similar to: swissvoice ip10s

I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Any ideas are appreciated. Robert mgcp.conf is: [general] port = 2427 bindaddr = 192.168.0.110 [ip10] host = 192.168.0.5 context = from-sip line => aaln/1 The portion of extensions.conf is: exten => 3001,1,Dial(MGCP/aaln1,20) exten => 3001,103,Hangup

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

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 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 had success getting the Swiss IP10S and the SIP ( IP10 SP v0.0.1 (Build 4)) firmware working with Asterisk? If so do you have copies of what worked in sip.conf and phone configuration files? I can't seem to get the phone to register, it tries but is denied with a Forbidden (which I am guessing is authentication). I tried without a secret, but the phone seems to use swissvoice

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 there, here some questions and experiences after playing for one day with 3 Swissvoice ip10s and the latest * CVS: QUESTIONS: - what is the user option "enter voice mail number" good for? It doesn't appear to be of any practical use - does anyone have some Swissvoice info that I cannot find on their web site like the guide to MGCP XML (.svd), guide to configuration file

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 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

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

Hi there, I have serious problems with MGCP and Swissvoice ip10s, and it appears that recent CVS also introduced trouble for other MGCP users. Please check and add comments in the bugtracker so that we can get a clearer picture - thanks! Also comment if things are working fine for you. http://bugs.digium.com/bug_view_page.php?bug_id=0001542

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

Hi there, after testing with a MGCP phone (Swissvoice ip10s) I found the following ASTERISK-based codes (VERTICAL SERVICE CODES) to work - I assume that most of those will also work with SIP, but haven't checked that yet: *67 - Calling Number Delivery Blocking *70 - Cancel Call Waiting *72 - Call Forwarding Activation *73 - Call Forwarding Deactivation *78 - Do Not Disturb Activation

Funny - I downloaded the latest Asterisk CVS, and it's pretty much working. Will report when I have some more success. PaulH -----Original Message----- From: Philipp von Klitzing [mailto:klitzing@pool.informatik.rwth-aachen.de] Sent: Tuesday, 28 September 2004 9:46 PM To: Paul Hales Subject: Re: [Asterisk-Users] Leader IP10S Hi! > I have been lent a Leader IP10S phone (SIP) for

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

Hi, everyone: I'm in the processing of deciding what IP phones we should use with our Asterisk server, and I wanted to get feedback from the user community on the quality, reliability and ease of operation of Snom phones. What do you have to say about these phones? Are there other phones you'd suggest along with or instead of Snom? Thanks, -Stephen-

I'm not sure that the problem lies in the NAT because the phone is talking to Asterisk. I'm hoping this is a simple config thing I've overlooked but I've tried all kinds of combos inside the [] in my mgcp.cfg file. The phone's IP is 192.168.1.116 (my comp is .110). The router to which the phone and my comp is plugged into has a WAN IP of 10.0.0.28. All the other comps (and SIP

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

I have this showing on my cli while being in a call. Then connection gets broken. Can someone tell me what it means ? Dropping frame of G.729 since we already have a VAD frame at the end Thank you in advance. Bartosz

I noticed when I call certain IVR systems, such as 1800calldhl, that Asterisk will not barge the prompt. Would this imply that Asterisk has an Early media detection problem. Is anyone else experiencing this problem. Is there a fix? Kurt