Hi,
We have a server with asterisk 1.4. We are upgrading to asterisk 12.4. Is
there a way to install 12.4 on the same machine? At any point we will only
run either 1.4 or 12.4.
Thank you,
Deepak
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but will there be any issues due to upgradation of dependencies? Is there a
way to install dependencies of asterisk 12.4 at different path so they
don't conflict with 1.4 dependencies?
On Mon, Sep 22, 2014 at 9:55 PM, Carlos Chavez cur...@telecomabmex.com
wrote:
On 9/22/14, 5:03 AM, Deepak
601sip:601@111.118.185.107;tag=209a8aa9
Regards
Deepak Bhatia
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Hello,
I have two sip phones (zoiper). Earlier these used to communicate using the
settings below for sip.conf and extensions.conf and now we asterisk
1.8.29.0, so these phones have stopped communicating. My question is that
does 1.8.29.0 release require any more changes to be done to the
://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
-- SIP/100-0015 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/101-0014' status is 'CONGESTION'
Regards
Deepak Bhatia
Software Consultant
Voxomos Systems Pvt. Limited
Mobile
Hi,
I am upgrading from Asterisk 1.4 to 12.4. I am able to authenticate the
user and call AgentLogin. But after that when I call AgentRequest I keep
getting Agent '1234' is busy.
If I put a delay of 5 second or more before calling AgentRequest then it
works most of the times. Here's my dialplan:
the agent logs
in for the first time. He/she is not supposed to be on any call. Is there
any command to find our where the agent is busy?
On Tue, Aug 12, 2014 at 9:27 PM, Richard Mudgett rmudg...@digium.com
wrote:
On Tue, Aug 12, 2014 at 1:33 AM, Deepak Rawat deepaksingh.ra...@gmail.com
wrote
On Mon, Aug 11, 2014 at 3:29 AM, Deepak Rawat deepaksingh.ra...@gmail.com
wrote:
Hi,
I modified the query in res/res_config_odbc.c.
Original: SELECT MAX(LENGTH(var_val)) FROM %s WHERE filename='%s'
Modified: SELECT MAX(LEN(var_val)) FROM %s WHERE filename='%s'
I rebuilt the code
backslash_is_escape
setting in res_odbc.conf maybe we can provide a database_name setting and
drive the query using this setting. If I get time, I will submit a patch
for review.
On Mon, Aug 11, 2014 at 8:01 AM, Matthew Jordan mjor...@digium.com wrote:
On Sun, Aug 10, 2014 at 5:02 PM, Deepak Rawat
deepaksingh.ra
, Matthew Jordan wrote:
On Sun, Aug 10, 2014 at 5:02 PM, Deepak Rawat
deepaksingh.ra...@gmail.com wrote:
On Mon, Aug 11, 2014 at 3:29 AM, Deepak Rawat
deepaksingh.ra...@gmail.com
wrote:
Hi,
I modified the query in res/res_config_odbc.c.
Original: SELECT MAX(LENGTH(var_val)) FROM %s
11 (I tried
compiling it in Asterisk 11 and it failed).
I may not be able to use DiaStar or i6net's VXI etc as the hardware is a low
end appliance with limited resources.
Would anybody be able to suggest how to go about it?
Warm Regards,
Deepak
Hi, we are experiencing a strange issue and I am hoping someone can point me
to the right direction or help out with some pointers.
We have asterisk 1.6.0.6 with a sangome a104DE card. We have basically 4
T1's for a total of DAHDI 96 channels.
We have an agi application (php) that acts as a kind
such a limit or what the default
is?
Thanks
On Thu, Jun 4, 2009 at 12:31 PM, David Backeberg dbackeb...@gmail.comwrote:
On Thu, Jun 4, 2009 at 12:15 PM, Deepak dlal...@gmail.com wrote:
Hi, we are experiencing a strange issue and I am hoping someone can point
me
to the right direction or help
BTW,we are using an ODBC connection to Microsoft SQL Server.
We are not using MySQL.
Would that be a possible cause?
Thanks
On Thu, Jun 4, 2009 at 12:31 PM, David Backeberg dbackeb...@gmail.comwrote:
On Thu, Jun 4, 2009 at 12:15 PM, Deepak dlal...@gmail.com wrote:
Hi, we are experiencing
...
