Time Warner providesan emta not an ATA and the technology is different. You do not even need internet connection for that and runs over their own private network through DOCSIS. You need to be a cable service provider to afford that. the good ATA that we use is Linksys Rt31P2-NA, make sure you
I am using the same Asterisk server for 2 different functions. I have users on
one side and have a calling platform on one side so I put in a context under
general but then only the context for a2billing (calling card platform works)
and the other extensions won't work. Below is how I have it
Try DIDx.net, I would not say they're best but at least they willing to help
you when there is problem and they have a large pool of numbers.
-- Original message --
From: Salvatore Giudice [EMAIL PROTECTED]
I have transitioned to other DID's. I think that company is
I have followed all the install note for A2billing and have everything
installed and configured and my asterisk works except the callingcard
application.
Added the following
[callingcard]
; CallingCard application
exten = 777,1,Answer
exten = 777,2,Wait,2
exten = 777,3,DeadAGI,a2billing.php
Thanks Rob, that helped a little bit but now getting a different kind of error:
-- Executing Dial(SIP/9614-3896, SIP/777|200|rt) in new stack
Feb 18 06:00:30 NOTICE[12236]: app_dial.c:1011 dial_exec_full: Unable to create
channel of type 'SIP' (cause 3 - No route to destination)
==
Do you have Argentina DIDs?
-- Original message --
From: Facundo Ameal [EMAIL PROTECTED]
Dear Asterisk Fans,
I'm an Asterisk consultant in Argentina and want to make an
spanish wiki (something like voip-info.org). I have the idea and some
concepts about this
This is related to asterisk database and in the process of installing a2billing,
am still in the install stages and not able to logon but know what the
problem. When I create the database and try to verify it, this what I get
a2billing= SELECT * FROM cc_ui_authen;
ERROR: relation
Had the same problem with them. I now use didx.net, and would not say they're
the best but atleast they have a good ticketing/help desk system and someone
does respond. They also have a large selection of numbers.
-- Original message --
From: Brad Templeton [EMAIL
Figured out myself, just sharing to help others
I have fixed the tables problem in the postgresql database
the parameter tcpip_socket is no longer used in version 8.0 soforget about
that. it has been replaced by listen_address.
and we only want postgres to listen on the localhost so the setting
I looked through the forums but could not find exactly what I needed. I need help setting up call forwarding in sip.conf, where the call forwards to PSTN number without a sip phone but just the channels in sip.conf without any hardware or softphone. Any help will be greatly appreciated.
I'm using it for virtual numbers. I have international virtual number from a DID provider and want to forward it to my cell phone.
In Sip.conf I have the channel
[4305]
type=friend
user=4305
secret=
;context=from-sip
callerid=
host=dynamic
nat=yes
canreinvite=no
dtmfmode=rfc2833
It sounds like a good idea, I tried it and get this error
Sep 8 09:52:17 WARNING[27193]: chan_sip.c:1968 create_addr: No such host: gafachi-o
Sep 8 09:52:17 NOTICE[27193]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
In
Tim, this is the way I have Gafachi set up in sip.conf and works well with channels that have anATA attached to it but not the virtual one. I have changed the host in extensions.conf to the .sip.gafachi.com. But I have calls on the server and cannot restart it yet. I'll keep you
Thanks all. It works fine now.
-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after
Does anyone know of a DID provider in Thailand?
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I have a follow up question. How do I pass on the caller ID of the call I'm forwarding to the other party? I can pass on the channels caller ID but prefer to pass on the forwarding party's number instead.
-- Original message -- From: [EMAIL PROTECTED]
Thanks all. It
I tried both of them but it still goes asID unavailable. First I commented it out, that did not work and left it blank and that did not work either. Below is the sample in sip.conf
[4305]type=frienduser=4305secret=xxx;context=from-sipcallerid= ; left it blank but did not get passed
Yes that works. I'm using Linksys adapter, is there a code I can put in the dial plan to prevent users from putting # after the number? I have a lot of people on the server and cannot ask them all to be pushing # after every call. Thanks for the tip and any help will be appreciated.
Thanks, I tried that and did not work for me. My users are calling US number and without the # at the end of the last digit dials it takes 11 seconds before it starts ringing.
-- Original message -- From: Alberto Sagredo [EMAIL PROTECTED] Yes you could script a dialplan
Try the Linksys ATA. I gave up on Granstream and have 4 sitting in around.
-- Original message -- From: "Zeeshan Zakaria" [EMAIL PROTECTED]
I am having hard time with grandstream phones for a30 phone setup. When a change in configuration is required, I have to change
for mass deployment the Linksys will allow you to update your routers with a tftp server.. You can have the routers always download their software from the tftp server, that way you have the latest on the server for upgrade software. The reason that I don't like granstream is their bad customer
I'm trying to install Asterisk billing server and when I put in su - postgres I
get this response instead of the password response.
