similar to: dropping extra frame..already have it????

Hi, I am using real-time, oh-0.7.2, G729 Calling from (SIP)UA through asterisk towards h323 devices or the other way round, I get only one-way audio. Called party can only talk, caller can only listen. Calling SIP to SIP is ok. All devices are on official IP addresses. (no NAT) Regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail

Hi list! Sorry for this slightly off-topic message but does anybody know if the standard for ISDN BRI is the same in Spain as it is in the rest of Europe (or the Netherlands). Will a standard HFC-S card work?

Hello: Has anyone on this list had to configure hairpinning on a Cisco gateway running IOS 12.2 or 12.3 and using a PRI for connectivity to the PSTN? If so could you tell me how it is done? I'm told this is the source of my call transfer problems and yet I cannot find clear instructions for how the configuration is done. Thanks,Steve -- ISC Network Engineering The University of

Dear Fellow *-ers, First, you guys are fantastic. Keep fighting the good fight. Second, it sounds like comments in the code are coming, which sounds welcome by all, even those of us who couldn't code their way out of a papersack, but who need to read the source. Last, I might be traveling to Europe (from US) & want to tow along hardware & haven't done this before & was

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

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

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 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 the following configuration and am obviously missing something small that is causing * not to work as expected. I have the following defined in sip.conf [ccme-in] type=peer host=10.0.9.1 context=devel_in disallow=all allow=alaw nat=no canreinvite=yes qualify=yes and [devel_in] is defined in extentions.conf However when I try to call via the dial peer I have configured on the cisco

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 All Has anyone managed to get the hint priority with Swissvoice IP10S phones working? I have 2 phones: a Snom 360, setup as the reception phone on extension 11, and a Swissvoice IP10S on extension 12. When calling each other (tested both ways) I can only ever see the Snom 360 in the Active State from 'show hints'. The Swissvoice stubbornly remains in the Idle State when on a call!

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/

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, 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

hello, anyone that know about this asterisk's message: frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Best REgards Ever -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060602/ea4abe0c/attachment.htm

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..... I am new to this software, and I want to implement a client (SIP or IAX) with PHP or at least to pass the main functions (connection,call, transfer, hangup, call id etc) to a CRM. Does anyone know if I could achive a project like that with AGI ? Any example using AGI with PHP ? Do I have all the functionality with AGI ? What about call id ? What is depend on ? (As I know * does not