Thanks in advance
On Thu, Jun 4, 2009 at 3:23 PM, Steve Edwards asterisk@sedwards.comwrote:
On Thu, 4 Jun 2009, Deepak wrote:
BTW,we are using an ODBC connection to Microsoft SQL Server. We are not
using MySQL.
Would that be a possible cause?
Probably not the cause, unless
I am trying to understand this from a CPU performance perspective.
We have SIP Phones/Asterisk using G729 codec. The calls are being forwarded
over T1 to the PSTN.
Thanks
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Hi, I am running asterisk 1.6.0.5 with a Sangoma A104DE (4 port T1 with Echo
Cancellation).
We are using DAHDI.
When I do a top, I see asterisk using up 460MB of VM which is huge compared
to Asterisk system not using the card.
We also notice a constant decrease in available VM memory size on
...@drogon.net wrote:
On Wed, 20 May 2009, Deepak wrote:
Hi, I am running asterisk 1.6.0.5 with a Sangoma A104DE (4 port T1 with
Echo
Cancellation).
We are using DAHDI.
When I do a top, I see asterisk using up 460MB of VM which is huge
compared
to Asterisk system not using
Hi,
Can someone please help to resolve the followinng issue:
We would like an asterisk user to call a number and when the called party
picks up the phone, we play a message (press 1 to accept call, 2 to reject
call). Only when the called party presses 1, do we bridge the call and the
two parties
Hi, I am in a predicament and any help/pointers would be appreciated.
We are using chanspy to listen in on conversations. We are doing this via a
web interface. The web interface lists all the ongoing calls. We click on a
call and then my local phone rings allowing me to spy on the session I
Hi, I require some urgent advice/help concerning chanspy.
We have a web interface that lists all bridged SIP calls (show channels
concise) When we click on the call, our local extension rings. When we pick
up to spy, parties involved in the call that we are spying on are suddenly
unable to hear
of fedora.
http://docs.fedoraproject.org/selinux-faq-fc3/
--
Deepak
[EMAIL PROTECTED] wrote:
Hi Dave,
I did make clean and then make. But then when I am giving make install its
giving error AVC access denied.
I am using Fedora.
What may be the problem?
Help me..
Thanking you,
Preeta Pandey
I use TE212P, it shoudl work without errors.
I use it with Asterisk 1.2.18 + zaptel-1.2.17.1
On RHEL 4.4
On Dell PowerEdge 850
It may be that the card is bad, try contacting Asterisk support.
I had one bad card when I first got it, the 2nd one worked .
--
Deepak
Jerry Geis [EMAIL
was just barking
... ha ha ha...
--
Deepak
Michael J. Liberatore [EMAIL PROTECTED] wrote: Hi all, i have been
using asterisk for a few years but i am about to do my first t1 setup. After
terrible quality issues between two business locations, we have decided to
purchase a point
how ur line feeds are setup. I just wanted to let u know that
there can be aproblem with transfer if u have multiple calls comming on same
line display.
Or, may be I am wrong in understanding ur email.
--
Deepak
Russell Brown [EMAIL PROTECTED] wrote:
Does anyone have any suggestions
Hi, I have a production asterisk-1.2.8 system with FreePBX PRI Digium card.
I am looking for a paging system to an external speaker. I can page to
internal Polycom 501 VoIP.
But, what hardware or system do I need to integrate with the asterisk to have
this acheived.
--
Deepak
Linux your
Thanks Jared, Yes I am using with Asterisk only. So I am using the inbuilt
music from Asterisk for onhold.
--
Deepak
Jared Smith [EMAIL PROTECTED] wrote: On Tue, 2007-07-31 at 06:36 +0100,
Deepak Naidu wrote:
I think we need to pay for the later, but I am not sure if we need to
pay
I am using TE212P with asterisk-1.2.18. It has echo DTMF in hardware to
support. I use it on Dell Power Edge 85 no IRQ's ...
Ya, just make sure that u get a good card I got the a broken card first time
which ddnt work for echo cancellor then RMA'ed it with new one.
