Displaying 20 results from an estimated 8000 matches similar to: "Pass a call to another switch"
2004 Oct 02
1
RE: Random disconnects
Re-sent because had wrong subject line on first post, sorry.
I have just installed * for a small office (P4 3Ghz, 1MB RAM, RH9).
It's replacing an analog PBX, and for now, all incoming calls arrive on
10 FXO's. Outgoing calls are via Voicepulse. Phones are SIP, Cisco
7940G's. My problem is random disconnects on both incoming and outgoing
calls. The phones are behind a firewall;
2004 Sep 04
3
Help Running am-main.pl Perl/CGI on Apache Server
Hi all,
I've installed Asterisk on Linux Red Had 9. Now, I was trying to set
up a GUI based system for the PBX.
I downloaded some packages, but I have to have Perl running CGI
scripts through the webserver. It does not allow me to.
I am able to run a basic script that just just prints out html
messages and nothing else. However, when I try to run am-main.pl or
config.pl or any other
2004 Sep 23
3
Help with strategy for echo cancellation.
I have just installed * (RH9, P4 3.0GHZ, 1G RAM) in a small office,
using three TDM400's with 4 FXO's each for incoming calls. Outgoing
calls are (for the moment) routed via VoicePulse. Phone sets are Cisco
7940G's using SIP. I'm getting intermittent echo on outgoing calls, and
my understanding, based on reviewing the wiki and several posts here, is
this:
>>>> The
2004 Sep 30
7
asterisk 407 Proxy Authentication Required
Hello,
I cannot accept any inboud calls from any provders in my asterisk
which tries to authenticate the provider and at the end rejects the
call with tthese message
407 Proxy Authentication Required
How do I turn off this message.
Thanks.
Ehsanul Karim
2004 Sep 27
1
Fedora2 and zaptel - using the udev
Hi,
I am sorry if this message has been reposted, but for some reason I am
having problems with posting it.
I configured asterisk and zaptel modules with fedora2.
I want to be able to load the zaptel wcfxo and wcfxs modules.
For now I will use only the Wildcard TDM400P card.
I am able to load the modules but I cant configure them using ztcfg or
zttool because the tools are compiled to use the
2004 Oct 05
1
Phantom calls on FXO
I'm getting these "calls" at 16 and 46 minutes after every hour. The
SIP phone rings, and if we pick up, we get a dial tone. If we don't
pick up, we get the dial tone in a voicemail message. An analog phone
connected to the incoming POTS line doesn't ring (whether or not *
remains connected to the line). It's like the horror movie where the
babysitter is getting
2004 Oct 04
0
using broadvoice and vonage hardware withAsterisk
So Asterisk can't send VOIP calls to Vonage -- but it is still possible
to use Vonage for flat-rate long distance by connecting the Vonage
AT-196 to an * FXS port, right? The price is an extra D/A <--> A/D
conversion.
Jim Shilliday
IT Director
Equal Justice Center
1315 Walnut St. Suite 400
Philadelphia PA 19107
215-238-6970
-----Original Message-----
From: Tim Petlock
2004 Dec 14
0
voicemail playback problem
My users are reporting that some voicemail messages are being cut off in
the middle of being played back. The recordings are OK (they play fine
when forwarded to e-mail, and they can often be accessed OK during a
later call to voicemail). I found nothing in the archives on this --
ideas anyone? RH9, P4, CVS-HEAD-09/02/04-08:44:34, aggressive echo
suppression turned on.
Jim Shilliday
IT
2004 Sep 27
0
Cisco 7940 -60 firmware upgrades
This for the archives in case it may help someone:
I was able to upgrade two Cisco 7940's from firmware P0030301MFG2 to SIP
7.1 as follows:
1. Installed 7.1 images from the Cisco zip file to the TFTP server.
2. Specified "image_version: P0S3-07-1-00" in SIP<MAC>.cnf and
SIPDefault.cnf
3. For the older of the two phones, renamed P003-07-1-00.bin to
P0S3-07-.bin, making it
2005 Mar 16
3
Cisco gateways and hairpinning
Hello:
Has anyone on this list had to configure hairpinning on a Cisco
gateway running IOS 12.2 or 12.3 and using a PRI for connectivity
to the PSTN? If so could you tell me how it is done? I'm told this
is the source of my call transfer problems and yet I cannot find
clear instructions for how the configuration is done.
