similar to: Call pickup across technologies (SIP, IAX, MGCP)?

Just by the way, do anybody knows if call pickup of a call to a sip extension from a mgcp phone is supposed to work (even if sip keeps ringing). The scenary is as follows: 3@mgcp02 (ext 136) calls sip/julia (ext 133) and after It starts ringing 2@mgcp02 (ext 135) dials *8. Nothing happens, only 135 gets congestion tone, 133 keeps ringing and in the asterisk console I get: --

Hi I have some question about mysql realtime. 1) Anyone know exactly if there is a specific order to declare sip table column for realtime ? In which file can I find that order ? 2) In my extconfig.conf, [settings] are : sipusers => mysql,general,siptable sippeers => mysql,general,siptable so means that I use realtime dynamic exactly ? Is it normal if some parameters from sip.conf still

I was running benchmark on IO performance using iozone3. In my build, the dom0 resides on a small usb stick and all the storage comes from a NFS mount. I test NFS performance on both dom0 && domU, mounting from the same server. The dom0 test works just well, but the domU run suffers from unstable NFS mount. Since this is a NFS root, the domU just appear to be freezed. The log from both

I was running benchmark on IO performance using iozone3. In my build, the dom0 resides on a small usb stick and all the storage comes from a NFS mount. I test NFS performance on both dom0 && domU, mounting from the same server. The dom0 test works just well, but the domU run suffers from unstable NFS mount. Since this is a NFS root, the domU just appear to be freezed. The log from both

I have this in my config: exten => _800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/tollfree/1${EXTEN}) exten => _1800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/tollfree/${EXTEN}) exten => _NXXNXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/trunk/1${EXTEN}) exten =>

Hi Is there any way to set the presence state of a peer to in-use in asterisk 1.8? The idea is to integrate DND buttons on phones to BLF. Regards -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street

Hello, I'm trying to figure out what was the return code of SIP for a call. The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to retrieve the peer name using ${CHANNEL(peername)}, I have an error message that CHANNEL does not have peername or it is not available to be used. I tried to print it with NOOP on a live channel, and also after hangup, both with the same error

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

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

I tried your extension definition as suggested: exten => *98,1,Verbose(0,${CHANNEL(peername)} calling voicemail) same => n,VoicemailMain(${CHANNEL(peername)}@VoiceMail,s) same => n,Hangup But there was no change in the prompts asked, ie. the voice first asked for 'mailbox', and then 'password' as before. The prompts are not removed. Please clarify what you mean by the

Hello, I am using Asterisk voicemail on a CentOS 7 server. I would like to be able to remove the 'mailbox' prompt and 'password' prompt when a user tries to access their voicemail. I can remove the 'password' prompt by not setting a password for the user, but the 'mailbox' prompt is always heard. Please let me know how Asterisk can be configured to remove these

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

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

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

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

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

Hi, I have the following situation: I've installed Asterisk at Machine 1 (M1 - IP: 192.168.1.145) and X-Lite (X_lite-Xten-Win32-1103m.exe from www.xten.com) at Machine 2 (M2 - IP: 192.168.1.100) and Machine 3 (M3 - IP: 192.168.1.200). I need to catch the SIP and MGCP messages that will appear when M2 calls to M3 and vice versa. The SIP messages are working (I don't have problems with the