--
Deepak
.
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, but the hardware DTMF didnt work,
so I wrote a mail, to the developer of the drivers he said they are still
working in the lab probably have one within a week.
Stephen Bosch [EMAIL PROTECTED] wrote:
Deepak Naidu wrote:
Hi,
I have a Dell Power Edge server planning yo buy Sangoma A101D
card
to get some
notes from user with custom install setup when used with
Asterisk+freepbx+Sangoma.
Also how do I enable DTMF hardware detection.
--
Deepak
Linux your Life, Don't Window it [[]]
{ All for the best }
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threashold value for DTMF(this is for hardware DTMF).
--
Deepak
Noah Miller [EMAIL PROTECTED] wrote:
i am using tdm400P in my office. i tested that TDMF generated by asterisk
is so bad. the sound is very soft and quality is so bad. i am using
asterisk 1.2.18. most of time, the # key
I am not sure what exactly you wish to achieve. Just a basic SIP--to--SIP call
or ?
I am not much into the configs, but ya I can tell you that you can try using
FreePBX or Trixbox kind of setup to write ur Asterisk config file, rather then
u editing them, as it has macros, context etc...
second line ie Ext 8555.
This is what I need, if I can dow it with Follow me, then how, if through ring
group how.
Deepak Naidu [EMAIL PROTECTED] wrote: Hi,
I ma using Asterisk 1.2.18 FreePBX 2.2.1. I have assigned every
users in office with Polycom with 2 extensions as below
555
8555
,DIRECTDIAL)
exten = ${VM_PREFIX}555,n,Hangup
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this is because we have 30+ queues and 150+ agents.
I will try it out again and debug this in the future.
Thanks and Regards,
Deepak
Tzafrir Cohen wrote:
On Fri, Jun 22, 2007 at 10:28:32AM +0530, Deepak Bhat wrote:
Yes I was aware of the MAX_INCLUDE_LEVEL define. Just wasnt sure about
The best person to check with is Digium support. They have support matrix for
Kernel hardware on which ur card will perform.
Please check the compatibility matrix. Should work fine with
http://www.digium.com/en/supportcenter/documentation/viewdocs/TE120P
Digium support. 256-428-6000
: Maaximum include level exceeded : 10*
Has anyone encoutered this before and does anyone know what it means ??
Any help will be deeply appreciated as I have been unable to find any
documentation on this.
Thanks
-Deepak
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!
Tzafrir Cohen wrote:
On Thu, Jun 21, 2007 at 12:35:30PM +0530, Deepak Bhat wrote:
Hi all,
I am using asterisk version 1.2.18.
I recently tried to change my asterisk configuration by using #include
statements to include other config files in my extensions.conf and
queues.conf files.
My
will try out your suggestions in the future when I need to make changes.
Will let you know of my findings then.
Thanks for your help.
Regards,
Deepak
Tzafrir Cohen wrote:
On Thu, Jun 21, 2007 at 04:07:03PM +0530, Deepak Bhat wrote:
Im sure its not a circular include.
Like you said its mostly
So, I am not sure whether its a zaptel issue. It have TE212P card which has
echo based hardware cancellor.
--
Deepak
Deepak Naidu [EMAIL PROTECTED] wrote: Hi,
I have Asterisk-1.2.18 install with FreePBX more than 75 extnsion,
daily I come accross an issue try resolving them its
are having trouble with DTMF detection, you can relax the DTMF
; detection parameters. Relaxing them may make the DTMF detector more likely
; to have talkoff where DTMF is detected when it shouldn't be.
;
relaxdtmf=yes
---
Deepak
it
--
Deepak
Deepak Naidu [EMAIL PROTECTED] wrote:
Hi, I was wondering if we can check the voicemails remotely from a cell or a
landline number.
We have SIP 3 Digit Extensions connected to Asterisk server.
If users are away from Desk need to access voicemails can they dial in to
Asterisk
through web link even have mailed. Aslo I have checked
regarding DISA, but I am not kind of OK in using DISA now for just voicemails.
Is their any other ways. I am using Free PBX so can I do any thing from
FreePBX to manager it, if not backend configs are fine.
