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 extension name=cisco condition field=network_addr expression=^192\.168\.2\.1$/ condition field=destination_number expression=^DID number action application=answer/ action application=sleep data=2000/ action application=ivr data=demo_ivr/ /condition /extension 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: freeswitch-users@lists.freeswitch.org Date: Wed, 2 Jul 2008 19:16:03 +0300 (EEST) To: freeswitch-users@lists.freeswitch.org 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 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 - !-- test -- 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 ___ 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
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
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)
---
!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --
profile name=cisco
!-- This profile is only for 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
-
!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --
include
context name=cisco
extension name=cisco1
condition field=network_addr expression=^xxx\.xxx\.xxx\.xxx$/
condition field=destination_number expression=^$
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$/
condition field=destination_number expression=^$
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$/
condition field=destination_number expression=^xxx$
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$/
condition field=destination_number expression=^xxx$
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@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
Re: [Freeswitch-users] How to Configure SIP DID to IVR (Maybe fixed)
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@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 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
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 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@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 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 ___ Freeswitch-users mailing list Freeswitch-users
