similar to: E911 SIP or IAX providers?

I have a Asterisk server setup and I am able to connect to the server using a soft client 'x-lite' and call and leave a message on my second extension 102. I have setup a Vitelity account and add what I believe to be the correct information to my sip.conf and extension.conf. I would like to setup incoming and outgoing calls with voicemail support. I've searched all over but many of the

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During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if

How exactly does Asterisk provide E911 service??

One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? ----- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part

Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Large pictures are at the bottom: http://www.hybsys.bg/img/ipph/IP5000_1.jpg http://www.hybsys.bg/img/ipph/IP5000_2.jpg -- Kyle Sexton

I have a client that would like to use asterisk to link their multiples locations together. However, if a person in the remote office dials 911, How can the 911 operator determine WHERE the emergency is?? Since all calss would be going out of the PRI in the main location, the police/fire trucks will show up at our COLO!! I know that there are some that are doing this multi site setup, how did

Started playing with 1.4 and I'm curious what uses people have come up with for the Jabber integration? So far I can think of presence based call routing, but I'm sure there are other ideas. How are YOU using the new Jabber features in 1.4? :) -- Kyle Sexton

Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Thanks in advance! Elder D. Arohuanca dCAP 1497 Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL:

Having a problem w/ not getting CID name from a PRI. CID Name appears in the PRI debug, but even after a Wait(4) it still appears after the phone is ringing. Here is the relevant info from my PRI debug output. Line 4 is a NoOp showing me trying to echo Name and Number. Line 6 dials the extension, and you can see callerid name get presented on line 29. Again, there is a Wait(4) before the

I believe the 911 is a serious issue if one does an asterisk installation in an office. How do you test 911? Won't they arrest you or something for dialing 911 for no reason and talking to one of their agents who could have taken a more important call? On the other hand what an emergency comes up (like someone got seriously injured) and on top of that asterisk crashed all of a sudden

I just set up a t1 with 2 camma mf 911 trunks on it, and I am having a issue with it. We can call 911 which is routed out these new trunks, and it goes to the 911 center, but they are not getting the ANI and hence "no record found". Our LEC is Embarq, and they say they can see the call come in and send: KP-911-ST and then KP-0-911-ST rather then KP-0-ANI-ST I turned on all the debug

Having a weird issue with some agents getting stuck busy on my system. Call will come into the queue and the agent will hit DND, or be DND when the call comes in (DND being the button on eyeBeam softphone, not a star code). After the agent comes back from DND they will be "stuck" as busy in the queue and I have to reload chan_agent.so in order to get them available. I'm running

Before I head down the path of converting voicemail to an ODBC backend, I have a couple questions that I was hoping someone would know. 1. Is the voicemail message stored in the datbase, or just it's location/filename? 2. Does MWI propagate when using an ODBC backend? 3. If it does both of those things, wouldn't it work well for a centralized voicemail system instead of a solution like

I'm wondering if anyone out there is running a community PBX for their local Asterisk User Groups or area Linux groups. I've been thinking of setting one up but am stuck as to what services to provide that people would actually find useful. I know that I could setup simple SIP->SIP to allow everyone to call each other, but that's not generally too fun. -- Kyle Sexton

All, Is razor only really useful for configuring mass numbers of hosts? I''ve been playing with it and one thing I''m not grasping is how to set a hostname if I''m only building one host, say wiki01.foo.com. For each server like that would I need a separate policy, or is there some other way people are using to pre-populate specific host information to be passed to

I have a Vitelity DID which generally works, but calls from a particular caller do not reach it. Vitelity has thus far disavowed any responsibility for working through this problem. I recognize that some action might be required by another provider which is outside Vitelity's control, but it seems that they should at least be trying to help resolve the problem by helping me determine

I'm setting rsync up for the first time and would prefer to have the exclude file defined in the conf file, but the exclusions aren't honoured when I define the parameter in rsyncd.conf - although they are when I specify the file in an argument. The server is the remote system and both rsyncd.conf and the exclude file are the same on both local and remote systems. I'm attaching the

Please see the attached file and let me know if you need any more information. /Sam Sam Sexton <mailto:sam.sexton@reuters.com> Reuters Coventry Automated Dealing Technologies Phone: +44 24 7625 6562 Fax: +44 24 7655 5203 --------------------------------------------------------------- - Visit our Internet site at http://www.reuters.com Get closer to the

On 2/5/2015 12:56 AM, Steffen Kaiser wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On Wed, 4 Feb 2015, George Sexton wrote: > >> I'm seeing two bogus messages appearing my Maildir/cur directory. >> They're dovecot.index.log and dovecot-uidlist. >> >> -rw------- 1 gsexton users 51 Feb 4 09:04 >> Maildir/cur/dovecot-uidlist:2,S