search for: voiptalk

...trick the above wiki page suggests; at first it seemed to work: picking up an extension and dialing any invalid extension would play the message (albeit it would play twice, can't understand why) and then hang up. Later I found the above configuration was interfering with my sip dial-out thru voiptalk.org: any call I place thru voiptalk (for example, dialing '8902' for the welcome message) is followed by the "invalid extension" message when the remote end hangs up. I'm running Asterisk 1.0.2, which I compiled from the source myself. Below are my extensions.conf and sip.con...

I am having difficulty configuring SIP behind NAT (using latest CVS). Using sip.conf: [general] port=5060 ; Port to bind to externip=ww.xx.yy.zz bindaddr=0.0.0.0 nat=yes register=>[userid]:[password]@voiptalk.org/2000 [voiptalk.org] nat=yes externip=ww.xx.yy.zz type=friend secret=[password] nat=yes reinvite=no canreinvite=no I fail to register. SIP Debug gives: SIP Debugging Enabled Jan 26 18:20:04 NOTICE[9226]: chan_sip.c:3126 sip_reg_timeout: Registration for '[userid]@82.145.32.73' timed o...

Trying now to set up the final part of my * switch. I must admit I've had great fun over the last week or so playing with it, and would like to thank you guys for all the assistance (past and present, since I've been trawling a lot of old posts!!!). Scenario - using voiptalk.org to supply the incoming gateway, tied to an 0845 number for convenience in testing. Internal 7960 -> 7960 calls successful. Internal -> External pstn calls also successful. However External -> internal calls (calls to the 0845 number) don't seem to work.... Using IAX as recommended...

Hi All, I have just installed Asterisk and am trying to connect it to a SIP account that I currently have with www.voiptalk.org but without any success. Although I know that voiptalk do provide asterisk accounts I don't want to convert the SIP account until am happy that it's gonna work for me. The asterisk box is currently behind a firewall and the following ports are being forwarded to it - UDP 5060, 10000-2...

I am trying to get asterisks to work with http://www.voiptalk.org 's IAX service. I have configured asterisks as per their instructions and am using the x-lite soft phone. When I get an incoming call the softphone rings but the caller (from pstn) gets a recorded message saying the number is unable to accept calls at present. Does anybody know what might b...

Why is it that newcomers always feel like inserting 'Answer' is a necessary step in their extension.conf entries? >[voiptalk.org] >;forwards any calls starting with an "8" thru voiptalk.org >exten => _8.,1,Answer >exten => _8.,3,SetCIDNum(55555555) >exten => _8.,4,SetCIDName(My Name And Surname) >exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk.org,,g) >exten => _8.,6,HangUp I fully...

I have several of incoming numbers on IAX from voiptalk and magrathea but have a problem with IAX.conf. If I follow the example from voiptalk [VoIPTalk Incoming Number] type=friend username=VoIPTalk Incoming Number context=[XXXXXXXX] and make incoming entries in IAX.conf for the numbers like below with a different entry for each number pointing to a...

There was some discussion on this list recently about the voiptalk silver service. I've just had an e-mail from them saying that the price has been reduced to 2.99 per month. However, they still only provide an 0870 number whereas pipecall provide a local call rate 0845 number in the fee. Chris

...dialing any invalid >> >extension would play the message (albeit it would play twice, can't understand >> >why) and then hang up. >> >> >;;;extensions.conf >> >[internal] ;;; context used by our internal SIP-phon >> >include => voiptalk.org ;include context below >> >exten => 11,1,Dial(SIP/gsbt100,20,tr);calling : dial our office phone >> >include => invalid_calls ;all ext numbers not handled above are invalid > >The 'separate context' approach _does_ work, but you've just c...

...xten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup [2002] exten => 2002,1,Dial(SIP/2002,15,t) exten => 2002,2,Voicemail(u2002) exten => 2002,102,Voicemail(b2002) exten => 2002,103,Hangup then also in extensions.cfg, I have also set these to allow connection to voiptalk: exten => _0[1-9].,1,Dial(IAX2/USERID@voiptalk/44${EXTEN:1}) exten => _00.,1,Dial(IAX2/USERID@voiptalk/44${EXTEN:2}) exten => _09XX,1,Dial(IAX2/USERID@voiptalk/$EXTEN}) Then, since I'm using an IAX connection to voiptalk : [voiptalk] type=peer username=USERID secret=PW h...

Hi, Anyone used this service, any comments on reliability/support? Thanks John

I've installed successfully freePBX with Asterisk, and got various internal extensions working, however. recently my internet facing IP address has been removed by my ISP (for various reason) and I'm not going to be able to get it back for a few weeks. Is there anyway in which I can successfully receive incoming calls from my Voip-Talk.org numbers (an 0845 number) without the static

Hello, I am experiencing a strange problem and I am wondering if anyone may have some pointers as to how to overcome it. I have an account with VoipTalk here in the UK which I have connected to my Asterisk server. VoipTalk supports IAX2 and SIP and I have connected to my Asterisk box using both methods. The problem is when I dial into my Asterisk box via my VoipTalk incoming PSTN phone number from a landline or mobile Asterisk does not consis...

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I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have the following dialplan entries, that I can dial into: exten => s,1,Answer exten => s,2,Wait(2) exten => s,3,S...

Hi all, Does anyone know of any providers that can offer local numbers based in the UK via IAX or SIP? We're looking at getting a number based in the UK. Thanks! -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

...going works. On the other one I can only get incoming - what ever combination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as: [voiptalk] type=peer secret=xxxxx username=xxxxxxx host=voiptalk.org [pipecall] type=peer secret=xxxxx username=xxxxx host=sipproxy.pipecall.com The first one works OK - I can dial out with no problems. The second one needs an extra field for the authuser - when I try to dial out I just get: May 17 01:03...

Hi, i am recording a message, but it is always cut off at 40 secs. There are no time out configured. Gabriel -- The educated person is not the person who can answer the questions but the person who can question the answer.

...olution for a group of teleworkers. Naturally, this exposes us to all sorts of fun, most of which we seem to have working properly. However, some NAT issues are still bugging us and we have noticed that often these situations didn't exist when users were connected directly to our VoIP provider, voiptalk.org. They have something which they call a NAT-Traversal Gateway (see item 6 at http://www.voiptalk.org/products/voiptalkfaq.html), which one configures as the outgoing proxy, using port 5065. Does anyone have any idea what this NAT-Traversal Gateway could be? Naturally, I'm asking this in th...

Hi, I use VoIPTalk as my provider and unsure of a minor issue. When people call me they get a US ring tone instead of UK. Is this a Asterisk configuration issue or one for VoIPTalk ? I am running 1.6.2.0 and IAX2 trunks. Thanks - Phil