Displaying 20 results from an estimated 9000 matches similar to: "Queues with unavailable members"
2008 Aug 22
4
set callerid with plus sign
Hi,
Is it possible to assign a plus sign on the callerid(num) ?
currently this is what i do CALLERID(num)=+6523450017
but telco is denying calls, coz they said they are seeing "bs523450017"
instead of +6523450017.
i tried putting it inside double quotes CALLERID(num)="+6523450017"
telco says the same thing.
is this possible? thank you
Regards,
nhadie
2010 Nov 18
2
exceeds the maximum size of ast_fdset error on Asterisk-1.8.0
Hi Friends,
i have installed and configure asterisk-1.8.0.
When i have tried asterisk start get below errors and not able to start
asterisk.
*FD 32767 exceeds the maximum size of ast_fdset!*
Thanks in advance.
--
Best Regards,
Rajnikant Vanza
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2008 Aug 29
5
Wi-SIP vs. SIP-DECT
Anybody care to muse on Wi-SIP vs. SIP-DECT?
My limited research indicates that none of the WiSip phones will ever be
able to match the performance of DECT phones. Maybe I'm wrong but a
Wi-SIP phone seems like a DIESEL sports car. There is nothing wrong
with the technology, but it seems like a shoe-horned fit into the
requirements of a wireless endpoint. DECT uses a wireless radio layer
2008 Feb 01
7
Enterprise or Fedora?
i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference
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2008 Jan 07
3
How to check if a SIP phone is forwarded without ringing it ?
Hi,
I feel I've read a thread about this previously but I couldn't find it.
Is there way for an Asterisk server to check if a sip phone is forwarded
without bothering phone's user ?
I was thinking of some Alert-Info option that would let the phone reply with
a 302 Moved Temporarily or 182 Queued message and not let the phone ring or
display anything on its screen.
So that, you could
2009 Jul 22
3
ExecIf and empty variables (early evaluation)
Imagine that you have this code:
exten => _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
If ${QueueName} happens to be unset, this will cause a warning:
[Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187
queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an
argument: queuename
The obvious solution:
exten => _X!,n,ExecIf($["${QueueName}" !=
2007 Feb 10
1
SIP retry time too low
I have a problem with asterisk-1.2.13, where it retries SIP INVITEs
too quickly. It happens when qualify is on, and the server it tries to
reach is only 1ms away according to qualify.
The time between the first SIP INVITE and the 7th (last) is then only
64ms, and that can be too short for the peer to react.
I reported this bug in much more detail in bugs.digium.com, but the
bug is gone now
2012 Jun 05
3
CDRs on multiple servers.
Hello guys,
I need to be able to throw cdrs on more than one servers at a time. Please let me know how this can be done.
Thanks
2009 Jan 06
1
"username mismatch, have <x>, digest has <y>"
I have two Asterisks connected using SIP. One is acting as a SIP
"server", the other as a SIP "client". This almost works; but calls
from 50607795 are rejected with this error:
check_auth: username mismatch, have <50607796>, digest has <50607795>
On the "client" I have these accounts configured in sip.conf:
register => 50607795:test at
2006 Oct 10
2
E164 caller ID
Is there a proper and accepted way to go about setting an E164 compliant
caller ID (ANI) ?
Currently, we're using just the Set(CALLERID(num)=XX) where XX is some E164
compliant number like 3539146632431 or some such.
Is there another way we should be doing that or is that proper?
N.
2009 Mar 06
5
work around the 64 pickupgroups limit
Hi!
What are the typical ways to work around the 64 groups limit?
thanks
klaus
2009 Aug 18
2
Channels don't go away with soft hangup
Hello List,
our setup:
Callcenter
IBM Hardware, 1x TE420, 1x xircom analog switch, 4x different cellular
providers on the xircom analog port, ~60 agents
Debian 5.0.1 (Lenny)
Asterisk 1.4.21.2 Debian Package recompiled with additional app_queue
segfault fix
Zaptel 1.4.11 Debian Package
My Problem is I have two channels (Zap/9-1 and Zap/6-1) which have a
duration of over 4 hours.
I am
2009 Aug 02
1
T.38 and reinvite
I have a setup with a number of customer Asterisks with T.38 enabled.
This works quite well for each customer sending faxes between branch
offices.
They all have a SIP trunk to a central Asterisk, which connects them to
the PSTN through various providers on dedicated lines. I cannot enable
reinvite on those SIP trunks, because that would allow calls from the
customer's phones to get
2012 May 07
6
using Wifi smartphones as SIP clients
All,
has anyone any experience in using Wifi smartphones as SIP clients? Does
this work properly? What models/brands are optimal for this (in terms of
ease of use, battery life etc)?
Thx!!
B.
2006 Dec 19
26
Match a Numer - then continue with dialplan
Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan?
I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature.
Doug.
2008 Feb 20
6
Coppercom and Asterisk
My provider has a Coppercom switch. I have included the authentication information they gave me. How would I structure this in Asterisk to the registration and the entry in sip.conf?
User Name - 8159093010
Password - XXXXX
No Pin
Proxy - sip.essex1.com (10.1.3.2)
Outbound Proxy - proxy.essex1.com (63.164.210.14)
Change setting to use "outbound Proxy"
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Mike Hammett
2008 Sep 05
2
Bridge 2 incoming calls
I think I've forgotten something obvious....
I've got 2 incoming calls, I want to bridge them - how can I do this ?
(assume I somehow know which calls should be paired up...)
I could dump them both in a meetme - but that seems wasteful
as i _know_ there will only ever be 2 parties. (And I need DTMF
to flow through). I may want to record the bridged call, but that isn't
vital.
2006 Mar 24
2
How to nice agi scripts?
Hi,
I have unpleasent short audio gaps when a
perl based agi scripts starts.
Thus, I now started to put all those things in C programmed
daemons for fast-agi.
Anyway I'm looking for another mean, which would help me
more quickly.
I noticed, that all agi scripts are running with system
priority -11, like asterisk does. This is really waste of
priority. I would like to have the AGI scripts
2012 Feb 08
4
SIP hardware phones
I'm trying to understand why vendors keep making 100Mbps integrated 1-port switches in their hardware SIP phones. Even the recently-announced D40 and D50 Digium phones are limited to 100Mbps. Only the more expensive models (like the D70) can run at 1000Mbps.
However, you can't expect a firm with hundreds of extensions to buy the most expensive model...
And gigabit speed is important when
2012 Aug 04
1
Suggestion of Server Specifications for Asterisk
What the minimum Server Specifications do I need to run
200 concurrent channels at the time with .WAV recording (MixMonitor)?
It will be connected via VOIP sip account.
Codec will be ulaw.
Which UK dedicated server provider do you recommend and how much bandwidth
do I need?
Thanks
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