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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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Call sip phones from gtalk / jabber
by kriko 17 Dec '08

17 Dec '08

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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general question about API command
by Carole O. 17 Dec '08

17 Dec '08

Hello, I have a general question about the API commands. Some of them are not available in the dialplan like uuid_transfer. I was wondering how to call an API command without using the CLI. Especially I would be interested in knowing if there is any way to call them from a phone, I mean bind a key to an API command. For instance I would like to transfer both members of a simple call into a conference by dialing *1. I have seen the transfer is possible from the CLI by doing: api uuid_transfer <uuid> -both 3001 but I do not know how to do it else. If somebody could give me an insight about the topic it would be great. Thanks, Carole -- View this message in context: http://www.nabble.com/general-question-about-API-command-tp21032754p2103275… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Speed Dial Emulation
by Michael S Collins 16 Dec '08

16 Dec '08

Do you need something just for one extension? Or system wide? If it's system wide then all you need is an extension that matches a condition like this: <condition field="destination_number" expression="^1$"> Anyone who dials just a single digit 1 will go to this Dialplan entry. From there you can have it bridge to whatever endpoint you like. -MC Sent from my iPhone On Dec 15, 2008, at 7:45 AM, jflowers <jflowers(a)ezo.net> wrote: > > How do I emulate a speed dial setup. That is, from extension 1003 I > dial > just a 1 ( or 2, or 3 etc.) and nothing else and freeswitch dials a > PSTN > number. Is there software to do this? > > > -- > View this message in context: http://www.nabble.com/Speed-Dial-Emulation-tp21016167p21016167.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > 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

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FreeSWITCH Infrastructure / Happy Holidays!
by Brian West 16 Dec '08

16 Dec '08

FreeSWITCers, As the project grows I felt the need for the project to own the critical infrastructure hosting things like SVN, Jira, Fisheye and various other things we as a project use. I have personally paid for two servers that Bandwidth.com is going to be colocating for the project in their data center. Teliax paid for and shipped me the two 3ware raid1 cards for the machines which was a great help. The servers are going into production after the first of the year and in the process we are rebuilding everything to allow for single sign on for every service on the FreeSWITCH project. The servers are in the hands of bandwidth.com to be racked up so we can start deployment, as you can see we have our work cut out for us with the integration and rebuild of our entire infrastructure. ;) If you wish to pitch in my personal paypal is brian(a)freeswitch.org or you can use chipin http://www.chipin.com/contribute/id/ddc094318ae5ab5b I would like to thank Bandwidth.com and Teliax for their support... I hope others will help out! Also we have another sounds order going in sometime near the end of January so any money that exceeds the server cost will go to that. The sound order I placed monday will be in this Friday. Happy Holidays and wish you the best for 2009! Thanks, Brian West FreeSWITCH

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