September 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH cluster with IM/Presence
by Even André Fiskvik 22 Dec '11

22 Dec '11

Hi. I currently have a project where I'm researching how to establish a clustered and scalable platform with: - registration - presence/blf between eyebeam/bria and Linksys SPA962 clients - IM between eyebeam/bria clients. - calling between the clients The first scenario I want to test is with two FreeSWITCH servers and shared ODBC db running in a "cluster". How the registrations and calls etc. are balanced between them is not decided yet, but I'm thinking that a UltraMonkey setup or DNS SRV records could to the trick here? The main challenge in this setup is sharing presence, and doing call-setup and IM between two FreeSWITCH servers (which can be scaled by adding n-nodes). I'm interested in user's that can confirm wether this solution is possible or not with FreeSWITCH and it's current state, and what it would take to make something like this work. If there's any part of FS that is lacking some implementation to make such a scenario work, we are interested in providing the funding to support those features. I've seen that the core db has added support for ODBC, but I'm not sure wether this is something that would be helpful in this scenario or not. This is only conceptual right now, so I'm interested in paying someone that could help me configure the two FS servers to support this scenario on an hourly basis to get a quick proof-of-concept up and running. The proof-of-concept has to be completed by March 19th, after which we will evaluate what technologies and setup we want to commit to. Some other variations I'm concidering is Opensips/Kamailio -> Asterisk, Opensips/Kamailio -> FreeSWITCH, Opensips/Kamailio -> Asterisk and Openfire + Asterisk-im plugin for IM/presence. I'm also open to other suggestions, but I'd like this topic to stay on how FreeSWITCH could facilitate such a platform. Best regards, Even André Fiskvik CTO - Oyatel AS

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Grandstream GXP 2000
by Jonas Gauffin 18 Dec '11

18 Dec '11

Hello I have problems with TCP and GXP2000 phones. They work fine using UDP, but no calls arrive if I switch to TCP. I've upgraded to the latest GS firmware today to see it that helped, but it didn't. Have anyone else had problems with TCP? Regards, Jonas

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Outbound calling with recorded message.
by Dave 08 Dec '11

08 Dec '11

Hi all, I have been using freeSWITCH for a while for inbound calls in which a person registers for a seminar via IVR. I simply use the DID, CID number and name to identify the person and put them in the Database. That part of the application is working wonderfully. The issue I'm presented with now is that we need automate making calls to a few of these registrants, after each event, whos caller_id_name comes in as "Unknown" or "Wireless Caller", Play a recording to let them know we need the name, and that they can record the information right then, or call the office. I am limited to using a Windows 7 machine. What would be the best tool to use to automate these calls with the use of an IVR? Dave Goodwin

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Need help
by Sam 08 Dec '11

08 Dec '11

Hi, I need some help with an idea I am trying to implement with Freeswitch. I am trying to work with a partner company which will forward a sip call to my freeswitch server and my freeswitch server is suppose to playback annoucements to the caller and then once they are finished listening they should be forwarded back to the partner company so that the call can continue without my freeswitch being in the call path. So does any one have any suggestions/ideas on the best way to implement this?

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get cepstral volume lower
by Madovsky 05 Dec '11

05 Dec '11

I'm trying the SSML example from mod_cepstral wiki <action application="speak" data="cepstral|david|<prosody volume=13>Thank you for holding</prosody>"/> but the voice doesn't say the sentence but says something like "slash blablabla...". Any idea why I can't embed SSML tag in the cepstral sentence ? Thanks

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google voice connection going offline (mod_dingaling)
by Federico Beffa 02 Dec '11

02 Dec '11

Hi All, I've setup mod_dingaling to connect to my google voice account according to the instructions on the wiki http://wiki.freeswitch.org/wiki/Google_Voice#Setup_FreeSWITCH_-_Dingaling_t… with two modifications: <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/> I use the same IP settings for SIP (sofia) with no problem. The setup works fine for the first hour or two, but then I can see from another google account that the connection established by freeswitch goes offline. If I try anyway to make an outbound call, then the other end does not ring (even if I wait for a long time). However, after some minutes the freeswitch account returns online and works again for a couple of hours. I'm behind a nat and firewall. Is there any port which I have to open to keep freeswitch online? As far as I understand only the google XMMP server needs to listen to port 5222 and port 5269. Or, is there any kind of "keep alive" timer available? The google talk client running in the same environment works flawlessly. Thanks for any advise. Fede

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Conference and Speex Wideband audio quality problem
by Richard Alam 23 Nov '11

