April 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Regarding SIP accounts creation
by Jagadish Thoutam 04 Apr '12

04 Apr '12

Hi all, this is silly but i am new to Freeswitch i need to know the configuration of sip account creation in sip_profile for local users,

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TTS/Cepstral choppy audio on Windows...
by Gerry Hull 04 Apr '12

04 Apr '12

I'm building a automation control application which has an audio stream. I want an IVR to allow the callee to manipulate the application while the stream is playing. (the audio is interfaced via soundcard/portaudio.) I'm using the latest Build of FSClient, and interfacing to it with EventSocket. I'm using EventSocket to gather digits and play TTS to the caller. The OS is Windows 7/64, and running all these apps in 32-bit. 6GB memory, dual-core processor. While I'm in a call, and playing TTS, CPU running at about 25% and memory about 60% (lots of stuff running on the box). It does not matter if TTS is flite or cepstral, I get a bad chop at about a 1Hz or more rate. I've tried flite and both 16 and 8-kHz cepstral voices, same issue. Am I doing something wrong, or is this a bug? TIA, Gerry

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double nat trouble
by Brett Wilson 04 Apr '12

04 Apr '12

Hey guys, We have just temporarily moved our office, so our phones are no longer just on the office LAN. They are on a remote lan, both sides behind NAT and I'm trying to get them working right but I am having trouble. I have things working to the point where I can call to the POTS or to an extension on the local network back home just fine. The problem is that the remote phones do not receive sip messages behind the nat. It seems to me that freeswitch is sending the messages to the sip port the phones are configured with, and not the NAT port that can be seen when viewing the registrations status. The nat ports are very high numbers. I suspect if FS was to send messages to those ports, it would get through the foreign nat and reach the phone. On the FS side, I am using pfsense as the firewall, and have port forwards set up to the fs box. I am using a dynamic ip with dyn-dns-like service and am using that domain name in the fs internal/external profiles for the ext-sip-ip and ext-rtp-ip variables. I have upnp turned off but it does not make any difference anyway. Strangely, I was actually able to place a call from one remote phone to another once earlier, but it must have been a fluke because I haven't had it working after that. What am I missing here? ******************************************* Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: Brett.Wilson(a)launch3.net AOL IM: Brett.Wilson(a)launch3.net <http://www.launch3.net/> www.Launch3.net <http://www.launch3telecom.com/> www.Launch3telecom.com ******************************************* <http://launch3telecom.blogspot.com/> Description: Description: Blogger-logo <http://www.facebook.com/pages/Launch-3-Telecom/165610820178720> Description: Description: FaceBook-Logo <https://twitter.com/launch3telecom> Description: Description: Twitter-Logo <https://plus.google.com/102713212642017290959> Description: Description: GPlus-Logo

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X-Lite and Video
by peter＠uringme.com 04 Apr '12

04 Apr '12

(Apologies to the list owner for originally sending this to the wrong address) I have two laptops running X-Lite 4. I have them registered to a FreeSwitch server (latest git) as extensions 7777 and 7778. I have a dialplan for each (quick and dirty) that just bridges them when one is dialed from the other: <extension name="7777"> <condition field="destination_number" expression="^7777$"> <action application="bridge" data="sofia/external/7777%test"/> </condition> </extension> (and vice-versa for 7778). I can dial between them just fine for audio calls -- bidirectional audio, etc, no problem. I'm trying to get video going. Both X-Lites have H.263 and H.263-1998 enabled in their settings. Freeswitch has the following in vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM,H263,H264,H263-1998"/> When I try to make a video call from one extension to the other, the calling extension seems to think it's in video, but the called extension doesn't. INVITE from freeswitch console: ------------------------------------------------------------------------ INVITE sip:7778@test SIP/2.0 Via: SIP/2.0/UDP 192.168.1.7:50350;branch=z9hG4bK-d8754z-9ea3618224c5cf23-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:7777@68.202.69.172:50350> To: <sip:7778@test> From: "Peter Test"<sip:7777@test>;tag=a7031d83 Call-ID: ZDUzZGE1YjUyOTQ2ZGNmZTY0Yjc5ODA5NTE4NDAzMGQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 681 v=0 o=- 12970176286658638 1 IN IP4 192.168.1.7 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.1.7 t=0 0 a=ice-ufrag:20fef6 a=ice-pwd:0a03863684bc5f16a9c862dcdccdd8eb m=audio 58632 RTP/AVP 107 0 8 101 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.1.7 58632 typ host a=candidate:1 2 UDP 659134 192.168.1.7 58633 typ host m=video 50994 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.1.7 50994 typ host a=candidate:1 2 UDP 659134 192.168.1.7 50995 typ host I do see freeswitch seeing the audio and video codecs: 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4683 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:2800 Set Codec sofia/external/7777@test PCMU/8000 20 ms 160 samples 64000 bits 2012-01-04 13:44:37.188657 [DEBUG] switch_core_state_machine.c:343 (sofia/external/7777@test) State NEW 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4797 Set 2833 dtmf send/recv payload to 101 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4856 Video Codec Compare [H263:34]/[H263:34] However, when freeswitch starts the bridge and calls the far-end party, it doesn't send along video information in the INVITE: INVITE sip:7778@68.202.69.172:32834;transport=udp;rinstance=677b87c43ee7970a SIP/2.0 Via: SIP/2.0/UDP 204.13.175.89:5080;rport;branch=z9hG4bK8jm56tcmZ6p6j2012-01-04 Max-Forwards: 69 From: "Peter Test" <sip:7777@204.13.175.89>;tag=ZeZUav5XXat5e To: <sip:7778@68.202.69.172:32834;transport=udp;rinstance=677b87c43ee7970a> Call-ID: fe64b2f5-b1a6-122f-a187-00144f49eecc CSeq: 22515274 INVITE Contact: <sip:mod_sofia@204.13.175.89:5080> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 207 X-FS-Support: update_display Remote-Party-ID: "Peter Test" <sip:7777@204.13.175.89>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1325690959 1325690960 IN IP4 204.13.175.89 s=FreeSWITCH c=IN IP4 204.13.175.89 t=0 0 m=audio 11718 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 So, am I missing something? Do I need to use something other than "bridge" in the dialplan, or do I need to add some variables to be able to pass on the video? All I'm trying to do is make a video call between two X-Lites that are locally SIP registered to freeswitch. Because I want to record the video at some point in the future, I don't want to divert the media -- I want it streaming/passing through freeswitch. When the call is connected, the caller shows a "Waiting for video", but the called doesn't show this. When I try to start the video, it says "Failed to Start Video".

