I am afraid i do not have a solution for you, but we also had this problem
occur, exactly the same. It happened overnight, with no changes to the server.
With help from our IAX provider, we did many tests, no solution, we then moved
to a SIP connection to our provider, problem solved.
Our *
Two Asterisk machines, different CVS, both say no
application MeetMe, show application does not show MeetMe, when I browse
to /asterisk/apps/ I notice that it is the only app that has not installed?
Do I need to install ZAPRTC first then try to install the
MeetMe application?
I do
We are on the lookout for a Firewall which is SIP aware, to
pass the voice stream to Asterisk.
We have looked at the Ingate Products, but they are very
expensive.
Can anyone point us to a well priced Enterprise SIP aware
Firewall?
SIP Phones - Firewall - Asterisk
Thanks
I have a problem with incoming
calls being recorded after a supervised transfer.
Incoming is CAPI BRI
- Asterisk - Supervised Transfer - SIP.
Call comes in, receptionist
answers, caller put on hold, Asterisk Monitor is recording, caller is on Hold,
Callee picks up the call, Asterisk
Is there a way to keep the incoming CallerID from the PSTN
and pass it onto the sip phone receiving the supervised call transfer?
The receptionist receives the PSTN callerID, performs a
supervised transfer, we get her local SIP callerID, not the original callers.
The main reason we
Hi,
I have the exact same problem, we have two Eicon DIVA Cards (BRI UK),
using chan_capi by Junghann.
The cards have been tested and work perfectly, if we make two outgoing
calls simultaneously, and someone calls us, they get a busy tone or call
failed, yet capi info says 2 channels are still
-digit presentation, and no ddi
numbers.
Best wishes,
John.
--- Craig Waddington [EMAIL PROTECTED] wrote:
Hi,
I have the exact same problem, we have two Eicon
DIVA Cards (BRI UK),
using chan_capi by Junghann.
The cards have been tested
Try this:
http://lists.digium.com/pipermail/asterisk-users/2004-March/039819.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ismaelg
Sent: 09 December 2004 12:07
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] A waning console error
Hello,
I
I have a problem with incoming
calls being recorded after a supervised transfer.
Call comes in, receptionist
answers, caller put on hold, Asterisk Monitor is recording, caller is on Hold,
Callee picks up the call, Asterisk Monitor Stops.
All recorded calls are named CallerID
to
It's the RTP Stream
Asterisk by default uses ports UDP 10,000 to 20,000
RTP = Audio
Open them on your firewall.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Aken
Sent: 07 December 2004 15:21
To: Asterisk Users Mailing List - Non-Commercial
Asterisk Monitor seems to be working fine. Though the
problem I am having is the two files (in out) muxing.
I added ,m to the string, yet the call records two files
still, and I get the resulting error, at the bottom.
monitor executing ( nice -n 19 soxmix
PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Craig Waddington
Verzonden: woensdag 1 december
2004 15:56
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users]
Asterisk Call Monitor and soxmix error
Asterisk Monitor seems to be working fine. Though the
problem I am having is the two files (in out
Have a strange problem.
2 different asterisk servers, running different CVS.
One behind Firewall, one not.
Cisco 7940 phones.
Over the past two weeks, users have had a problem with one
way audio, after about 2 minutes into a call, they can hear the other person,
but the other
Sorry forgot to mention this is with IAX2
only, SIP works fine.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington
Sent: 29 November 2004 10:46
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Audio
Drops out at Random - one way
Have a strange
Of WipeOut
Sent: 29 November 2004 13:01
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Audio Drops out at Random - one way
Craig Waddington wrote:
Have a strange problem.
2 different asterisk servers, running different CVS.
One behind Firewall, one
I am getting quite a few of these warnings lately, and audio
is sometimes dropping to one way.
Is this some way related? Latency to my IAX provider is
minimal, and no major packet loss.
Nov 29 15:04:00 WARNING[1095035200]: chan_iax2.c:1464
attempt_transmit: Max retries exceeded to
On say 2 out of 10 calls, when on a call, the Audio at our
end will drop for about 5 seconds, we can hear them, they cant hear us.
It doesnt happen every call, random, which is making
it very hard to trouble shoot, I am guessing it has something to do with RTP
stream?
Nothing has
Recently our provider had an issue, so we couldnt make
VOIP calls.
We currently have a BRI which we use for incoming calls, at
the moment I have the below in my dialplan, so if our VOIP provider or our
internet drops, the outgoing calls are sent through the ISDN Bri.
The problem is,
I have a 1751 with a BRI Wic, I would like it to pass
incoming calls to Asterisk.
After spending a lot of time on this, I cannot get it to
work. I can see the incoming call and the callerID, yet the router
doesnt seem to pass the call to asterisk.
In SIP.conf
[213.137.185.150]
I have a 1751 with a BRI Wic, I would like it to pass
incoming calls to Asterisk.
After spending a lot of time on this, I cannot get it to
work. I can see the incoming call and the callerID, yet the router doesnt
seem to pass the call to asterisk.
