Displaying 20 results from an estimated 8000 matches similar to: "Monitored outbound dialing via Zap interface?"
2005 Mar 22
1
Is there a way to get inserted into an LEC's CLIDB?
> -----Original Message-----
> From: Robert Goodyear [mailto:me@jrob.net]
> Sent: Tuesday, March 22, 2005 1:21 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's
> CLIDB?
>
>
> Does anyone know if there's a service out there to -- for a fee --
> inject our DID into the
2004 Sep 07
3
Maximum tollerable lag/jitter for IAX2 w/o jitterbuffer enabled?
I'm having a problem with intersite calls over IAX2 being abruptly
terminated. Nothing odd shows in any of the logs for Asterisk or the host.
The only think I can think it might be is a lag-spike on the site to site
connection.
How sensitive is IAX2 to lost frames, lag spikes or large variations in
jitter with the GSM codec and:
bandwidth=low
jitterbuffer=no
trunkfreq=100 ; Raised from
2004 Aug 31
1
Why is it called 'Comedian Mail?
Inquiring (management) minds want to know. I'm assuming it's because 'it's
funny how simple it really is to write a really decent voicemail system'?
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
2005 Aug 11
1
PRI dropped calls w/ asterisk dropped betweenpstn & norstar
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Gary Reuter
> Sent: Thursday, August 11, 2005 12:59 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] PRI dropped calls w/ asterisk dropped
> betweenpstn & norstar
>
>
> I poured over my logs most of
2004 Aug 19
6
How to run different codecs between the same endpoints on an IAX trunk?
Or perhaps how to configure and refer to two parallel IAX trunks with
different codecs?
I have a situation where I'm using G.729A as my IAX trunking codec. Now I
need to push some short duration, low bitrate modem traffic over the link (a
credit card terminal). Obviously the modem audio isn't going to survive the
G.729 codec process intact, so for the times the device is used I'd like
2004 Aug 27
3
Digit detect during a Background() inside a Macro wrongly jumps b ack to the calling context to match digits?
Consider this dialplan fragment, where the call is being dialed into
[macro-process-routing] over an iax2 channel from another (same build)
Asterisk server:
[macro-process-routing]
; This is the entrypoint of the debug call but is also refered to by
Macro(process-routing) elsewhere in the dialplan
; XXX-NNN-6800
exten => _6800,1,Macro(6800-interceptor)
; This is matched when 8 is
2004 Aug 19
1
Inband announcement of parking slot from app _parkandannounce?
Couldn't see the forrest for all the fascinating tree-like applications that
are out there:
For future reference, see:
http://www.voip-info.org/wiki-Asterisk+call+parking
:-)
-----Original Message-----
From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca]
Sent: August 11, 2004 1:10 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Inband announcement of parking slot from
2004 Sep 16
2
No Caller Name sent from Asterisk over National or DMS100 PRI to a Norstar MICS?
I have a PRI link up and running between Asterisk and a Nortel Norstar MICS
v4.1 . I'm having a problem getting the textual Caller Name across the link
from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns
to Ast both elements come through fine. I'm forcing dummy values for testing
using:
exten => s,1,SetCIDName(Test)
exten => s,2,SetCallerID(1234561234)
2004 Sep 07
4
Maximum tollerable lag/jitter for IAX2 w/oji tterbuffer enabled?
> -----Original Message-----
> From: Chris Shaw [mailto:chriss@watertech.com]
> Sent: September 7, 2004 4:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2
> w/ojitterbuffer enabled?
>
{clip}
>
> If you can reproduce it, this smells like a bug... IAX runs over TCP and
TCP
>
2004 Sep 07
0
Monitored outbound dialing via Zap interface ?
> -----Original Message-----
> From: Adam Goryachev [mailto:mailinglists@websitemanagers.com.au]
> Sent: September 7, 2004 8:05 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Monitored outbound dialing via Zap
> interface?
{clip}
> Have you considered adding the r option to the Dial command, so they
> might hear ringing
2004 Sep 27
0
SNMP instrumentation and/or talk path health monitoring?
