What's with the errors on the voicemail sound files?
-MC

On Thu, Jun 30, 2011 at 9:12 PM, Max Alex <max.asterisk@gmail.com> wrote:
Hi,
Can any one provide their suggestions and help for this issue,
I really need to resolve and get this working.

Thanks,
Max Alex
Voip Developer



On Tue, Jun 28, 2011 at 6:01 PM, Max Alex <max.asterisk@gmail.com> wrote:
Hi,
Thanks for your reply.
I have enabled logger as per your help.
I have given completed log on following link of pastebin.
http://pastebin.freeswitch.org/16616

You can see this line as it is showing answered and this call is answered on my cell phone but on softphone it is ringing and i have rejected from there.
2011-06-28 17:47:52.909657 [NOTICE] mod_freetdm.c:1953 Channel [FreeTDM/2:1/01234567890] has been answered    

It is pre answering the cell phone when it is ringing on phone
Waiting for your help.

Thanks,
Max Alex
Voip Developer



On Mon, Jun 27, 2011 at 9:15 PM, Michael Collins <msc@freeswitch.org> wrote:
Get a complete, unedited, unabridged console debug log w/ siptrace and put it on pastebin w/ "FreeSWITCH Log" for the syntax highlighting. Use "sofia global siptrace on" to make sure you can all SIP traffic.

-MC


On Mon, Jun 27, 2011 at 6:05 AM, Max Alex <max.asterisk@gmail.com> wrote:
Hi,
Thanks for reply,
I have tried the same way and reloaded freeswitch, but still it is answered on first ring of the call.
When it is ringing the call on 1001 and the same time it is answered on cell phone, so something is done when it is ringing on 1001.

Here is the dialplan for the same

    <extension name="Local_Extension">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
        <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>-->
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=30"/>
          <action application="set" data="hangup_after_bridge=true"/>
         <action application="set" data="continue_on_fail=true"/>
        <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
        <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>

       <action application="bridge" data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/>
       <action application="sleep" data="1000"/>
        <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>


Please help me for this issue.


Thanks,
Max Alex
Voip Developer



On Fri, Jun 24, 2011 at 11:38 AM, Michael Collins <msc@freeswitch.org> wrote:
Are you using the default dialplan? I think you might just need to ignore early media on your bridge app. If you are using the default.xml file then locate "Local_Extension" and the bridge line:

    <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>

Change it to this, then try again:

    <action application="bridge" data="{ignore_early_media=true}user/${dialed_extension}@${domain_name}"/>

If I understand correctly, the "symptom" you are experiencing is the normal operation of the bridge app (and it's cousin, the originate API command). When the b-leg sends back media, including ringing, the bridge (or the originate) will be considered "successful," and in the case of bridge, the b-leg's audio (the early media) will be connected to the a-leg. If you set ignore_early_media=true then you are explicitly telling the bridge app that you only want to connect the b-leg to the a-leg if the b-leg actually answers.

I hope that made sense...

-MC



On Thu, Jun 23, 2011 at 9:32 PM, Max Alex <max.asterisk@gmail.com> wrote:
Hi,
Thanks for reply.
Current scenario is below.

PSTN call -> sangoma device -> freeswitch incoming context -> routed to 1001 -> ringing (Answered on cellphone)
Here when it is routed to 1001 the call it is started ringing, but on phone that call is answered and hearding sound of ringing.

Required flow:
PSTN call -> sangoma device -> freeswitch incoming context -> routed to 1001 -> ringing (Ringing on cellphone)

I hope it is understable, the call should not be answered until 1001 answer it, right not when it is started ring it is answered on cell phone.
that should not be happen as it is not answered yet.

Waiting for your reply.


Thanks,
Max Alex
Voip Developer



On Fri, Jun 24, 2011 at 5:48 AM, Michael Collins <msc@freeswitch.org> wrote:
I'm not sure I understand the problem. What is happening vs. what you believe should be happening?
-MC


On Thu, Jun 23, 2011 at 3:31 AM, Max Alex <max.asterisk@gmail.com> wrote:
Hi,
Thanks for your reply.
Here is my configuration and log
http://pastebin.freeswitch.org/16571

I am using A200 analog sangoma device with freeswitch, it is working fine but when it is routing call to 1001 then it is answered.
Please provider your suggestions.

Thanks,
Max Alex
Voip Developer




On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <msc@freeswitch.org> wrote:
I thought the A200 was an analog card? Maybe I have my numbers mixed up...

Go ahead and collect a debug log of this call. It might help to have your configs posted as well. Use pastebin.freeswitch.org. See this wiki article for tips on how to collect information:

-MC

On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <max.asterisk@gmail.com> wrote:
Hi,
I have installed freeswitch latest version with sangoma card A200 as well,
I have installed and configured freetdm module with wanpipe drivers for freeswitch,
We are properly receiving the inbound calls in public context and then we are routing that call to 1001 extension,
it is properly routing to 1001 as well, but we have one issue while routing on 1001.

Here is the issue description.
I am calling from my cell phone to that DID number of pri line, and then it will start ringing on 1001 extension,
When 1001 extension start ringing the call is answered on my cell phone,
it is something like freeswitch preanswer or autoanswer the call, how can i stop this answer call when it is ringing on 1001 extension,
Waiting for good reply.

Thanks,
Max Alex



_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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



_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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



_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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



_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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



_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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



_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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



_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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



_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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




_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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