[Freeswitch-users] Freeswitch-users Digest, Vol 37, Issue 90
Kareem Hamdy
Kareem.Hamdy at trustvesta.com
Wed Jul 15 08:29:49 PDT 2009
Thanks Michael, but I'm setting up a T1, not a PRI. I should be able to use all 24 channels.
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of freeswitch-users-request at lists.freeswitch.org
Sent: Wednesday, July 15, 2009 1:06 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Freeswitch-users Digest, Vol 37, Issue 90
Send Freeswitch-users mailing list submissions to
freeswitch-users at 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 at lists.freeswitch.org
You can reach the person managing the list at
freeswitch-users-owner at 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: 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 <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] OpenZAP and FreeSWITCH w/ Sangoma
To: freeswitch-users at lists.freeswitch.org
Message-ID:
<87f2f3b90907141724q2735fac1jdacea3994db62782 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
See inline comments
On Tue, Jul 14, 2009 at 5:04 PM, Kareem Hamdy
<Kareem.Hamdy at trustvesta.com>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
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/da16d89e/attachment-0001.html
------------------------------
Message: 2
Date: Tue, 14 Jul 2009 19:20:19 -0700
From: DigiLord <digilord at me.com>
Subject: [Freeswitch-users] GXW4104 & FreeSwitch
To: freeswitch-users at lists.freeswitch.org
Message-ID: <8DC39E34-A395-42D5-B299-070605A2DCEE at me.com>
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 <brian at freeswitch.org>
Subject: Re: [Freeswitch-users] GXW4104 & FreeSwitch
To: freeswitch-users at lists.freeswitch.org
Message-ID: <085EE9F4-A513-45FD-89E9-C66A0BE3715F at freeswitch.org>
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/eed9a9d7/attachment-0001.html
------------------------------
Message: 4
Date: Wed, 15 Jul 2009 10:19:48 +0600
From: Muhammad Shahzad <shaheryarkh at googlemail.com>
Subject: [Freeswitch-users] SIP Trace Option at Runtime
To: freeswitch-users at lists.freeswitch.org
Message-ID:
<b16156850907142119s26dd5e2cu3cdd416926b9217b at mail.gmail.com>
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: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/ca7ce7b2/attachment-0001.html
------------------------------
Message: 5
Date: Wed, 15 Jul 2009 14:32:25 +1000
From: Jason White <jason at jasonjgw.net>
Subject: Re: [Freeswitch-users] SIP Trace Option at Runtime
To: freeswitch-users at lists.freeswitch.org
Message-ID: <20090715043225.GA21117 at jdc.jasonjgw.net>
Content-Type: text/plain; charset=us-ascii
Muhammad Shahzad <shaheryarkh at googlemail.com> 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 <elihayun at gmail.com>
Subject: Re: [Freeswitch-users] Get voicemail messages
To: "freeswitch-users at lists.freeswitch.org"
<freeswitch-users at lists.freeswitch.org>
Message-ID: <4A5D5FC3.4050701 at savion.huji.ac.il>
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
<elihayun at gmail.com> 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/80671 at 132.64.3.86] has been answered
2009-07-15 07:44:51.933664 [WARNING] mod_voicemail.c:2072 Can't find
user [80671 at 132.64.3.86]
2009-07-15 07:44:52.533435 [NOTICE] switch_core_state_machine.c:179
Hangup sofia/internal/80671 at 132.64.3.86 [CS_EXECUTE] [NORMAL_CLEARING]
2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1085 Session 3
(sofia/internal/80671 at 132.64.3.86) Ended
2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1087 Close
Channel sofia/internal/80671 at 132.64.3.86 [CS_DESTROY]
------------------------------
Message: 7
Date: Wed, 15 Jul 2009 16:05:24 +0800
From: Brad Tuan <brad.tuan at gmail.com>
Subject: [Freeswitch-users] How to set the IVR of VM menu??
To: freeswitch-users <freeswitch-users at lists.freeswitch.org>
Message-ID:
<f2179ba0907150105gab17105t19cedc58040d6a7b at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
How to set the date format , and the IVR flow ........??
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/c35eaf18/attachment.html
------------------------------
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
End of Freeswitch-users Digest, Vol 37, Issue 90
************************************************
More information about the FreeSWITCH-users
mailing list