similar to: MGCP issue

Hello all, Is X-lite soft phone support G.729 ? I actually use it but there is no G.729 support. Anyone know where to have it? Regards. Daniel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050209/8cdbeeec/attachment.htm

Hello, This is the scenario i want to setup: Cisco ATA 186 -----------> SER -------------> Asterisk I want the Cisco ATA to register to Asterisk through SER. when the Cisco ATA place a call, SER querry a data base (MySQL or else), and if there is a valid Account for the ATA, the call go to Asterisk. Did someone know how to set SER to work like this with Asterisk? which version of SER

Hi all, I am configuring asterisk in a cable modem network, using a motorola TM401A. I can make calls from the MTA but I can receive, display the following error: -- Executing [1500 at alberti:1] Dial("OSS/dsp", "MGCP/aaln/1 at 0-13-11-82-bd-a.ssw.intercal.net|30") in new stack [Sep 2 00:10:53] NOTICE[28062]: chan_mgcp.c:3572 mgcp_request: Asked to get a channel of

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

Hi, Anybody out there running Adit600s? I have in my care an Adit600 channel bank connected to an old (version 1.0.6) Asterisk instance with MGCP. When trying a more recent Asterisk (1.4.21.2~dfsg-3+lenny1, Stock current Debian) calls fail. I have attempted to add the "slowsequence = yes" line to mgcp.conf. (It seemed to be the only likely candidate in the example files I found

Good Morning, I'm having an issue with callerid display when calles are placed _from_ an mgcp device (8x8 ata w/mgcp firmware). Internally, there are several different sip devices and one mgcp device. Calls from any of the sip devices to any other device (sip or mgcp) have name/number displayed properly by the called party's phone. Calls from the mgcp device to any other device display

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 ...

Hello I got the similar error while trying a call. -- Executing Answer("SIP/8000104-86ef", "") in new stack -- Executing Wait("SIP/8000104-86ef", "2") in new stack -- Executing AGI("SIP/8000104-86ef", "areskicc.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/areskicc.php areskicc.php:

I am having problems trying to get R to graph data input that is log-normal on the horizontal (x) axis. The data is log (base 10), and I am more interested in viewing the tails of the distribution. The closest I can get with this is log on the vertical (y) axis and linear on the horizontal axis. Note that the x-axis is Centile (0%-100%) and is at this time linear, however I want it to be

R 2.1.1 Win 2k Would someone suggest a method (or methods) that can be used to determine ntile cutpoints of a vector, i.e. to determine values that can be used to divide a vector into thirds such as 0-33 centile, 34-66 centile, 67-100 centile. If for example I had a vector: 1,2,3,4,5,6,7,8,9 and wanted to divide the vector into thirds I would have cut-points of 3, and 6. Thanks, John John

I'm wondering if there is a simple way to assign a quantile to a vector in a data frame, much like one could in Stata using centile. Let's say I want 100 slices in my assignation. I can easily see what the limits of each slice by using quantile: quantile(my.df$my.var, probs=seq(0, 1, 0.01)) But how do I assign the appropriate value to each row/record in my data frame? Clearly the

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 5870 bytes Desc: image002.jpg Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050129/e53b0567/attachment.jpeg

I am having problems trying to get R to graph data input that is log-normal on the horizontal axis like the example I have attached and is also below. The data is log (base 10), and I am more interested in viewing the tails of the distribution. The closest I can get with this is log on the vertical axis and linear on the horizontal axis. This is what I have so far: plot(x,y,type="l",

Hello, We had a strange thing happen today - we shut down our system (virtual machines) so that some work could be done in the server closet and everything was shut down gracefully but when everything was started back up I can no longer access my file server from windows. I get the following message in windows: \\fs1 is not accessible. You might not have permission to use this network

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 =

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

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

Hi, I would like to customise an end user application like Centiles's callpad software (http://www.centile.com/solutions-applications-callpad.php ). Its purpose is to allow users to set or read various personal phone-related parameters (call history, voicemail settings, conference, ...) instead of using phone keys combinations. Are you aware of any software that could be used for this ?

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})

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) --