similar to: Asterisk recieves "11" when pressing "1" from SIPphone

My DTFM is not working in current CVS-stable *-1.0.6 and *-1.0.7 but it works in version 1.0.5 (was working with 1.0.3). I'm using SPA-3000 and dtmfmode=inband -- #Joseph

We have some guys working in the US who can't always dial back to our company in Europe easily (lots of clients require authorization to make international calls), so I set up the following: - ipkall.com number links to a FWD number - office Asterisk box registers with FWD Then I programmed Asterisk to accept office extension number using DTFM tones. This works OK. Then I programmed

Hi! I'd like to ask whether someone did test Opus in real-world VOIP (SIP). Did someone e.g. some characterization about sending faxes or DTFM through Opus? Does it work and if yes for which bitrates? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20130518/907c5cbf/attachment.htm

Hi Gang Not a specific Asterisk Question. But I wonder, if the called party replies with 183 + SDP indicating support for telephony-event. Should the caller be able to send DTFM Tones? Swiss Railways uses an IVR that kicks in before the call is answered. So far I have found no SIP Phone which would allow sending RFC4733 during the early audio phase (so I cannot test if Asterisk would forward

I'm having a very annoying problem with access my asterisk system from work. Our phone system here only produces very very short DTMF tones. The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE => CALLUK.COM 0870 => IAX Inbound The

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When attempting to call one of the example numbers, like 17474745000, I only get "488 Not Acceptable Here". It works fine when I configure the softphone (Xten X-Lite) to use sipphone's server directly. Am I missing something? Here's my relevant config sections: sip.conf: in [general]: register => 17472442457:mypassword@proxy01.sipphone.com [sipphone] type=friend

Hi, Does anyone know how to dial toll-free (800) numbers through FWD or Siphone? Using the configuration below, I can dial out to SIPphone.com users by simply dialing their number (1747XXXXXXX) and can dial out to FWD users by dialing 1383<FWD#> However, when I dial 18005551212 through SIPphone, or through FWD (depending upon which line is selected in "; 800 Toll Free Numbers"

for inbound calls, i can register context = from-sipphone register => 1747xxxxxxx:passwd@proxy01.sipphone.com but how do i configure to make outbound calls to them? exten => _1747XXXXXXX,1,GoTo(dial-sipphone,${EXTEN},1) .... [dial-sipphone] ; ; SIP to sipphone.com ; exten => _X.,1,Dial(SIP/${EXTEN}@??????) ^^^^^^

Hello All. I started setting up my Asterisk system yesterday and everything was going well, i have registered with sipphone.com and set-up my Asterisk system to register with sipphone per the sip.conf file below. It was registered perfectly but I could not receive calls so I added in the line "insecure-very" and I then used the Washington DC access number to test and the phone

Hello, can anyone comment on how one could use SIPphone's $89 All-in-One adapter with Asterisk? Sounds to me like it should work as both a FXO and FXS. It would be a cheap way of getting started with Asterisk and PSTN. Any comments on the SIPphone FX200? Any comments on SIPphone in general? Thank you for your help

Hello when i dial a toll free no using sipphone i get this error message. How do i solve this? Any help will be appreciated. console message: Starting simple switch on 'Zap/2-1' -- Executing SetCallerID("Zap/2-1", "17473863282") in new stack -- Executing SetCIDName("Zap/2-1", "Deepak JV") in new stack -- Executing

Has anyone else had problems registering with Sipphone over the last few weeks? Previously, this had worked fine. I contacted Sipphone technical support, but they're not much help. register => 17471234567:password@northamerica.sipphone.com/123

Hi all. I found a connection error from sipphone.com. It seems 'realm based authentication' by sipphone.com. any ideas? Regards. mack

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

Should SipPhone support ISN routing for their 747 ITAD? Cast a vote: http://forums.gizmo5.com/viewtopic.php?t=10197 Meanwhile if you're interested, you can use the Nerd Vittles 'bandit' ITAD #1089 to call a SipPhone/Gizmo5 subscriber via ISN, which I think is clever (Karl tips his hat to Ward Mundy) and it's also really, really funny.

Does anyone else have DTMF issues with SIPPhone? When calling into my DID, and entering, say, 1002. Sometimes it will recognize it properly (rarely), other times it will receive something different. Such as, 1102 or 1000, etc. Has anyone else been having these issues? I'm only accepting ulaw and alaw, and my relevant sip.conf information follows: [sipphone] type=peer

All of a sudden, my account doesn't appear to work, or even perhaps exist with SIPPhone. Is anyone else having trouble?

Any sip softphone will work.? Linphone is free. I have tested many . All works well with audio.? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131102/21e533b0/attachment.html>

I have a lot of experience, all of it pretty good, with various Sipura products, Grandstreams, Zultys, IAXy, and numerous SIP/IAX soft phones connecting into Asterisk as clients. Good sound quality, great reliability. I've tried two of the units named in the subject line, and frankly I'm frustrated. Calls usually start out OK, but within a brief period the sound goes totally to