Thanks Michael, but I'm setting up a T1, not a PRI.  I should be able to use 
all 24 channels.
 


-----Original Message-----
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Sent: Wednesday, July 15, 2009 1:06 AM
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Subject: Freeswitch-users Digest, Vol 37, Issue 90

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Today's Topics:

   1. Re: OpenZAP and FreeSWITCH w/ Sangoma (Michael Collins)
   2. GXW4104 & FreeSwitch (DigiLord)
   3. Re: GXW4104 & FreeSwitch (Brian West)
   4. SIP Trace Option at Runtime (Muhammad Shahzad)
   5. Re: SIP Trace Option at Runtime (Jason White)
   6. Re: Get voicemail messages (Eli Hayun)
   7. How to set the IVR of VM menu?? (Brad Tuan)


----------------------------------------------------------------------

Message: 1
Date: Tue, 14 Jul 2009 17:24:18 -0700
From: Michael Collins <[email protected]>
Subject: Re: [Freeswitch-users] OpenZAP and FreeSWITCH w/ Sangoma
To: [email protected]
Message-ID:
        <[email protected]>
Content-Type: text/plain; charset="iso-8859-1"

See inline comments

On Tue, Jul 14, 2009 at 5:04 PM, Kareem Hamdy
<[email protected]>wrote:

> Hello:
>
> I would like to connect a Sangoma T1 card to FreeSWITCH.  Most of the docs
> I see pertain to a PRI.  When I leave out the d-chan notation, I get errors
> regarding not able to get the d-chan up and running in the CLI.
>
> Here's my info:
>
> [span wanpipe T1]
> trunk_type => t1
> b-channel => 1:1-24


b-channel => 1:1-23
d-channel => 1:24

>
>
> [span wanpipe T2]
> trunk_type => t1
> b-channel => 2:1-24
>

set up like span 1 example

>
> ----
>
> #================================================
> # WANPIPE1 Configuration File
> #================================================
> #
> # Date: Wed Dec  6 20:29:03 UTC 2006
> #
> # Note: This file was generated automatically
> #       by /usr/local/sbin/setup-sangoma program.
> #
> #       If you want to edit this file, it is
> #       recommended that you use wancfg program
> #       to do so.
> #================================================
> # Sangoma Technologies Inc.
> #================================================
>
> [devices]
> wanpipe1 = WAN_AFT_TE1, Comment
>
> [interfaces]
> w1g1 = wanpipe1, , TDM_VOICE_API, Comment
>
> [wanpipe1]
> CARD_TYPE       = AFT
> S514CPU         = A
> CommPort        = PRI
> AUTO_PCISLOT    = NO
> PCISLOT         = 1
> PCIBUS          = 6
> FE_MEDIA        = T1
> FE_LCODE        = B8ZS
> FE_FRAME        = ESF
> FE_LINE         = 1
> TE_CLOCK        = NORMAL
> TE_REF_CLOCK    = 0
>
> TE_HIGHIMPEDANCE        = NO
> LBO             = 0DB
> FE_TXTRISTATE   = NO
> MTU             = 1500
> UDPPORT         = 9000
> TTL             = 255
> IGNORE_FRONT_END = NO
> TDMV_SPAN       = 1
> TDMV_DCHAN      = 0


TDMV_DCAHN = 24


>
> TDMV_HW_DTMF    = YES
> TDMV_HW_FAX_DETECT = NO
>
> [w1g1]
> ACTIVE_CH       = ALL
> TDMV_HWEC       = YES
> MTU             = 80
>
>
> ---
>
>
>
> In freeSWITCH's openzap.conf.xml, I've tried pri_span as well as analog.
>
> I cannot find a straight up T1 wiki anywhere.  Would someone please provide
> an example?
>
>
> Thanks,
> Kareem
>
> _______________________________________________
> Freeswitch-users mailing list
> [email protected]
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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Message: 2
Date: Tue, 14 Jul 2009 19:20:19 -0700
From: DigiLord <[email protected]>
Subject: [Freeswitch-users] GXW4104 & FreeSwitch
To: [email protected]
Message-ID: <[email protected]>
Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes

Hello all,
        I am getting my feet wet with FreeSwitch by migrating my Asterisk box  
over.  I have run into a few things that I am not sure how to  
accomplish.

        I have a Grandstream GXW4104 with one analog line connected.  I have  
it connected and I am able to receive calls on my Polycom 501 (ext  
2101) that is registered to the FreeSwitch server.  The one problem is  
that CallerID is not the CallerID from the caller, it's the CallerID  
from the Grandstream device (ext 2100).

        On the same device there is HORRIBLE echo.  I have set echo  
cancellation on the device to enabled and disabled to no avail.  Under  
Asterisk there was no echo.

        I setup the device as a provider.  Was that the right way to  
accomplish connecting this device to FS?

        Is there a way to enable sending an e-mail containing my voicemail  
messages like Asterisk does?

Thanks in advance for any help you can give!

