March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

xmpp
by Peter P GMX 19 Oct '12

19 Oct '12

I saw that xmpp is supported in Fresswitch. See wiki: http://wiki.freeswitch.org/wiki/Mod_xmpp_event Has anybody already set this up? I have found no mod_xmpp neither in my mod directory nor in the source? There was also a question: " Q: Is it possible to send commands to fs via xmpp? Answer: Yes. " Anybody knows what can be done here and how to do this? Best regards Peter

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mod_dingaling picking wrong IP address / no audio?
by Mark Campbell-Smith 14 Sep '12

14 Sep '12

Hi! I am trying to call from my corporate network (firewalled) using Gtalk to Freeswitch. I am not getting any audio. In the logs I see that mod_dingaling is using my internal corporate IP address which is not publically addressable. 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates 146.xx.xx.xx:50320 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable Candidate 146.xx.xx.xx:50320 Further on in the log, I can see GTalk sending a new candidate IP address to use: 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 name=rtp type=local protocol=udp username=e+JTkVHT1xEkqXGD password=fAxU6Pr1oF9Zq48U address=192.168.1.102 port=50322 pref=1.00 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 name=rtp type=stun protocol=udp username=RBqyF2XNMYLfJNoU password=DQMjon1fSVoJIRTp address=124.xxx.xxx.xxx port=50323 pref=0.90 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=relay protocol=udp username=62L5zs2FHbcUdeCJ password=KxmNgkUmZsLfuX6S address=209.xx.xxx.xxx port=19295 pref=0.50 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] Because of this, I never get audio. Any ideas how to fix this? Thanks!

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ESL problem
by Thangappan.M 31 Jul '12

31 Jul '12

Dear all, In the previous post, I got the information that using event outbound socket we can implement the IVR and also see the example in libs/esl/perl/server2.pl. I have seen it and understood the flow of the script.But when I was running that script it tells the following error. Can't locate loadable object for module ESL in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at ESL.pm line 11 Compilation failed in require at server2.pl line 1. Then I checked the 11th line in the ESL.pm that line is (bootstrap ESL;) But in perl directory there is no directory called ESL. What would be the issue?. Is ESL necessary is necessary for implementing IVR using event outbound socket? -- Regards, Thangappan.M

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How to break a playback with loops
by Daniel Liang 24 Jun '12

24 Jun '12

Hi, I am trying to playing back a wav file a few times, and on receiving a dtmf event, it stops playing the file. The command I sent is: sendmsg call-command: execute execute-app-name: playback execute-app-arg: <the wav file> loops: 4 On receiving a dtmf event, I send a break command: sendmsg call-command: execute execute-app-name: break However, that only breaks one loop of the playback. And I have to press a dtmf for 4 times to stop the file. I tried sending 4 break commands all at once. But it didn't work. And even worse, my program stops receiving dtmfs anymore. My guess is that the first break stops the first loop, but before the second loop starts, the following 3 breaks stop something else that I don't know what they are. Any help is appreciated. Thanks. Daniel Liang

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Facebook...
by Klaus Hochlehnert 27 May '12

27 May '12

Hi Guys, just saw that you already have a Facebook entry. Wouldn't it be possible to connect the news/twitter feeds somehow with Facebook? I'd really like to see all news on Facebook... Cheers, Klaus

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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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Hardware phones that do 48Khz Celt: Was Re: Is there any license G729?
by Craig Askings 29 Sep '11

29 Sep '11

Are there any hardware phones that support 48 Khz Celt and automated/mass deployment? Craig. 2009/6/30 Jason White <jason(a)jasonjgw.net>: > Brian West <brian(a)freeswitch.org> wrote: >> Everyone has this need for lower bandwidth calls... I tend to march the >> other way. 48kHz baby! (btw you can do 48kHz in the same bandwidth as a >> single ulaw call) > > 48khz Celt (celt@48000 in your codec preferences) sounds wonderful with > FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio. > This also works well in 48khz conferences. > > I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711, > G.722, G.722.1 and (my current favourite) Celt all the way. > > > _______________________________________________ > 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 > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig(a)overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365

