On Sun, May 29, 2011 at 3:18 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
And f/w POS3-07-4-00
That is strange that Asterisk is not sending anything back in response
to the register. Have you looked at the Asterisk console or logs to
see why it is rejecting the register. You might have
COT, 05/30/2011
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3
On Sun, May 29, 2011 at 3:18 PM, Ian S. Worthington
ianworthing...@usa.net
On Mon, May 30, 2011 at 2:45 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
Console is showing the following. Looks like it doesn't like the format of the
REGISTER message???
--- SIP read from UDP:192.168.1.114:5060 ---
REGISTER sip:192.168.1.41 SIP/2.0
Via: SIP/2.0/UDP
-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3
On Mon, May 30, 2011 at 2:45 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
Console is showing the following. Looks like it doesn't like the format of
the
REGISTER message
On Mon, May 30, 2011 at 5:18 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
Many thanks for that.
I tried pedantic=no (adding it directly to the [702] section in
sip_additional.conf: I'm using the freepbx frontend and it doesn't seem to
have a way to enter that through the gui), but it
--
Received: 05:11 PM COT, 05/30/2011
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3
On Mon, May 30, 2011 at 5:18 PM, Ian S. Worthington
-- Original Message --
Received: 09:03 PM COT, 05/28/2011
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3
On Sat, May 28, 2011 at 5:18 PM, Ian S
And f/w POS3-07-4-00
i
--
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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I am having a problem registering my cisco phones which is exactly like that
described in
http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html
except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00
The symptoms are:
o 7960 lines show [X]
o Outbound calls can be
On Sat, May 28, 2011 at 4:08 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
I am having a problem registering my cisco phones which is exactly like that
described in
http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html
except that I am on Asterisk 1.8.3.3 and using sip
-- Original Message --
Received: 03:45 PM COT, 05/28/2011
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3
On Sat, May 28, 2011 at 4:08 PM, Ian
On Sat, May 28, 2011 at 5:18 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
I too had heard that 1833 did NOT have the 184 problem, which makes me
suspicious that it's not that.
I don't think its a NAT problem. Neither a sip trace not tcpdump show any
response at all to the incoming
On Tue, Sep 19, 2006 at 12:12:03AM +0900, Gary Guthary wrote:
This is going to be an exercise in 'Networking' for sure...
The only catch is that per the phone's network settings:
The phone uses a static IP of something like 192.168.0.220 with a Gateway of
192.168.0.1. - Standard class
I know this isn't directly Asterisk related. - But I do appreciate the
responses I get from you folks. - At least I don't get flamed like I've seen
on the Java Perl-Mod lists (geesh!).
Okay - Here's what I've got.
A Cisco (used) 7940 that's loaded with MGCP and I want to load SIP so it'll
work
You'll need to change your XP box to the default router of the phone (or, just change your XP to a static on your local net, then add a static ipunder advanced for the default route of the phone, that way you still have access to the internet from your XP box), and then add (under advanced) the IP
A Cisco (used) 7940 that's loaded with MGCP and I want to load SIP so it'll
work on my Asterisk box (outside of the FXO FXS modules on the TDM card in
the Asterisk server, I only run SIP on the hardphones).
I don't know the phone's password (sound familiar?). - Have tried
everything, cisco,
I've recently setup Asterisk. Calls are routed to Asterisk through a
Cisco 1760 router. Some calls originate at Cisco 7960 phones connected
to the router, some originate at other phones that are switched by a
legacy PBX.
My problem is that calls that begin at the 7960s do not seem to transmit
Do you have a SIPDefault.cnf ? if yes check there. The phone is
unprovisioned b/c you didn't give any configuration settings.
On Tue, 08 Mar 2005 08:08:42 +0100, Thomas Trepper
[EMAIL PROTECTED] wrote:
Hi all,
i have a problem with some cisco 7960. Yesterday i did a
firmware-upgrade from
the
phone with the correct line settings.
