similar to: Looking for a good VoIP Provider in the UK-

Is it possible for Asterisk to set the ptime attribute on outbound calls in SDP invite? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060405/7e1e68d1/attachment.htm

Hi, Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? Ta. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040421/3d91c7f6/attachment.htm

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

Hi, I have access to two providers. On one of them the authuser is the same as the username, so outgoing 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:

Have anyone tried to interface BT's Broadband Voice with asterisk? Kannaiyan

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Hi all My * implementation is working brilliantly with only one small fault left to kill. I'm using IAXTalk from Telappliant for my incoming/outgoing calls to the pstn network; if I set my codec to GSM everything works great - no pauses but quality is a bit poor. If it set the codec to alaw (I think I'm using the correct one - I'm in the UK) I get intermittent pauses on the call.

I have installed asterisk @ home 2.6. I am using a Telasip VOIP account. When I make outbound or inbound calls the calls seem to connect and then get hung up. I was wondering if there was something that I am misisng. I have tried several different sip.conf configurations. Here is what they are currently. telasip-gw canreinvite=yes context=telasip-in dtmfmode=rfc2833 fromuser=jrasxxx

Hi everyone I'm interested in using the Telappliant/voip-talk offering as an alternative to my DDI analog problem. (see [Asterisk-Users] Multiple DDI & Hunting on Analog Lines (UK) for details) Does anyone on the list have any recent comments on reliability etc? I would really appricated some positive and negative comments. Cheers Matt

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

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 different context, incoming numbers always

Hi all, I've just setup a sip line with Telasip and when they route the calls to my asterisk box, they include an extension along with the context that is defined in sip.conf for that DID. At first, I couldn't figure why they were getting 404 error from my asterisk box, but then figured out that they are sending the call to an extension that matches my number with them, in the

Hi, I am looking for voip termination all over the world especially based on IAX or SIP. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040124/25e16736/attachment.htm

Probably just a simple syntax issue, but does anyone know how to dial a number and the once phone has been answered, play DTMF tones and then disconnect. I am trying to use this for page notification. Ive been trying the following string with out luck: exten => s,2,Dial(SIP/TelaSip-gw4/5198881212|D(12345678) Any help would be greatly appreciated! -------------- next part -------------- An

Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a "Service Unavailable"

Does anyone know of a BYOD provider that terminates calls to NCFA numbers (UK 'national rate'). I enjoyed broadvoices unlimited to those numbers, but this is getting silly now, it doesnt work and no answer if after switching to a new provider it will ever work. Can anyone suggest an alternative provider that serves NCFA? -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605

Has anyone out there used the PipeMedia/PipeCall PSTN gateway? Anything good/bad to say about it? I'm considering using them for a new customer. They seem to have good rates, good provisioning tools and (better still) give commission on usage to dealers. -- David Gurr Congruity Ltd. Fax: 0871 661 1756 Hemel Hempstead UK

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service. Is there a way to be able to transfer calls to other extensions of my asterisk server FROM the cell phone/ This isn't a

I can't seem to get consistant outbound caller ID working correctly. I have set the fromuser and callerid field in my sip.conf for my TelaSIP peer, but half the time it shows up as "No Caller ID" on my cell phone, other times it shows it correctly. Using asterisk CVS. Any ideas? Doug

I am new to Asterisk so can I start by apologising if this question has been asked and answered already. I'm in the UK using BT for two incoming lines, one on Wildcard TDM400P and the other on Wildcard X100P. I also have a SIP connection to voiptalk.org. Incoming calls via SIP/broadband ring on extensions immediately. However, an incoming call via PSTN is displayed on the CLI as an