October 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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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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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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Stream audio file/live to multiple SIP endpoints with IP multicast
by Sluschny, Thomas 30 Aug '10

30 Aug '10

Hi, i want to stream audio (from file and live) from FreeSwitch to multiple SIP endpoints (these are proprietary). Because of bandwith limitations we have to use IP multicast for audio data. Now the question: how can i do this smartly in FreeSwitch? I have read some Wiki entries: - mod_esf: seams to use multicast, but i couldn't find usage information - mod_shout: streaming, but from internet ? MP3, but i want wav-files to stream - mod_local_stream: seams to play files in a loop, i want to play a whole file once. - mod_conference: seams not to support IP multicast Im searching for a simple way to send a SIP INVITE to all my endpoints and then stream audio (live from soundcard or from file) over IP multicast to all these endpoints. Thanks in advance, Thomas

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OSP Interop w/ Trans Nexus
by Gregory Boehnlein 02 Dec '09

02 Dec '09

Anyone know if Freeswitch can work w/ OSP and use something like Transnexus for CDR/Rating?

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freeswitch as a session border controller
by Robert McNaught 30 Mar '09

30 Mar '09

Hi, I have been reading the recent news articles about freeswitch. I am particularly interested in using it to replace a session border controller we have (kagoor voiceflow 1000). Has anyone actually used this in production as a session border controller for NAT traversal? Thanks Robert McNaught

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Inbound calls
by whudson05 26 Jan '09

26 Jan '09

Hi, I'm experiencing a very specific problem to do with inbound calls through a gateway. When either end of the phone call hangs up then the other end will remain connected as if still in the conversation. If anyone has any idea why this would occur it would be greatly appreciated. Thanks -- View this message in context: http://www.nabble.com/Inbound-calls-tp18694165p18694165.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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busy tone detection
by Gopal krishnan 05 Jan '09

05 Jan '09

Hi, Is it possible to have busy tone detection thru console dialing? -- Thank you with regards, Gopal,

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OpenZap Not working Stable
by palani sivagurunathan 17 Dec '08

17 Dec '08

Hi, I have installed freeswitch through SVN in centos.OutBound is working fine .But not always .Once we make 8 to 10 call from a sip extension we get service unavailable error in xlite . Error and configuration are pasted below .we observed the problem is getting rectified after 5 to 10 mins .and we are able to make calls and again after 8 to 10 calls Errors happen how to check. what will be the issue my configuration is: *wanpipe1.conf * [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 9 PCIBUS = 0 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE = NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TDMV_HW_DTMF = NO [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES MTU = 80 * openzap.conf* [span wanpipe] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 *openzap.conf.xml * <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <pri_spans> <span id="1"> <param name="mode" value="user"/> <param name="dialect" value="euro"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> </configuration> *default.xml* <?xml version="1.0" encoding="utf-8"?>  <include> <context name="default"> <extension name="Long Distance - wanpipe"> <condition field="destination_number" expression="^9(\d{5,15})$"> <action application="set" data="dialed_ext=$1"/> <action application="bridge" data="OpenZAP/1/a/${dialed_ext}"/> </condition> </extension> </context> </include> After starting the freeswitch . In console i type oz dump 1 a *my status before calling is:* span_id: 1 chan_id: 16 physical_span_id: 1 physical_chan_id: 16 type: DQ921 state: UP last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE * my (oz dump 1 a) status After calling is:* span_id: 1 chan_id: 16 physical_span_id: 1 physical_chan_id: 16 type: DQ921 state: UP last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE But while i try to make a outbound call continously, i am getting following error [08 02 00 00 4e 18 03 a1 83 81 79 01 80] 2008-09-23 01:05:50 [DEBUG] Span:0 Q.931() Sending message to Layer4 (size: 119) 2008-09-23 01:05:50 [DEBUG] ozmod_isdn.c:319 zap_isdn_931_34() Yay I got an event! Type:[46] Size:[119] CRV: 0 (0, CTX: Originator) 2008-09-23 01:05:50 [DEBUG] ozmod_isdn.c:351 zap_isdn_931_34() zchan 0 (-1:-1) source isdn_data->channels_remote_crv[0] 2008-09-23 01:05:50 [DEBUG] ozmod_isdn.c:399 zap_isdn_931_34() Changing state on 1:1 from DOWN to RESTART 2008-09-23 01:05:50 [DEBUG] Span:0 Q.931() Sending message to Q.921 (size: 119) 2008-09-23 01:05:50 [DEBUG] Span:0 Q.931() Creating Q.931 Message Header: ProtDisc 8 (0x8), CRV 0 (0), CRVflag: 0 (0), MesType: 78 (0x4e) 2008-09-23 01:05:50 [DEBUG] ozmod_isdn.c:1499 q931_rx_32() WRITE 13 -------------------------------------------------------------------------------- [08 02 00 00 4e 18 03 a1 83 81 79 01 80] 2008-09-23 01:05:50 [DEBUG] ozmod_isdn.c:796 state_advance() 1:1 STATE [RESTART] 2008-09-23 01:05:50 [DEBUG] mod_openzap.c:1368 on_clear_channel_signal() got clear channel sig [RESTART] 2008-09-23 01:05:50 [DEBUG] ozmod_isdn.c:882 state_advance() Changing state on 1:1 from RESTART to DOWN 2008-09-23 01:05:50 [DEBUG] ozmod_isdn.c:796 state_advance() 1:1 STATE [DOWN] 2008-09-23 01:05:50 [DEBUG] zap_io.c:1088 zap_channel_done() channel done 1:1 -- Thanks & Regards, Palanivel.S

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Freeswitch wedges in voicemail?
by Marc Lewis 13 Nov '08

13 Nov '08

I have a strange problem that I've been dealing with for the last week or two. Once or twice a day, Freeswitch will "wedge", in what appears to be voicemail. The switch is sort of still responding, meaning I can issue various commands at the console, but no new sofia channels will be handled, registrations start failing on the internal profile and no new calls can come in from the external profile. When it happens, existing channels stay up, but once they are completed, no new channels can be created. There is always one channel that remains that looks something like (DID obfuscated): api show channels Content-Type: api/response Content-Length: 394 uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate ea758bcf-25f5-4cde-9f6c-5e3f1226130f,2008-10-27 10:02:27,1225126947,sofia/external/253941xxxx@204.11.200.165,CS_EXECUTE,253941xxxx,253941xxxx,199.245.214.150,328,voicemail,default pgm.cisvp.com 246,XML,pgm,PCMU,8000,PCMU,8000 1 total. I've been updating every weekend with the latest code, this is currently trunk rev 10146. Anyone have any thoughts on this? Its driving me crazy. - Marc -- Marc Lewis Avvatel Corporation

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