[Freeswitch-users] create a external FXO Trunk - A200

Thu Jul 1 00:23:32 PDT 2010

I may be wrong, but it looks like it is not detecting the dial tone, and
therefore not dailing the number. 

You should check that connection to the
line is correct, and that the tones are correctly configured for the
carrier you use. 

François 

On Thu, 1 Jul 2010 16:16:36 +1000, Deya M 
wrote:   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, 
wrote:
 Send FreeSWITCH-users mailing list submissions to

freeswitch-users at lists.freeswitch.org [2]

 To subscribe or unsubscribe via
the World Wide Web, visit

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [3]
 or, via
email, send a message with subject or body 'help' to

freeswitch-users-request at lists.freeswitch.org [4]

 You can reach the
person managing the list at
 freeswitch-users-owner at lists.freeswitch.org
[5]

 When replying, please edit your Subject line so it is more specific

than "Re: Contents of FreeSWITCH-users digest..."

Today's Topics:

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

---------- Forwarded
message ----------
From: Deya M 
To: freeswitch-users at lists.freeswitch.org
[7]
 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,  wrote:
 Send FreeSWITCH-users mailing list submissions
to
 freeswitch-users at lists.freeswitch.org [9]

 To subscribe or unsubscribe
via the World Wide Web, visit

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [10]
 or, via
email, send a message with subject or body 'help' to

freeswitch-users-request at lists.freeswitch.org [11]

 You can reach the
person managing the list at
 freeswitch-users-owner at lists.freeswitch.org
[12]

 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 
To:
freeswitch-users at lists.freeswitch.org [14]
 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  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  wrote:

On Wed, Jun 30, 2010 at 6:02 PM, Sameer Khan  wrote:
 > Mr Giovani what
was that ?

 just jokin, nevermind

 --
 Sincerely,

 Giovanni Maruzzelli

Cell : +39-347-2665618

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-- 
Anthony Minessale II

FreeSWITCH
http://www.freeswitch.org/ [26]
ClueCon http://www.cluecon.com/ [27]

Twitter: http://twitter.com/FreeSWITCH_wire [28]

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[32]
googletalk:conf+888 at conference.freeswitch.org [33]
 pstn:+19193869900

---------- Forwarded message ----------
From: Vladimir Elizarov 
To:
freeswitch-users at lists.freeswitch.org [35]
 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_dtmfs${dtmf}}sofia/internal/${real_dialed_number}@192.168.50.53:5061
[36]"/>

 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 [37]

 --
 Best regards,
Vladimir Elizarov

---------- Forwarded message ----------
From: "Tim St.
Pierre" 
To: freeswitch-users at lists.freeswitch.org [39]
 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 at communicatefreely.net [41]>> 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 at lists.freeswitch.org [42]
 >
FreeSWITCH-users at lists.freeswitch.org [43]>
 >
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [44]
 >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[45]
 > http://www.freeswitch.org [46]
 >
 >
 >
 >
------------------------------------------------------------------------

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---------- Forwarded message
----------
From: Deya M 
To: freeswitch-users at lists.freeswitch.org [52]

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 
To:
freeswitch-users at lists.freeswitch.org [54]
 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  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

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ClueCon http://www.cluecon.com/ [61]

Twitter: http://twitter.com/FreeSWITCH_wire [62]

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[65] #freeswitch

FreeSWITCH Developer
Conference
sip:888 at conference.freeswitch.org
[66]
googletalk:conf+888 at conference.freeswitch.org [67]
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