similar to: 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

Hey list, Does anyone have a current working config example of a cisco ubr924 and * ? I think the 924 only supports MGCP. I want to get VoIP on this device, I was wondering if anyone has already tackled the problem, if not, I'll go in blind :) Thanks Duane Cox -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch the hangup. I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like it such that I call from * via E1/PRI to the cisco, and call out via R2 to

I repost a message I put a week ago: I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? Also I recently

Hey all, I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get a bit of help here. First I'll explain my setup, and then my problem. Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO ports. I have an analog phone line plugged into the first port (voice-port 1/0/0). I've got it setup so that calls coming into that analog line are

Dear Researchers, I wrote two function to fit a circle using noisy data. 1- the fitCircle() is derived from MATLAB code of * zhak Bucher* from the link http://www.mathworks.com/matlabcentral/fileexchange/5557-circle-fit/content/circfit.m 2- the CircleFitByPratt() from MATLAB code of *Nikolai Chernov *from the link

I have a problem moving from multistratum aov analysis to lmer. My dataset has observations of ampl at 4 levels of gapf and 2 levels of bl on 6 subjects levels VP, with 2 replicates wg each, and is balanced. Here is the summary of this set with aov: >> summary(aov(ampl~gapf*bl+Error(VP/(bl*gapf)),hframe2)) > >Error: VP > Df Sum Sq Mean Sq F value Pr(>F) >Residuals

Hi folks I seted up the asterisk with an active ISDN B1 AVM Card (german vendor) and it works fine, various SIP clients (IP fon snom, xlite under MacOSX) and also incoming and outgoing connectins. Ok. No problem. After that I configured a CP7940G with a MGCP IOS. It's connected to the asterisk too via switch. No NAT deivce between. And of course, I can call out to PSTN and also to one of the

Hi there, I'm brand new to R, so let me know if this question is not appropriate for this list. I've been reading through the documentation and have tried a number of things, but am pretty much stuck so far. Here's the session info: > sessionInfo() R version 2.6.2 (2008-02-08) i386-apple-darwin8.10.1 locale: C attached base packages: [1] stats graphics grDevices

Full_Name: Martin O'Gorman Version: OS: Submission from: (NULL) (84.176.63.149) I have been looking at the PPC package and have a question. As the input data is comma separated, shouldn?t the command to read in the raw (no batch) mass spec data indicate that sep=?,? (marked below) ? Otherwise, the data read in is the pair of values (m/z,intensity). It is not obvious why that should be.

Hi Ben, I've read your message in the syslinux forum and I see that you got the IOS c820-k9osy6-mz.122-15.T5.bin, for Cisco 827 Series IOS IP/FW Plus IPSec 3DES IP/FW Plus/ IPSec 3DES c820-k9osy6-mz 8 MB 24 MB I need to use URLFILTER on my Cisco 827 and that is the one of the few IOSs that can do it. There is any way of you e-mail me a copy of that IOS?

Does ctrl+c work? My experience with (non klibc) init=/bin/bash is that getting ctrl+c to work requires some jumping through hoops, because running as pid 1 doesn't really work. If I recall correctly you have to call setsid, and open a tty as a controlling tty, you can't call setsid as pid 1. The hack that I got to work (for reference, I'm using this for raw kernel testing

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

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

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

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

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

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

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

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