- Original Message -
From: Thomas Trepper [EMAIL PROTECTED]
To: Asterisk-Users@lists.digium.com
Sent: Tuesday, March 08, 2005 1:08 AM
Subject: [Asterisk-Users] Cisco 7960 Problem - Phone Unprovisioned
Hi all,
i have a problem with some cisco 7960
Hi all,
i have a problem with some cisco 7960. Yesterday i did a
firmware-upgrade from 3.1 (1.2) with P0S30203.bin as described in the
most documents. Now i get the message Phone unprovisioned and in
TFTP-Log i find the following line:
07.03.2005 19:58 :Timeout error sending P0S3-07-3-00.bin
I am trying to connect a Cisco 3640 terminating a PRI to * with SIP.
When I call into the PRI, the Cisco answers the call and sends it on to *,
however there is no audio. The clue is, the following message out of *:
Oct 15 07:50:58 NOTICE[1094289728]: chan_sip.c:2679 process_sdp: Content
is
PROTECTED]
Sent: 15 October 2004 15:50
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco to * problem
I am trying to connect a Cisco 3640 terminating a PRI to * with SIP.
When I call into the PRI, the Cisco answers the call and sends it on to
*, however there is no audio. The clue
Hi everyone.
I've just tried installing the SIP image and i'm getting a very odd error.
It says:
Configuring VLAN
then
Configuring IP
then
Protocal Application Invalid
Help
Matt
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Forget that desprate plea! It turns out my image file was corrupt;
re-dowloaded it and all is ok.
Cheers
Matt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: 13 June 2004 20:42
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 7960
PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Matt
Gesendet: Sonntag, 13. Juni 2004 21:42
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] Cisco 7960 Problem
Hi everyone.
I've just tried installing the SIP image and i'm getting a very odd error.
It says:
Configuring VLAN
then
Configuring IP
Hi.
I have set up * in our lab here at work and got the 7940 up and running
OK. But the 7912 wont work. It registers sometimes. When I call it it
rings, but when I lift off the handset it just keeps ringing and the call
is not set up. When I try to make a call from the 7912 I get no dialtone,
and
?yvind Johnsen ([EMAIL PROTECTED]) wrote:
Hi.
I have set up * in our lab here at work and got the 7940 up and running
OK. But the 7912 wont work. It registers sometimes. When I call it it
rings, but when I lift off the handset it just keeps ringing and the
call
is not set up. When I try to
On Wed, 25 Feb 2004, [iso-8859-1] Øyvind Johnsen wrote:
I tried at first Theo's chan_sccp. (No Dialtone)... I have now tried the
lambda-solution and asterisk keeps crashing every time I have used the
7912... I guess that skinny / asterisk have some way to go before it is of
any use other than
I tried at first Theo's chan_sccp. (No Dialtone)... I have now tried the
lambda-solution and asterisk keeps crashing every time I have used the
7912... I guess that skinny / asterisk have some way to go before it is of
any use other than playing with.
please do start asterisk with
Hi all !!!
I m trying to setup a cisco AS5300 and I ve got some problem !!!
During a call test I m getting this error message all the time.
NOTICE[15371]: File rtp.c, Line 263 (process_rfc3389): RFC3389 support
incomplete. Turn off on client if possible
[general]
port = 5060
to select the preffered codec and when I change this to G.729/A-law/U-law all works
perfectly for me.
-Original Message-
From: Areski [mailto:[EMAIL PROTECTED]
Sent: 29 September 2003 14:02
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cisco AS5300 : problem configuration
Hi all
September 2003 14:02
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cisco AS5300 : problem configuration
Hi all !!!
I m trying to setup a cisco AS5300 and I ve got some problem !!!
During a call test I m getting this error message all the time.
NOTICE[15371]: File
I'd suggest you comment out the bindaddr line altogether.
-Original Message-
From: Areski [mailto:[EMAIL PROTECTED]
Sent: 29 September 2003 17:08
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] cisco AS5300 : problem configuration
Hello,
Below the IOS config file.
Should
: [Asterisk-Users] cisco AS5300 : problem configuration
Hello,
Below the IOS config file.
Should I disable RFC3389 ??? If yes HOW ??
Show running-config
-
version 12.2
service timestamps debug datetime msec
service timestamps log datetime msec
service
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