Displaying 20 results from an estimated 10000 matches similar to: "[Asterisk-Dev] Asterisks"
2004 Sep 01
2
Lucent iMerge
I've read the wiki and other resources on how to connect Vonage / Voicepulse
and all these other services to Asterisk... We are attempting a connection
to a Lucent iMerge. Lucent has told us that it won't work - but we feel
confident that it will. Has anyone worked with the Lucent iMerge - or would
be willing to help lend a hand?
It is capable of H323 / MGCP. Even if I could make the
2004 Aug 25
4
YAAN (Yet Another Asterisk Newbie)
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2004 Sep 10
1
No DTMF or Audio
I have built latest Asterisk w/ OpenH323 channel driver. We have a SIP
softphone registered to the Asterisk. We can place outbound calls from the
SIP phone to the PSTN via OpenH323 connection to our gatekeeper. Everything
works okay - DTMF and Audio...
But in the reverse - if we call from a cellphone or landline the PSTN number
we can get the SIP phone to ring - we answer and can hear the
2004 Aug 25
0
Asterisks
Take mercy on me - I'm a newbie w/ Asterisks... Here's what I'm trying to do
- and please someone let me know if this can be done...
We have a commercial VoIP network (we are a communications carrier)...
The gatekeeper (Lucent iMerge) supports MGCP/H.323 and
allows for calls to be made to the PSTN cloud via GR303 links.
I would like to build Asterisks with H323 (or MGCP if need be -
2005 May 20
2
MGCP 1.0 / NCS 1.0
I'm trying to use a 4 port ATA that was designed for use with MGCP 1.0 / NCS
1.0 -- is the channel_mgcp compat with NCS -- anything I can do to make it
compat?? This is what happens - below
*CLI> mgcp reload
Reloading MGCP
== Parsing '/etc/asterisk/mgcp.conf': Found
Use EXIT or QUIT to exit the asterisk console
== MGCP Listening on 10.1.22.39:2427
== Using TOS bits 0
mgcp
2003 Oct 12
6
SIP phone
I have a Cisco 7940
when you call in from outside and dial the Cisco phone extension I get
this
Read_channel ## vpb/1-3: Setting record mode, bridge = 0
WARNING[18451]: File chan_sip.c, Line 1111 (sip_write): Asked to
transmit frame
type 8, while native formats is 4 (read/write = 4/4)
== Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3'
-- hangup on vpb (vpb/1-3)
2004 Dec 23
1
Premature DRQ
I have a problem where an Asterisk server is sending a premature DRQ... Not
sure why..
Here's the setup - Asterisk using inAccess networks H323 replacement channel
driver
Connecting to a Lucent iMerge...
The call connects fine - I get the out of the box greeting - but after
exactly one
Minute - the call terminates.
I have had this problem on multiple different Asterisk configs...
I'm
2004 Sep 05
2
GRQ / RRQ
All - NEED HELP BADLY..
Using Asterisk with H323 Channels..
Everything is installed but we cannot register w/ Lucent gatekeeper.
We ran ethereal and found that it was making GRQ (Gatekeeper discovery
requests)..
We had provided the name of the Gatekeeper (it's IP) and cannot determine
why
it's trying to do a GRQ. We want it to go straight into RRQ or ARQ and skip
the
GRQ.. Netmeeting does
2004 Sep 09
2
Dial Out w/ OH323
Due to the format of the message coming from the H323 channels included w/
Asterisk we were unable to use our gatekeeper.
For a quick solution we tried the OH323 channel drivers and can receive
inbound calls from the parent gatekeeper.
We are trying to do a dial to gatekeeper...