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T.38 via UPDATE request
by Peter P GMX 19 May '11

19 May '11

Hello, we have the following problem. 2 Fax machines are communicating via Freeswitch. One is externally attached via a Telco who is able to handle T.38. The other one is attached locally. When 2 Fax machines start syncing each other, the Telco sends a SIP UPDATE message with T.38 SDP, as it detects fax during the fax negociations. Freeswitch answers with an SIP OK message back to the telco, and I can see the T.38 SDP on the debug console of freeswitch. Then nothing happens any more until one of fax machines detects timeout. We have set proxy-media to true. However is was done during call setup when both machines communicated with G711 SDP. The UPDATE message was commited by FS to the telco, but was not sent to the other fax, so I think in this case Freeswitch is supposed to transcode between T.38 and G711 which it cannot do, as we know. How can I overcome this scenario? Is this a defect, should freeswitch send the UPDATE message to the other fax? Or is there a workaround? Best regards Peter Here's the UPDATE message: UPDATE sip:mod_sofia@82.115.xx.xxx:5080 SIP/2.0. Call-ID: 5283fe4e-334f-122d-d1b9-001517956764. Contact: <sip:212.91.xxx.xxx:5060>. Content-Type: application/sdp. CSeq: 16340063 UPDATE. From: <sip:06912345678@sip.telco.de>;tag=00-08135-017e041a-4d21f6037. Max-Forwards: 31. Route: <sip:82.115.96.165;lr;ftag=pH663F4S02erm>. To: "030987654321" <sip:030987654321@82.115.xx.xxx>;tag=pH663F4S02erm. User-Agent: Cirpack/v4.41e (gw_sip). Via: SIP/2.0/UDP 212.91.xxx.xxx:5060;branch=z9hG4bK-178C-1923B31. Content-Length: 300. . v=0. o=cp10 125551618103 125551618105 IN IP4 212.91.xxx.xx. s=SIP Call. c=IN IP4 212.91.xxx.xx. t=0 0. m=image 6860 udptl t38. a=sendrecv. a=T38FaxVersion:0. a=T38MaxBitRate:9600. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:1000. a=T38FaxMaxDatagram:200. a=T38FaxUdpEC:t38UDPRedundancy.

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xml_curl and gateways
by Peter P GMX 21 Feb '11

21 Feb '11

According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to set up gateways dynamically, but I do net get it to work: I always get : "Invalid profile" My assumptions for a right xml answer back to FS are as follows, but I think at least one of it is false: 1. ) I start with <document type='freeswitch/xml'> <section name='configuration'> <configuration name='sofia.conf' description='RailsAdditionalConf'> 2.) I do not want to change settings, so I do NOT add a <settings> tag 3.) I want to enhance an existing profile, so I add a profile tag (<profile name='external'>) 4.) I assume that I can have an existing profile "external" in the xml conf files and can mix additional parts dynamically into the same profile 5.) Within the profile I want to add gateways, so I add a gateway tag (<gateways>) 6.) Then I add the gateways 7.) I add the closing tags Which one of the assumptions if wrong? The request I receive is as follows: {"key_name"=>"name", "action"=>"directory", "tag_name"=>"configuration", "key_value"=>"sofia.conf", "controller"=>"xml_curls", "hostname"=>"freeswitch", "section"=>"configuration"} I only answer once at startup with the follwing: <document type='freeswitch/xml'> <section name='configuration'> <configuration name='sofia.conf' description='RailsAdditionalConf'> <profile name='external'> <gateways> <gateway name="freeswitch2"> <param name="username" value="1007"/> <param name="realm" value="192.168.0.56:5061"/> <param name="password" value="xxxx"/> <param name="proxy" value="192.168.0.56"/> <param name="expire-seconds" value="30"/> <param name="register" value="true"/> <param name="register-transport" value="tls"/> <param name="retry-seconds" value="30"/> </gateway> </gateways> </profile> </configuration> </section> </document> Does anybody have a correct XML answer for a gateway and can post it? Best regards Peter

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