Displaying 20 results from an estimated 8000 matches similar to: "How to set Name/username to something like 229/john instead of 229/229"
2009 Jan 19
1
Need help registering Cisco 7960 Phones on Asterisk
Hi everyone,
I googled this followed the instructions, but it hasn't work for me yet.
I have universal setting in SIPDefault.cnf and phone specific settings in
SIPXXXXXXXXXX.cnf. But it doesn't get registered.
I need to register it on two different asterisk boxes. So my
SIPXXXXXXXXXX.cnf looks like this:
phone_label: "Zeeshan A Zakaria"
line1_name: "523"
2006 Nov 01
0
AW: Which IP phones have best voice quality, preferably under $150
snom 300 :">
CS
-----Urspr?ngliche Nachricht-----
Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von Kristian Kielhofner
Gesendet: Mittwoch, 1. November 2006 12:01
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Which IP phones have best voice quality,preferably under $150
Zeeshan Zakaria
2006 Nov 01
0
[SPAM HEADER] - Which IP phones have best voice quality, preferably under $150 - Email found in subject
I'd recommend any of the following, which are all in your price range
Snom 300
Polycom IP430
Polycom IP501
Aastra 9112i
Linksys SPA-922
Grandstream GXP-2000
Cory Andrews
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Zeeshan
Zakaria
Sent: Wednesday, November 01, 2006 11:17 AM
To: Asterisk
2010 Jul 20
0
asterisk-users Digest, Vol 72, Issue 49
sorry for typo mistake in my last post. as from my orignal post two
registration of the same user are as follows
SIP/XYZ at 119.68.0.90:5060
SIP/XYZ at 202.16.34.10:5678
so dial command with unique-id i want to use will be
Dial(SIP/XYZ at 192.168.0.20:5062-096afee8,30,rtT)
Dial(SIP/XYZ at 192.168.0.12:64290-0966ab80,30,rtT)
and not
Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)
2007 Mar 30
0
forwarding loop not detected
Asterisk 1.2.16
I have an extension "102" with a Polycom 430
I am trying to protect against forwarding loops
If I set the phone to forward the line to itself, extension 102 I get
the following
-- Got SIP response 302 "Moved Temporarily" back from 206.83.240.18
-- Now forwarding Local/102@mycontext-b2ee,2 to
'Local/102@mycontext' (thanks to
2010 Oct 13
0
innomedia ATA's
We are testing the innomedia ATA's to possibly replace our current line up
of ATA's that we are using. Has anyone used their product? What is their
track record on stability, voice quality, DTMF talkoff, T.38
Thanks
Bryant
----------------------------------------
From: "Zeeshan Zakaria" <zishanov at gmail.com>
Sent: Wednesday, October 13, 2010 10:41 AM
To:
2020 Aug 06
2
Is it possible to use Stasis to control both legs of a Local channel created using ARI?
I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis.
I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter.
[mycontext]
exten =>
2007 May 18
5
Phone losing IP address for a few seconds but doesn't drop call
Hi,
Recently I've noticed on a customer's GXP-2000 phone that it loses its IP
addresse for a few seconds, audio goes blank obviously, and after about
30-60 seconds get the same IP addresse back and resumes the call. This shows
that call was not dropped but phone lost connection with the server, whereas
the caller on the other end was still talking. This is just unacceptable as
this is
2006 Nov 06
1
Audio goes one way during the call for a fewseconds. Is it RTP, NAT, dyndns, or what it is?
We had very similar problems to this which drove us insane for ages.
Basically we use VoIP trunks (SIP) for all our inbound + outbound calls.
Call quality was good however we would get random problems where people
could not hear us or us hear them for about 5-10 seconds at a time.
After weeks of trying to get to the bottom of the problem it appeared
our VoIP trunk provider (sentiro/sip2go) had
2009 Jul 11
2
Suggestions for web based soft phones
For a while now I've been looking for a good web based soft phone solution,
but so far no luck. A few solutions which I've tried, both Java based and
Flash based, either don't work, or had bad sound quality. I need something
which I could put on my productions server for my clients.
