Displaying 20 results from an estimated 300 matches similar to: "Has the "allow=all" function changed in sip.conf?"
2003 Sep 09
0
Snom200 -> C7960 noisy?
When a Snom 200 (v2.1l) calls a C7960 (v4.4), both using g711u as default,
the conversation is extremely noisy from the Snom to the Cisco, but clear
in the reverse direction. Using a sniffer, I see packets from the Snom to
the Cisco of 87 bytes and Cisco to Snom of 214 bytes. Asterisk is CVS from
Saturday.
The communications between the two was working fine on Saturday, however
something has
2003 Sep 10
9
Free World Dialup (FWD).
Hi,
Is it possible to use asterisk with Free World Dialup (FWD) ?
Did someone manage to make it work? how?
Best,
-P
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2003 Sep 11
1
How much to charge for Asterisk installations?
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I have a medium sized business that is interested in implementing *
as their PBX system. They currently have a Panasonic system with
Panasonic handsets that they are going to replace Asterisk with, as
the current system is maxed out, and they don't even have voicemail
capabilities.
I have been considering using an Adtran Atlas 550 with FXO and
2003 Sep 11
1
newbie - sip, pxb, ata, nat
hi all,
I don't know how to setup asterix to work as PBX.
If I want just basic configuration with 2 SIP phones (snom, ata), what
all I have to write in the configuration files, or respectively in the
configuration of ata and snom ?
If there is any good documention available, send me URL too.
All (ata, snom) are behind firewall (nat) and astrix is on the public
IP, but I can move for
2004 Jan 31
4
rtp sound quality?
pstn -> sip gw -> * -> C7960
When I dial into * via the pstn, I hear the ivr menu just fine (good
quality). I press 3000 (valid extn), and I begin to hear ringing however the
ring back is very very choppy.
I answer the C7960, and speech is clear in both directions. Place the C7960
extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates
both the sip gw
2004 Jun 18
2
C7960 g729 question
I have multiple voiceage g729 licenses installed on a RH9 box, and have
a remote C7960 configured to use it (low bandwidth). In calls like:
Remote C7960 -> g729 -> asterisk -> g711 -> C7960
the audio is oftentimes rather choppy. Changing the remote 7960 to use
g711 seems to eliminate/reduce the choppyness. Any ideas on what might
be behind this?
2003 Sep 06
7
OT: Creating documentation using a web interface
Hello all,
I would like to document some things I am doing with asterisk, but would
prefer to do this from a web interface. I am unfamiliar with any
software that allows you to create online documentation from a web
interface. Ideally I will be able to create documentation online from a
browser, which then when saved, is immediately ready to be read online.
Perhaps I can setup different authors
2005 Oct 07
1
Distorted VM with iax2 with ilbc and jitterbuffer - bug?
Two asterisk boxes 150 miles apart, both cvs-head as of this morning
(and since Sept 27th), connected via iax2 with low-utilized ds3 internet,
C7960 calls exten on remote system (also C7960), and call goes to VM.
No other calls in either system (eg, no load).
Both boxes have iax config'ed as:
trunk=yes
allow=ilbc
jitterbuffer=yes
Recorded VM messages are very distorted.
Changing only
2006 Mar 18
1
Polycom IP600 - no ring?
Have a strange problem...
When a C7960 calls the Polycom ip600, the ip600's first line button
blinks, the ip600 display shows the proper callerid, but the phone does
not ring at all.
If I call the same ip600 from a bt102, the ip600 rings properly.
If I call the same ip600 from another C7960, the ip600 rings properly.
All phones and asterisk are on the same lan within a few feet.
The
2006 May 24
2
DHCP configuration for Cisco 7960?
