similar to: MGCP and dialing out

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 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

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

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

Does MGCP transverse NAT? Seeing as the only decent yet cheap IP phone is the Swissvoice, it would be rather helpful.

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, 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

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

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

Hi folks I seted up the asterisk with an active ISDN B1 AVM Card (german vendor) and it works fine, various SIP clients (IP fon snom, xlite under MacOSX) and also incoming and outgoing connectins. Ok. No problem. After that I configured a CP7940G with a MGCP IOS. It's connected to the asterisk too via switch. No NAT deivce between. And of course, I can call out to PSTN and also to one of the

Hi there, from my logfile (with Swissvoice ip10 phones): WARNING[23575]: chan_mgcp.c:847 mgcp_indicate: Don't know how to indicate condition 14 Anyone has a clue as to what this means? Cheers, Philipp

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 All, I'm trying to hook up asterisk (CVS-HEAD-02/09/05-13:44:11 ) to a ADIT 600 via MGCP. Got it working really nice but now have a pretty bad problem: 1. When I perform a flash on the telephone, I usually get a second dialtone, but when I dial, dialtone doesn't break. If I flash back and forth a few times, it will eventually give me no dialtone.. here if I dial, it successfully

Hi. I'm new in Asterisk with MGCP. I set up a MGCP user agent and start asterisk with the next configuration files. '--------------- extensions.conf ---------------------------------------------------- [general] static=yes writeprotect=yes [globals] ap1 => mgcp/aaln/ap200@64.76.148.186 [macro-apl1] exten => s,1,Dial(${ARG1},30,Ttmr) ;exten => s,2,Voicemail(u${MACRO_EXTEN})

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

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've read the Wiki on using asterisk's built-in transfer options (#8 and #6). They work fine but how does one cancle an attended transfer? Example: I have person on phone, I hit #6 to being att-transfer. I enter Sally's extension. I let it ring for a few seconds. Sally never picks up but her voicemail does. How do I hangup her voicemail and resume the previous call? The example on the

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

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

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/