Displaying 20 results from an estimated 1000 matches similar to: "Integrate a CPE with Asterisk in MGCP"
2007 Apr 19
1
AudioCodes MP-104 MGCP?
Greetings;
We are trying to get Asterisk up and happy at our site-we tried VOIP
using Sphere about a year ago, spent a *boodle* on expensive hardware
and services from a local "expert", but it never was happy.
I'm brand-spanking new at VOIP, and I've learned a *ton* getting
Asterisk breathing in the last couple of days. I have three Polycom
Soundpoint IP 500 SIP phones, which
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 Jan 22
2
MGCP Problem.
Hi.
I'm new in Asterisk with MGCP. I set up a MGCP user agent and start asterisk
with the next configuration files.
'--------------- extensions.conf
----------------------------------------------------
[general]
static=yes
writeprotect=yes
[globals]
ap1 => mgcp/aaln/ap200@64.76.148.186
[macro-apl1]
exten => s,1,Dial(${ARG1},30,Ttmr)
;exten => s,2,Voicemail(u${MACRO_EXTEN})
2003 Sep 24
3
Dlink DG-104S (chan_mgcp) and configuration w/Asterisk
I have a DG-104S (which I reset to factory settings, it's DHCP'ing off my
network, plugged into the WAN port). The system comes up, and I through the
web browser set under Call Agent IP Address to:
Notify Entry: dlinkgw@[192.168.1.1]:2427 (192.168.1.1 is the * server)
I have RGW Name: and DNS IP addres the DNS IP of the MGCP box and DNS State
disabled (not sure what to set it to) --
2003 Oct 20
1
mgcp transfer takeback with ata186 (logs with comments - long post)
Hi,
in following of a recent discussion I got to work on MGCP with the Cisco
ATA186 again, and got it to work very nicely. However, there is a little
thing with transfers I would like to get comments on:
Call comes in from PSTN and goes to an ATA186 (MGCP)
Call is answered and then, using flash, transferred to another extension
If the extension is available, there can be an announcement and
2004 Jul 31
3
MGCP & Cisco ATA 186 Help
Does anybody has the expirience configuring Asterisk with Cisco ATA 186
MGCP firmware ?
I have Cisco software v3.1.1 atamgcp (Build 040629A)
Asterisk 1.0-RC1
On ATA i only put domain test.
mgcp.conf looks like this
[test]
host = 192.168.195.55
context = default
line => aaln/2
line => aaln/1
Asterisk CLI shows this:
Jul 31 16:05:40 WARNING[135449600]: chan_mgcp.c:485
2004 Oct 15
1
Asterisk crashes on special Transfer with MGCP/ATA 186
Hi all,
i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco
ATA-186 3.1.1 atamgcp
We are used to make an special ;) blind transfer like
(Flash)Number(Hangup before anyone answers or ring).
Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp
If one waits until the last one rings, then hangup, everything is fine.
If one waits until the last one
2004 May 19
2
MGCP error dialing
I am trying to dial a mgcp extention from my sip phone and i am getting this
error message. anyone got any idea?
error
I> -- Executing Dial("SIP/2204-5dc2", "MGCP/aaln/1@10.0.1.150") in new
stack
May 19 22:30:01 NOTICE[1251156800]: chan_mgcp.c:1104 find_subchannel: Gateway
'10.0.1.150' (and thus its endpoint 'aaln/1') does not exist
May 19 22:30:01
2005 May 10
3
MGCP : chan_mgcp.c:1509 find_subchannel
When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just
keep getting this message every 30 seconds or so :
May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its
endpoint '*') does not exist
Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets
to
2010 Mar 05
0
MGCP FXO endpoint
I have a fxo endpoint installed in a Cisco router. I would like in my
dialplan to get an extension call a telephone number through that fxo
endpoint.
Since with zaptel channels it is done like:
exten => 0999,1,Dial(DAHDI/2-1/111) --> being 111 the phone number I
want to call.
I thought that for mgcp it would be the same, and I did:
exten => 5200,1,Dial(MGCP/aaln/S0/SU3/0 at
2003 Dec 24
8
G729 troubles
Hello,
I've successfully installed Asterisk from last CVS and configured it
for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip
server.
All are work fine at G711 codecs, but then I disable all codecs except
g729 some calls failed (Not all calls. Some calls passed at g729
succesfully).