In SIP.conf
[213.137.185.150]
I cannot get incoming calls to sip phones behind a PIX to
work, outgoing is fine.
Asterisk (Public IP) Internet PIX (NAT) Sip Phones
I have tried no fixup protocol sip, I have punched a hole in
the Pix allowing anything from the Asterisk box into the network, still no
incoming.
message.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington
Sent: Saturday, September 25, 2004
8:17 AM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco
PIX and Asterisk
I cannot get incoming calls to sip phones behind a PIX to
work, outgoing is fine
Go into modules.conf
Comment out chan_modem.so=yes
Make it look like this:
[global]
chan_capi.so=yes
chan_modem.so=yes
;space here
Hope that helps
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 13 September 2004 21:22
To:
In my sip extensions context I have
include = parkedcalls
In extensions.conf I have
[parkedcalls]
Exten = 2000,1,Answer
In parking.conf I have the same.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Antkowiak
Sent: 29 June 2004 22:56
To:
I am also having the same problem. Latest CVS Latest Capi
When it does work and you pick up the phone, CAPI disconnects the call.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ePyron Felix
Deierlein
Sent: 28 June 2004 18:34
To: [EMAIL PROTECTED]
Thanks I will give that a try.
Looks like this may need a bug report? We are all getting the same
errors.
Outgoing is fine for me.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas
Anderson
Sent: 28 June 2004 23:26
To: [EMAIL PROTECTED]
Subject:
I am trying to have the ring types different for internal
and external incoming calls.
I have followed the guide on the wiki, the SetVar executes,
in extensions.conf I have it as s,1,
Yet it doesnt work?
When the phone rings, the ring type is the one I chose
on the phone, it rings
When I saw the update for Cisco Phone RTP issue I thought I would
try it.
Unfortunately chan_capi wont compile on this update.
Can anyone recommend a good * release for Capi, Bri ISDN and
Cisco 7940s SIP 6.3.
Or will CHAN_CAPI also be updated ?
Running Eicon Diva Bri Cards.
.html
- Original Message -
From: Craig Waddington [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 22, 2004 12:24 PM
Subject: [Asterisk-Users] Chan CAPI and Latest CVS wont compile
When I saw the update for Cisco Phone RTP issue I thought I would try
it.
Unfortunately
Voiptalk provide an excellent service and great support.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Potkin
Sent: 10 May 2004 23:51
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP in the UK
On Mon, May 10, 2004 at 08:58:23AM +0100,
I am using ISDN with CAPI and Eicon Diva card.
On ISDN calls in and out, some people are saying they find
it hard to hear us. Its only the odd few though, not everyone. We can hear them
no problem.
Do I just increase the txgain?
What is the limit for txgain, or are there any
Hi,
Currently using voiptalk.org and the quality is getting
really bad.
I would like a second provider preferably in UK,
anyone got any suggestions?
Ta.
PROTECTED]
Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider
On Wed, Apr 21, 2004 at 03:24:06PM +0100, Craig Waddington wrote:
Currently using voiptalk.org and the quality is getting really bad.
I would like a second provider preferably in UK, anyone got any
suggestions?
That's
. However,
this was down to a bug in asterisk
(http://bugs.digium.com/bug_view_page.php?bug_id=0001374) and cvs
updating fixed the problem.
Tan
Telappliant.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig
Waddington
Sent: 21 April 2004 15:38
To: [EMAIL
Hahahhaaa your right there Tan.
List, don't get me wrong, voiptalk are very good, service, support,
price, I am just having some issues which may be my end.
I was just wanting to try some iax providers out to see what worked best
for us.
Hopefully will get sorted.
-Original Message-
resolved it
-gcc
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington
Posted At: Friday, April 16, 2004
1:04 PM
Posted To: Asterisk User Group
Conversation: Cisco 7940 no audio
Subject: [Asterisk-Users] Cisco
7940 no audio
Kudos to the CAPI developers.
I would like to have multiple MSNs on my ISDN Bri
lines.
I see all the cool features but cannot find any examples or
guides to build from.
Currently running Diva Eicon Cards with CAPI from http://www.junghanns.net
I would like to route calls to sip
When we receive or make a call to the outside they can
hear us, but we cant hear them.
It may work 1 of 20 times. I have set canreinvite=no and
looked at many sites but cannot track down this problem.
Current setup:
Isdn Eicon Diva card / Capi - Asterisk network.
I have tried
Phone 20
accountcode=20
qualify=yes
context=sip
Thanks.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Cuthie
Sent: 16 April 2004 18:37
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7940 no audio
Craig Waddington wrote:
When we
, 2004, at 10:04 AM, Craig Waddington wrote:
When we receive or make a call to the outside - they can hear us, but
we cant hear them.
With SIP, missing audio is *usually* either a firewall or NAT issue.
Check firewall logs and make sure that you aren't seeing packets being
lost. Do you have
7940 no audio
On Fri, Apr 16, 2004 at 06:04:04PM +0100, Craig Waddington spake thusly:
When we receive or make a call to the outside - they can hear us, but
we
cant hear them.
