- FreeSWITCH-users - FreeSWITCH Mailing Lists

Reloading - restarting FS ?
by Henk Oegema 03 Jul '08

03 Jul '08

When I make a change in a XML file, must I reload (how?) or stop and start FS ? Or don't I need to do anything ? (wait for a while ?) Rgds Henk

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Re: [Freeswitch-users] Compiled freeswitch for Windows Vista, any beginner's DOC avaialble?
by Kin Quek 03 Jul '08

03 Jul '08

Thanks Jeff. The problem was Visual Studio 2005 + Windows SDK Platform was not supported on Vista. I installed Visual Studio 2008 and ignored the error messages and FS built OK. Thanks for the help. Can you please point me to how to test out the new built FS? Is there any beginner's DOC available? Thanks, Kin > > From: "jeff sacksteder" <jsacksteder(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 2 Jul 2008 12:02:38 -0400 > Subject: Re: [Freeswitch-users] Compile freeswitch > for Windows Vista > > You can disregard those 'solution folder' messages. > The source > includes accommodations for building on linux, > vs2005 and vs2008 in > one location. > > In VS you can either build in release mode or debug > mode by setting a > build variable. > > On Wed, Jul 2, 2008 at 5:56 AM, Kin Quek > <kin_quek(a)yahoo.com> wrote: > > 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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Meeting at 5PM
by Brian West 02 Jul '08

02 Jul '08

Don't forget. Brian West sip:brian@freeswitch.org

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

02 Jul '08

Strange I changed regex to <DID> not ^<DID> and it worked?! >-------- Оригинално писмо -------- >От: Hristo Benev >Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR >До: freeswitch-users(a)lists.freeswitch.org >Изпратено на: Сряда, 2008, Юли 2 21:00:59 EEST >Here is the output: >--------------------------------------- >2008-07-02 13:48:47 [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel sofia/cisco/@ [c0d8586f-f6b9-4108-8676-c49e66f32e6d] >2008-07-02 13:48:47 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing ->@cisco >2008-07-02 13:49:12 [ERR] sofia_glue.c:450 sofia_glue_ext_address_lookup() Stun Failed! stun.freeswitch.org:3478 [Timeout] >2008-07-02 13:49:12 [NOTICE] mod_sofia.c:386 sofia_answer_channel() Hangup sofia/cisco/@ [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] >2008-07-02 13:49:12 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 1 (sofia/cisco/@) Ended >2008-07-02 13:49:12 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/cisco/@ [CS_HANGUP] >--------------------------------------- >CallinfNumber is the number I call from >CiscoIP is IP of Cisco AS >DIDNumber is DID I have > >Thanks > >I'm doing something wrong, but what? >Again Here are the files >/conf/sip_profiles/cisco.xml (just copied external.xml and changed sip port) >----------------------------------- > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >---------------------------------------------------------- >/conf/dialpaln/cisco.xml >--------------------------------------------------------- > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >---------------------------------------------- >Sensitive data is obfuscated > > > > >-------- 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 20:22:31 EEST > > >Most likely its not actually matching the extension or it runs out of > >actions to perform, can you post the full debug logs from the console? > > > >Mike > > > >On Jul 2, 2008, at 1:14 PM, Hristo Benev wrote: > > > >>> -------- 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 > >> > >> _______________________________________________ > >> 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 > > > >_______________________________________________ >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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Re: [Freeswitch-users] How to Configure SIP DID to IVR
by Hristo Benev 02 Jul '08

02 Jul '08

Here is the output: --------------------------------------- 2008-07-02 13:48:47 [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel sofia/cisco/<CallingNumber>@<CIscoIP> [c0d8586f-f6b9-4108-8676-c49e66f32e6d] 2008-07-02 13:48:47 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing <CAllingNumber>-><DIDNumber>@cisco 2008-07-02 13:49:12 [ERR] sofia_glue.c:450 sofia_glue_ext_address_lookup() Stun Failed! stun.freeswitch.org:3478 [Timeout] 2008-07-02 13:49:12 [NOTICE] mod_sofia.c:386 sofia_answer_channel() Hangup sofia/cisco/<CallingNumber>@<CiscoIP> [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-07-02 13:49:12 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 1 (sofia/cisco/<CallingNumber>@<CicoIP>) Ended 2008-07-02 13:49:12 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/cisco/<CallingNumber>@<CiscoIP> [CS_HANGUP] --------------------------------------- CallinfNumber is the number I call from CiscoIP is IP of Cisco AS DIDNumber is DID I have Thanks I'm doing something wrong, but what? Again Here are the files /conf/sip_profiles/cisco.xml (just copied external.xml and changed sip port) -----------------------------------  <profile name="cisco">  <gateways> <X-PRE-PROCESS cmd="include" data="cisco/*.xml"/> </gateways> <aliases> <alias name="cisco"/> </aliases> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="5"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="cisco"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> ---------------------------------------------------------- /conf/dialpaln/cisco.xml ---------------------------------------------------------  <include> <context name="cisco"> <extension name="cisco1"> <condition field="network_addr" expression="^xxx\.xxx\.xxx\.xxx$"/> <condition field="destination_number" expression="^xxxxxxxxxxxx$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> <extension name="cisco2"> <condition field="network_addr" expression="^xxx\.xxx\.xxx\.xxx$"/> <condition field="destination_number" expression="^xxxxxxxxxxxx$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> <extension name="cisco3"> <condition field="network_addr" expression="^xxx\.xxx\.xxx\.xxx$"/> <condition field="destination_number" expression="^xxxxxxxxxxx$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> <extension name="cisco4"> <condition field="network_addr" expression="^xxx\.xxx\.xxx\.xxx$"/> <condition field="destination_number" expression="^xxxxxxxxxxx$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> </context> </include> ---------------------------------------------- Sensitive data is obfuscated >-------- 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 20:22:31 EEST >Most likely its not actually matching the extension or it runs out of >actions to perform, can you post the full debug logs from the console? > >Mike > >On Jul 2, 2008, at 1:14 PM, Hristo Benev wrote: > >>> -------- 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 >> >> _______________________________________________ >> 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 >

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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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