similar to: MGCP RFC (2705) vs. PacketCable MGCP spec

We are using an MGCP configuration. There seems to be some incompatibilities between our Netergy T2 VOIP chip and Asterisk. This is how our Vendor sees it: Here's what I see. 1. The first call is initiated. (CRCX) The interesting thing here is that the CA (Call Agent) tells us to go directly into sendrecv mode which means that we start streaming audio immediately. All other CAs that

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:

Hi group! I'm very interested on VoIP and trying to make some tests with 2 eMTA motorola (SBV5120) but with the PackectCable architecture I need a softswitch. My question is: is Asterisk and PacketCable compatible? If they are...., how can I configure it to act as a Call Management Server into my LAN? Which I have read from this mailing list is Asterisk with SIP but I believe that Asterisk

Hi group! I'm trying to make the first VoIP call in our LAN but I have some doubts. We have a Motorola CMTS (BSR1000) which has packetcable, and we have 2 eMTA Motorola to make some tests. I know I need a softswitch to make call management. My question is if this software is the appropiate to install it on my bussiness with packetcable to make some tests and the first call VoIP and say

Hi everybody, I have a MTA from Terayon that I try to make run with Asterisk using MGCP channel. The device is running with MGCP 1.0 NCS 1.0 Each time Asterisk try to send a Request (Request Notify, Audit Endpoint....) the device returns error 510 "Protocol Error" Does anybody have already meet this problem and provide me support to make run it ?! (I have already try to change

Sirs, According to RFC 2705 (MGCP), these are the parameters that are used in the transactions: ReturnCode, Connection-parameters <-- DeleteConnection(CallId, EndpointId, ConnectionId, [Encapsulated NotificationRequest,] [Encapsulated

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:

https://bugzilla.mindrot.org/show_bug.cgi?id=2705 Bug ID: 2705 Summary: Allow configuring syslog facility for the SSH client Product: Portable OpenSSH Version: -current Hardware: Other OS: Other Status: NEW Severity: enhancement Priority: P5 Component: ssh Assignee:

Hi Lately I've been having problem with the handsets on the polycom phones. The agents are complaining that the humming/buzzing can be heard on random days. When the agents touch their computer the buzzing may increase louder or reduced. Also when they grab the handset cable, the buzzing noise is less loud and is loud again when I let it go. I have tried changing the handsets, problem not

I have an intermittent problem with the one FXS line that I have. On most calls, the first ~5 seconds of the call has a loud buzzing noise on the line. After 5 seconds or so, it fades off to nothing, and the sound quality is great. Searching for "buzzing" on the list doesn't give a whole lot to work with. The buzzing happens on calls that are routed over both my FXO line and

Hello rsync people Today I was recovering data from a beginning-to-fail external USB hard disk. I started with my usual 'rsync -av --ignore-errors <source> <dest>', and that was fine until it got to the first I/O errors. It paused but continued after the first couple of errors, but then the disk started buzzing and rsync gave error messages for every file (I'm afraid

I've got a ShoreTel 210 MGCP phone drops calls. My packet capture indicates that the phone may be trying to renew its registration with *, but reports Restart Method of Disconnected (frame 2), then * seems to take that as a sign that it has lost the connection and closes things down. The phone, meanwhile, seems to think it can continue the conversation until a few ICMP "port

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, We have 15 Soundpoint IP 500 phones and recently call quality has deteriorated. On some calls there is a static-buzzing of sorts that occurs when users talk. It can be picked up on SIP-SIP calls and SIP-ZAPTEL (Channel BANK<<-T100P). It basically sounds really weird whenever someone talks, it sounds like a bee buzzing or something. Very hard to explain. Also, there will be echo

hi all I'm being spammed by these messages in the console (see below) and sound doesn't work with today's cvs. I rolled back a week, and it works fine. In addition to the sound problems, I had to enable inband dtmf squelch on the dilnk mgcp phones. if not, each pressed key was counted twice NOTICE[245776]: File chan_mgcp.c, Line 710 (mgcp_rtp_read): MGCP ast_dsp_process

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,

Hello everybody! I have an Asterisk 1.2.12.1 server with SIP as the VoIP protocol. What I want to do: I want to talk to the "outside world" via MGCP. I suppose I must set an MGCP peer to route outgoing calls. So, I must set the endpoint syntax of the Asterisk server (Asterisk will act as an MGCP gateway and will talk with an MGCP Gatekeeper) and with other MGCP gateways via

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

I have 1 IP phone (Swissvoice IP10S) and 1 POTS phone. When I dial the number for the IP phone off the POTS phone, the IP phone rings. But when I pick up the handset on the IP phone, I get a busy signal and this message on *: Aug 23 09:38:57 NOTICE[1142106560]: chan_mgcp.c:2243 handle_response: Terminating on result 502 from svip10@00059002042b-1 Here is the entire session. svip10 is the 1 and