similar to: DTMF is being MUTED by asterisk to/from SIP channels from SIP or ZAP

I'm trying to get inbound calls from a TNT working but get 407 errors from the TNT. This is what I have in sip.conf: [maxtnt] type=friend host=x.x.x.x dtmfmode=rfc2833 callerid="MaxTNT" <maxtnt> context=demo qualify=yes disallow=all allow=g729 allow=ulaw insecure=very This is what the TNT is spitting out: Jul 24 14:55:12 tnt1 1/17: Releasing

Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip >< sip TNT pri >< pri asterisk The TNT is running 11.0.6 and the asterisk servers are running 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to asterisk but not the other way. The call from asterisk to pri to tnt is good, the TNT is passing SIP invite to the

Dear R help forum, I am using the function 'coxph' to obtain hazard ratios for the comparison of a standard treatment to new treatments. This is easily obtained by fitting the relevant model and then calling exp(coef(fit1)) say. I now want to obtain the hazard ratio for the comparison of two non-standard treatments. >From a statistical point of view, this can be achieved by dividing

Dear all, I'm trying to connect to an MSAccess database (ArcGIS personal geodatabase). I keep getting an error about the channel when using sqlQuery(). However, sqlTables() does not complain about the channel and lists all tables in the database. If I try sqlFetch(), then R crashes. I'm happy to hear suggestions on how to solve this. Best regards, Thierry > MDB <-

Hello, I know that my Kphone question may be a bit off topic, but I have been busy with this again and again for about one month now, sent three mails to kphone@wirlab.net (the contact address mentioned on http://www.wirlab.net/kphone/index.html), asked for a solution in a german ip phone forum and tryed many things by myself. I try to compile KPhone 4.0.3 (tryed CVS Version as well) but

I trying to receive a SIP call and have ring a analog phone. I have a TDM11B card with FXS(green) module in line 1. I have Sip server "SER" setup to accept a SIP call, add a 970 extension to uri and set to asterisk SIP server on port 5065. When I send a SIP call from "kphone a soft SIP phone" running to sip://wally.world@cci.net "SER" picks call ok and changes uri

I'm having trouble with kphone on our system. It's using ulaw on an internal network. No NAT. I had it working fine with very similar hardware (an old Dell Optiplex GX1) running as an LTSP terminal. But then I put the same sound card in an Optiplex G1. Kphone will answer the line fine when I call it (call coming from the * machine), but when we try to get kphone to dial, each GUI

Hi All, posted last week but didn't get any responses. I'm trying to figure out why some of our analog phones aren't showing CID when hooked up to asterisk. To recap, I have an Aastra Powertouch 350, which shows caller ID fine when connected to the PSTN, but when hooked up to asterisk, CID does not show. I've hooked up another phone to the same * port that the Aastra phone is on,

Dear Asterisk experts, I've just downloaded Asterisk CVS version (since I'd like to manage its configuration from RealTime). Next, I have kphone on the same Linux host, and VmWare virtual machine with Windows and X-Lite IP phone inside. I successfully tested the demo's with X-Lite, but failed to hear something with kphone (v4.1.1). There were NO problem with this kphone and stable

Hi there, I have two Samba servers running within an NT domain. One Samba (F-Samba) server runs samba-1.9.18.10 on top of FreeBSD-2.2.8 and the other (L-Samba) runs samba-1.9.18p2-C6 on a RedHat Linux variant. All of our NT machines run NT4,SP4. One is workstation and the rest run server. The PDC is an NT Server (NT1). F-Samba and L-Samba only show up in the browsers of about half of the

-------- Original Message -------- Subject: Re: [Asterisk-Users] Troubles with Kphone Date: Tue, 25 May 2004 15:44:15 +0530 From: Murali Krishnan <murali@bksys.co.in> Reply-To: ismk@myrealbox.com Organization: bk SYSTEMS (P) LTD., To: asterisk-users@lists.digium.com References: <200405250652.46370.klky3@fibertel.com.ar> enano wrote: >Hi , > > > >I'm triying to use

All the clients that I'm aware of for IP telephony have drawbacks. Some won't work at all. KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad to dial tones during the middle of the call, so the demo that * comes with can't be run. Kphone (3.1, the latest) also has a habit of crashing if you do something even mildly stressful, such as hang up while Kphone is

Hello, I'm currently testing Asterisk over a T1 cross connect to a MaxTNT chassis that we have. It is working fine switching the calls through, but there is about a 10 second delay from the time Asterisk initiates the call until the TNT accepts it. It appears to be a ANI issue, I've changed several settings and formatting options on the T1 between the two, as well as turning on/off the

I am clearly doing something ridiculously wrong. Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are unable to register. They keep trying and then time out. With the sip debug on in Asterisk nothing is logged. Here is the trace from one of the phones (kphone): (192.168.100.13 is kphone, 192.168.100.3 is Asterisk) sipclient: sending: 21:47:45.454

Hey all, Has anyone ever had problems with flaky behaviour when registering softphones dynamically with asterisk? I'm working with Kphone and having problems getting consistent results with registration. When Kphone is able to register, asterisk reports -- registered SIP '2001' at XXX.XXX.XXX.XXX port 5060 where asterisk sees [2001] in sip.conf as [2001]

Hi, i just installed latest asterisk on fedora rc2 and on the same machine i installed a sip soft phone called kphone. Kphone complains about /dev/dsp being used and can't place/answer calls (/dev/dsp is obviously used by asterisk) . how can "share" my sound card with these two programs? or can i disable the sound card in asterisk so i can use kphone to place/answer calls? BTW kphone

Im new to Astererisk. I compiled the latest CVS and setup the server. It looks like things are working. I'm running kphone, x-lite and sjphone to test things out. The kphone (local to the asterisk server) can call and receive calls from any of the 2 windows machines. The first windows phone I start I can send/receve calls the second one I cannot. I. No matter which one I start first only

I just can't get kphone to register with asterisk, i can make calls to the demos and even get into the mailbox but kphone cannot register. Here's my story. Can you help me?? Please I have installed asterisk on debian using apt-get install asterisk. I have configured an extension in extensions.conf as follows exten => 8003,1,Dial(Sip/8003,${RINGTIME},rt) exten =>

i tried asterisk ilbc codec against kphone. when the call got connected, i started to immediately get these kind of message to the console: WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)? WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of

Hi , I'm triying to use kphone 4.02, but when i'm make a call the programs doesn't respond any command, so i can't hear any sound .. in sip.conf that's my codec config: disallow=all allow=gsm allow=ulaw allow=ilbc and the kphone give the follow : SipClient: Sending: 06:46:28.116 -------------------------------- ACK