Displaying 20 results from an estimated 30000 matches similar to: "Testing 911 call"
2010 Feb 15
2
insecure=invite - not working for different dial plan
I'm using "insecure=invite" with two different dial plans, it it working with one dial plan but not with the other.
What other parameters could influence "insecure=invite"
In sip.conf below "insecure=invite" is working OK
[pstn-1270]
type=friend
secret=spa3k
username=voice-1270
mailbox=369
host=dynamic
insecure=invite
canreinvite=no
disallow=all
allow=ulaw
2010 Feb 19
3
splitting sip.conf to two files
Is it possible to split sip.conf into two files (sip1.conf sip2.conf)?
I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP
and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (=> one single IP with different SIP ports), the last entry
into my
2010 Feb 16
1
call is not going to wrong "context"
I've Audiocodes MP-114 registered per-endpoint (2x FXO / 2x FXS) but when call comes on pstn-4444 it goes to context "fax-incoming"
in sip.conf:
[pstn-4444]
type=friend
context=incoming
...
[pstn-9998]
type=friend
context=fax-incoming
...
the device register per end point just fine, so it can find "secret=xxx" correctly but why the call is not forwarded to correct
2010 Jan 12
1
AudioCodes MP-114 - SAS (Stand Alone Survivability) configuration
I have AudioCodes MP-114 and I'm trying to configure SAS (Stand Alone Survivability); when Asterisk is down the MediaPack gateway should forward the call
IN/OUT through the gateway (without asterisk in the middle), but it is not working.
I'm working with tech. support from the source I purchase the unit from they we are just emailing back and forth and the unit is still not working.
Can
2009 Dec 31
1
AudioCodes Caller ID
I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO)
AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to interpret it as authentication:
[Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553 check_auth: username mismatch, have <pstn-5665>, digest has <pstn-1270>
[Dec 31 11:41:07] NOTICE[9752]: chan_sip.c:14316 handle_request_invite:
2010 Oct 14
6
Audiocodes firmware
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body text="#000000" bgcolor="#ffffff">
<font size="+1">Does anyone have links to the most recent audiocodes
2009 Dec 28
1
AudioCodes MP-114 making calls via FXO
I was able to setup AudioCodes MP-114 to rote calls form FOX to Asterisk and make internal calls:
Routing Tables -> Tel to IP Routing:
*, *, 10.0.0.109 (my asterisk IP)
But I'm not sure how to setup AuioCodes to make calls out via FXO?
In extensions.conf
[Globals]
pstn-5665=10.0.0.157
Whenever, I try to call out I get a busy signal.
--
Joseph
2009 Oct 19
1
Cisco 1751 setup with asterisk
How hard is to setup Cisco 1751 w/2x FXO with asterisk?
I was googling but couldn't find much information; how to access unit interface for programming?
It might be a good replacement for Linksys.
--
Joseph
2006 Dec 24
1
Voicemail hangup by gateway?
Hi,
I have a spiffy new gateway which seems quite promising.
It's the Audiocodes MP114 FXS_FXO (2 of each).
I have got it configured and working reasonably well, but have a couple
of issues.
1) Asterisk 1.2.13 voicemail seems to be hung up on by the gateway
after 10 seconds. This isn't asterisk saying it's quiet for 10
seconds, it's the gateway deciding it's time to go
2009 Nov 06
2
Question about callerid?
Hello again Asterisk people.
I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
deployed for several years now, with pretty good results.
Recently I added a callerid service to my landline (qwest).
I am using the audiocodes MP114 2fxo/2fxs gateway, which is an
outstanding piece of hardware once it's configured (lol).
Anyhow, I can see that the gateway is passing
2007 Jan 16
4
Audiocodes GPL
I have some Audiocodes units which appear to be running Linux,
according to the unit's own "System Log"
kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006
However my contact at Audiocodes claims otherwise
On 12/4/06, Yaniv Nizan <Yaniv.Nizan@audiocodes.com> wrote:
>
>
>
> I doubt that we are running Linux on the MP-202. Perhaps there is a
2004 Jan 06
1
Re: 911 and lawsuits and redundancy
Hi,
Most companies we work with, have 'designated' crisis management teams.
These vary from the insignificant crisis', through to life-threatening
crisis'. There is always an assigned emergency services contact, whose
job it is in an emergency, to maintain communication with the emergency
services.
One of our corporate functions is crisis management - so we have to
consider
2010 Mar 19
0
SPA3102 + asterisk drop call and loop (was SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?) )
Ok,
I downgraded spa3102 to 3.3.6. Now when I make a call from pstn and call is
established asterisk seems to drop the call.
However I still hearing ringback on pstn side, call is established again,
and asterisk drops the call again, like a loop.
-- Executing [preat_admin at nodo:1] Playback("SIP/PSTN-08214948",
"horario-atencion/our-business-hours-are") in new stack
2004 Jun 18
2
Testing UK emergency dialing and LCR.
Hiyall.
Just wondering how people test your emergency dialing in the UK.
Obviously you need to dial the 999 for emergency services, but am a bit
unsure if this would go down too well with the operator with a 'sorry
just testing' call. (you do all /test/ your emergency dialing dont
you!?:-) )
As another thing, what is the correct method when using least cost
routing... If you have a
2005 Sep 30
7
911 Q
OK, got a question on 911.
Looking into setting up a couple asterisk servers at a country club,
with VOIP phones in each of 100 short-term residential rental units.
Approx 100 extensions, approx 24 outside lines.
Since everything is geographically at one location, reaching 911
correctly shouldn't present a problem. However, the club wishes to
ensure that 911 authorities are able to identify
2003 Oct 01
1
Audiocodes gateway and asterisk
Is anyone on the list using an Audiocodes gateway with asterisk and SIP?
I'm looking at that platform, but I have a couple of issues:
1) Echo cancellation. The echo that I'm hearing with an X100P is
unacceptable. Does the Audiocodes do better?
2) Line signalling. I'm using Kewlstart with the X100P, but it looks like
the audiocodes uses loopstart only. How does this work with
2006 Oct 22
3
Audiocodes MP-20x
Has anyone used the AudioCodes MP-20x?
http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf
Seems like a good device, but I can't seem to find anyone actually using
them...
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2005 Jun 28
1
audiocodes
Is anyone on this list using and audiocodes FXO gateway? I have
Asterisk(1.07 on OS X) setup and working fine, including SIP phones
and IAX2 phones - I can make outbound calls just fine and receive
inbound calls just fine. However, I can't seem to find the right
series of DTMF settings on the AudioCodes to allow DTMF tones to be
sent after an outbound call is connected(phone banking,
2004 Jun 16
3
911 emergency service and VoIP
I understand that most VoIP providers allow for 911 calling but that 911
service is not the same as that available to PSTN.
2012 Jan 07
2
Asterisk 10.0 & 1.4 - iax codec are not compatible
I'm trying Asterisk 10.0 (as 8.x is not passing PSTN CallerID) and Asterisk 10.0 is no better.
I'm still getting:
WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have <11>, digest has <pstn-1270>
NOTICE[12295]: chan_sip.c:22769 handle_request_invite: Failed to authenticate device "KMIEC Z" <sip:7804715665 at 10.0.0.110>;tag=1c1222950155
Anybody