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Kareem.Hamdy at trustv...
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 Posted: Wed Jul 15, 2009 10:31 am Post subject: [Freeswitch-users] Freeswitch-users Digest, Vol 37, Issue 90 |
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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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of freeswitch-users-request@lists.freeswitch.org
Sent: Wednesday, July 15, 2009 1:06 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Freeswitch-users Digest, Vol 37, Issue 90
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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@freeswitch.org>
Subject: Re: [Freeswitch-users] OpenZAP and FreeSWITCH w/ Sangoma
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<87f2f3b90907141724q2735fac1jdacea3994db62782@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@trustvesta.com>wrote:
| Quote: | 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
| Quote: |
[span wanpipe T2]
trunk_type => t1
b-channel => 2:1-24
|
set up like span 1 example
| Quote: |
----
#================================================
# 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
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------------------------------
Message: 2
Date: Tue, 14 Jul 2009 19:20:19 -0700
From: DigiLord <digilord@me.com>
Subject: [Freeswitch-users] GXW4104 & FreeSwitch
To: freeswitch-users@lists.freeswitch.org
Message-ID: <8DC39E34-A395-42D5-B299-070605A2DCEE@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@freeswitch.org>
Subject: Re: [Freeswitch-users] GXW4104 & FreeSwitch
To: freeswitch-users@lists.freeswitch.org
Message-ID: <085EE9F4-A513-45FD-89E9-C66A0BE3715F@freeswitch.org>
Content-Type: text/plain; charset="us-ascii"
On Jul 14, 2009, at 9:20 PM, DigiLord wrote:
| Quote: | 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?
| Quote: | 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?
| Quote: |
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
| Quote: |
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 <shaheryarkh@googlemail.com>
Subject: [Freeswitch-users] SIP Trace Option at Runtime
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<b16156850907142119s26dd5e2cu3cdd416926b9217b@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@hotmail.com
Email: shaheryarkh@googlemail.com
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------------------------------
Message: 5
Date: Wed, 15 Jul 2009 14:32:25 +1000
From: Jason White <jason@jasonjgw.net>
Subject: Re: [Freeswitch-users] SIP Trace Option at Runtime
To: freeswitch-users@lists.freeswitch.org
Message-ID: <20090715043225.GA21117@jdc.jasonjgw.net>
Content-Type: text/plain; charset=us-ascii
Muhammad Shahzad <shaheryarkh@googlemail.com> wrote:
| Quote: | 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@gmail.com>
Subject: Re: [Freeswitch-users] Get voicemail messages
To: "freeswitch-users@lists.freeswitch.org"
<freeswitch-users@lists.freeswitch.org>
Message-ID: <4A5D5FC3.4050701@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@gmail.com> wrote:
| Quote: | | Quote: | 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@132.64.3.86] has been answered
2009-07-15 07:44:51.933664 [WARNING] mod_voicemail.c:2072 Can't find
user [80671@132.64.3.86]
2009-07-15 07:44:52.533435 [NOTICE] switch_core_state_machine.c:179
Hangup sofia/internal/80671@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@132.64.3.86) Ended
2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1087 Close
Channel sofia/internal/80671@132.64.3.86 [CS_DESTROY]
------------------------------
Message: 7
Date: Wed, 15 Jul 2009 16:05:24 +0800
From: Brad Tuan <brad.tuan@gmail.com>
Subject: [Freeswitch-users] How to set the IVR of VM menu??
To: freeswitch-users <freeswitch-users@lists.freeswitch.org>
Message-ID:
<f2179ba0907150105gab17105t19cedc58040d6a7b@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
How to set the date format , and the IVR flow ........??
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brian at freeswitch.org
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