11

10 Mar '11

Hello, Whats the command to get the participants count in the conference ? Rwgards Sam

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Not able to install gsmopen module on FC 13
by ovvenkat 10 Mar '11

10 Mar '11

Hi to all, I have followed this wiki page to intall gsmopen module http://wiki.freeswitch.org/wiki/GSMopen#Linux Installed following prerequisites packages # yum -y install alsa-lib-devel alsa-utils then, go to gsmopen directory #/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen/gsmlib/gsmlib-1.10-patched-12ubuntu1 I wanted to compile only SMS support, no voice call. When, I compiled the source code , I am getting following error , #make ...... ...... gsm_unix_serial.cc:89: error: 'strerror' was not declared in this scope make[1]: *** [gsm_unix_serial.lo] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch/ src/mod/endpoints/mod_gsmopen/gsmlib/gsmlib-1.10-patched-12ubuntu1/gsmlib' make: *** [install-recursive] Error 1 When, I enter into following directory and trying the compile the source code, #/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen # make I am getting following error gsmopen_protocol.cpp:5:28: error: gsmlib/gsm_sms.h: No such file or directory gsmopen_protocol.cpp:9:36: error: gsmlib/gsm_unix_serial.h: No such file or directory gsmopen_protocol.cpp:11:30: error: gsmlib/gsm_me_ta.h: No such file or directory gsmopen_protocol.cpp:16: error: ‘gsmlib’ is not a namespace-name ... ... gsmopen_protocol.cpp:3223: error: ‘DCS_DEFAULT_ALPHABET’ was not declared in this scope gsmopen_protocol.cpp:3225: error: ‘DCS_SIXTEEN_BIT_ALPHABET’ was not declared in this scope gsmopen_protocol.cpp:3226: error: ‘bufToHex’ was not declared in this scope make[1]: *** [gsmopen_protocol.o] Error 1 make: *** [all] Error 1 I do not know, what I am doing wrong. Anyone please help me regarding this. -- Regards Venkatesan OV.

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sip auth challenge HACKING ???
by Erkan Ünlü 10 Mar '11

10 Mar '11

Hi FS Users, in last time i see in my console of FS this kind of error messages. [WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile 'internal' for [5828@?????????] from ip ???7?.1??.7??.??? i check my config files ever again and again, but today the console is only given this kind of messages. Maybe 20 messages per second. i see the ip that given in the console "from ip xx.xx.xx.xx" i block this ip in my firewall and everything is fine. Now i understand that this a trying to hacking my server. The blocking of the ip is a solution but can not handle this in Freeswitch, because i see this problem sometimes on different FS servers also and in normally the FS server maybe must can handle this problem. For example with automatic black lists if an ip trys more than 20 times with wrong login. So the ip will be banned for 1 hour or so. i'm interesting in if other users have the same problems and ideas in how we can handle this. Kind regards Erkan

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SIP Trunk Registration
by Mark Holloway 10 Mar '11

10 Mar '11

Does FreeSwitch support single phone number registration for a Trunk Group? For example, in Broadsoft Broadworks I can build a Trunk Group with 100 numbers but I identify one number as the Pilot Number. I have a SIP endpoint (such as a Cisco ISR or Adtran TA900) that registers with Broadworks using the main Pilot Number only (Business Contact) so I don't have to register all 100 numbers. It's like a parent-child relationship where the other 99 numbers are a child of the main parent (pilot) number. Thanks, Mark

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mod callcenter
by Sam 10 Mar '11

10 Mar '11

Hello, Was testing callcenter module and found out that at times it gives error " invalid application callcenter " and after reloading the module it works fine. Some time also happens that if I reload the module it do not reloads the parameters of the callcenter like the agents & tires. It just unload & reloads even if there are changes to the specifications of agents. Regds Sam

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Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem
by Michel Habib 10 Mar '11