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adding leg C for external recorder
by Bote Man 04 Apr '12

04 Apr '12

What are the ways that I can set up an outbound call to send an additional leg to an external logging recorder? Right now I'm trying to use mod_conference, but it just ends the call as soon as it begins, even though I have another dialplan extension that sets up several endpoints in a conference bridge just fine. I thought about using eavesdrop, but that is initiated by and sends audio to leg A. Plus it joins an existing call whereas in this application I'm establishing a new call, so that won't work. I need to send the audio to leg C from the start. I'm hoping I don't have to write a script, because I'm on the customer's site and I'm running out of time. Any ideas? Thanks. Bote

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Session-wise Language Setting
by Sharad Garg 04 Apr '12

04 Apr '12

Hi All Just wondering whether we can define the language from beginning of the call means when a call is originated or landed to Freeswitch, can we define the language of all the prompts whether the call should be processed in English or Russian or so on.? Regards Sharad ----- Original Message ----- From: <freeswitch-users-request(a)lists.freeswitch.org> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Thursday, March 29, 2012 7:33 AM Subject: FreeSWITCH-users Digest, Vol 69, Issue 282 > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > -------------------------------------------------------------------------------- > Today's Topics: > > 1. Re: Open Bugs on Jira and Call for help (Ken Rice) > 2. mod_callcenter and moh-sound (Vik Killa) > 3. Re: mod_callcenter and moh-sound (Vik Killa) > 4. Registration VIA TCP (Rob Moore) > 5. Forward calls from opensips to freeswitch (Sherif Omran) > 6. Re: Registration VIA TCP (Mitch Capper) > -------------------------------------------------------------------------------- > _______________________________________________ > 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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Re: [Freeswitch-users] FreeSWITCH-Billing
by David Villasmil 04 Apr '12

04 Apr '12

Also, remember to set the correct info in the "variables.conf" file. David On Wed, Apr 4, 2012 at 12:26 PM, David Villasmil < david.villasmil.work(a)gmail.com> wrote: > Hello Dihn, > > There is a script called "install.pl", you need to execute that. > > David > > > On Wed, Apr 4, 2012 at 7:02 AM, dinh thuc <dinhthuc1303(a)gmail.com> wrote: > >> Hi David !! >> >> We need to solution about billing in my system. I have read link >> https://github.com/davidcsi/FreeSWITCH-Billing >> >> You have script for install all ? >> >> I want to install and test >> >> B&R >> >> Dinh Thuc >> > >

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Configurations external and intenal calls in FreeSwitch
by sardor 04 Apr '12

04 Apr '12

Hello everyone, Could you help me to confgure out my *.xml files for receiving calls to SIP and making external calls to regular phones. I've done a lot of changes in this side, but no success. Please if you can advise smth or send copy of xml files, then it would be great help for me. Thanks in advance... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Configurations-external-and-i… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Fisheye and Jira Upgraded!
by Ken Rice 04 Apr '12

04 Apr '12

Just a quick announcement. As you may have noticed end of the Day US Time on Tuesday April 3rd, Fisheye and Jira were offline for a while. However, I am pleased to report that both have now been upgraded, and a Hardware replacement has happened on both hosts. Please report any issues to me or BKW via the mailing list or join us on the IRC Channel at #FreeSWITCH on irc.freenode.net Thanks for your Support! K

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FS shutdown not closing TCP port causes TIME_WAIT problems
by Mario G 03 Apr '12

03 Apr '12

If FS is shutdown and started too quickly, the internal profile ip/port stays in use and FS will fail to start it. After searching on the wen I found many references to TIME_WAIT but all fixes were to the program source. So I am wondering if when FS/sofia shuts down they are not releasing the ip/port for internal. If it's true, I found some source changes to solve it. Google "TIME-WAIT" and you'll find lots of good stuff like http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-t…. Apparently there is no way to free up the port manually. Help/comments anyone? Mario G

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