search for: 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...

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 witho...

hallo, does anybody successfully managed to get swissvoice ip10s with h323 firmware work with asterisk ? mgcp firmware works fine, but with h323 i'm still getting one way audio. regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------...

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...

...st 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 evaluation. > No luck with making or receiving calls - the phone registers OK though. > Has anyone had any experience with these phones? Sounds like a re-branded Swissvoice IP10S. Search the Wiki and/or the list for Swissvoice. You...

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 f...

....24.112] context=local host=10.1.24.112 callerid = "Brad Chilton <7726>" callgroup=0,2-5 canreinvite=no pickupgroup=0,1 nat=no threewaycalling=yes transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer ;callwaiting=yes ; this might be a cause of trouble for ip10s ;cancallforward=yes line => aaln/1 EXTENSIONS.conf =============== exten => 7726,1,Dial(MGCP/aaln/1@10.1.24.112,20,tr)

...1 (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 the conference mode. Most often, with the different partner we have, the conference capability is managed by a MCU on the network, then on the phone side, it's transparent. Regards, Jean-Francois If you want you can o...

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

Hi, Is there a way to use swissvoice IP10 in MGCP mode behind NAT. I already use this fonctionnality with a PBX other then asterisk, and it work very well. In fact, I need to use this settings in the IP10 telnet session set xgcp rgw_name MAC_ADDR set tcid 0 notify_entity ca@[@IP PBX]:2427 In fact, the phone register with the MAC@ and not with the IP@, and the PBX reply the phone in the UDP or

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

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 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:

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 http://bugs.digiu...

...lt host=192.168.0.241 wcardep=* line => * ; ; Line 1 ; callerid = "John" <123> callgroup=0 pickupgroup=0 nat=no threewaycalling=yes transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer callwaiting=yes ; this might be a cause of trouble for ip10s cancallforward=yes line => aaln/1 ; ; Line 2 ; callerid = "Jane" <124> callgroup=0 pickupgroup=0 nat=no threewaycalling=yes transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer callwaiting=yes ; this might be a cause of trouble for ip10s...

...2727 bindaddr = 10.95.20.1 disallow=all allow=g729 allow=alaw 020202020202] context=mgcp host=dynamic canreinvite=no dtmfmode=rfc2833 nat=yes threewaycalling=yes transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer callwaiting=yes ; this might be a cause of trouble for ip10s cancallforward=yes line=aaln/1 - Place outgoing calls through a SIP PROXY : OK, but the call goes down after some seconds : -- Executing Answer("MGCP/aaln/1 at 020202020202-1", "") in new stack -- MGCP mgcp_answer(MGCP/aaln/1 at 020202020202-1) on aaln/1 at 02020202020...

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 Deacti...

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@...

Scenario: [Asterisk Server] on routed/public IP \/ /\ \/ /\ \/ /\ \/ /\ \/ [Draytek Router] --> internal IP's \/ /\ \/ /\ \/ /\ \/ /\ \/ [internal Network 192.x.x.x] [IP10s] + [IP10s] + [Softphones] Everything is good as long as only *1* phone is registered from the internal network. The minute another phone goes online registration is dropped and the sip debug complains: [Feb 9 18:11:00] NOTICE[10381]: chan_sip.c:12687 check_auth: Correct auth, but based on stale...

...cations, internet connections in each one of them and would like to use it via asterisk to do telephone calls. Hardware-wise I have a couple of dual Opterons and some classic i386 servers (P4). As far as VoIP phones are concerned - I have a couple of Grandstream's, a couple of SwissVoice IP10s's and some Cisco's and Kirk's. I'd like to setup a network that uses those phones (it's not a must to use ALL of them), and to have VoIP telephony between the phones in the company, and if it's not "within the company" - to use ISDN lines to connect to "out...