--
Deepak
phones which I dial no
echo. So ya dont know whats wrong.
Thanks all for your inputs sharing ur experience.
--
Deepak
Darryl Dunkin [EMAIL PROTECTED] wrote:
This should only be for TDM to TDM calls, SIP to SIP calls don't use the
zaptel driver
possibility of Asterisk failing to cancel the echo.
OK, one question here howz the call flow when a SIP---SIP call is established
ie. is the connection between 2 phones when an Internal call is made or does
the SIP call goes via Asterisk once the SIP--SIP call is establised.
--
Deepak
Matthew
We are planning to buy a wifi SIP phones to work with Asterisk 1.2-18 setup. I
would like to get feedback views regarding Linksys WIP300 WIFI IP Phone or
any other wifi phones which has been stable.
Thanx for any updates.
--
Deepak
-
The all
this card, which you can
install and check what exactly the settings were before.
This is an entie new setup by me, the old one was using 1.4 build I am
using 1.2 build both are different server.
--
Deepak
Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Once upon a time I used to have
I blame on any network. B'cos for sure
we have a spegati of networks no QoS. Also the intresting thing is if I call
from one extension to other dialing the main line then extension the call is
crystal clear. but when dialing a direct extension its a hell of echo.
--
Deepak
Stephen
Yeah I have made sure its the correct port. We have 75 polycoms currently.
? the SIP-to-SIP echo is there.
--
Deepak
Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote:
Deepak Naidu wrote:
Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to
PRI we were till
reinvite is disabled. Also its a Dell PowerEdge 850 server running asterisk
connected to a Cisco switch. other network in company have Cisco Switch.
Also we have approx 75 Polycoms all over.
canreinvite=no
--
Deepak
Steve Totaro [EMAIL PROTECTED] wrote:
v
resolve this mess. Feels bad when one does best in aggregating
things some louzy device screws up... Oh my frustation is comming on mail :
--
Deepak
C F [EMAIL PROTECTED] wrote:
Are the config files you are using with the phones what was meant with
that firmware? or did you upgrade
through this issue.
--
Deepak
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pridialplan=national
signalling=pri_cpe
faxdetect=incoming
usecallerid=yes
echocancel=yes
callerid=asreceived
echocancelwhenbridged=no
echotraining=128
;rxgain=-3.0
;txgain=-7.0
group=0
channel=1-23
--
Deepak
Alex Balashov [EMAIL PROTECTED] wrote: On Sat, 9 Jun 2007, Deepak Naidu wrote:
But now when we
then parse for the IP address(use
grep /or awk).
--
Deepak
Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
An answer to your original question: if you can get someone _to_ the
phones, on the polycom 30x's you would hold the VolDn, VolUp, DND, and
Hold buttons for a while to reboot
: Setting yellow alarm on span 1
SPAN 1: Primary Sync Source
VPM400: Not Present
VPM450: Not Present
Completed startup!
wct2xxp: Clearing yellow alarm on span 1
Zaptel Transcoder support loaded
Has any one had this issue with RHEL4-Update 4. Please let me know your views.
--
Deepak
I think the best way is to conact Digium Hardware support. it seems there may
be an IRQ problem.
--
Deepak
Francois Deppierraz [EMAIL PROTECTED] wrote:
Hi,
I'm trying to get a TE212 working on a Dell PowerEdge 1850 running
Debian etch using the latest release of libpri (1.4.0), zaptel
Asterisk 1.2.18 latest version of zaptel drivers.
Hope if someone had the same issue, I what has done to resolve it would be
much appreciable.
--
Deepak
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Yahoo! Answers - Got a question? Someone out there knows the answer. Tryit now
So Steven, did the echo problem stopped once the Hardware echo cancellor card
was installed out of the box, or you needed to do some configuration changes
like Rx Tx etc.
Thanks for sharing your experience.
--
Deepak
»Steven Ringwald« [EMAIL PROTECTED] wrote:
Deepak Naidu wrote
A small way to make little easy, I dont know it people are ok to that, try
integrating freepbx asterisk so you know what the sip configs should look
like when things are all well.
Things might stop working if there is a bug or change in configs.
