RE: [Asterisk-Users] Junction Networks
Title: Junction Networks These guys seems to have Canada DIDs. They do not explicitly say that there is a per minute charge for incoming on DIDs.Are their DIDs flat fee ? -Original Message-From: Adam Collard [mailto:[EMAIL PROTECTED]Sent: Monday, May 23, 2005 2:01 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Junction Networks I'm using them right now. I would have to say great. Call me at 800-757-5669 x4861 for more info. That's the Junction Networks trunk. Adam Collard General Manager, ER Wireless (800) 757-5669 x4861 (517) 242-1800 Cell Nextel DC 131*256784*19 [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley SilerSent: Monday, May 23, 2005 8:32 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Junction Networks Anyone have experience with these guys? If so, good, bad, average? http://www.junctionnetworks.com Thanks, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voice pulse connect - no dtmf
Hi, I've got bunch of VP connect lines, and a day back two LA area numbers stop sending DTMF. They are IAX2. So, simply my customers can dial in, it hit my IVR but when they punch-in the number, my * running 1.0.7 cannot identify the dtmf. IAX debug does not show dtmf being sent to me. Just want to know whether any of you had this experience, and if so how that was fixed. Funny thing is this happened on two dids and others are OK. Cheers DH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 10-10 dial around
Folks, This may be not directly related asterisk, but hope some experts can help here. How would one start offering a calling program based on 10-10 dial around basis. Are there companies who could provide a 10-10 number just like a 800 DID. What kind of infrastructure needed for this kind of service. DH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Experiences with Termination Providers?
-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. Well, you must be dreaming :) It all depends on your buying power, if you have at least 2-3 million minutes goto Level3 or broadvox. If you are just starting up and no commitments, then you have to stick with one of the two categories that you mentioned below. I chose to use first type you mentioned. BTW:- if you find a provider which could give those points mentioned and still go with no commitments, please let me know. Cheers Dough -Original Message- From: Me [mailto:[EMAIL PROTECTED] Sent: Saturday, November 27, 2004 11:08 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Experiences with Termination Providers? 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 of them to be offer a quality service with most of the features I want but horrible customer service/support and response times to my questions etc. The other two seem to respond quickly and have great customer service but have awful connections to the web and basically unusable services. Can someone recommend a termination partner for our VOIP Venture that can provide reliable services, good features/DID's and GOOD customer service? Price is important as well but comes last in line after the items mentioned above. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astcc not working
Hi, I posted to this list couple of days ago, that my astcc is not writing the card balance to the mysql database. http://lists.digium.com/pipermail/asterisk-users/2004-September/061645.html I just want to ask this question one more time before creating a bug note in "mantis". Since the application is still alpha, I belive that it would be helpful to improve this app for the benifit of all of us. There seems to be situations that it still can hit problems, like what I see here. Cheers DH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] astcc dont write to the table cdrs or cards
Hi, I did a cvs update on 03 Sep. How do I find out all available variables (to agi) in a particular code version. I tried show agi get variable, but that wouldnt give me much info. Cheers dh -Original Message- From: Areski [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 08, 2004 1:38 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] astcc dont write to the table cdrs or cards Variables DIALSTATUS: added to CVS head in june/july 2004! What is your CVS version? Areski On Wed, 2004-09-08 at 03:44, Doug Harris wrote: Hi, I have set-up astcc with outgoing sip channel. Call processing works fine but after the call tables, CDR and Cards does not get updated. At the beginning it goes to the database and fetch card details and correctly provides the card balance etc. Also it indeed write the inuse field (so writing and reading from database works fine). I've inserted a break point as such in the code; $dialstatus = $AGI-get_variable(DIALSTATUS); print STDERR dial status $dialstatus\n; It seems like dialstatus is not returned (which prints nothing). So obviously later part of the agi does not go through database updating portion (which only happens if dialstatus = Answerd). I am using deadagi to call the astcc.agi script as explained. Can someone explain why this happens ? Cheers dh __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astcc dont write to the table cdrs or cards
Hi,
I have set-up astcc
with outgoing sip channel. Call processing works fine but after the call tables,
CDR and Cards does not get updated. At the beginning it goes to the database and
fetch card details and correctly provides the card balance etc. Also it indeed
write the inuse field (so writing and reading from database works
fine).