I have had this problem several times and so far no resolution. However
for me it has always been with IAX. I have been
I am looking to buy a wireless sip phone, probably the
IPC5000, I have looked at Wisip phone and read tons of posts regarding that
phone.
Do any * admins have any feedback on this phone?
Is there any major differences between the phones, besides
looks?
The site has very limited
Thanks for the info. Sounds good.
Does that mean I can contact them for a test
unit also, to try before I buy?
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Michael Devenijn
Sent: 11 March 2004 18:25
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
What sort of phone are you trying to call?
I use SIPPS and * and it works fine, it just wont work when you call
Windows Messenger for some reason. I can call X-lite, POTS, GS phones no
problems.
I use the same config as X-lite, in SIPPS if you click on the spanner or
press F9 to go into
: [Asterisk-Users] Ahead SIPPS and Asterisk
Date: Fri, 5 Mar 2004 18:25:05 -
From: Craig Waddington [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
What sort of phone are you trying to call?
I use SIPPS and * and it works fine, it just wont work when you call
Windows
Nice one thanks for sharing, I look forward to it.
This will be very handy for SIP call transfers. At the moment I blindly
transfer on sip.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer
Sent: 05 March 2004 19:49
To: [EMAIL PROTECTED]
Title: Re: [Asterisk-Users] New Wiki page: Dimensioning an Asterisk system
Is the article correct in saying:
g729 codecs licenses
can be purchased for Asterisk (not for SCSI systems!)
I thought people had this working on SCSI
now?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Those phones look good, but, only have 10 milliwatt output.
Have you looked at these:
http://www.spectralink.com/products/nl-wts.html
100mw output.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Miguel
Cavazos
Sent: 15 February 2004 12:39
To:
SIP.CONF
[general]
; Codecs your choice
disallow=all
;allow=gsm
allow=ulaw
allow=alaw
;allow=ilbc
;allow=spx
allow=g723
allow=g729
register=1234:[EMAIL PROTECTED]/5000
[fwd.pulver.com]
type=friend
secret=password
username=1234
host=fwd.pulver.com
context=sip
You can add:
; Say Current Date and Time;exten = 13,1,DateTime()exten = 13,2,Wait(1)exten = 13,3,DateTime()exten = 13,4,Hangup
into exten.
maybe that helps
http//www.ntfs.org
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Deepakumar JV
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: 30 January 2004 16:39
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Music on Hold Warnings
On Friday 30 January 2004 04:33, Craig Waddington wrote:
1.Warning, flexibel rate not heavily tested
Hi.
I am having the following warning when using music on hold.
It works from X-Lite to Grandstream. I get a lot of errors
and warnings.
1.Warning, flexibel rate not heavily tested!
2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread:
Request to schedule in the past?!?!
Me too :(
100% CPU.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of F.G.Testa
Sent: 14 January 2004 20:01
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] 100% of cpu in an out of the box *
Hi all!
I'm newbie, so here goes my situation:
I have
Hi
I am attending the tutorial day, i am looking forward to it.
See you there.
Craig.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Klaus-Peter
Junghanns
Sent: 13 January 2004 10:31
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: [Asterisk-Users]
January 2004 10:35
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk behind NAT How to do it.
i like the idea of not requiring to open 1 ports
in the firewall.
Do i need to change rtf.conf to from 1 - 2 to
16384 and 16394.
thanks,
-B
- Original Message -
From: Craig
Customizing the Cisco SIP IP Phone Ring Types
The Cisco SIP IP phone ships with two ring types: Chirp1 and Chirp2. By
default, your ring type options will be those two choices. However,
using the RINGLIST.DAT file, you can customize the ring types that are
available to the Cisco SIP IP phone
Thanks for the info. I would like to go.
Is it in German or English?
I only speak English.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver
schmidt
Sent: 04 January 2004 18:10
To: Asterisk User List
Subject: [Asterisk-Users] OT: Anyone going
Hi
I have SIP working on NAT using X-lite phones.
On my Cisco 827H ADSL router I forwarded ports 5060, 16394, 16384 to my
* (10.1.0.0).
16394,16384 being RTP.
In X-lite set the RTP port to use 16394 instead of the default 8000.
Works great over the internet. Didn't need patches or anything
Balaji,
I also have the
same issue. Works fine on any phone except GS for me.
After a bit of
research I found a post saying set the phone to offer only one codec set.
It looks like we
have to set the phone to use one codec GSM
I am concerned
that you cant use passwords when
Sorry for the late reply.
I try port 5060 and it just knocks me back straight
away, I cant see it even try to authenticate in the CLI.
X-lite works both inside the LAN and outside using
SIP.
Messenger version = 4.7
John I will try your suggestion with sip.conf thanks
for the
I have read the guides on using Messenger to connect
via SIP.
I just cant get it to connect, even inside the LAN.
I enter local ip address:5036, it trys to
sign in, but times out and says Service Unavailable.
Do I need anything extra in my configs for Messenger
to work?
Have *
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