10 Mar '11

Yes, I get the Audio from FS in regular calls - I already disabled all possible firewalls - all 3 machines [softphone, freeswitch, Speech Server (and mrcp connector) ] are on a switch. 192.168.5.107 is the freeswitch server 192.168.5.110 is the MRCP-Connector/Speech Server 2007 server I made too many iterations on the configuration below: <include> <profile name="mrcp-connector" version="2"> <param name="client-ip" value="192.168.5.107"/> <param name="client-port" value="5090"/> <param name="server-ip" value="192.168.5.110"/> <param name="server-port" value="5070"/>  <param name="sip-transport" value="udp"/>   <param name="rtp-ext-ip" value="192.168.5.107"/> <param name="rtp-ip" value="192.168.5.107"/> <param name="rtp-port-min" value="4000"/> <param name="rtp-port-max" value="5000"/>  <param name="rtcp" value="1"/>  <param name="rtcp-bye" value="2"/>  <param name="rtcp-tx-interval" value="5000"/>  <param name="rtcp-rx-resolution" value="1000"/>    <param name="codecs" value="PCMU PCMA L16/96/8000"/>  <synthparams> </synthparams>  <recogparams>  </recogparams> </profile> </include> ---------- Forwarded message ---------- > From: Christopher Rienzo <cmrienzo(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 7 Mar 2011 09:32:51 -0500 > Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server > [using MRCP Connector] - Audio Problem > Do you get audio between FS and your SIP client when not using ASR/TTS? > > Show me the MRCP profile configuration and your FreeSWITCH logs during the > call. > > > > On Mon, Mar 7, 2011 at 4:55 AM, Michel Habib <michelhabib(a)gmail.com>wrote: > >> Hello All, >> I have MS OCS Speech Server 2007 [working correctly, as i can make SIP >> calls and use its ASR and TTS Services successfully] >> I am also using MRCP Connector from AumTech - which allows me to use ASR >> and TTS Services through an MRCP Client . >> Now, i am using Freeswitch mod unimrcp to use ASR and TTS. >> >> for TTS, I can successfully make the call, the Audio RTP of the TTS voice >> is transferred succesfully from Speech Server [through MRCP Connector] back >> to the Freeswitch Server. >> However, Freeswitch is not sending back the Audio RTP to the SIP client. >> >> for ASR, I can successfully define the grammar and start recognition, but >> the audio RTP sent to speech server [through MRCP Connector] is silent >> [empty]. >> >> I am suspecting something is wrong with the RTP Configuration - can you >> help me? >> >> Let me now if you need any specific logs/scripts/configuration? >> >> Thank you, >> Michel. >> >> >>

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incompatible destination
by Sam 09 Mar '11

09 Mar '11

Hello, The extension 7006 was working few days ago and now it not working,rest all the phone are working properly. U 192.168.2.190:5060 -> 192.168.2.14:5060 INVITE sip:7006@192.168.2.14:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..Max-Forwards: 70..From: "" <sip:9322273640@192.168.2.190>;tag=4QggZ1XcNZD3F..To: < sip:7006@192.168.2.14:5060>..Call-ID: 7bf ae496-c324-122e-bc85-0026b976aef7..CSeq: 9403059 INVITE..Contact: < sip:mod_sofia@192.168.2.190:5060>..User-Agent: NOVANET..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-Dis position: session..Content-Length: 203..X-FS-Support: update_display..Remote-Party-ID: <sip:9322273640@192.168.2.190>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1299446773 1299446774 IN IP4 192.168.2.190..s=FreeSWITCH ..c=IN IP4 192.168.2.190..t=0 0..m=audio 31474 RTP/AVP 8 101 13..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ptime:20.. U 192.168.2.14:5060 -> 192.168.2.190:5060 SIP/2.0 488 Not Acceptable Here..Via: SIP/2.0/UDP 192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..From: < sip:9322273640@192.168.2.190>;tag=4QggZ1XcNZD3F..To: "7006" < sip:7006@192.168.2.14:5060>;tag=983B5A59-EB953B22..CSeq: 9403059 INVITE..Call-ID: 7bfae496-c324-122e-bc85-0026b976aef7..Contact: < sip:7006@192.168.2.14:5060>..User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.2.2.0477..Accept-Language: en..Warning: 488 SDP "NotAcceptableHere"..Content-Length: 0.... I tried all the possibilities with codec priorities on U/A . http://pastebin.freeswitch.org/15577 Any help. Regards Sam

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FreeSWITCH Conference Call Followup - Recordings Online
by Michael Collins 09 Mar '11

09 Mar '11

FYI, We had a nice conference call with Jim Gettys and Dave Täht from the Bufferbloat project. I've made the recordings available in their usual spot: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls Use the links next to "FS_weekly_2011_03_09" to download the audio format of your choice. Thanks to Jim, Dave, and all those who participated today! -MC

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