We're running a multi server Asterisk private network, including some PBXs
via private network PRI. As the system is growing and users are coming to
rely on it more we'd like to integrate it into our network management
environment to provide some basic health monitoring. The prefered method
would be SNMP as we're already monitoring the underlying Linux host. Has
anyone implemented
2004 Sep 09
4
IAX2 dropping call?
Hello all,
I updated from CVS 3 days ago and now my IAX2 gateway is dropping
calls without warning.
It happens right in the middle of a conversation with no pattern. I
never had this
Problem before and am usually talking 2-3 hours a day.
Is their a bug? Should I rollback?
Cheers,
Paul Seniuk
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Name: Paul
2004 Aug 29
1
Bridging audio in cmd_dial() before connect completes?
Is it possible to make cmd_dial() bridge the audio going out to the network
back to the calling party as soon as dial() starts? Put another way, is it
possible to have the caller hear the outside dialtone and subsequent DTMF
digits? I notice that there is an option 'r' to dial(), thus:
r: Generate a ringing tone for the calling party, passing no audio from the
called channel(s) until one
2004 Sep 08
0
T100P calls with playback starts speaking be fore pickup
> -----Original Message-----
> From: Jerry Geis [mailto:geisj@pagestation.com]
> Sent: September 8, 2004 2:19 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] T100P calls with playback starts speaking
> before pickup
>
>
> Hi
>
> I am using a T100P connected to a panasonic phone switch using T1 and the
> switch has 4 analog lines
2004 Aug 27
0
Hangup() doesn't always when talking to Nortel Norstar over CT1 E &M wink-start trunk line?
I've noticed a problem with calls to Hangup when talking to my Norstars over
channelised T-1 E&M trunk lines - it's been present since I started to
fiddle with Asterisk last December and it's still present in 'Asterisk
CVS-HEAD-08/13/04-10:37:13'.
Specifically, when a call is connected to Asterisk from the Norstar DTI card
to my T100p I get the following conditions
2004 Sep 07
2
Maximum tollerable lag/jitter for IAX2 w/o j itterbuffer enabled?
Unfortunatly no on both counts.
The arrangement right now has:
PSTN Trunks & Stations <-> Nortel Norstar#1 <-CT1-> Asterisk#1 <-IAX2->
Asterisk#2 <-CT1-> Nortel Nortstar#2 <-> Stations
The Asterisk boxes provide Voicemail to their sites Norstars and intersite
calls over IAX. Local Voicemail works flawlessly at each site but there have
been reports of PSTN calls
2005 Oct 11
6
PRI echo issues: solvable?
Hello,
After solving the other "low hanging fruit" audio issues in our Asterisk
PBX, we are left with occasional cases of severe echo which we have not
found a solution for yet.
Our system:
- Asterisk 1.2.0-beta1
- TE110P on a PRI
- TDM04 and TDM40, but these are unrelated to current echo issues
- Fedora core 3
- Echo canceller KB1
Most calls have minimal, acceptable echo levels. But
2004 Dec 06
3
PRI/Zap premature dialing problem
The originating PRI system passes the entire dialed number in the d-channel
setup frame, thus the concept of a wait time for additional digits is
meaningless. Progressive digit gathering implies that the signalling is
occuring 'in-band' as would be the case with DTMF signalling on analog
lines.
You need to look in the Ascom and find the configuration table that lays out
the dialplan for
2005 Jun 07
2
Help! Zap echo on bridged calls
I've been going nuts lately trying to get rid of an annoying echo
problem that makes my asterisk server unusable when clients try to call me.
Here's the breakdown of the issue - Hoping that someone can throw me a
clue:
My setup is as such:
Single AMD Athon machine with X100P clone card and voip through multiple
providers .
* Inbound calls through the X100P that do not bridge to
2004 Sep 02
5
Any way to _always_ execute certain commands in a dialplan context?
I've got a need to do something like the following:
[foo-context]
exten => _.,1,SetCIDNum(123)
exten => _.,2,SetCIDName(XYZ)
include => local
include => tollfree
But of course, this example won't work. The goal here is this: if a call
ends up being handled by the "local" or "tollfree" contexts, I want
those SetCID*** commands executed. Otherwise, I