similar to: Asterisk & MGCP / NCS

Hello guys, I ask you to share your experience with your BudgeTone 100.... I have my asterisk @ work and I've bought a GrandStream BudgeTone (SIP phone) and I usually use X-Lite I have plugged my BudgeTone into my home network because I want to be called even at home. I succeed to register my X-Lite with Asterisk from home but I can't do that with my BudgeTone. (I don't know

Perhaps stupid question but, is Asterisk a statefull or stateless proxy? Ignace

Hi everybody, I would like to create SIP call flow Diagram under Windows. Is anybody know a program to perform it? I have already Ethereal and I would like an explicit diagram just to show where something have problems... Thanks Ignace

I try to connect an MGCP device(Terayon) to asterisk. I have found many example BUT the Terayon always return error 510 ! "Verb:'510' Identifiers :'2' Endpoint: 'Error' Version'(null)' Accoding to RFC: 510 = The transaction could not be executed because a protocol error was detected. BUT I have read that somebody ( Roy Sigurd Karlsbakk) have

Does anybody can explain me what is early B3? Thanks! Ignace

Hi, I've installing a AVM Fritz Card in my ASterisk Box I've configured everything and its running perfectly. The problem is that everybody is allow to call through it. Explaination: All users registered in Asterisk can make a call towards the ISDN network But, everybody from the Internet, knowing the extension of CAPI in the dialplan, can call through my Asterisk to any phone

Hi guies, I know I know this subject have been The most written subject about VoIP Right... but I just want to make clear, just one time ! If Asterisk is on a Public IP Address and a softphone behind the nat, sip.conf must contains for this phone: nat=yes .... Now if I want to configure my sipphone (X-Lite) placing behing the NAT, it must have in "Domain/Realm" the external IP

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

Hey, I''m looking for a way to get the content of my div in my Rails view. I would like the literal HTML content, so only HTML and no Rails I thought inner_html would work, but I get a RJS exception (TypeError: $("content").innerHtml is not a function) Does anyone know how I can get the content? Thanks! In my controller I use this code: [code] def save_test render :update

Hi group I wrote 2 years ago to know if there is some workaround for PacketCable. Since then I got no answer and now I hope there's something about. Is there any chance to use Asterisk as softphone with cable modem technology using Packetcable? Thanks in advanced Carlos Bernat -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've looke throught the archives and can't find much to help my situation. I have a Slackware 10 server running Samba 3.0.4 on a Win2k/2K3 Active Directory Domain. My win2k clients can connect to the samba shares but the WinXP clients cannot. I've tried the enableplaintextpassword hreg hack but that doesn't solve the issue. Here is my [global] section of my smb.conf Can anyone

We are working on a new implementation of asterisk. We are using a fiber-served WorldWide Packet switch at the home that incorporates a VOIP T2 switch that feeds 2 POTS connections. We are told that the T2 is programmed with code that follows the PacketCable spec. This version has a problem with the 'congestion' message which is based on the MGCP RFC (2705) spec. This causes the T2 to

Hi, I'm trying to connect asterisk on FreeBSD and mtas. I have registered endpoints, but after I pick up the receiver I hear nothing, there is no signal. There is a message: -- MGCP mgcp_new( endpoint ) created in state : Down I'm concerned about messages like chan_mgcp.c:372 __mgcp_xmit: mgcp_xmit returned -1: Address family not supported by protocol family Can asterisk work without

Hello, Is there someody who can explain me the meaning of these sentence. "Sip is philosophically horizontal and H.323/MGCP are vertical" Thank U (if you have some links to share about this protocols, share it :) ) Ignace

Hi, I tried to establish NCS call b/w 2 endpoints of same PacketCable MTA using asterisk-10.0.0. I observed that MDCX sent to aaln/1 contains its own SDP. Some I observed with aaln/2. So voice path is not established b/w aaln/1 and aaln/2. My Configurations: mgcp.cong: [mta84.globaledgesoft.com] host = mta84.globaledgesoft.com wcardep = aaln/* callwaiting = 1 ;canreinvite = 1 dtmfmode

Hi, I have cable modems Arris with MGCP protocol. And I need PacketCable NCS patch for Asterisk. http://asterisk.urtho.net/ doesn't work! -- Pagarbiai, Giedrius Augys Siauliu Universitetas, IST IP telefonijos inzinierius Tel. 8 41 590408 Mob. Tel. 8 678 05790 el. pastas voipas@gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the

Hi, I picked up a router with 8 voice ports that supports MGCP, but it has several options that I am not familiar with or do not seem apparent in the mgcp.conf. Enter the default IP address for the Notified Entity: [0.0.0.0] Enter the listening port of the Notified Entity: [2427] Enter the IP address for MGCP signalling (Data): [192.168.0.210] Enter the local port for MGCP signaling (Data):

I have been fooling around with DynExtenDB and run into two glitches. 1) The code is looking for (chan->dnis) and in my case I find (null). I forced (chan-dnis) to be the same as (chan->exten). So far so good. Now I can connect and talk. This lead me to the second glitch. 2) As soon as the call ends by hanging up, the code issues a (ast_spawn_extension). This causes asterisk to drop

Hi, We have been working with samba 3.0.4 and MIT 1.3.6 on SUSE SLES 8 (SP3) for some time. We use security=ads and our samba is joined to the W2K3 domain. We use this with winbind to use the authentication from w2k3 instead of local users (telnet, squid, samba and via smbauth also apache). When preparing a new server with suse SLES 9 and heimdal 0.6.1 RC3 (suses package) we started running