similar to: Early media dectection problem

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

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/

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

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

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

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, 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, 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-

Hi All Can anybody help me for dial plan to barge or Spy(ExtenSpy) specificor selective extemsions among 20 extension in my office. lets say my office extension range is 301-320 & i want to barge only 3 extension say 320, 302,314. is this possible to barge specific extension? . Plz help me for this.I am using Asterisk 1.4.9 & SIP channels. Regards Amit -------------- next

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

HI folks, I have set a barging facility with our production box.Client able to barge a agent but client raise a requirement, they want talk to barge agent but that communication is not listen by customer. It is possible with asterisk or not. thanks in advance. Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, OS OpenSuSE 10.3 * ver 1.4.26.2 zaptel ver. 1.12 Digium TE122 I have a request for remote users to be able to dial through the system so that the sales managers can barge/chanspy on the sales force. I have the DISA part working with authentication(rather straight forward) but what I can not figure out is how to enable the supervisors to be able to barge on these calls. Is there a

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

-----Original Message----- From: woody+asterisk@solutionsfirst.com.au [mailto:woody+asterisk@solutionsfirst.com.au] Sent: January 12, 2004 11:25 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] GUI client for windows for live monitoring/barge > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com]

I have a couple of questions on the zapbarge: 1) zapbarge asks for a channel - how would a manager know what channel to enter ? Is there any way of being able to enter an extension number instead ? I know that you can get the information from the manager interface, but I wouldn't want to give my users access to this, or have to install / write a system just to get an extension number from a

Sorry that got accidentally sent incompleted, here's the full post: OK, here is the long drawn out description of how I am using Zap Barge and Monitor: Zapbarge(listen in on live calls): Very simple actually I just added this to my dial plan(extensions.conf): ; barge monitoring extension exten => 8159,1,ZapBarge exten => 8159,2,Hangup Then when you dial 8159 on

Hi All, I encountered a DTMF problem. We have an IVR built on Asterisk 1.0.7 with RedHat 9. When the caller calls into our IVR, and IVR plays the first prompt and asks caller to dial four-digit extension. Caller has to dial slowly, otherwise, Asterisk cannot recognize the extension number. I look at the trace on Asterisk CLI and there are missing digit in the middle of string. ex, caller

I've seen a few but can't get them to work. I need one where I can drop a call into a conference without them knowing it to us it as a live monitor and barge function, anyone doing this are know of a gui client for windows I can use? Thanks,

I'm running Asterisk 1.6 and was wondering if anybody have a workig "barge in" solution running. I was thinking of using chanspy, but i would like that the original call would be dropped, and the new call would be the only one there. -------------- next part -------------- An HTML attachment was scrubbed... URL: