similar to: US DID

I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. I've worked with Coredial, Broadvox, and Broadvoice and have had some bad experiences with each of these providers. Broadvoice offers low cost service, however I have constant issues with Broadvoice blocking my customers due to Asterisk

Hi, Inspired by a recent rant about one particular provider, I am getting very curious about something I've never mastered. I'd like someone to explain this here or at least post a link or two that can educate me and probably countless others who have no knowledge in this area. I'm sure there are several of you reading this that know all about the subject. What are the various

Hi, Is there a way to limit only one registration for each user at a time? meaning if a user tries to register, but that user is already registered. i will deny? or is it possible to for a single user at the same time, and when someone calls that user, it will ring both phones? Just want something whereby a user can assign his extension on an IP phone in the office, and assign the same

Hi, Is it possible to detect where the call came from and route it out to different sip trunks. e.g. i have user 100300 when that user calls outbound i will make him use of [sip-trunk-100] another user, 101300 when that users calls outbound i will make him use of [sip-trunk-101] actually the 100 and 101 at the beginning of the username is the accountcode i used for cdr. hope my question

We currently use asterisk 1.4.x with two Zaptel cards connected to POTS lines. So we make "outbound" calls from their softphones (using ulaw format), which go over a dedicated DSL line to the asterisk server in our office, which then converts the calls to POTS. This all works fine, assuming there aren't any unusual problems. It sounds as good as POTS on both ends. However, we

Hi All, is it possible to choose outbound route by checking the extension of the caller? e.g extension that starts with 3 goes to outbound route 1 extension that starts with 4 goes to outbound route 2. Basically, i'm hosting two(2) office, extension 3XXX is office 1 and extensions 4XX is office 2, they both have the same dialling pattern so i need to choose route based on source.

if i have this setup: [sip users] -- [asterisk] --- [as5300] --- [pstn] asterisk will talk to as5300 using sip. i will use as5300 as a trunk on the asterisk so sip users can call out to pstn. what i would like to is do prepaid on those trunks, not on the sip users. sip users can call any other sip users . i want to do it that way coz i'm trying to build a multi-tenant pbx, and i will use

Hi All, I'm trying to use the orginate cmd. I have it working if originate is from a user e.g. SIP/7777 originate SIP/7777 extension 987654321 at outbound-route What i'd like to be able to is instead of a local extensions i would call an outside number then connect it another outside number. e.g. originate SIP/85431210 at outbound-route extension 987654321 at outboudn-route is this

Just received: Cognigen is very proud to announce the official launch of Broadvox Direct, a new VOIP service. Broadvox Direct offers unlimited calling plans to anywhere in the US and Canada for a low monthly payment starting at $29.95 and basic accounts as low as $12.95. http://cognigen.net/broadvox/?mu Anyone know who's behind that? It's not BroadVoice, is it?

We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second

Guys, For what it's worth, after months of trying to troubleshoot issues with them, and after paying them around $2500 for setup and a down payment (it's unclear what of that will be refunded, if any) BroadVox -- http://www.broadvox.net/ -- decided to terminate our contract without any valid reason, and the only explanation they could cite was "it's because of the software

Greetings. I did some digging with Google, the wiki, and on the archives, but didn't find a recent conclusive answer. If this is answered in the wiki or archives somewhere, please point me to it. I'm in the process of setting up an Asterisk box for home use. I've got a X100P card on the way. I've not decided what analog adapter(s) to get yet. The only phone service to hook up

What are your experiences with faxing through Asterisk to the PSTN? We are using g.711u as a codec, and are originating/terminating with Broadvox as well as through our own PSTN gateways. We have had some luck with incoming faxes coming into our network from Broadvox DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody that is using FAX is in our local rate centers.

Howdy, My first forray into using res_mysql.conf for realtime access of sip users and extensions. I have the following relevant section of extensions.conf: --- [trunklocal] exten => _NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [local] include => trunklocal include => trunktollfree [longdistance] include => local include => trunkld [international] include

Nothing has changed in my Asterisk configuration and now outbound US is getting nothing, but 403's. Anyone else having the same problem? Inbound calls to my DID's are working fine. Thanks, SG -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070325/1f30a3d3/attachment.htm

Hi, Is it possible to use realtime for Music On Hold? Is it also possible to store the music/audio files on the database, same way a voicemail can be stored on the database? Thank You Regards, Nhadie

Hello everybody, A client wants to install a FreePBX infrastructure, but have all calls forward to their cell phones rather than buying VoIP phones. They would be doing SIP trunks over a Comcast business line. Probably maximum 6 simultaneous calls. Any gotchas we should warn them about? Thanks! noam Noam Birnbaum El Presidente http://www.desksidemanner.com 415-854-0885 x89 tweet @noamb

OK, been messing with Asterisk for a long time and I have my opinion on where the issues lies but sometimes it's just nice to see what others think that can relate :-) Here goes.. Inbound calls flow like this:Tier 1 Provider (SIP) > Asterisk 1.8 > Name Brand PBX - Calls work fine Outbound calls flow like this:Name Brand PBX > Asterisk 1.8 > Tier 1 provider (SIP) - Calls work fine

Hello to all! I apologize in advance, for the following could be too simple, but I can't figure it out. I configured a vps using centos 8, all is working fine but I'm not able to send email from this server. I tried $ telnet smtp.mailtrap.io 25 but it doesn't work yet $ telnet www.google.ca 80 does work and particularly $ telnet smtp.mailtrap.io 2525 does work too. I temporarily

Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting