Hi,

Managed to ge

I added the following to the dialplan:

No ring.

I get the following:

2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:403 Set codec PCMA 20ms

2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:1317 Connect outbound channel OpenZAP/1:4/

2010-07-01 11:07:04.627394 [NOTICE] switch_channel.c:776 New Channel OpenZAP/1:4/ [3ca74e5b-7e26-4a0a-b5f7-72d7a2f33fbf]

2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:1331 (OpenZAP/1:4/) State Change CS_NEW -> CS_INIT

2010-07-01 11:07:04.627394 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK]

2010-07-01 11:07:04.627394 [DEBUG] ozmod_analog.c:59 Changing state on 1:4 from DOWN to DIALING

2010-07-01 11:07:04.627394 [DEBUG] ozmod_analog.c:293 ANALOG CHANNEL thread starting.

2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_INIT

2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:4/) State INIT

2010-07-01 11:07:04.628392 [DEBUG] mod_openzap.c:431 (OpenZAP/1:4/) State Change CS_INIT -> CS_ROUTING

2010-07-01 11:07:04.628392 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK]

2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:4/) State INIT going to sleep

2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_ROUTING

2010-07-01 11:07:04.628392 [DEBUG] switch_channel.c:1471 (OpenZAP/1:4/) Callstate Change DOWN -> RINGING

2010-07-01 11:07:04.627394 [DEBUG] ozmod_wanpipe.c:608 Enabled DTMF events on chan 1:4

2010-07-01 11:07:04.628392 [DEBUG] ozmod_analog.c:464 Executing state handler on 1:4 for DIALING

2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:4/) State ROUTING

2010-07-01 11:07:04.628392 [DEBUG] mod_openzap.c:454 OpenZAP/1:4/ CHANNEL ROUTING

2010-07-01 11:07:04.628392 [DEBUG] switch_ivr_originate.c:64 (OpenZAP/1:4/) State Change CS_ROUTING -> CS_CONSUME_MEDIA

2010-07-01 11:07:04.628392 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK]

2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:4/) State ROUTING going to sleep

2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_CONSUME_MEDIA

2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:4/) State CONSUME_MEDIA

2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:4/) State CONSUME_MEDIA going to sleep

2010-07-01 11:07:20.335200 [DEBUG] ozmod_wanpipe.c:1036 1:4 wanpipe returned event 5

2010-07-01 11:07:20.335200 [DEBUG] ozmod_wanpipe.c:1061 1:4 rxhook, state 2

2010-07-01 11:07:21.150195 [DEBUG] ozmod_wanpipe.c:1036 1:4 wanpipe returned event 5

2010-07-01 11:07:21.150195 [DEBUG] ozmod_wanpipe.c:1061 1:4 rxhook, state 1

2010-07-01 11:07:21.150195 [DEBUG] ozmod_analog.c:802 EVENT [ONHOOK][1:4] STATE [DIALING]

2010-07-01 11:07:21.150195 [DEBUG] ozmod_analog.c:838 Changing state on 1:4 from DIALING to DOWN

2010-07-01 11:07:21.166199 [DEBUG] ozmod_analog.c:464 Executing state handler on 1:4 for DOWN

2010-07-01 11:07:21.166199 [DEBUG] mod_openzap.c:1556 got FXO sig 1:4 [STOP]

2010-07-01 11:07:21.166199 [DEBUG] switch_channel.c:2261 (OpenZAP/1:4/) Callstate Change RINGING -> HANGUP

2010-07-01 11:07:21.166199 [NOTICE] mod_openzap.c:1577 Hangup OpenZAP/1:4/ [CS_CONSUME_MEDIA] [NONE]

2010-07-01 11:07:21.166199 [DEBUG] switch_channel.c:2277 Send signal OpenZAP/1:4/ [KILL]

2010-07-01 11:07:21.166199 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK]

2010-07-01 11:07:21.166199 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_HANGUP

2010-07-01 11:07:21.166199 [DEBUG] zap_io.c:1388 channel done 1:4

2010-07-01 11:07:21.166199 [DEBUG] ozmod_analog.c:778 ANALOG CHANNEL 1:4 thread ended.

2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:500 (OpenZAP/1:4/) State HANGUP

2010-07-01 11:07:21.168191 [WARNING] mod_openzap.c:517 VETO Changing state on 1:4 from DOWN to HANGUP

2010-07-01 11:07:21.168191 [DEBUG] mod_openzap.c:556 OpenZAP/1:4/ CHANNEL HANGUP

2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:46 OpenZAP/1:4/ Standard HANGUP, cause: NONE

2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:500 (OpenZAP/1:4/) State HANGUP going to sleep

2010-07-01 11:07:21.168191 [DEBUG] switch_ivr_originate.c:3369 Originate Resulted in Error Cause: 19 [NO_ANSWER]

2010-07-01 11:07:21.168191 [INFO] mod_dptools.c:2382 Originate Failed. Cause: NO_ANSWER

2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:333 (OpenZAP/1:4/) State Change CS_HANGUP -> CS_REPORTING

2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK]

2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_REPORTING

2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:591 (OpenZAP/1:4/) State REPORTING

2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:53 OpenZAP/1:4/ Standard REPORTING, cause: NONE

2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:591 (OpenZAP/1:4/) State REPORTING going to sleep

2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:327 (OpenZAP/1:4/) State Change CS_REPORTING -> CS_DESTROY

2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK]

2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1175 Session 71 (OpenZAP/1:4/) Locked, Waiting on external entities

2010-07-01 11:07:21.168191 [NOTICE] switch_core_session.c:1193 Session 71 (OpenZAP/1:4/) Ended

2010-07-01 11:07:21.168191 [NOTICE] switch_core_session.c:1195 Close Channel OpenZAP/1:4/ [CS_DESTROY]

But no ringing is heard.

thanks,

-:D

On Thu, Jul 1, 2010 at 2:49 PM, <freeswitch-users-request@lists.freeswitch.org> wrote:

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When replying, please edit your Subject line so it is more specific
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Today's Topics:

1. Re: Default Dial Plan: action application bridge (Deya M)

---------- Forwarded message ----------
From: Deya M <deya787@gmail.com>
To: freeswitch-users@lists.freeswitch.org
Date: Thu, 1 Jul 2010 14:49:26 +1000
Subject: Re: [Freeswitch-users] Default Dial Plan: action application bridge

I generated samples, and started from scratch.

I also made a copy of the conf after installation, rm it, and copied the original conf before testing!

-:D

On Thu, Jul 1, 2010 at 1:37 PM, <freeswitch-users-request@lists.freeswitch.org> wrote:

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freeswitch-users@lists.freeswitch.org

To subscribe or unsubscribe via the World Wide Web, visit
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
freeswitch-users-request@lists.freeswitch.org

You can reach the person managing the list at
freeswitch-users-owner@lists.freeswitch.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-users digest..."

Today's Topics:

1. Re: Query !! freeswitch, dingaling, jingle, skypopen
(Anthony Minessale)
2. continue_on_fail and hangup_after_bridge with transfer
(Vladimir Elizarov)
3. Re: SIP header on only one fork of a bridge (Tim St. Pierre)
4. Default Dial Plan: action application bridge (Deya M)
5. Re: Default Dial Plan: action application bridge
(Anthony Minessale)

---------- Forwarded message ----------
From: Anthony Minessale <anthony.minessale@gmail.com>
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 30 Jun 2010 17:40:27 -0500
Subject: Re: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, skypopen
If he didn't like to joke, he'd never survive in telephony.