All my devices configred to use only g729 and I don't see other codecs
at mgcp or sip
2005 Mar 16
1
MGCP Channel Lockup and other probelms
Hi All,
I'm trying to hook up asterisk (CVS-HEAD-02/09/05-13:44:11 ) to a ADIT
600 via MGCP. Got it working really nice but now have a pretty bad problem:
1. When I perform a flash on the telephone, I usually get a second
dialtone, but when I dial, dialtone doesn't break. If I flash back and
forth a few times, it will eventually give me no dialtone.. here if I
dial, it successfully
2003 Apr 24
3
new mgcp patch errors
see below
I tried to call 98013356 from the following phone (from mgcp.conf)
[iptlf03]
host = 192.168.33.3
context = default
inbanddtmf = 1
callerid = 22545062
line => aaln/1
Console output:
== Spawn extension (capiring, 9988001133335566, 1) exited non-zero on
'MGCP/aaln/1@iptlf03-1'
-- MGCP mgcp_hangup(MGCP/aaln/1@iptlf03-1) on aaln/1@iptlf03
-- Delete connection 4
2009 Jun 30
0
Asterisk & Adit 600 Configuration
Has anyone ever gotten an Adit 600 to work with Asterisk1.4 via MGCP.
Asterisk keeps giving me the following error in the LOGs:
[Jun 30 08:32:59] NOTICE[26785]: chan_mgcp.c:1726
find_subchannel_and_lock: Gateway 10.0.0.245' (and thus its endpoint
'*') does not exist
MGCP Config:
[AFSWestAdit600]
host = dynamic
context = default
canreinvite = no
threewaycalling = yes
2003 Jul 11
3
mgcp problems
I strange error messages when using mgcp and ata186 .
This session is simply dial into 600 demo extension - echo test
...
Handling request 'NTFY' on aaln/1@10.0.1.19
Transmitting:
200 29 OK
to 10.0.1.19:2427
-- Endpoint 'aaln/1@10.0.1.19-1' observed '0'
-- MGCP Asked to indicate tone: on aaln/1@10.0.1.19-1 in cxmode:
sendrecv
Posting Request:
RQNT 306
2010 Feb 25
1
Deadlock while using MGCP on Asterisk
Hello all,
I'm running Asterisk 1.2.35 with chan_mgcp activated.
The process host around 2,4K users.
Along the day I've got some debug reports like :
Feb 24 22:25:42 DEBUG[28546] channel.c: Avoiding deadlock for
'MGCP/aaln/1 at 028421223635-1'
Feb 24 22:29:04 DEBUG[28670] channel.c: Avoiding initial deadlock for
'MGCP/aaln/1 at 028421223635-1'
Then, at
2004 Aug 19
1
No Success with SwissVoice.
I'm not sure that the problem lies in the NAT because the phone is talking
to Asterisk. I'm hoping this is a simple config thing I've overlooked but
I've tried all kinds of combos inside the [] in my mgcp.cfg file.
The phone's IP is 192.168.1.116 (my comp is .110). The router to which the
phone and my comp is plugged into has a WAN IP of 10.0.0.28. All the other
comps (and SIP
2004 Dec 22
1
MGCP Transaction identifiers
I know this is not the most appropriated list to this, but I will try:
Does anyone know what is the criteria to the generation of the transaction identifiers in MGCP? I mean, are they generated by a randomic method?
I'm using Asterisk and MGCP eyeP Phone and observed that the RSIP and NTFY (methods created by the gateway) use high values in the transaction identifiers, while the RQNT, AUEP,
2003 May 19
1
MGCP and Cisco ubr924
I've been trying to figure this one out for a while, but to no avail.
I have my cisco ubr924 setup for MGCP with Asterisk as the call-agent. I have manually registered the endpoint in mgcp.conf. When I pick up the phone, I get no dialtone and debug shows errors. IOS on the ubr924 is 12.2.
Any help is appreciated.
from mgcp.conf:
[ubr924]
host=65.37.86.203
context = from-sip (just as a
2005 May 12
1
Asterisk with ShoreTel 210 (MGCP)
Okay, so I'm a noob.
Asterisk looks very promising, so I say "thanks" and "good job" to all who
contribute.
My * test box is up and running with soft phones using IAX and SIP, so now I'm
on to testing hard phones.
I borrowed a couple ShoreTel 210 phones from somebody who had them on hand but
they only support MGCP. I see that there's an mgcp.conf in