December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Question about running Freeswitch in the background
by mark morreny 18 Dec '08

18 Dec '08

Hi, I have a small question about running Freeswich in the background using -nc option. Does freeswitch writes the log in any log file when running in background mode? If I have some print out statement in my custom mod, can I still see the output of those logs somewhere? Thanks, Mark

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Re: [Freeswitch-users] Call sip phones from gtalk / jabber
by Kirk Bateman 18 Dec '08

18 Dec '08

Kriko, Have a look at this, I used it to get my gtalk to fs working. http://chesterton.id.au/blog/2008/01/02/freeswitch-google-talk-dingaling-ji… Cheers Kirk Date: Thu, 18 Dec 2008 09:37:38 +0100 > From: kriko <kristjan.ugrin(a)gmail.com> > Subject: Re: [Freeswitch-users] Call sip phones from gtalk / jabber > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <op.umcnkzr2atcbto(a)linux.site> > Content-Type: text/plain; format=flowed; delsp=yes; charset=utf-8 > > I'm not sure if we understood each other correctly. > I meant calling from jabber to other sip phones. I'm not sure you can just > add a sip phone as > a buddy into e.g. gtalk or any other jabber service and call it. > So that why a bot (and only one but), which you would have as a buddy and > you would feed him with numbers. > And he would estabilish a call between jabber user and typed in number. > Was I clearer? > > I'll take a look how the component mode works, hopefully there is more > documentation than only this: > http://wiki.freeswitch.org/wiki/Dingaling > > Also if this functionality is not possible with fs as it is, I could maybe > write a java program which interact with fs > interface and do that as intended? > > Cheers > >

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Re: [Freeswitch-users] Call sip phones from gtalk / jabber
by Kirk Bateman 18 Dec '08

18 Dec '08

Kriko, I have been looking at the same sort of thing, but I'm planning to implement an ejabberd bot component (so I can hopefully use the new mod_erlang_event freeswitch interface). It seems to me that bits of the current dingaling / jingle interface are having problems, like not liking sending messages to other domains, its generally working if they are all .gmail.com users but when you have some on googlemail.com etc it starts breaking, and doesn't use the whole JID in the from attribute for sending messages. When I get a chance I'll try and narrow down the problem. Cheers Kirk Date: Wed, 17 Dec 2008 15:46:16 +0100 > From: kriko <kristjan.ugrin(a)gmail.com> > Subject: [Freeswitch-users] Call sip phones from gtalk / jabber > To: "freeswitch-users(a)lists.freeswitch.org" > <freeswitch-users(a)lists.freeswitch.org> > Message-ID: <op.uma9zdmxatcbto(a)linux.site> > Content-Type: text/plain; format=flowed; delsp=yes; charset=utf-8 > > I have an idea which is takes too many characters for irc. > I'm relatively new to telephony and such stuff, I managed to get > freeswitch running, but I don't fully uderstand > my problem in detail and how to solve it, so I need a bit of directions. > > Briefly, my idea is to have a jabber contact, which would gtalk user add > as their buddy. Let's call it callbuddy(a)somejabbersrv.com. > This would be a bot that would accept commands - I've already made a small > xmpp java bot which just logs into gtalk and send echo > messages to users - like a parrot, nothing serious. Also I've set up > openfire jabber server. > What I would like him to do, is when user would tipe "call 0189432443" it > would initiate a call between contact who tiped in this number (command) > and this number. > > Is this possible, and what would be the best implementation (dingaling > acting as client...)? > > Thanks. > > -- > kriko > >

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ignore dtmf
by stephen at stephenjc 18 Dec '08

18 Dec '08

I have a click to click system written in javascript, so i call out both legs of the call then bridge them. I am looking for a way to ignore dtmf tones on 1 leg of the call. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print.

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multi domain issue
by Tjapko Smits 17 Dec '08

17 Dec '08

Hi, Fresh freeswitch user. Installed freeswitch with default installation a week ago. Need more information on multi domain usage. Followed the wiki pages with multi tenant examples. All working well but I do have a problem when calling to an internal extension. Scenario: domain_A created by copying the default.xml to domain_A.xml following steps from the wiki multi tenant informaion. Than created the domain_A directory and copied the /default directory extensions there. Same steps for Domain_B. For the rest all is like default. Registered 3 phones to Domain_A and 3 phones to Domain_B Domain_A -> 1000 1005 and 1006 Domain_B -> 1002 1005 and 1006 When I call from 1002 inside Domain_B to extension 1000 in Domain_A phone rings and this is what I do not like to happen. Trace shows that when called to 1000 or 1005 or 1006 only the IP addresses from those endpoints in Domain_A are addressed. The first INVITE shows an OK invite with correct domain name but after the 100 trying the re-invite makes an IP address out of most certainly because it followed the rules in the dialplan. Can anybody point me in the direction how to make it possible that calls from DOmain_A stay in Domain_A- and Domain_B stay in B etc. All the rest like inbound and outbound calls on dedicated gateways are working fine. I assume that I need to configure acl for this but still not very clear. Any tip will be most appreciated. -- Tjapko

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Help with routing sound locally through FS
by mszlazak＠aol.com 17 Dec '08

17 Dec '08

I'm making a call internally from a soft phone to an extension that is suppose to bridge the call internally to another application on the same computer. The applications logs indicate that a connection was made but sound is not being passed back from the application through freeswitch to the softphone. There maybe an issue with rtp timing and associated ports but I'm very new at diagnosing this and fixing the problem. I've attached both a copy of the FS log and an associated pcap file. It's all on Windows XP. Could someone please take a look. Thanks.

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Cisco contest
by freeswitch＠davidnicol.otherinbox.com 17 Dec '08

17 Dec '08

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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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Zaptel Error!!!
by fidibus83 17 Dec '08

17 Dec '08

Hello, Im a newbie in FS and my English isnt very good but I try to explain my problem. Hopefully you can understand me. :-) * I have a Linux-Server with a Digium Wildcard TE110P. I install and configure openzap (PRI/ISDN) and zaptel. But I have an error when I execute ztcfg vv: 31 channels configured. ioctl(ZT_LOADZONE) failed: Invalid argument Notice: Configuration file is /etc/zaptel.conf line 288: Unable to register tone zone 'de' ZAPTEL.CONF span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = de defaultzone=de What do I wrong? Please help me!

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DNS RV faiover
by Tamas Cseke 17 Dec '08

17 Dec '08

Hello, We'd like to use DNS SRV for failover. if we are bridge sofia/profile/whatever(a)domain.with.srv it works perfectly but with gateway wich has this record in its proxy parameter it doesn't work. Once we set up an A record too it works, so we assume dialing gateway doesn't use SRV records. Is there any differences between the 2 method? Thanks any help, Tamas

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