23 Nov '11

Hi Everyone, Hopefully, the answer to my problem isn't in the book as I've yet to order it. :) First off, I'm one ot the devs of BigBlueButton (http://code.google.com/p/bigbluebutton/) We currently use Asterisk for voice conference. Leif Jackson made BBB work with FreeSWITCH. Now, from BBB, a user can join a voice conference in FS. We have a java SIP client (based on Red5Phone) that calls FS. All is working well for Ulaw. I am now trying to make Speex WB work since Flash Player (where our client runs) supports Speex WB. Using Speex WB frees the java SIP client from transcoding between Nellymoser and Ulaw. The java SIP client just takes the speex audio data from RTMP and sends it to FS through RTP. However, the sound quality is bad. It is choopy and robotic. Let me describe what I have done and what I found from debugging so far. I modified vars.xml to add the speex codec. ================ <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@16000h@20i,speex@8000h@20i,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@16000h@20i,PCMU,PCMA,GSM"/> Setup the conference in dialplan/public.xml ================== <extension name="bbb_conferences"> <condition field="destination_number" expression="^(7\d{4})$"> <action application="answer"/> <action application="conference" data="$1@wideband"/> </condition> </extension> When I start up freeswitch, it looks like it's loading the speex codec fine. Full console log is here http://pastebin.com/rnHfgQYj =============== 2010-07-30 16:04:10.620375 [NOTICE] switch_loadable_module.c:251 Adding Application 'AvoidingDeadlock' 2010-07-30 16:04:10.620658 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_g723_1.so **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file: No such file or directory** 2010-07-30 16:04:10.620726 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_g729.so **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No such file or directory** 2010-07-30 16:04:10.621189 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_amr.so **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No such file or directory** 2010-07-30 16:04:10.621547 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_ilbc] 2010-07-30 16:04:10.621618 [NOTICE] switch_loadable_module.c:185 Adding Codec 'iLBC' (iLBC) 8000hz 30ms 2010-07-30 16:04:10.621647 [NOTICE] switch_loadable_module.c:185 Adding Codec 'iLBC' (iLBC) 8000hz 20ms 2010-07-30 16:04:10.622043 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_speex] 2010-07-30 16:04:10.622108 [NOTICE] switch_loadable_module.c:185 Adding Codec 'SPEEX' (Speex) 32000hz 20ms 2010-07-30 16:04:10.622135 [NOTICE] switch_loadable_module.c:185 Adding Codec 'SPEEX' (Speex) 16000hz 20ms 2010-07-30 16:04:10.622160 [NOTICE] switch_loadable_module.c:185 Adding Codec 'SPEEX' (Speex) 8000hz 20ms 2010-07-30 16:04:10.622513 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_h26x] When I call into the conference, it looks like mod_conference does not use speex codec but L16@16000hz. ====== 2010-07-30 16:05:56.874138 [DEBUG] sofia.c:4317 Channel sofia/external/75115(a)192.168.0.166 entering state [completed][200] 2010-07-30 16:05:56.874138 [DEBUG] switch_core_session.c:647 Send signal sofia/external/75115(a)192.168.0.166 [BREAK] 2010-07-30 16:05:56.874138 [DEBUG] switch_channel.c:2494 (sofia/external/75115(a)192.168.0.166) Callstate Change RINGING -> ACTIVE 2010-07-30 16:05:56.874138 [NOTICE] mod_dptools.c:746 Channel [sofia/external/75115(a)192.168.0.166] has been answered EXECUTE sofia/external/75115(a)192.168.0.166 conference(75115@wideband) 2010-07-30 16:05:56.875481 [DEBUG] mod_conference.c:5237 Raw Codec Activation Success L16@16000hz 1 channel 20ms 2010-07-30 16:05:56.875481 [DEBUG] mod_conference.c:5282 Raw Codec Activation Success L16@16000hz 1 channel 20ms 2010-07-30 16:05:56.877940 [DEBUG] switch_core_codec.c:116 sofia/external/75115(a)192.168.0.166 Push codec L16:10 tport_wakeup_pri(0x95bfd98): events IN Is this the correct behavior? I've tried using xlite to connect to the conference, and the audio is good. Also, when connecting from our client, the audio file that gets played by FS to notify how many users are in the conference is good. It's my voice that's bad. I tried to use the echo application instead of the conference. When I call from the BBB client, the sound quality is good. I appreciate any tips on how I can fix the problem or proceed to figure out how to fix it. Thank you very much. Richard -- --- BigBlueButton http://www.bigbluebutton.org http://code.google.com/p/bigbluebutton

