Displaying 20 results from an estimated 1000 matches similar to: "Not call pickup for call to sip from mgcp phone"
2003 Dec 29
1
transfer with MGCP
Hello,
I`m try to make the attended transfer work Dlink DG-104S via FLASH, when
somebody calls my phone I pickup and press flash to get a second line to
call another extension. When I press flash I hear no dialtone, and only
a long and then small beep. When I try to dial digits I hear again those
long+short beeps, but the extension dialed is not ringing. If I pres
flash again I get back to
2004 Jun 29
0
MGCP and call waiting, doesn't work.
Hey guys, can you shead some light on this?
I will copy my mgcp.conf and post below, but here is the problem.
I can't get call waiting to work with my MGCP device. I already have one call going, and I can hear the second call come in, I flash over to it, but all I get is a dial tone, * puts the 1st call on mute/hold, but I never get the second, and it terminates. I flash back over and pick
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 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 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
...
2003 May 22
0
MGCP NOTICE message and WARNING messsage
> Hello all,
>
Can someone help me on the problem which I have on MGCP phone test . I test
mgcp - asterisk- zap. But I got several NOTICE message from rtp.c.
> NOTICE[20501]: File rtp.c, Line 221 (process_rfc3389): RFC3389 support
> incomplete. Turn off on client if possible
>
> -- Endpoint 'aaln/1@VG101-1-1' observed '9'
> NOTICE[20501]: File rtp.c,
2003 Oct 22
0
MGCP error for Cisco 7750 FXO card
Can anyone tell me what MGCP error that I'm getting means?
The hardware is a MRP200 in a Cisco 7750 PBX. (Its a FXO blade with 2 slots, first one has a 4 port FXO card and the second has 2 port FXO card. It recognises those correctly, at least to the point of this error.)
MGCP Debugging Enabled
MGCP read:
NTFY 13 aaln/S0/SU0/0@MRP200-S1 MGCP 0.1
X: 1adace42
O: L/hd
from
2005 Aug 17
1
Comfort Noise incomplete - No translator path exists for channel type MGCP (native 4) to 256
I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC. I listed my conigs below
I go off hook
2009 Dec 17
2
Integrate a CPE with Asterisk in MGCP
Hello all,
I'm looking for some help to try to understand why my CPE doesn't work
good with Asterisk in MGCP.
Here is what I want to do :
- Register a TECOM AH4021 on Asterisk in MGCP with the following profile
in mgcp.Conf :
[general]
port = 2727
bindaddr = 10.95.20.1
disallow=all
allow=g729
allow=alaw
020202020202]
context=mgcp
host=dynamic
canreinvite=no
dtmfmode=rfc2833
nat=yes
2005 Aug 15
2
No translator path exists for channel type MGCP & Comfort noise support incomplete
ONLY ON MONDAY!
Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this?
2004 May 17
0
mgcp with busy tone
Hi there,
::: i have an mgcp phone (iph-90) using the sample mgcp.conf for it
(the dlink section). i've tried both asterisk stable and development
release but i'm getting the following error when i lift the receiver:
. .. in stable branch:
-- MGCP mgcp_new(MGCP/aaln/1@gw52302432-1) created in state: Down
while the phone is giving me busy tone
. .. in development release:
2003 Sep 13
0
# during ringing causes Asterisk to crash!
*This message was transferred with a trial version of CommuniGate(tm) Pro*
Hey all,
Just noticed something that might be an issue. I have just made
asterisk crash consistently by doing the following.
I have a D-Link DG1102s running MGCP into asterisk and an extension *9
setup which dumps me into my inbound context to simulate calls coming in
from my X100P. This usually works with no hassles
2006 Mar 10
0
Flash call transfer problem
Hi,
I have some problems transfering call from phone to phone with my Asterisk. When I dial Flash I can hear for half a second the dial tone, but it stops suddenly. The other phone hear the on hold music and pressing flash key another time I get back to the previous channel.
On the asterisk consolle seems to be all ok, this is whant I can read:
asterisk1*CLI>
-- Swapping 0 for 1 on
2004 Apr 12
2
SwissVoice IP10S not able to dial calls
I have set up a new SwissVoice phone and it can receive calls but I cannot
make calls out from it. The setup is simple for now, 2 phones: SwissVoice
is ext 7726 and Cisco 7960 (SIP) is ext 7999.
I can call from the Cisco phone and it rings on the SwissVoice phone but
when I dial from the SwissVoice phone I get a busy tone upon dialing the
second digit. The log reads as follows:
-- Endpoint
2004 Aug 18
2
Festival Installation - Asterisk 1.0-RC2 && Debian Woody
Hey All,
Thought I'd take a bash at trying to get Festival to work here on my lab
system with the aim of using it to create our IVR menu prompts. I've
spent most of the afternoon searching through the Wiki, the Festival
website and Google and I've got a couple of questions.
First one is that the 'Asterisk+festival+installation' page on the Wiki
mentions the RedHat 9 RPMs
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 29
1
Can't place a call with MGCP Phone
Hello,
I have just received an MGCP Phone for test purpose and I can't place a
call from my MGCP Phone.
I can call my MGCP phone from a SIP Phone. Here is my mgcp.conf:
;
; MGCP Configuration for Asterisk
;
[general]
;port = 2427
;bindaddr = 0.0.0.0
;[dlinkgw]
;host = 192.168.0.64
;context = default
;line => aaln/2
;line => aaln/1
[192.168.10.10]
host = 192.168.10.10
context =
2004 Sep 23
0
asterisk-1.0.0 woes
Is this a new bug, or am I doing something wrong here... ?
Same config files as I have always been using, maybe there is a config file
change needed?
I have been running CVS for a while and just upgraded to 1.0.0
Now I get an RTP error with MGCP, MGCP dies afterwards.
Sep 23 11:28:33 WARNING[98310]: rtp.c:711 ast_rtp_offered_from_local: rtp
structure is null
-- MGCP
2004 May 13
0
MGCP channel problem
Hello
I have a problem with my MGCP voice gateway.
I use D-Link DG104S
Boot PROM Version 3.0B38-D
Firmware Version 3.0T86-D
I tried asterisk v 0.7.2 and I am using latest CVS version now.
When I dial a number very fast, or when I use a redial function, my asterisk receives coupled digits.
My co-worker called number 245005111, these are a few lines of my debug.
The identifier of first digit
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,