--
Deepak
Ken Williams [EMAIL
I have Vista on my new HP laptop X-lite soft phone works like charm with it,
I tried sjphone, I couldnt get that working, its gets hung.
--
Deepak
Chris Bagnall [EMAIL PROTECTED] wrote:
Greetings list,
I've noticed over the last couple of weeks that, unsurprisingly, nearly every
new
Can anyone help on this.
--
Deepak
Deepak Naidu [EMAIL PROTECTED] wrote:
Hi,
I am in the process of planning a dial plan, In regards to the
requirement, I am confused how to go about the dial plan.
The scenario is like below.
BRANCH - A - (COMPANY
(extension 700)
to accept support calls vice-versa, I dont know how this is possible what
would my dial plans be.
It would be much appreciated if someone can help me resolve this dial plan
support issue.
Thannks,
Deepak
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Hi friends !
I want to add stun functionality in asterisk.
can anybody give me some hint that how can i start that.
thanks in advance
Deepak Dhiman
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in not listening to ser or ser is not
forwarding to asterisk.
Thanks
Deepak Dhiman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: Thursday, April 28, 2005 12:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
, but
can assure abt the ser that is is running well and also forwarding
packets to asterisk server but * is not getting these packets. Can
anybody tell me that what`s wrong with my Asterisk server? Do I need to
change /add something in sip.conf? Please help me .
Regards,
Deepak Dhiman
in not listening to ser or ser is not
forwarding to asterisk.
Thanks
Deepak Dhiman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: Thursday, April 28, 2005 12:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
, but can assure abt
the ser that is is running well and also forwarding packets to asterisk
server but * is not getting these packets. Can anybody tell me that what`s
wrong with my Asterisk server? Do I need to change /add something in
sip.conf? Please help me .
Regards,
Deepak Dhiman
Software
, but can assure abt
the ser that is is running well and also forwarding packets to asterisk
server but * is not getting these packets. Can anybody tell me that what`s
wrong with my Asterisk server? Do I need to change /add something in
sip.conf? Please help me .
Regards,
Deepak Dhiman
Software
, but can assure abt
the ser that is is running well and also forwarding packets to asterisk
server but * is not getting these packets. Can anybody tell me that what`s
wrong with my Asterisk server? Do I need to change /add something in
sip.conf? Please help me .
Regards,
Deepak Dhiman
Software
, but can assure abt
the ser that is is running well and also forwarding packets to asterisk
server but * is not getting these packets. Can anybody tell me that what`s
wrong with my Asterisk server? Do I need to change /add something in
sip.conf? Please help me .
Regards,
Deepak Dhiman
Software
that already have worked well.
thanks
Deepak Dhiman
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hi friends !
my asterisk is giving one error while running.
it says that unable to write audio data, module codec_speex.so is not
loaded.
have anybody face this kind of problem than plz tell me the solution.
thanks
Deepak Dhiman
___
Asterisk-Users
or that kind of facility is
given in zapata.conf.
tell me in detail abt the configurations of the sip.conf and
extensions.conf.
thanks
Deepak Dhiman
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Hi Bacon
Thanks for the quick response.
Actually I want to confirm that whether it is possible to divide logical
channels into group just like physiacl channels in zapata.
Deepak Dhiman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent
wrong.
Thanks
Deepak Malhotra
This message was sent using IMP, the Internet Messaging Program.
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Hello
I setup Mediatrix 1124, I am able to make incoming call but unable to make
outgoing call. When ever I tried it just gave me a beep sound.
I appreciate any help on this.
Thanks
Deepak Malhotra
This message was sent
Hello
I setup Mediatrix 1124, I am able to make incoming call but unable to make
outgoing calls. When ever I tried it just gave me a beep sound.
I appreciate any help on this.
Thanks
Deepak Malhotra
This message was sent
Hello
I am getting message "Call not Approved" while
using xlite SIP phone. Phone is registered correctly and I am able to get
incoming call but on out going calls I am getting "Call Not Approved" message.
Please through some id
Use only g729 and not g729a. I tied it and it works greate with Granstream
Phone.