Thanks,Steve
--
ISC Network Engineering
The University of
2010 Aug 25
1
Asterisk 1.6.1.17 ACK/BYE question
We're running Asterisk 1.6.1.17 for our campus voicemail server and Juniper M120s as our SBC. Unanswered calls, which arrive via the SBC, are diverted to voicemail using a 302 redirect when the called party doesn't answer. In this case the caller is able to hear the greetings and begin to leave a message only to have Asterisk terminate the call mid-recording.
We're uncertain why
2004 Sep 11
1
call park question
I can part a call (dial #700 it is parked on 701) but if I dial 701 I am told it is not a valid extension?
I have include => parkedcalls in my local extension context. I have Ttr on all extensions and the incoming pots line.
It parks, plays MOH but I can't retrieve it.
--john
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2005 Jan 28
1
Command to light MWI on 7940 /7960
We have several agents on queues, and want to indicate to them that they
are logged in or logged out. We have tried several different ways, from
changing the screen to presenting different service menus, but cannot
get anything to be "in their face" (their words, not mine).
One of our team has suggested, as the agents do not have voicemail, is
to use the MWI on the 7940 phones to
2005 Feb 08
3
Looking for FXS device - CISCO ATA 186
I was looking for something to connect a couple of POTS handsets to my
asterisk server and found this on ebay
http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&category=162&item=5162868118
&rd=1
The documentation says that it does SIP - therefore will it work in an
asterisk environment.
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus
2005 Feb 14
1
Uptime/reliability with SER, Asterisk
Could anyone shed any light on how SER and/or Asterisk (stable branch)
has held up for them in that last while?
Are you using SER and/or * in a production environment? Do you ever
restart the software or reboot the system? How many users are
utilizing the system? How many calls per day/concurrently?
I read some uptimes and such on the mailing list from long ago, so I
was wondering what some more
2005 Sep 12
1
Other Voicemail systems
Since I can't seem to get anything figured out for the Comedian system, are
there any other systems out there that we can hook asterisk into?
Sherwood McGowan
ViaTalk
Level 2 Support
VOIP System Engineer
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2006 Jan 06
1
server recommendations
OK all. I need some help. Looking to deploy asterisk servers and want
to get a recommendation on what server to buy. I love Dell's, but from
what I see on the list they seem to have some issues. I would like to
stay with one brand and need systems for small offices (20 users),
medium (50 users) and large (100 users) systems. Thanks for the help.
Keith
2006 Mar 14
2
OT - force Cisco phones to reboot
Hello to all
Does someone knows how to force the Cisco IP phones (7940 and 7960) to
reboot weekly or monthly?
I think this would be useful because sometimes we change the
configuration settings in the TFTP, but the phone just check the TFTP
when he restarts...
Thanks
Jo?o
2005 Feb 10
1
SER Asterisk Voicemail
Hi all,
I have SER and Asterisk set up together with ser handling user
registrations and asterisk providing voicemail services. When I ring
a phone and it doesnt answer after a designated amount of time, the
request is forwarded to asterisk, and I can leave a message.
Now, this may seem a ridiculous question but how can I listen to my
message afterwards? I have read about a solution by Java
2005 Jan 27
2
CISCO 7905 Phone Weirdness
It seems on my phone, which is hooked up to a large pbx network powered by
an asterisk server, that it will randomly start ringing with a callerid#
of 2013 which is its username for that phone. I have looked and been
watching on the asterisk command line with the -vvvvvvr switch and nothing
has been seen that indicates a reason for this random ringing. This leads
me to think that this trouble