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external sip profile
by Chad Vogel 18 Nov '11

18 Nov '11

hello, I'm trying to switch from asterisk to freeswitch; however i'm wondering how can I create a sip profile because the sip profile i created doesn't seem to function with level 3. here is the sip profile i created that isn't working: <include> <gateway name="Level3"> <param name="username" value="USERNAME"/> <param name="password" value="PASSWORD"/> <param name="proxy" value="4.55.35.60:5070"/> <param name="expire-seconds" value="3600"/> <param name="register-transport" value="udp"/> <param name="register" value="true"/> </gateway> </include> here is my asterisk profile (it works): [level3_out] type=peer nat=no host=4.55.35.60 username=***Username*** secret=***Password*** dtmfmode=rfc2833 port=5070 [level3_in] nat=no insecure=very dtmfmode=rfc2833 disallow=all context=from-trunk canreinvite=no allow=ulaw&alaw host=4.55.35.60 type=peer port=5070 How can I create a sip profile that will function the same in freeswitch?

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Help with choppy audio after attended transfer
by Chris Cureau 17 Nov '11

17 Nov '11

I'm having some issues with extremely choppy audio after a call has been sent to another extension via an automated transfer. The audio is great when the call is answered. Shortly after, the transfer button is pressed and the incoming call hears music on hold. The music on hold is sent to the caller sounds fine as does the conversation between extensions. When the transfer is completed, the caller hears what sounds like someone speaking through a fan (though slower) but incoming audio sounds fine. Since it's such a large log, I posted it to the FreeSWITCH pastebin: http://pastebin.freeswitch.org/16911 I'm thinking that it has something to do with the transition from MOH to the internal extension, but I can't figure out what is happening. Any ideas?

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Problem 'adding' video to an audio initiated call, fmtp
by Tyler Winter 07 Nov '11

07 Nov '11

Hi, I'm trying to add video to a non-video initiated call. I should mention that I have no problems initiated and answering a call as video (using Bria softphone and H.264 codec). I captured several test calls to compare the SDP. I found that adding video doesn't work only when initiating the call as audio. In other words, I can initiate the call as video on the A-leg, and answer the call as audio on the B-leg and then add video on the B-leg and video starts working. I think problem is due to there being no fmtp "media attribute" on the 200 OK w/SDP response from FS to the A-leg when I initialize the call as audio, then add video to it. I believe I narrowed down the problem to be the following: - Caller (a-leg) initiated the call as audio. - Callee (b-leg) answered the call as video - Caller (a-leg) "adds" video which sends an INVITE to the switch to negotiate the video codecs - FS responds to Caller (a-leg) with 200 OK (SDP) but the "fmtp" media attribute is missing. On a working video call the fmtp attribute is appended to the SDP along with the rest of the video codec details. Here's an example: this is the INVITEs SDP information to FS from the caller (a-leg) on a working video call (initiated as video): Media Description, name and address (m): video 55752 RTP/AVP 123 124 Bandwidth Information (b): TIAS:1574400 Media Attribute (a): rtpmap:123 H264/90000 Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1 Media Attribute (a): rtpmap:124 H264/90000 Media Attribute (a): fmtp:124 profile-level-id=42801f;packetization-mode=1;level-asymmetry-allowed=1 Media Attribute (a): sendrecv 200 OK from FS to the Caller (a-leg): Media Description, name and address (m): video 16454 RTP/AVP 123 Media Attribute (a): rtpmap:123 H264/90000 Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1 Now, when I try to initiate a call as audio, then add video: INVITE from the caller (a-leg) to FS ("adding" video during call): Media Description, name and address (m): video 64698 RTP/AVP 123 124 Bandwidth Information (b): TIAS:1574400 Media Attribute (a): rtpmap:123 H264/90000 Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1 Media Attribute (a): rtpmap:124 H264/90000 Media Attribute (a): fmtp:124 profile-level-id=42801f;packetization-mode=1;level-asymmetry-allowed=1 Media Attribute (a): sendrecv 200 OK from FS to the caller (a-leg): Media Description, name and address (m): video 23678 RTP/AVP 123 Media Attribute (a): rtpmap:123 H264/90000 So, it appears that the "fmtp" attribute is missing on the 200 OK response: Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1 I can upload the captures somewhere if the full details will be helpful to anyone. If anyone has had any luck with this feature, I'm interested in how you accomplished it. I also tried using the setting "sip_force_video_fmtp" to no avail. I appreciate any help. Regards, Tyler

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