Displaying 20 results from an estimated 5000 matches similar to: "Anyone else having problems with the list"
2007 Jun 06
4
Slow list
Wow. My message made it to the list after more than 3 hours.
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? -> http://www.das-asterisk-buch.de
Gesch?ftsf?hrer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
2008 Jan 10
3
OT - Is handover included in DECT GAP ?
Hi,
Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support
roaming and handover and are these functions transparent for handset (then,
these functions are implemented in DECT base stations) ?
Regards
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2009 Nov 25
6
How many lines do you use.
Just for some information really : How many of you use multiple sip lines on
a phone ?.
I'm sitting here looking at my 7960, with it's 6 lines. I've every only used
one line, and I was wondering if I was a weirdo ;)
The only time I've ever found a use was when I had two systems (production
and test) and it caused so much grief (could have been asterisk or cisco) I
simply use a
2005 Jul 20
5
Grandstream GXP2000 resetting all the time
All,
I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones.
All seems well other than the phones have to be reset up to 5 times per day.
It is like they lose thier ip connection or maybe thier SIP connection. Has
anyone else experienced this issue? I have the phones set for static IP
addresses and that doesnt seem to help either. Any help would be greatly
2005 Jul 25
4
Fritz PCI card in ptp mode with chan_misdn
Hello !
I would like to get working a Fritz PCI card using chan_misdn
operating in ptp mode.
Afer compiling mISDN into the kernel and building chan_misdn
Asterisk stops loading with :
[chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri))
== Parsing '/etc/asterisk/misdn.conf': Found
UnLocking config_mutex
== Registered channel type 'mISDN' (This driver enables
2007 Oct 23
5
Asterisk under VMWare
Anyone had any experience with an Asterisk server as a VMWare virtual
machine?
2005 Sep 02
6
Looking for better "Follow Me"
Hi everybody :)
I am a new member here and hope that someone gives me a hint for my problem:
Let's say I am at work and my SIP phone (KPhone in my case) is connected to my
private Asterisk. I want to call my wife at home so her SIP phone rings. She
does not pick up the phone (maybe she is somewhere in the house and has to
run to the phone) so after 15 seconds her cell phone should ring.
2006 Mar 03
9
Program Buttons on Cisco 79xx Phones
Does anyone have a good resource to learn how to program the soft and
hard buttons on a Cisco 7940 or 7960 phone? Using SIP
Firmware...thanks.
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2005 Sep 21
7
add 0 (zero) to incoming callerID - how?
I have an asterisk box and SIP / IAX2 phones.
To call out, users have to add 0 (zero) before a real telephone number.
That means, that if they want to call someone that has a number 123456,
they have to call 0-123456.
Simple, right?
This has a serious drawback though - when someone calls us from the
number 123456, we see the callerID 123456, and we're unable to use the
callback/redial
2005 Aug 15
2
No translator path exists for channel type MGCP & Comfort noise support incomplete
ONLY ON MONDAY!
Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this?
2006 Jun 28
9
Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
Hello,
Here is a breakdown of the issue I am experiencing. I have three remote
employees, in various states, who have Polycom 501 phones. They are
unable to receive incoming calls after a few minutes of the phones being
plugged in. They work immediately after being plugged in, but they lose
the ability shortly thereafter. They can always make outbound calls, but
only to real phone numbers, not
2008 Sep 12
2
Setup speed dials on Cisco 7921
I've added lines like this:
speeddial = 123,test
speeddial = 260,Bob
in the [device] section for my 7921, but the speed dials do NOT appear on
the menu (click right from the main screen). Am I missing something obvious
here?
Thanks
MD
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2006 Feb 25
2
sipgate.de question
Hi,
Anyone here using sipgate.de ?
It worked for months, but for a couple of days now I'm
unable to register with them.
My account is ok, because I can login to the website.
Asterisk keeps showing me:
Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n)
I looked at the sip debug stuff, and all I
2005 Aug 17
8
DECT gateways
Heya list,
I need some advice/experience.
Some of our customers are asking us about DECT solutions for
their asterisk install. Some others will not go to asterisk
if there won't be a DECT solution.
They now have a Siemens or a Samsung PBX. Those PBX-es come
with a DECT basestation and optionally repeaters etc.
All those basestations speak some own protocol to the PBX,
so we cannot use them
2007 Feb 28
1
extensions.conf & sccp.conf howto call external number
Hello @List,
i'm using a Cisco 7970 / 7914 phone with Asterisk 1.4 & sccp
part of my sccp.conf
type = 7914 (Cisco 7970 with 7914 phone extension)
autologin = 117
description = Test
speeddial = 10,Test (10),10@mycompany
my speeddial line is for internal call
howto call a external number eg. +xx467584933 ??
i want to press a button and call directly out
thank you
greetings,
Daniel
2007 Jul 19
1
Idefisk softphone - official 2.0 release - Zoiper
Hello guys,
The so expected 2.0 release of Idefisk 2.0 softphone is a fact.
Idefisk and Zoiper became one - Zoiper 2.06.
Here are some of the features: SIP and IAX, TCP, TLS support,
Multi-language support, Automatic provisioning (XML), URL handling,
Outlook Integration, Native conferencing, API, Changeable number of
lines....
You could read the complete Press Release here:
2007 Dec 23
1
Asterisk 1.2.26 badly broken?
After upgrading from 1.2.25 to 1.2.26 i noticed that IAX -> IAX calls
always result in Asterisk just exiting without any message.
Asterisk also seems to die when using a TDM400 with 2 FXO modules, placing
2 outgoing calls on both lines as Zap/g2 and then trying to make a 3rd
call.
Went back to 1.2.25 for the moment
2006 Jun 26
3
This is getting really annoying - re: POSTFIX
What on earth is going on with the list?!?! Some of my messages
never make it... then days later I get something like this back:
Final-Recipient: rfc822; asterisk-users@lists.digium.com
Action: failed
Status: 5.0.0
Diagnostic-Code: X-Postfix; mail forwarding loop for
asterisk-users@lists.digium.com
2006 Oct 19
1
bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now
Bristuff has been updated;
http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz
--
Vidar
2006 Oct 25
2
Multiple queue_log files based on queue - is it possible??
Hello List,
Question: Has anyone been able to create multiple queue_log files in
/var/log/asterisk for multiple queues?
We are designing a multi-tenant system and separating the log files
would be useful, instead of dropping all queue actions into one file.
Is it possible this is a user configurable option I am missing?
Cheers,
--
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Christopher T Aloi
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