July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

PD: SIP -> XMPP calls
by Muszalski Michał - Korpo TP 12 Aug '14

12 Aug '14

> Hello, > > We are trying to make some kind of a gateway between SIP and XMPP domain. > We have an environment with FreeSWITCH and OpenFire (FreeSWITCH is > registered as a component in OF). Calls from XMPP to SIP are working fine. > The problem is with calls from SIP to XMPP. The caller (SIP) has a ringing > tone, a the callee (XMPP) has a 'connecting...' message after answer the > call. > Do you have any experience in making calls between SIP and XMPP users? Is > it possible to establish the video calls also? > > Best Regards, > > Michał Muszalski > Senior Specialist > > Orange Labs Poland > Obrzeżna 7, 02-691 Warsaw >

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Dingaling In Component Mode "How To Call JingleTo SIP"
by Ali Jawad 11 Aug '14

11 Aug '14

Dear All This might be a duplicate post, I am sorry if it is. Normally I get direct feedback from the list. But this time I did not get any reply. So I was afraid it might not have been delivered. First I would like to state that dingaling is working in client mode with gtalk perfectly. I can call from gtalk to my sip extension "1000". I have setup component mode now I am using FS with Openfire, open does show the following: 1 fs.jabber2.mydomain.net LibDingaLing gateway However stats on the jabber server are Session Statistics: Packets Received/Sent: 1/1 So the component is REGISTERED, so far so good my problem now is how to initiate a call from XMPP to a SIP extension. Using client mode I would do the following: 1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings I don't know how to do the same using component mode. Adding sip+1000@mydomain to the jabber client "Jabbin" does not popup an "add contact" on XTEN. Anything I missed here ? Thanks for any hints or suggestions The Freeswitch server logs are below: 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" to="fs" version="1.0"></stream:stream> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" from="fs.jabber2.MyDomain" id="69fcaa7b"></stream:stream> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <handshake>72dac833190326bb82dddedae558b2b794df5e1b</handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <handshake></handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <iq type="get" id="970-504" from="component.jabber2.MyDomain" to="fs.jabber2.MyDomain"> <query xmlns="http://jabber.org/protocol/disco#info"></query> </iq> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <iq from="fs" to="component.jabber2.MyDomain" id="970-504" type="result"> <query xmlns="http://jabber.org/protocol/disco#info"> <identity category="gateway" name="LibDingaLing"></identity> <feature var="http://jabber.org/protocol/disco"></feature> <feature var="http://jabber.org/protocol/disco#info"></feature> <feature var="jabber:iq:version"></feature> <feature var="vcard-temp"></feature> <feature var="http://www.google.com/xmpp/protocol/voice/v1"></feature> <feature var="jabber:iq:last"></feature> </query> </iq>

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Compiling Freeswitch for Android?
by GillesToo 25 Apr '14

25 Apr '14

Hello Searching the archives only seems to return information on how to use SIPDroid as an SIP client to connect to a Freeswitch server. Since Android is built on Linux, and Freeswitch has been compiled for compact devices, I was wondering if someone had tried compiling it for Android? I could use a small IVR on a smartphone to handle incoming calls. Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compiling-Freeswitch-for-Andr… Sent from the freeswitch-users mailing list archive at Nabble.com.

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DID providers - any thoughts? (jan 2013)
by Cal Leeming [Simplicity Media Ltd] 22 Apr '14

22 Apr '14

Hello everyone, I've been recently reviewing several DID providers, and looking for some feedback. I appreciate this isn't totally on topic for FreeSWITCH, but over the last few months I've found the community feedback to be invaluable - so I'm hoping it might open up a good discussion. I would be really interested to hear other peoples experiences with their DID providers, as much detail as possible. There are several providers we've looked at, but they are either reselling others services, or do not deliver the service quality we are looking for. The criteria I've been looking for is; * Able to provide either US or UK numbers * Exposes an API that allows control of purchasing and routing * Guarantees premium call quality, good latency, and inbound CLI. * Does NOT operate on a market/exchange basis * Trading as a telecoms provider for more than 5 years * Has a reputable name within the community or industry So far, I have only found the following providers that meet this criteria; * Hello Telecom - UK numbers only, ITSPA registered, no charges for minutes, rental or concurrent channels. No minimum charges. Excellent call quality/latency, no CLI problems, and almost no downtime in the last 6 months. However, they do not offer a sane purchasing API which makes automation difficult, and interop with FS has been somewhat painful. * Simwood - UK numbers only, ITSPA registered. Slightly more expensive but still very fair. Currently at inter-op stage, so can't comment further - but they seem very good so far. * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, difficult to justify unless you already have existing traffic. * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over 9000 pages of inter-op and around £10k in set up fees. Thanks in advance Cal

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Cisco phone registration
by andpe 20 Apr '14

20 Apr '14

Hi I have a problem with the registration of the Cisco 79xx phones. The SIP Message Header (MH) which sends the phone, in the fields "TO" and "FROM", places the IP address instead of the domain name. The configuration of the phone is set to the domain name (xx.yy.com.) Is there any way to set the phone to send the domain name in the field instead of an IP address? Is there a firmware version that sends MH correctly? Andy

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avmd from mod managed
by Malay Thakershi 24 Mar '14

24 Mar '14

Hello, Is there a way to do AVMD from mod_managed? I see JavaScript example technique from http://wiki.freeswitch.org/wiki/Mod_avmd but it seems signature of StreamFile is different in mod_managed. In JavaScript, second argument to StreamFile is a function (OnInput) where detection is done while is mod_managed, it is sample start number. So is AVMD possible from mod_managed? Thank you. Malay

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FreeSwitch Proxy + RTPProxy Media server
by adahary 09 Feb '14

09 Feb '14

I would like to integrate FS with RTPproxy like openSips and kamailio are well integrated with it. FS should handle the SIP signaling and the RTPproxy should relay the RTP stream from A to B: A.sip <=> FS <=> B.sip FS = PASS-THRU A.rtp <=> RTPproxy <=> B.rtp. I understand that FS should ask the RTPproxy to allocate UDP ports for both endpoint and then pass-thru-bridge them to cummunicate directly through the RTPproxy. What I cannot yet figure out is how replace both A and B ip/port sets in SDP with the RTPproxy ip/port-udp. I'v read about switch_r_sdp (Leg.A) and switch_m_sdp (Leg.B !?) but couldn't figure it out. how should I use switch_r_sdp (or/and switch_m_sdp) in my dialplan when when A calls B? Tried the following with no help. <action application="set"> </action> <action application="set"> </action> Any tip/help/advise will be appriciated. Regards Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Med… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Lenny's killing it.
by Gabriel Gunderson 26 Jan '14

26 Jan '14

No way! Lenny is on a roll! It's too perfect (listen in order)! 1) http://itslenny.com/recording.php?file=4d8cde0cc5a4d86ccf221aa5b2903e95 2) http://itslenny.com/recording.php?file=4fdd3f445fff8e1b73456af602367b6e Classic! Gabe

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TLS error?
by Federico Beffa 27 Dec '13

27 Dec '13

Hi, I'm trying to setup FS as a gtalk client. I'm working on Linux Ubuntu 10.04 and have cloned the git FS repository about 1 week ago. I can compile everything with no errors. However, after configuring client.xml and starting FS I see the following error message in the freeswitch.log file: 2010-07-20 23:38:56.392853 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_ dingaling.so **/opt/freeswitch/mod/mod_dingaling.so: undefined symbol: gnutls_global_init** TLS comes from the following Ubuntu packages: libgnutls26, libgnutls-dev. I would appreciate any suggestion on how to investigate further the problem. Could it be a gnutls library version incompatibility? A bug in FS? In the ubuntu gnutls packages? Or simply an error in the configuration? Thanks for you help! Fede

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TLS Audiocodes
by Szasz Szabolcs 26 Dec '13

26 Dec '13

Hi, Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got to work TLS between freeswitch and a softphone (phonerlite), but I have problem with Audiocodes during the TLS authentication. I've loaded the certification but it still doesn't work. Can I debug the tls in freeswitch? Please help me. Thank you in advance. Szasz Szabolcs

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