Dan



------------------------------

Message: 3
Date: Tue, 14 Jul 2009 21:31:23 -0500
From: Brian West <[email protected]>
Subject: Re: [Freeswitch-users] GXW4104 & FreeSwitch
To: [email protected]
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"


On Jul 14, 2009, at 9:20 PM, DigiLord wrote:

> Hello all,
>       I am getting my feet wet with FreeSwitch by migrating my Asterisk box
> over.  I have run into a few things that I am not sure how to
> accomplish.
>
>       I have a Grandstream GXW4104 with one analog line connected.  I have
> it connected and I am able to receive calls on my Polycom 501 (ext
> 2101) that is registered to the FreeSwitch server.  The one problem is
> that CallerID is not the CallerID from the caller, it's the CallerID
> from the Grandstream device (ext 2100).

How is the callerid passed on this device?

> On the same device there is HORRIBLE echo.  I have set echo
> cancellation on the device to enebled and disabled to no avail.  Under
> Asterisk there was no echo.

If it didn't have echo on asterisk it shouldn't have echo on  
FreeSWITCH, Can you describe the echo better? Are you using speaker  
phone?  What codecs?

>
>
>       I setup the device as a provider.  Was that the right way to
> accomplish connecting this device to FS?
>
>       Is there a way to enable sending an e-mail containing my voicemail
> messages like Asterisk does?

Yes check the mod_voicemail page on the wiki.

/b


>
>
> Thanks in advance for any help you can give!
>
> Dan

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Message: 4
Date: Wed, 15 Jul 2009 10:19:48 +0600
From: Muhammad Shahzad <[email protected]>
Subject: [Freeswitch-users] SIP Trace Option at Runtime
To: [email protected]
Message-ID:
        <[email protected]>
Content-Type: text/plain; charset="utf-8"

Hi,

Is there any CLI command to enable  / disable SIP packet trace at runtime. I
do know an option in SIP profile which enables / disable SIP trace but it to
apply it i have reload mod_sofia, which at many times fail due to a running
call.

Thank you.


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: [email protected]
Email: [email protected]
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------------------------------

Message: 5
Date: Wed, 15 Jul 2009 14:32:25 +1000
From: Jason White <[email protected]>
Subject: Re: [Freeswitch-users] SIP Trace Option at Runtime
To: [email protected]
Message-ID: <[email protected]>
Content-Type: text/plain; charset=us-ascii

Muhammad Shahzad <[email protected]> wrote:
> Is there any CLI command to enable  / disable SIP packet trace at runtime. 

sofia profile <profilename> siptrace on
sofia profile <profilename> siptrace off

sofia help would have answered your question.




------------------------------

Message: 6
Date: Wed, 15 Jul 2009 07:49:07 +0300
From: Eli Hayun <[email protected]>
Subject: Re: [Freeswitch-users] Get voicemail messages
To: "[email protected]"
        <[email protected]>
Message-ID: <[email protected]>
Content-Type: text/plain; charset=ISO-8859-1

did you bind your lua script to directory lookups in addition to the
dialplan? On Tue, Jul 14, 2009 at 7:02 AM, Eli Hayun
<[email protected]> wrote:

> > Hi
> > I am not using fixed xml files for the extension registration. I have
> > LUA script to return an XML string to FS.
> > Everything goes fine until I am trying to get the voice messages.
> > When  am entering my id, FS (or voicemail module) try to get the xml for
> > that id, but it cant find it. My lua script did NOT recieved any xml
> > request at that point.
> > What should I do to solve the problem.
> >
> > Thanks
> > Eli Hayun
>
>
>   
Yes I did bind it: my lua.conf.xml is like this

<configuration name="lua.conf" description="LUA Configuration">
  <settings>
    <param name="xml-handler-script" value="GenXml.lua"/>
    <param name="xml-handler-bindings" value="directory"/>
  </settings>
</configuration>


When an extension tried to register, I have no problem. But when I want
to use VoiceMail to retrieve my messeges, I got a problem.

Here is the partial log:

2009-07-15 07:44:49.373089 [INFO] mod_dialplan_xml.c:252 Processing
Phone2->*98 in context default
2009-07-15 07:44:49.386466 [NOTICE] mod_dptools.c:649 Channel
[sofia/internal/[email protected]] has been answered
2009-07-15 07:44:51.933664 [WARNING] mod_voicemail.c:2072 Can't find
user [[email protected]]
2009-07-15 07:44:52.533435 [NOTICE] switch_core_state_machine.c:179
Hangup sofia/internal/[email protected] [CS_EXECUTE] [NORMAL_CLEARING]
2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1085 Session 3
(sofia/internal/[email protected]) Ended
2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1087 Close
Channel sofia/internal/[email protected] [CS_DESTROY]






------------------------------

Message: 7
Date: Wed, 15 Jul 2009 16:05:24 +0800
From: Brad Tuan <[email protected]>
Subject: [Freeswitch-users] How to set the IVR of VM menu??
To: freeswitch-users <[email protected]>
Message-ID:
        <[email protected]>
Content-Type: text/plain; charset="iso-8859-1"

How to set the date format , and the IVR flow ........??
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