similar to: Passing called number in SIP

I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. So far I have tested 4 providers which I will not mention here. I have found two

I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or

I have been looking around for VoIP providers but have not found a good listing. Is there no "yellow pages" for VoIP providers? Google mostly returns services like Vonage, Packet8, NuFone, ect. None seam to be very reseller friendly and none offer LNP or local DID's for my area. Anyone know of a list (even a partial one) Jeromie Reeves

Hi, all. My company is setting up a branch office in Germany, and I'm very interested in a VoIP provider over thataway. However, I'd need a few things: - Reliability. Can't have my branch office's DID's just going down. A company with a proven track record would be very, very good. - English. I speak great English, decent Spanish... and zilch German. So, as provincial as

Anyone having problems with LiveVoIP lately? I am seeing failed outgoing calls. Calls that are being routed to wrong numbers. DID's that ring busy. For the pass 2 days I am unable to pass CID. Is anyone else have these problems? Can anyone recommend a Quality VoIP provider?

How do you get the called number on incoming SIP calls? I've never had multiple DID's via SIP from one provider before and somehow I never realized that with IAX it just works, and SIP is different. If I don't set an extension in the register command the incoming invite has <sip:s@me.com> in the To field. Now if I have multiple DID's that I want routed to different

Hi, I'm looking for a multi-line sip/ip phone that can answer multiple incoming paths. IE a secretary sits at a front desk and can answer multiple incoming lines/DID's. Is there something like this? Thanks, Brent -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041202/5b685d69/attachment.htm

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"

Did Linksys really use Asterisk for the SPA9000 software? -- Andres Technical Support http://www.telesip.net

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I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines. Have a recently installed Asterisk system, with a dedicated T1 line. (Well, it's really a fonality system). What would I need to do, or where is the reading material, for what I need to do, to convert the Hylafax server to use the T1 line? Reliably. Preferably to use DID's as well. The current FAX works

I have always been able to block toll-free numbers by catching them with a line similar to this for each DID I have on my system: exten => 5554441212/_888NXXXXXX,n,Playback(GoAway) Where 15554441212 is one of the DIDs that rings into our Asterisk box. The problem with this approach that I have to create a line like this for every pattern I want to block multiplied by every DID on my system,

I've just had the most appalling performance from * ever. Dialling: Cisco 7960 => asterisk => IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 => asterisk => IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be

Hello - I currently have 10 DID's coming into one Asterisk server, I seem to be having some difficulty routing based on the DID dialed and am hoping someone on the list can assist me. Here's the relevant info: Ingress SIP trunk: IP: 123.45.45.3456 DID's XXX-XXX-XX00-XX10 sip.conf: [general] useragent=Asterisk port=5060 context=default tos=lowdelay disallow=all allow=ulaw

Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK

Hi, Here is a general question, not applying to asterisk so much, but in the application of asterisk. I have purchased a few IAX DID's through VoicePulse and am interested in a service provider who has the ability to provide me with one number (reliable, as I wish to publish), and the capacity to redirect those calls to my IAX DID's (is this even possible)? Also, with IAX

Hi, I have two asterisk machines side by side both running debian sarge, one running sarge's version of asterisk (1.0.7.dfsg.1-2) and the other running the version of asterisk from www.backports.org (1:1.2.1.dfsg-2bpo1). I also have a SIP provider who is routing blocks of DID's to both machines. The sip.conf is nearly identical on both machines (the general section, and the section

I'm getting 403 forbidden errors when attempting to register to a certain provider. I've tried just about every combination of configuration settings I can think of with no luck. Following is what I would think should work (and one of the settings I have tried). Rather then list every combinaton I've tried, what are the common causes of a 403 forbidden on a register attempt? Other

I need 100 DID's in a specific rate center (916-854-xxxx). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas

Are there any service providers offering Cape Town DID's? -- - Eric Smith