I've inserted a
break point as such in the code;
$dialstatus =
$AGI-get_variable("DIALSTATUS");print STDERR "dial status
$dialstatus\n";
It seems like
dialstatus is not returned (which prints nothing).
So obviously later
part of the agi does not go through database updating portion (which only
happens if dialstatus = Answerd).
I am using deadagi
to call the astcc.agi script as explained.
Can someone explain
why this happens ?
Cheers
dh
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RE: [Asterisk-Users] FINALLY! a good book about Asterisk.
we should buy it and encourage everyone to do so, that will support whoever took the initiative to write a book on Asterisk, which has been long overdue. DH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Totaro Sent: Tuesday, June 08, 2004 8:16 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FINALLY! a good book about Asterisk. I think I will pass. $49 for something free on the wiki seems too expensive. A cheaper PDF would would save a tree and probably be more reasonable in cost. - Original Message - From: Joe Babstock [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Thursday, July 08, 2004 10:05 AM Subject: [Asterisk-Users] FINALLY! a good book about Asterisk. There is finally an introductory book about Asterisk! It looks like Paul Mahler at www.signate.com wrote it with a lot of help from Digium. I looked at the sample pages, it looks great. __ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] quantumvoice
http://quantumvoice.com Anybody using this company. They have all you can eat toll free service. Don't see any reference to asterisk, but can use your own Cisco or Sipura. If there is any known working config, appreciate if it could be posted here. DH
RE: [Asterisk-Users] quantumvoice
I wanted to know how ATA is configured, so that I can get some clue whether it would work with asterisk. They told me that they do not provide config instructions but they provide a script that needs to be loaded to ATA. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk Sent: Tuesday, July 06, 2004 4:18 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] quantumvoice Interesting service, VERY interesting pricing. I talked to their support about Asterisk and would you believe they NEVER HEARD ABOUT IT!! Anyway, I offered to give them config info/installation instructions in return for a test-account. We'll see what happens. -Original Message- From: Doug Harris [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 06, 2004 12:02 PM To: [EMAIL PROTECTED] Digium. Com Subject: [Asterisk-Users] quantumvoice http://quantumvoice.com Anybody using this company. They have all you can eat toll free service. Don't see any reference to asterisk, but can use your own Cisco or Sipura. If there is any known working config, appreciate if it could be posted here. DH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quantumvoice
Well what they bring in is a cheap toll free service. It would be a nice back-up for many of us needing an alternative. DH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk Sent: Tuesday, July 06, 2004 9:06 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] quantumvoice Well, I'm pretty sure they're using SIP, since the SIPura SPA-2000 is also supported. I asked them outright if they would provide SIP credentials -- and I'd hope they would. I currently have broadvoice, sipgate and vonage's SoftPhone service working on my asterisk, and since all are SIP based, I don't see a problem configuring quantumvoice. The question will be how reliable they are. -Original Message- From: Doug Harris [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 06, 2004 10:19 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] quantumvoice I wanted to know how ATA is configured, so that I can get some clue whether it would work with asterisk. They told me that they do not provide config instructions but they provide a script that needs to be loaded to ATA. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk Sent: Tuesday, July 06, 2004 4:18 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] quantumvoice Interesting service, VERY interesting pricing. I talked to their support about Asterisk and would you believe they NEVER HEARD ABOUT IT!! Anyway, I offered to give them config info/installation instructions in return for a test-account. We'll see what happens. -Original Message- From: Doug Harris [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 06, 2004 12:02 PM To: [EMAIL PROTECTED] Digium. Com Subject: [Asterisk-Users] quantumvoice http://quantumvoice.com Anybody using this company. They have all you can eat toll free service. Don't see any reference to asterisk, but can use your own Cisco or Sipura. If there is any known working config, appreciate if it could be posted here. DH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] prepaid application
how could any prepaid application be good if it does not update the balance :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hekuran Doli Sent: Wednesday, June 30, 2004 9:58 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] prepaid application Hello! I have installed the modified prepaid application and its working god. the only problem is that when I finish the call it does not update the balance of the card. any one has any idea how this could be fixed? best regards Hekuran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vonage and Asterisk integration
Vonage does not allow any other device other than their own to be hooked up to their system, period. There are whole bunch of service providers who allow you to hook-up your own device. So why split hairs, use someone else other than Vonage. Their is nothing extraordinary about Vonage, except they have some advertising dollars. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk Sent: Tuesday, June 29, 2004 8:54 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Vonage and Asterisk integration I do. I decided not to bother with Vonage's sub-par and unmotivated customer service(*) and plugged my ATA186 into an FXO port. (*) Examples: Had three lines on two ATAs. Asked if I can moved one of the lines off to a third (new) ATA -- they couldn't do it. Asked if I can move an existing number to a Softphone line. Nope, couldn't do it. Can I make an existing number a virtual number? Nope, can't do. Apparently, they can utilize LNP to move numbers from you CLEC to themselves, but they can't move numbers around inside Vonage. Ba! It cost them two lines and about $45/month in services. -Original Message- From: Jerry Roy [mailto:[EMAIL PROTECTED] Sent: Monday, June 28, 2004 12:51 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Vonage and Asterisk integration All, I have been thru the archives and all the relevant URL's sent to me. I have sent e-mail to those who have gone before me and are attempting to accomplish the same goal - no one has it working?. Doesn't anyone have a WORKING asterisk pbx that hooks into vonage? Thanks, Jerry Roy 562-305-9545 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk console mode
Hi folks, I use safe asterisk to startup and run asterisk in the background. In Safe_asterisk script, there is a parameter (right at the top ), CONSOLE which I can set to no or something. If it is no asterisk startup as asterisk -vvvg , if it is set to something the asterisk startup as asterisk -vvvg -c. Now I am running an agi script when calls get hung-up. That is in my extensions.conf I call myagi.agi like h,1, agi, myagi.agi. When I have asterisk started in console mode everything works fine, however if I start -vvvg, soon after the agi completes asterisk shut it down. == Spawn extension (fwd-out-test, 613, 3) exited non-zero on 'SIP/-081467b0' -- Executing AGI(SIP/-081467b0, updatecb_post.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/updatecb_post.agi == Spawn extension (fwd-out-test, h, 1) exited non-zero on 'SIP/-081467b0' asteriskremote*CLI Disconnected from Asterisk server Executing last minute cleanups Asterisk ending (0). Any Idea why this is happening. ??? What are the pros and cons running asterisk in console mode in safe asterisk ? Cheers DH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk console mode
Hi
folks,
I use safe asterisk
to startup and run asterisk in the background. In Safe_asterisk script, there is
a parameter (right at the top ), CONSOLE which I can set to no or something. If
it is no asterisk startup as asterisk -vvvg , if it is set to something the
asterisk startup as asterisk -vvvg -c.
Now I am running an
agi script when calls get hung-up. That is in my
extensions.confIcall myagi.agilike h,1, agi,
myagi.agi. When I have asterisk started in console mode everything works
fine, however if I start -vvvg soon after the agi completes asterisk shut it
down.
== Spawn extension
(fwd-out-test, 613, 3) exited non-zero on 'SIP/-081467b0'
-- Executing AGI("SIP/-081467b0", "updatecb_post.agi") in new
stack -- Launched AGI Script
/var/lib/asterisk/agi-bin/updatecb_post.agi == Spawn extension
(fwd-out-test, h, 1) exited non-zero on
'SIP/-081467b0'asteriskremote*CLIDisconnected from Asterisk
serverExecuting last minute cleanupsAsterisk ending
(0).
Any Idea why this is
happening. ??? What are the pros and cons running asterisk in console mode
in safe asterisk ?
Cheers
DH
RE: [Asterisk-Users] NuFone?
Hi, Seems like there arn't any alternative to NuFone either ? Toll free Number + IAX/SIP + 2.9 Cents per minute, no strings attached. Doug Message: 2 Date: Wed, 17 Mar 2004 08:34:25 -0800 (PST) Subject: RE: [Asterisk-Users] NuFone? From: Robert Hajime Lanning [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] quote who=[EMAIL PROTECTED] I have used both VoicePluse and Nufone. I have to say that the support and the service I have gotten from NuFone is second to none. They are quick to respond, they had me up in no time. I have Nufone and I would have to say, their network is top. I have not had any network outages, delays, or otherwise. Their business side (and trouble shooting) is not ready for prime time. Issues: o NO BILLING! o no detailed accounting o no way to check your account other than emailing a request. - I may setup a cron job to request my account ballance once a week. o I couldn't dial 800 numbers via Nufone (IAXTel and PSTN worked) - I had forwarded him all pertinent information from my configs - All I got from support was, Everyone else can. and I can't reproduce it and we treat 800 the same as all other US calls, even after I had suggested that it wasn't him, it was the carrier he passes the call to. - I finally figured out that his carrier requires exactly 10 digits in the callerID, for tollfree numbers. This requirement does not exist for any other US number. I am wondering when Nufone will get serious about being a business with customers to be responsible to. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NuFone?