Seems like good web based solutions are all paid ones, nobody is giving it
for free. Any ideas,
2010 Oct 18
5
Same extension registering over eth0 and eth1
Hello list,
I need to know how to deal with a redundant network with only one asterisk
server, which is receiving registrations from the end points on both of its
ethernet ports. This means extension 201 is registering both from eth0 and
from eth1.
Is there a way/software which can act as a middle man between asterisk and
the ethernet ports, and by default sends registrations to asterisk only
2006 Nov 29
0
Something similar or better than HUD Pro?
Is there something similar or better than HUD pro out there for asterisk
PBX. HUD pro is wonderful thing, but they require complete Fonality product
to be purchased first, and don't sell it as a stand alone product. If
someone is not interested in Fonality product but is ready to purchase some
good interface like HUD, what are the options?
--
Zeeshan A Zakaria
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2010 Oct 20
4
Recommendation for a new server
Hello list,
What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and
a not much busy website, i.e. getting 500-1000 hits a day.
Thanks,
Zeeshan A Zakaria
--
www.ilovetovoip.com
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2014 Jan 31
2
callfiles.call
hello list,
i have created a callfiles with my asterisk 1.4.43 like:
Channel: SIP/watara/06xxxxxxxx
MaxRetries: 10
RetryTime: 5
WaitTime: 20
Context: mycontext
Extension: s
Priority: 1
extensions.conf
mycontext
exten => s,1,Ringing()
exten => s,n,Playback(hello-world)
exten => s,n,Dial(SIP/105)
exten => s,n,Hangup()
it works with one number how can i do in order to create a
2010 Oct 23
3
Cepstral voice quality not good
Hello list,
I have been using Cepstral's 8KHz voices for my text-to-speech service for
some time now, and have been noticing that the voice quality is really poor,
doesn't matter what phrase I give it to convert. None of the other 8KHz
voices I have ever used were this bad. It doesn't seem good enough system to
be used in a commercial system. Is there any better quality text-to-voice
2010 Mar 18
6
Asterisk DIES with no trace. PLEASE
Thanks Zeeshan,
SAngoma told me that the asterisk problem is unrelated to wanpipe drivers,
they told me to reinstall asterisk again
But, i still having doubts about the problem :(
Thanks in advance
>
> Message: 10
> Date: Thu, 18 Mar 2010 00:21:06 -0400
> From: Zeeshan Zakaria <zishanov at gmail.com>
> Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE
>
2009 Jul 14
3
Why CDR is recording dst value = h?
For a new project, I have written a dialplan and it is pretty straight
forward: The [dialout] context dials out a number, and h extension in this
context writes the CDR. But what is happening is that if the callee hangs up
first, all values in the CDR are fine, but if the caller hangs up first, the
'dst' column is always 'h'. I put a NoOp right in the begining of this macro
to
2004 Jul 17
1
voicemail broadcast feature
Using CVS from 7/12/04 and trying to get the voicemail broadcast feature
to work.
Voicemail.conf has
[mycontext]
3722 => 1234,BroadCast Test,,,cc=*@mycontext
.
then many other voicemail boxes.
-----
whenever I leave voicemail at box 3722, only box 3722 gets the
voicemail. It is not expanding it to other voicemail boxes in the
[mycontext] context.
Even if I replace the cc= line with
2005 Sep 08
0
How to cascade dial status back through IAX
On machine A I have something like the following in extensions.conf:
[iax-extensions]
exten => _9.,1,Dial(IAX2/machineB/${EXTEN:1}@mycontext)
exten => _9.,2,NoOp(DIALSTATUS=${DIALSTATUS})
exten => _9.,3,Hangup
On machineB I have something like this:
[mycontext]
exten => 2002,1,Dial(SIP/2002,60)
exten => 2002,2,NoOp(DIALSTATUS=${DIALSTATUS})
exten => 2002,3,Hangup
If I use a
2011 Jan 21
4
Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?
Hi list,
For a client I am setting up a system which will use T1 PRI from Primus, who
offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only
used switchtypes euroISDN and National. Although the documentation says
Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you
have used these protocols on an Asterisk box and if there were any things to
consider. If