July 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] How to Configure SIP DID to IVR
by Hristo Benev 02 Jul '08

02 Jul '08

>-------- Оригинално писмо -------- >От: Michael Jerris >Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR >До: freeswitch-users(a)lists.freeswitch.org >Изпратено на: Сряда, 2008, Юли 2 19:24:22 EEST >"^" seems like an invalid regex. is that literally what >you have there or you have some number? > >Mike > >On Jul 2, 2008, at 12:16 PM, Hristo Benev wrote: > >> Hi, >> >> I'm new to FS and trying to configure DID only configuration. >> >> Here is the setup: >> PSTN Cisco AS(realIP/maybe multiple ones in production) >> FS(realIP) >> >> Cisco box is configured to send SIP to IP (real IP nor 192.168.x.x >> type) and I do not have much control over it. No authentication is >> needed. >> >> I'm using FS 1.0.0 >> >> What I need to configure to send incoming PSTN calls to demo IVR >> What I've changed? >> Created cisco.xml file in /conf/directory/default >> ---------------- >> >> >> "/> >> "/> >> "/> >> >> >> ------------------ >> Added to /conf/dialplan/default.xml >> ----------------------------- >> >> >> "> >> >> >> >> >> >> ------------------------------ >> When I call DID it just rings. >> If I connect to FS with SoftPhone on extension and I dial DID. >> >> I was able to get this configuration working with Asterisk(but had >> some sound quality issues and wanted to try something else) so there >> is no HW problem. >> >> Where is my misconfiguration(hopefully just this)? >> >> Thanks >> >> _______________________________________________ >> 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 > > >_______________________________________________ >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 Yes there is an actual number that I do not wanted to disclose. I have some progress now call are accepted by FS, but something is wrong after dialplan_hunt() is executed it hangs up. Thanks

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How to Configure SIP DID to IVR
by Hristo Benev 02 Jul '08

02 Jul '08

Hi, I'm new to FS and trying to configure DID only configuration. Here is the setup: PSTN <->Cisco AS(realIP/maybe multiple ones in production) <-> FS(realIP) Cisco box is configured to send SIP to IP (real IP nor 192.168.x.x type) and I do not have much control over it. No authentication is needed. I'm using FS 1.0.0 What I need to configure to send incoming PSTN calls to demo IVR What I've changed? Created cisco.xml file in /conf/directory/default ---------------- <include> <gateway name="Cisco1"> <param name="extension" value="<DID number>"/> <param name="realm" value="<CiscoIP>"/> <param name="proxy" value="<CiscoIP>"/> </gateway> </include> ------------------ Added to /conf/dialplan/default.xml -----------------------------  <extension name="cisco"> <condition field="destination_number" expression="^<DID number>"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> ------------------------------ When I call DID it just rings. If I connect to FS with SoftPhone on extension and I dial DID. I was able to get this configuration working with Asterisk(but had some sound quality issues and wanted to try something else) so there is no HW problem. Where is my misconfiguration(hopefully just this)? Thanks

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Compile freeswitch for Windows Vista
by Kin Quek 02 Jul '08

02 Jul '08

Hi All, I am new to FS and I tried to compile FS for Windows Vista and I will appreciate help if anyone has this experience. I have installed VS 2005 Express Edition and Platform SDK in Vista. I downloaded the FS source and unzip it to freeswitch directory. I found Freeswitch.2008.sln and Freeswich.sln. I tried to start Freeswitch.sln and got many error messages. Any suggestions? Thanks, Kin

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seeking a few hours (paid) help with Python <-> FS
by Brian B 02 Jul '08

02 Jul '08

We have a FS box on Lylix/CentOS connected to les.net and making and receiving calls, running commands from the console etc. However, my tech resource is not available for the time being, and I need some help setting up a basic Python "hello world" which will capture the event of an incoming call, and execute a couple custom conference commands as each person enters. Is anyone available for an hour or more of dedicated help -- paid? We're a pre-funding start-up aiming for a non-profit market segment ... so don't have a huge budget, but we *do *have a small budget, a ton of energy, a perhaps-better-than-most chance of having a positive impact on the world, and a whole HEAP of gratitude. Thanks for all your help! I updated the Wiki with all of the tips I got last round, btw. -- =========================== Brian Burt 415-308-4258 mobile (best) freeswitch(a)brian-burt.com ===========================

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Re: [Freeswitch-users] openzap / dialplan / Sangoma A204 questions
by John Wehle 02 Jul '08

02 Jul '08

> Can you try upgrading to the latest svn build of FS. > There are several fixes to openzap in there that I know will fix your issue. Yes, that works much better. A couple minor changes required so the openzap code would compile on FreeBSD 6.2: a) The header for select needed to be included. b) A comment needed to be fixed. I'm now able to dial out through the fxo as well as have incoming calls answered. What would my dialplan look like so that dialing 551 just bridges the call to the tip ring line from the PBX? I.e. dialing 551 gets me a PBX dialtone without actually dialing an extension on the PBX. -- John -----------------------8<-----------------------8<----------------------- *** libs/openzap/src/include/openzap.h.ORIGINAL Tue Jul 1 19:07:52 2008 --- libs/openzap/src/include/openzap.h Tue Jul 1 19:20:12 2008 *************** *** 127,132 **** --- 127,133 ---- #include <strings.h> #endif #include <assert.h> + #include <sys/select.h> #include "zap_types.h" #include "hashtable.h" #include "zap_config.h" *** libs/openzap/src/isdn/Q931.c.ORIGINAL Tue Jul 1 19:07:51 2008 --- libs/openzap/src/isdn/Q931.c Tue Jul 1 19:17:25 2008 *************** *** 346,352 **** /* Protocol Discriminator */ m->ProtDisc = Mes[IOff++]; ! /* CRV */add m->CRVFlag = Mes[IOff + 1] & 0x80; m->CRV = Q931Uie_CRV(pTrunk, Mes, m->buf, &IOff, &ISize); --- 346,352 ---- /* Protocol Discriminator */ m->ProtDisc = Mes[IOff++]; ! /* CRV add */ m->CRVFlag = Mes[IOff + 1] & 0x80; m->CRV = Q931Uie_CRV(pTrunk, Mes, m->buf, &IOff, &ISize); ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Re: [Freeswitch-users] openzap / dialplan / Sangoma A204 questions
by John Wehle 02 Jul '08