On Wed, Jun 30, 2010 at 3:39 PM, Sameer Khan <sameer2k3t@gmail.com> wrote:

no problem. but did't expect from the creator of mod_skypopen, the wonder i would say

On Wed, Jun 30, 2010 at 9:19 PM, Giovanni Maruzzelli <gmaruzz@celliax.org> wrote:

On Wed, Jun 30, 2010 at 6:02 PM, Sameer Khan <sameer2k3t@gmail.com> wrote:
> Mr Giovani what was that ?

just jokin, nevermind

--
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900

---------- Forwarded message ----------
From: Vladimir Elizarov <xengelpublicx@gmail.com>
To: freeswitch-users@lists.freeswitch.org
Date: Thu, 1 Jul 2010 03:29:09 +0400
Subject: [Freeswitch-users] continue_on_fail and hangup_after_bridge with transfer
I'm trying to make a dialplan:

expression="^(^([0-9]{10})$|^([0-9]{11})$|^([0-9]{12})$|^\+([0-9]{11})$)$">

data="hangup_after_bridge=true"/>

application="set"data="continue_on_fail=NO_ROUTE_DESTINATION"/>

break="on-true">

data="dtmf=WWWWWWWWWWWWWWWWWWWWW111222#WWWWWW89$2#@100"/>

data="sofia/gateway/gw1/89$2|sofia/gateway/gw2/89$2|sofia/gateway/gw3/89$2"/>

data="{execute_on_answer=send_dtmf\s${dtmf}}sofia/internal/${real_dialed_number}@192.168.50.53:5061"/>

The logic of his work: if unavailable gateway to the next. if not
available all the gateway to go to the extension transfer.
A problem in If we get a code busy here, it is not satisfied
hangup_after_bridge. The call goes to the transfer. Why is this so?

trace:
http://pastebin.freeswitch.org/13244

--
Best regards, Vladimir Elizarov

---------- Forwarded message ----------
From: "Tim St. Pierre" <fs-list@communicatefreely.net>
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 30 Jun 2010 22:45:19 -0400
Subject: Re: [Freeswitch-users] SIP header on only one fork of a bridge
Thanks all,

I read that bit, but didn't fully understand it's implications.

I'll give that a go.

-Tim

Steven Ayre wrote:
>
> data="{this_is_global=true}[this_is_gw1_only=true]sofia/gateway/gw1/$1|[this_is_gw2_only=true]sofia/gateway/gw2/$1"/>
>
>
>
> On 28 June 2010 19:45, Tim St. Pierre <fs-list@communicatefreely.net
> fs-list@communicatefreely.net>> wrote:
>
> Hello list,
>
> I would like to bridge a call to multiple SIP endpoints, but add
> different headers to each.
>
> I'm not entirely sure how to do this. I have no problem exporting a
> SIP header that does what I
> want for one destination, but I'm not sure how to set it for two.
>
> My application is that I want two IP phones to ring - one with the
> internal ring-ring splash, the
> others with a group-answer (single ring, then lamp flash only), for
> administrative assistants, etc.
>
> How do I export different variables to each branch?
>
> Thanks!
>
> -Tim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users@lists.freeswitch.org
> FreeSWITCH-users@lists.freeswitch.org>
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>
>
>
> ------------------------------------------------------------------------
>
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> http://www.freeswitch.org

---------- Forwarded message ----------
From: Deya M <deya787@gmail.com>
To: freeswitch-users@lists.freeswitch.org
Date: Thu, 1 Jul 2010 06:19:51 +0300
Subject: [Freeswitch-users] Default Dial Plan: action application bridge

Hi,

In the default dial plan, with two extensions defined, 1000 and 1001, when I call from 1000 to 1001, I always get Voicemail, using the default config files: conf/dialplan/default.xml

I changed the following from :

TO

and it did work! Not sure if the first / default one,   should work, but something is missing ?

New:



















































Old:



















































:-D

---------- Forwarded message ----------
From: Anthony Minessale <anthony.minessale@gmail.com>
To: freeswitch-users@lists.freeswitch.org
Date: Wed, 30 Jun 2010 22:37:15 -0500
Subject: Re: [Freeswitch-users] Default Dial Plan: action application bridge
you must have changed more than you think.

you might want to diff you configs against the in-tree ones.

On Wed, Jun 30, 2010 at 10:19 PM, Deya M <deya787@gmail.com> wrote:

Hi,

In the default dial plan, with two extensions defined, 1000 and 1001, when I call from 1000 to 1001, I always get Voicemail, using the default config files: conf/dialplan/default.xml

I changed the following from :

TO

and it did work! Not sure if the first / default one,   should work, but something is missing ?

New:



















































Old:



















































:-D

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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900

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