I am trying
exten => 5551212,1,Wait,2
exten => 5551212,2,Dial,OH323/5551212
But I am not sure if this is the
2003 Jun 03
2
Asterisk Works on Linux on Sparc
I have built Asterisk on SuSe Linux 7.3 on an Ultra 2 Sparc WorkStation. I am listing the modification I had to do for the benefit of anybody else who wants to use Asterisk
This workstation is equipped with one 400 MHz RISC UltraSparc II CPU, 256 MB RAM, Two 9 GB 10,000 RPM UltraSCSI Disks. I have a gatekeeper running on this machine,
I had to do the following modification to build * on Sparc:
2003 May 22
2
authentication h323
Hi all,
i want to use authentication in h323.conf.
How can i use it ?
In the h323.conf is writed :
"If you wish to use Authentication you need to set the appropriate auth
keyword above"
Where and how i have to set this keyword ???
I've tried "auth=yes",but it does not work.
Thanks for Help,
Thomas.
2004 Jan 19
3
Residential services
Hi folks,
The obligatory newbie disclaimer:
"Hi, I'm new to Asterisk and I have a couple questions..."
OK, now that that's over with:
I've just started working for a small CLEC, and I'm trying to sell * to
my boss as a replacement for some expensive/inflexible/closed-source
software he's been using to provide residential dialtone with for a
couple years now.
2004 Sep 20
2
H.323 call problemm (no sound)
Hi all,
I'm having trouble with H.323 outbound calls, * connects but there is no
sound in both ways.
I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which
using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729
licenses installed and this is onli one call.
I tested my * with another ITSP over SIP and G.729 codec and there was
all ok
Here is my configs
2004 Jan 22
3
MGCP w/8x8 DTA-310 and as5300 pstn gateway
Hello folks,
I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no
dialtone & can't get it to ring. My mgcp.conf says:
;
; MGCP Configuration for Asterisk
;
[general]
port = 2427
bindaddr = 0.0.0.0
[172.16.2.25]
host = 172.16.2.25
context = default
line => aaln/1
And here's the interesting bits of extensions.conf:
[globals]
...
TRUNK=H323/BYEXTENSION@pstn_gw
...
2004 Jul 29
4
One More IP Phone for interoperability with Asterisk
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2005 Jun 28
1
Fw: Shoutcast Music On Hold problems?
----- Original Message -----
From: "hank" <hanksmith4@earthlink.net>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: Tuesday, June 28, 2005 10:52 PM
Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems?
>I am using asterisk@home 1.0
> my mp3 is called
> mp3
> it has nothing before it
2007 Oct 05
1
[asterisk-dev] oh323.conf, extentions.conf
Send these questions to Asterisk-Users mailing list.
h323.conf
##################################################
;
; Configuration file of OpenH323 channel driver
;
[general]
listenAddress=W.X.Y.Z ; local ip
listenPort=1720
tcpStart=10000
tcpEnd=20000
udpStart=10000
udpEnd=20000
fastStart=yes
h245Tunnelling=yes
h245inSetup=yes
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=100
2005 May 19
7
Cisco Call Manager & Asterisk for Voicemail
Has anybody successfully (or I guess unsuccessfully for that matter)
implemented Cisco Call Manager and used an * box for voicemail? I
checked the wiki and google and I see some references to Call Manager
Express and *, but CME is completely different than CM. If anybody has
done this or has any insight, it would be appeciated. We are trying to
migrate ~ 300 users off of Cisco Unity and
2006 Oct 11
1
MGCP stuff
Hello everybody!
I have an Asterisk 1.2.12.1 server with SIP as the VoIP protocol.
What I want to do: I want to talk to the "outside world" via MGCP.
I suppose I must set an MGCP peer to route outgoing calls. So, I must
set the endpoint syntax of the Asterisk server (Asterisk will act as an
MGCP gateway and will talk with an MGCP Gatekeeper) and with other MGCP
gateways via
2005 Jun 30
5
wi-fi phone advice
Hi:
I want to connect a wi-fi phone to my Asterisk box
through a wi-fi AP so I can make voip calls.
please send me your recomendation about what wi-fi
phone I should be looking for. Anybody tried the
HOP1502 Wi-Fi IP phone. Its listed price $39.
Regards;
Chawki
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