02 Jul '08

> Can you try upgrading to the latest svn build of FS. > There are several fixes to openzap in there that I know will fix your issue. Note I do get the following messages: [WARNING] zap_zt.c:356 zt_open() Echo training not available for 1:2 [WARNING] zap_zt.c:642 zt_next_event() Unhandled event 17 Any idea what they mean? Should I be concern? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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How to dial out via Voipbuster ?
by Henk Oegema 01 Jul '08

01 Jul '08

I have created a gateway to Voipbuster in conf/sip_profiles/external/voipbuster.xml: <include> <gateway name="voipbuster"> <param name="username" value="henkoegema"/> <param name="realm" value="voipbuster.com"/> <param name="from-user" value="003216980xxx"/> <param name="password" value="xxxxxx"/> <param name="extension" value="1000"/> </gateway> </include> I suppose this is correct since I can register with Voipbuster: freeswitch@ubuntu> sofia status API CALL [sofia(status)] output: Name Type Data State =============================================== internal profile sip:mod_sofia@192.168.1.101:5060 RUNNING (0) external profile sip:mod_sofia@78.21.64.194:5080 RUNNING (0) voipbuster gateway sip:henkoegema@voipbuster.com REGED nat profile sip:mod_sofia@78.21.64.194:5070 RUNNING (0) default alias internal ALIASED 192.168.1.101 alias internal ALIASED outbound alias external ALIASED ================================================ 3 profiles 3 aliases Then I added the following lines to conf/dialplan/default.xml  <extension name="voipbuster"> <condition field="destination_number" expression="^(0031\d*})$"> <action application="bridge" data="sofia/gateway/voipbuster/$1(a)voipbuster.com"/> </condition> </extension> Something is still not correct: Get always busy tone when I dial 003134464135x (?) No Route !! 2008-07-01 17:47:21 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting What am I doing wrong ? Thanks Henk

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openzap / dialplan / Sangoma A204 questions
by John Wehle 01 Jul '08

01 Jul '08

I have a A204 with hardware echo cancelling and two FXO modules on FreeBSD 6.2 connected to tip-ring lines from a PBX. ztcfg reports: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) Hooking a plain old telephone to the tip-ring lines from the PBX works fine. On startup freeswitch reports: [DEBUG] zap_io.c:1951 load_config() found config for span [DEBUG] zap_io.c:1978 load_config() created span 1 of type zt [DEBUG] zap_io.c:1991 load_config() span 1 [name]=[OpenZAP] [DEBUG] zap_io.c:1991 load_config() span 1 [number]=[551] [DEBUG] zap_io.c:1991 load_config() span 1 [fxo-channel]=[1] [DEBUG] zap_io.c:2020 load_config() setting trunk type to 'FXO' start(KEWL) [WARNING] zap_zt.c:135 zt_open_range() this ioctl fails on older zaptel but is harmless if you used ztcfg [device /dev/zap/channel chan 1 fd 26 (Inappropriate ioctl for device)] [INFO] zap_zt.c:170 zt_open_range() configuring device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:25 Ultimately I want freeswitch to run a script when any of the FXO lines receive a call. Playing around produced some questions: 1) I have a dialplan of: <extension name="outgoing-fxo"> <condition field="destination_number" expression="^55[1-4]$"> <action application="set" data="dialed_ext=482"/> <action application="bridge" data="openzap/1/1/${dialed_ext}"/> </condition> </extension> which I'm assuming will cause freeswitch to use the fxo to dial 482 on the PBX when routing a call to 551. When I dial 551 from a VoIP phone I see: [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel OpenZAP/1:1/482 [edf17e96-0247-dd11-9800-001fc6ab49e2] [WARNING] zap_analog.c:52 analog_fxo_outgoing_call() VETO Changing state on 1:1 from DOWN to DIALING [WARNING] zap_zt.c:356 zt_open() Echo training not available for 1:1 however I don't hear anything on the VoIP phone (i.e. no ringing) and extension 482 which is right next to the VoIP doesn't ring. 2) What would my dialplan look like so that dialing 551 bridges the call to the FXO with the FXO just going off hook ... not dialing? I.e. dialing 551 just gets me a PBX line with dialtone. 3) What condition would I use in my dialplan to match an FXO line ringing? I.e. when the FXO line rings I want to invoke javascript. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Conference call 5PM Central UTC-6 on Wednesday
by Brian West 01 Jul '08

01 Jul '08

Everyone is invited... we need more people to get involved in the wiki/ doc project. I would also like to go over usability issues with the website... Download, getting started guide to make sure everyone can get started with minimal effort. We also need to have someone monitor and update the FAQ. I add questions and answers but we need to pay closer attention and make sure we get them all on there. I hope to see everyone there... http://wiki.freeswitch.org/wiki/Wiki_meet_2008_07_02 Please add to this page anything you wish to discuss on the agenda... ASAP. Thanks, Brian West

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