Some don't do g.729 and per second billing. These are the other things when you have to compare. would you ? From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] NuFone? Date: Wed, 17 Mar 2004 20:17:47 - Organization: TelAppliant Ltd Reply-To: [EMAIL PROTECTED] Since everyone is offering their services then: USA - =A30.016 (~ 2.9c) UK - =A30.016 (~ 2.9c) Europe - =A30.02 (~ 3.6c) UK 0800 - FREE SIP / IAX termination. auto-provisioning, web-based billing, call history, on-line top-up, credit-card payments. Not US-based though :-( Tan www.voiptalk.org www.iaxtalk.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BETA RADIUS support for Asterisk
Hi, http://bugs.digium.com/bug_view_page.php?bug_id=0001193 First of all thanks for doing this. Now we can play with any VoIP g/w in the same level field. Being a new user always wondered why there is no radius support in asterisk. Sorry for the stupid question; Why is this in bug note. Is there a bug that needs to be fixed before we use this, or is this the general way that one could publish an add-on to asterisk. Are we ready to go with this ? What is the minimum asterisk version needed to run this ? Bug note says : 4) modify extensions.conf to use the radius application How thanks a lot. Dough
Re: [Asterisk-Users] BETA RADIUS support for Asterisk
Thanks Steven Dough Fairly appropriate sig. Didn't you notice my email handle :) Well I will wait for dbruce for my specific questions on how to use it. Thanks D.Harris (new sig.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] new2agi -php
Hi
firiends,
Sorry for a basic
question here.
I am trying to write
an agi script using php. Nothing fancy just simple script
first.
I call the php
script from the extensions.conf
exten =
91234/1001,1,Wait,1exten = 91234/1001,2,AGI,test.agiexten =
91234/1001,3,Hangup
php script is
;
#!/usr/bin/php
-q?phpob_implicit_flush(true);set_time_limit(6);$stdout =
fopen('php://stdout', 'w');fwrite($stdout,"STREAM FILE demo-congrats"
);fflush($stdout);
?
* telles me "Error
in Argument 1, char 3, option not found. ".
script can be run
from command line.
Appreciate some help
to get going here.
Dough
Re: [Asterisk-Users] Best ATA 186 Firmware - my mistake - btw gs 486 is coming
Silly me, I was dreaming last night I guess. When I saw the post, I thoughtthat this is Grandstream device. BTW: Have you guys seen new announcement from Grandstream. http://www.grandstream.com/HT486.pdfat 85 MSRP. DH Message: 1 From: "Dan" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Best ATA 186 Firmware Date: Thu, 4 Mar 2004 10:34:10 +0200 Organization: Personal Reply-To: [EMAIL PROTECTED] Hi, - Original Message ----- From: Doug Harris Where can I download this version ? Cant find it here http://www.grandstream.com/TEMP/FIRMWARE/ Do you mean ATA-18x (from Cisco) or ATA-286 from Grandstream??? BR, Dan
Re: [Asterisk-Users] Best ATA 186 Firmware
Where can I download this version ? Cant find it here http://www.grandstream.com/TEMP/FIRMWARE/ Thanks Doug Message: 10 Date: Wed, 03 Mar 2004 23:50:40 +0200 From: NetOne Administrator [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Best ATA 186 Firmware Reply-To: [EMAIL PROTECTED] v.3.0 works fine too James Coberly wrote: v2.16.2 ata18x Works Fine for me.
[Asterisk-Users] can * support ANI
Hi, I am very new to *. I did searched the list to see whether asterisk has a way to support ANI for known caller-IDs. I know there is a function called authenticate. I thought of modifying authenticate to support ANI. Where could I find the code for authenticate function in the code base ? Has anybody done this, in that case I do not need to re-invent :) Thanks Doug Harris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
