Hi, thank you very munch for reply, this is working fine, when we configure <condition destination_number="^1003" <action application="bridge" data="sofia/gateway/gatewayname/$1"/> but in my scenario, i dont want to route call based on extensions( eg, here 1003) routing, i just want to route the calls when the destination domain is defferan from local domain, example: INVITE packet from registered extension to sipserver like this. From: 1000@gw.proxy.com:5060 To : 1003@gateway.com:5060 here from uri and to uri is different. any help Srinivas On Fri, Feb 26, 2010 at 5:44 AM, < freeswitch-users-request@lists.freeswitch.org> wrote:

Send FreeSWITCH-users mailing list submissions to 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: Retrieving voicemail without entering user ID (extension) (Joseph Puchalski) 2. Freeswitch/Openzap dials out on disconnected FXO line (Robert Hadley)

---------- Forwarded message ---------- From: Joseph Puchalski <joseph.puchalski@personalcyberspace.com> To: "freeswitch-users@lists.freeswitch.org" < freeswitch-users@lists.freeswitch.org> Date: Thu, 25 Feb 2010 23:31:18 +0000 Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user ID (extension)

Lars,

Thanks!

Joe

*From:* Lars Zeb [mailto:larclap@yahoo.com] *Sent:* Wednesday, February 24, 2010 10:09 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Retrieving voicemail without entering user ID (extension)

Joe,

I used the extension below, but I think that Brian said it was too insecure. Being a total beginner, I removed the condition.

<extension name="Local_Extension_Abe">

<condition field="destination_number" expression="^(101[0-9])$">

<action application="set" data="dialed_ext=$1"/>

<action application="export" data="dialed_ext=$1"/>

</condition>

<condition field="destination_number" expression="^${caller_id_number}$">

<action application="set" data="voicemail_authorized=${sip_authorized}"/>

<action application="answer"/>

<action application="sleep" data="1000"/>

<action application="voicemail" data="check default $${domain} ${dialed_ext}"/>

<anti-action application="bind_meta_app" data="1 a a execute_extension::dx XML features"/>

<anti-action application="bind_meta_app" data="2 a a record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>

<anti-action application="bind_meta_app" data="3 a a execute_extension::cf XML features"/>

<anti-action application="set" data="transfer_ringback=${us-ring}"/>

<anti-action application="set" data="call_timeout=15"/>

<anti-action application="set" data="sip_exclude_contact=${network_addr}"/>

<anti-action application="set" data="hangup_after_bridge=true"/>

<!--<anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,BUSY,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->

<anti-action application="set" data="continue_on_fail=true"/>

<anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/>

<anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>

<anti-action application="bridge" data="user/${dialed_ext}@ $${domain}"/>

<anti-action application="answer"/>

<anti-action application="sleep" data="1000"/>

<anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/>

</condition>

</extension>

Lars

*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Rupa Schomaker *Sent:* Wednesday, February 24, 2010 4:01 PM *To:* freeswitch-users *Subject:* Re: [Freeswitch-users] Retrieving voicemail without entering user ID (extension)

Look at the end of:

http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail

Advisable? With it enabled, I can walk up to anyone's phone and retrieve their VM w/out authentication. It was removed on purpose due to that reason as far as I remember.

On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski < joseph.puchalski@personalcyberspace.com> wrote:

I’m trying to modify my dialplan so that I can press a single button on my phone, be connected to voicemail, and enter only a password to gain access.

Currently I use a programmable key to dial 4000. I am prompted for my ID, and then password.

I’ve poked around “mod voicemail” on the wiki and searched the mailing list and web, but haven’t found enough info. I have discovered that this behavior seems to have been available in previous versions of the default dialplan.

Is it still possible? Is it advisable? Was this feature/behavior removed for security reasons?

I apologize ahead of time if the answer is somewhere in plain sight that I haven’t looked yet. If so, I’d much appreciate being pointed in the right direction.

As always, thanks for any help,

Joe P.

_______________________________________________ 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

-- -Rupa

---------- Forwarded message ---------- From: "Robert Hadley" <robert.hadley@teotech.com> To: <freeswitch-users@lists.freeswitch.org> Date: Thu, 25 Feb 2010 16:13:10 -0800 Subject: [Freeswitch-users] Freeswitch/Openzap dials out on disconnected FXO line

When dialing out, Freeswitch/Openzap is not detecting that an analog FXO channel is disconnected and tries dialing out on the channel anyway. No error is reported. The call doesn’t timeout until a minute later. Shouldn’t Freeswitch/Openzap skip over a disconnected channel to the next connected channel?

I have configured a Sangoma A200 FXO card as a FXO span.

[span wanpipe FXO]

name => PSTN Line 1

number => 4253491059

fxo-channel => 2:3

name => PSTN Line 2

number => 4253491058

fxo-channel => 2:4

The wanpipe driver does detect and report when a CO line is connected or disconnected (in /var/log/messages), and Freeswitch/Openzap gets an event as reported in the log.

/var/log/messages: Feb 25 15:23:10 roberth-c53 kernel: wanpipe2: Module 3: FXO Line is disconnected!

FS_CLI: 2010-02-25 15:23:10.711604 [DEBUG] ozmod_analog.c:788 EVENT [ALARM_TRAP][3:1] STATE [DOWN]

/var/log/messages: Feb 25 15:23:44 roberth-c53 kernel: wanpipe2: Module 4: FXO Line is connected!

FS_CLI: 2010-02-25 15:23:44.901979 [DEBUG] ozmod_analog.c:788 EVENT [ALARM_CLEAR][3:2] STATE [DOWN]

I have the dialplan configured to use the next available port in the FXO span (there will be more than 2 channels later).

<extension name="OutgoingFXO">

<condition field="destination_number" expression="^7(\d+)$">

<action application="bridge" data="openzap/FXO/a/$1"/>

</condition>

</extension>

Here is a portion of the log when that shows dialing out on a disconnected analog FXO channel.

EXECUTE sofia/internal/5410@192.168.72.45:5060bridge(openzap/FXO/a/93491045)

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1257 Connect outbound channel OpenZAP/3:1/93491045

2010-02-25 15:26:17.891443 [NOTICE] switch_channel.c:642 New Channel OpenZAP/3:1/93491045 [3c8f46f5-77a8-498f-a51c-015837746cb7]

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1269 (OpenZAP/3:1/93491045) State Change CS_NEW -> CS_INIT

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:59 Changing state on 3:1 from DOWN to DIALING

2010-02-25 15:26:17.891443 [WARNING] switch_core_session.c:486 OpenZAP/3:1/93491045 does not support the proxy feature, disabling.

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread starting.

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:450 Executing state handler on 3:1 for DIALING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/3:1/93491045) Running State Change CS_INIT

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/3:1/93491045) State INIT

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:394 (OpenZAP/3:1/93491045) State Change CS_INIT -> CS_ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/3:1/93491045) State INIT going to sleep

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/3:1/93491045) Running State Change CS_ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/3:1/93491045) State ROUTING

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:417 OpenZAP/3:1/93491045 CHANNEL ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_ivr_originate.c:66 (OpenZAP/3:1/93491045) State Change CS_ROUTING -> CS_CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/3:1/93491045) State ROUTING going to sleep

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/3:1/93491045) Running State Change CS_CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/3:1/93491045) State CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/3:1/93491045) State CONSUME_MEDIA going to sleep

Thanks,

Robert

_______________________________________________ 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

-- Srinivasula Reddy K