- Original Message -
From: Jefferson Carvalho [EMAIL PROTECTED]
To: Digium/ Asterisk-Users [EMAIL PROTECTED]
Sent: Saturday, October 16, 2004 3:27 AM
Subject: [Asterisk-Users] G729 and Sipura.
Hello All,
the call.
Please help me to fingure out this
issue.
Thanks
Deepak
Extension.conf :
exten =
_9NXX,1,ChanIsAvail(${TRUNK})exten =
_9NXX,2,NoOP,${AVAILCHAN}exten =
_9NXX,3,Cut(TheChannel=AVAILCHAN,,1)exten =
_9NXX,4,NoOP,${TheChannel}exten =
_9NXX,5,Dial(${TheChannel}/${EXTEN
I faced this issue when i was using April code of Asterisk, after upgrading
to latest and greatest this problem goes away.
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 07, 2004 7:01 AM
Subject: Re: [Asterisk-Users] TDM FXO port
and Asterisk. I will appreciate if you tell me the way to talk to Citel
Link using Hyper Terminal.
Thanks
Deepak
version of Asterisk updated on June 24th.
Thanks
Deepak
but it again appear
after some time.
Thanks
Deepak
This message was sent using IMP, the Internet Messaging Program.
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[EMAIL PROTECTED]
http
but it again appear
after some time.
Thanks
Deepak
This message was sent using IMP, the Internet Messaging Program.
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[EMAIL PROTECTED]
http
but it again appear
after some time.
Thanks
Deepak
This message was sent using IMP, the Internet Messaging Program.
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http
Hello
I am trying to setup Directory service for incoming
call but it is not working. As per the document on voip-info.org, * should exist
from directory but that is not happening.
Is it a bug or I am doing something
wrong?
Please help.
Thanks
Deepak
HelloI
have an interesting situaltion and not sure if I am doing something wrong
orit is a BUG. I Installed Rhino Channel on T1 line and connected Analog
Phone onRhino's Zap Channels.When I pickup analog phone and
hangup without dialing anynumber , I am getting extra ring after hangup and
I will keep in mind, but still no solution.
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 03, 2004 1:13 AM
Subject: Re: [Asterisk-Users] (no subject)
USE SUBJECTS!!!
On Wed, Jun 02, 2004 at 06:18:20PM -0700, Deepak Malhotra wrote:
Hello
:14:06 DEBUG[1234379840]: chan_zap.c:1118
update_conf: Updated conferencing on 1, with 0 conference
users -- Hungup 'Zap/1-1'
Thanks in advance for any help.
Regards
Deepak
:06 DEBUG[1234379840]: chan_zap.c:1118 update_conf: Updated
conferencing on 1, with 0 conference users
-- Hungup 'Zap/1-1'
Thanks in advance for any help.
Regards
Deepak
This message was sent using IMP, the Internet Messaging
to
configure soft Phone defined in PBX200 to dial out side using PBX300 Zap
devices.
I am geting error message " Rejected connect
attempt from PBX200".
Please help if this is possible.
Thanks
Deepak
help if this is possible.
Thanks
Deepak
This message was sent using IMP, the Internet Messaging Program.
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to configure soft Phone defined
in PBX200 to dial out side using PBX300 Zap devices.
I am geting error message Rejected connect attempt from PBX200.
Please help if this is possible.
Thanks
Deepak
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idea.
Thanks
Deepak
SoftPhone.
Any working examples of configuration files is highly appreciated.
I mentoned followin lines in /etc/zaptel.conf file.
fxsks=1
fxoks=2
Thanks
Deepak
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lines in /etc/zaptel.conf file.
fxsks=1
fxoks=2
Thanks
Deepak
This message was sent using IMP, the Internet Messaging Program.
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but there is no voice
exchange ( i can't hear any think except ring). Here is the portion of sip.conf
and extensions.conf.
Let me know if i missed something.
Thanks
Deepak
sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP
Tony
Are you able to make this configuration work with 2 sip phone on same Asterisk
server? I am also trying to do the same using xlite softphone abailable on
www.xten.com site.
Please let me know wgat you did?
Thanks
Deepak
Quoting Tony [EMAIL PROTECTED]:
On Sun, 2004-05-09 at 18:51, [EMAIL
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