(Apologies to those Toronto Asterisk Users' Group folks who have seen
this message... I figured I'd have more success with a wider audience)
I'm trying to boot a Cisco 7960 from an ISC DHCPD server (3.0.3 on
FreeBSD 4.11), so far unsuccessful, and getting some odd behaviour on
the wire. I wonder if anyone has done this before and therefore can
validate whether or not the traffic I am
2004 Mar 31
2
C7960 "busy" notification
Using the following defnitions with a C7960:
exten => 3001,1,Dial(SIP/3001,15,r)
exten => 3001,2,Voicemail2(u3001)
exten => 3001,102,Voicemail2(b3001)
exten => 3001,103,Hangup
If someone is on this phone (real conversation) and another call comes in,
the second call goes through the 15 second timeout and is dropped into the
2-priority as "unavailable" (not the 102 busy as
2003 Sep 19
1
built in dial functions?
Someone recently posted the following list as functions built into *
*0# sends flash
*8# remote call pickup (pickup phone in your group)
*67# disable caller id
*70# no call waiting
*78# do not disturb on
*79# do not disturb off
*72# enable call forwarding
*73# disable call forwarding
*82# enable callerid
I'm running a CVS from a couple of weeks ago with multiple C7960's,
snom 200,
2004 Apr 06
3
Problems with IAX2?
Are there open problems/issues with iax2 and jitter (quality)?
Just upgraded to today's dev cvs about an hour ago, and it seems the iax
conversations are lower quality then a month or two ago. iax2 show firmware
says version 13. (Test call originated from C7960 with g711.)
Using the demo as an example,
iax2 show channels
Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter
2004 Jul 25
2
Incoming SIP gateway context?
I just started service with Broadvoice.com and everything seems to work.
However, apparently my understanding of incoming sip contexts is less
then what I thought it was. Could someone point me in the right direction?
(* on a public address, CVS-HEAD-07/12/04, C7960 phones)
In my sip.conf I have:
[general]
port = 5060
bindaddr = 0.0.0.0
allow=ulaw
tos=0x18 ;sets ip tos bits (=lowdelay and
2005 Oct 12
0
Notice message meaning for C7960?
Asterisk cvs-head compiled 2005-10-07 11:
Oct 12 18:35:12 NOTICE[21740]: chan_sip.c:10685 handle_request_register: Registr
ation from 'sip:301495906@204.212.194.101' failed for '208.5.218.28' - Not a lo
cal SIP domain
The sip phone is a Cisco 7960 with one line defined, and registration
with * is occuring just fine. Calls to/from the phone are fine. The
phone is on a distant
2003 Oct 17
0
X100P Echo - not resolved
I still have echo during the first 15 seconds (or so) of each call, and
the exact same thing is happening with either of two X100P in the same
2.2ghz system.
asterisk# cat /proc/interrupts
CPU0
0: 198457690 XT-PIC timer
1: 60 XT-PIC keyboard
2: 0 XT-PIC cascade
3: 0 XT-PIC usb-uhci
8: 1
2003 Sep 15
3
SOME QUESTIONES (LOG, MySQL, Extensions)
Hi all.
I have some questions:
1) Is there a way to get a full log of the calls (incoming and outgoing)
2) How is the intregation of Mysql and Asterisk. At witch Aplicattions.
3) And of the Extension
a) I have a Support Call Center. Almost all the time all the
extensions are busy, and some calls at hold. Is there a way that when some
2003 Sep 08
0
Is this use of DISA secure?
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OK, so I have a local extension that a phone can call to take it to
voicemail. I don't want it to exit out to a fast busy tone, as I
would rather it allow the user to simply continue on and call a new
number (without having to physically release the line first). The
[intern] context is where everything goes by default (sip.conf for
example has
2003 Sep 10
1
MOH - White noise, static
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Hi all,
I am using a TDM40B, and have managed to compile mpg123 and turned on
MOH. Problem I am having is that it is choppy, staticy, and sounds
like white noise pretty much. I have search the archives to see if
this problem had been resolved, but I haven't found anything yet.
Has anyone had this problem and resolved it? I am calling from
2004 Apr 01
4
CISCO 7940 and directory/services problem
I have quite successfully set up the Services button to work on the 7940
running SIP.
I have a metric-imperial converter, a foreign exchange rate calculator, a
calendar etc available to users.
The XML is really fussy though.
Simon
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Rich Adamson
Sent: Thursday, 1