similar to: No Success with SwissVoice.

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

I had the telnet window to the phone open by chance and noticed this line twice when I tried to call the IP10: WARNING: may need to undo rtp port operation here The warning line appeared immediately when I picked up the handset. I have no idea what this means. I also tried calling the phone from a POTS phone and I got the same warning. What is even better is that this coincides with the

Does anybody has the expirience configuring Asterisk with Cisco ATA 186 MGCP firmware ? I have Cisco software v3.1.1 atamgcp (Build 040629A) Asterisk 1.0-RC1 On ATA i only put domain test. mgcp.conf looks like this [test] host = 192.168.195.55 context = default line => aaln/2 line => aaln/1 Asterisk CLI shows this: Jul 31 16:05:40 WARNING[135449600]: chan_mgcp.c:485

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

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

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,

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Does anyone know what I need to put in the mgcp.conf to connect to an adit 600? Also if you know what I need to configure on the Adit600 itself, that would help too. - --Tod -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

Hi I've a mgcp fon (swissvoice IP10s) behind a NAT router. Configured is NAT for both in/out going on port 2427. Now I got the following mgcp debug messages when i try "mgcp audit endpoint <endpoint>" ---------------------------------- from 172.16.98.57:2427 Verb: 'RSIP', Identifier: '5346', Endpoint: 'aaln/1@[192.168.2.3]', Version: 'MGCP

I could use some skinny/Cisco help here. Was finally able to get the phone registered to * but whenever someone tries to call that phone it freezes and reboots itself. Same thing happens when you pick the handset up off the 7960G; it locks and reboots about 5 sec later. Here is what * shows when I plug the phone in: -- Starting Skinny session from 64.72.107.1 Device SEP000F3442E4A7 is

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) --

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

Hello, Now that I have a nearly working configuration for my IP10S with * I wonder if anyone has done call transfert with this Phone. In the IP10S documentation they talk about the 'service key' wich is the key with the white dot on it. With this Key, it should be possible to have a menu with call transfert entries. This menu should (accordingly to the documentation) depend on the

Hi there, tonight I upgraded one of my ip10 phones to "Appli 1.0.0 build 3" with "Boot 0.3.6" This firmware is dated Dec.19, 2003. For Robert (aka "info lists") this fixed the multiplied digits problem (which I never had). Since there is no website of Swissvoice I guess you'll have to e-mail them if you want upgrade... Unfortunately the upgrade did

taken from bug 881 (now resolved) :-( ---------------------------------------------------------------------- markster - 05-19-2004 09:21 CDT ---------------------------------------------------------------------- As it turns out the 10S cannot conference on the device. From Jean-Francois at Swissvoice: Hi Mark, IP10S have not the capabilities to mix by itself 2 RTP flows, that why it refuses

Hi, Does anybody know the place to download the firmware for swissvoice ip10s I have several phones with application IP10 H3 v1.0.0 (Build 1) I'm looking for newer H.323 and also MGCP firmwares Are the SIP firmware available, according to web its targeted to Q1 2004, but we have week left in Q2 I sent several email to swissvoice support,, no answers Regards Juri

I need to purchase approx. 10 phones for a small office implementation. Nothing fancy is required besides a full-duplex speakphone, in the sub-$200 range. I am currently looking at the Polycom Soundpoint 500, Uniden UIP-200 and SwissVoice IP-10. I have searched around and found various posts regarding each phone's ability to work with asterisk (SIP, I probably should have mentioned), but

Hi Does anyone have a howto on how to set one of these up on Asterisk or Trix box please? I can make it SIP or MGCP so whatever you have ;-) I have found one page but it isn't really a howto setup Thanks in advance Paul -------------- next part -------------- An HTML attachment was scrubbed... URL:

Has anyone used the Swissvoice IP 10S (www.swissvoice.net) VoIP Phone with *? Adrian -- Adrian Walker adrian@digitaltraffic.co.uk ======================================================================= This email has been scanned for Virus infection by MessageLabs For more information please contact messagelabs@atomwide.com

I have set up a new SwissVoice phone and it can receive calls but I cannot make calls out from it. The setup is simple for now, 2 phones: SwissVoice is ext 7726 and Cisco 7960 (SIP) is ext 7999. I can call from the Cisco phone and it rings on the SwissVoice phone but when I dial from the SwissVoice phone I get a busy tone upon dialing the second digit. The log reads as follows: -- Endpoint