Re: [Freeswitch-users] Compiled freeswitch for Windows Vista, any beginner's DOC avaialble?
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" <[EMAIL PROTECTED]> > To: freeswitch-users@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 > <[EMAIL PROTECTED]> 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 > > ___ Freeswitch-users mailing list Freeswitch-users@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] Meeting at 5PM
Don't forget. Brian West sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to Configure SIP DID to IVR (Maybe fixed)
The ERR stun failed below is killing your call. On Jul 2, 2008, at 3:08 PM, Hristo Benev wrote: > > Strange I changed regex to not ^ and it worked?! > > >> Оригинално писмо >> От: Hristo Benev >> Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR >> До: freeswitch-users@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@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@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@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@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 acc
Re: [Freeswitch-users] How to Configure SIP DID to IVR (Maybe fixed)
Strange I changed regex to not ^ and it worked?! > Оригинално писмо >От: Hristo Benev >Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR >До: freeswitch-users@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@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@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@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@lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>
Re: [Freeswitch-users] How to Configure SIP DID to IVR
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@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@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@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@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@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
Re: [Freeswitch-users] How to Configure SIP DID to IVR
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@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@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to Configure SIP DID to IVR
> Оригинално писмо >От: Michael Jerris >Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR >До: freeswitch-users@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to Configure SIP DID to IVR
"^" 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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to Configure SIP DID to IVR
You don't need a extension created for the cisco... Just set it up to forward the DID to the freeswitch boxes IP on its dial peer.. Then on freeswitch you set up a profile w/ auth calls turned off then have a separate context for that profile that does IP auth for the cisco something like this Setting up gateways is ONLY required if you are going to have to register and use sip username/password auth K > From: Hristo Benev <[EMAIL PROTECTED]> > Reply-To: > Date: Wed, 2 Jul 2008 19:16:03 +0300 (EEST) > To: > Subject: [Freeswitch-users] How to Configure SIP DID to IVR > > 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@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@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] How to Configure SIP DID to IVR
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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Using Mod_fifo
if you add these 2 extensions: 7010 will be for agents who will hear music till someone calls and 7011 will be the customer who will hear hold music until an agent is free. On Wed, Jul 2, 2008 at 1:05 AM, Faraz R. Khan <[EMAIL PROTECTED]> wrote: > Is there any example on how to use mod_fifo? > > I am trying to implement a call centre queue as follows (much like > Asterisk queues) : > > Inbound call-> press 0 for operator -> mod_fifo -> 3 agents of whom any > one can get the call (doing round robin or whatever) > > I checked out: > http://wiki.freeswitch.org/wiki/Mod_fifo > > I understand how to park the call, I understand how to retrieve the call > from the fifo- but I dont understand how this will happen automatically. > Is the expected way to write some kind of JS to run periodically, check > if any of the 3 specified agents are free and send them (originate) to > the fifo 'pop' extension? > > Pointers would be appreciated! If anyone has sample JS to do something > close to the above it would give me a great headstart. > > Thanks! > > -- > Faraz R Khan > Chief Architect > Emergen Consulting Pvt Ltd > +92.21.529.0381 x200 > www.emergen.biz > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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 <[EMAIL PROTECTED]> 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 > > > ___ > Freeswitch-users mailing list > Freeswitch-users@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@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] seeking a few hours (paid) help with Python <-> FS
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) [EMAIL PROTECTED] === ___ Freeswitch-users mailing list Freeswitch-users@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] Compile freeswitch for Windows Vista
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 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Reloading - restarting FS ?
You can add users (people who register to you) but not gateways without restarting the sip profile. On Jul 2, 2008, at 3:14 AM, Anton wrote: > Sorry, just a little not sure to understand correctly - > there is no way to add a new SIP account without restarting > the SIP profile, and so any such changes will cause a call > drop? > Am I correct? > > On Wednesday 02 July 2008 02:38, Brian West wrote: >> Please take the time and make sure the wiki is updated so >> others can learn also. I know some of this info is there >> already but the usability factor is wrong. >> >> /b >> >> On Jul 1, 2008, at 4:22 PM, Henk Oegema wrote: >>> On Tuesday 01 July 2008 23:10:18 Brian West wrote: You'll need to restart the sip profile. sofia profile external restart reloadxml The reloadxml arg is option it'll just make sure it reloads the XML before it restarts the profile. >>> >>> Thanks Brian. That works ! >>> I 've learned enough for today. :) >>> >>> Rgds >>> Henk >>> >>> >>> ___ >>> Freeswitch-users mailing list >>> Freeswitch-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch >>> -users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options >>> /freeswitch-users http://www.freeswitch.org >> >> Brian West >> sip:[EMAIL PROTECTED] >> >> >> >> >> ___ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-u >> sers >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/f >> reeswitch-users http://www.freeswitch.org > > ___ > Freeswitch-users mailing list > Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Reloading - restarting FS ?
Sorry, just a little not sure to understand correctly - there is no way to add a new SIP account without restarting the SIP profile, and so any such changes will cause a call drop? Am I correct? On Wednesday 02 July 2008 02:38, Brian West wrote: > Please take the time and make sure the wiki is updated so > others can learn also. I know some of this info is there > already but the usability factor is wrong. > > /b > > On Jul 1, 2008, at 4:22 PM, Henk Oegema wrote: > > On Tuesday 01 July 2008 23:10:18 Brian West wrote: > >> You'll need to restart the sip profile. > >> > >> sofia profile external restart reloadxml > >> > >> The reloadxml arg is option it'll just make sure it > >> reloads the XML before it restarts the profile. > > > > Thanks Brian. That works ! > > I 've learned enough for today. :) > > > > Rgds > > Henk > > > > > > ___ > > Freeswitch-users mailing list > > Freeswitch-users@lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch > >-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options > >/freeswitch-users http://www.freeswitch.org > > Brian West > sip:[EMAIL PROTECTED] > > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-u >sers > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/f >reeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