-bash-3.00$
-bash-3.00$
-bash-3.00$
Anyone seen this before? I'm using Fedora core 4 and have the same on a local
machine that works fine.
Better to get one provider that does origination and termination and has no minimum requirements. Most companies will require a deposit or minimum usage requirements. Make sure for origination you know the diffference between metered and unmetered DIDs. codec 729 will be pushing it a little bit.
Its a trickish business, when they say unlimited and you make more than 2500 minutes they cut you off.
-- Original message -- From: Christopher Corn [EMAIL PROTECTED]
I spoke to a voip provider today who mentioned that though they offer an unlimited plan, if we use it
Thanks for all the help from Collin, I figured out the postgres database creation. I now have a different problem trying download files. I get this error. Maybe wget does not work in Fedora.
[EMAIL PROTECTED] a2billing]# wget http://www.areski.net/Open_A2Billing_version_Racoon.tar.gz--05:52:17--
I figured it out, I had old install manual.
-- Original message -- From: [EMAIL PROTECTED]
Thanks for all the help from Collin, I figured out the postgres database creation. I now have a different problem trying download files. I get this error. Maybe wget does not work
I was successful in getting the tarball for a2billing
[EMAIL PROTECTED] a2billing]# ls -alltotal 4872drwxr-xr-x 2 root root 4096 Sep 11 06:22 .drwxr-xr-x 20 root root 4096 Sep 10 21:28 ..-rw-r--r-- 1 root root 165 Sep 11 06:16
Yes, I put the filename "Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz" and got that error.
-- Original message -- From: "Jamin W. Collins" [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I was successful in getting the tarball for a2billing [EMAIL PROTECTED]
You're right. How did I miss that?
-- Original message -- From: [EMAIL PROTECTED]
Yes, I put the filename "Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz" and got that error.
-- Original message -- From: "Jamin W. Collins" [EMAIL PROTECTED]
Everything was going well, I got the tarball, unpacked the tarballs, created the postgre user and password, database is created and checked ownership and even got a list of database users. I even imported the data schema into the new database. My problem now is verification of database
This is a questions about database verification and not a2billing. Asterisk also uses database for such things as cdr and sometimes you call dial plans from database. Someone might have seen a similar situation while installing postgres for Asterisk. It is Asterisk related.
--
You're right Voxee support sucks. But I think they do well and provide good rates. I'm using Gafachi, a little expensive and have Voxee. I'm using LCR so the termination will try Voxee first and when not available will use Gafachi. You can set up something like that with a least cost routing.
Thats good news for us.
-- Original message -- From: Doug Lytle [EMAIL PROTECTED] Interesting article I found linked from Groklaw: "Sam Houston State University replaces Cisco CallManagers, Nortel PBXs with Linux-based VoIP and messaging servers"
I rebooted the server on which the Asterisk is hosted on. The * did not come back up and I get this message when I attempt to use CLI
[EMAIL PROTECTED] ~]# asterisk -rUnable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
___
Thanks everyone it is working now.
-- Original message -- From: Tzafrir Cohen [EMAIL PROTECTED] On Sun, Sep 17, 2006 at 10:40:16AM -0400, Steve Totaro wrote:you're right, one should proof, under which user asterisk runs... Besides security reasons, running asterisk
I saw this termination company, www.BuyMin.comthe rates looks good. Has anyone any experience with this company? I use Gafachi, very reliable but expensive.
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To
When I started Asterisk I get this error but it is working fine and should I be concerned. Error below:
[EMAIL PROTECTED] asterisk]# /etc/rc.d/init.d/asterisk startStarting Asterisk PBX: FATAL: Module ixj not found.
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I have created a context in extensions.conf and when I dial, it is suppose to ask me to enter pin number but instead this the error I get.
Sep 18 18:11:54 WARNING[6514]: chan_sip.c:1968 create_addr: No such host: 4035Sep 18 18:11:54 NOTICE[6514]: app_dial.c:1011 dial_exec_full: Unable to create
The only load I have is,
load = chan_modem.soload = res_musiconhold.so
[global]chan_modem.so=yes
-- Original message -- From: "Justin Tunney" [EMAIL PROTECTED] Check /etc/asterisk/modules.conf and see if there is a line trying to load it. On 9/18/06, [EMAIL PROTECTED]
Thanks I'll give them a trial.
-- Original message -- From: "Insider KT" [EMAIL PROTECTED] I've used this company now for over a year. It is part of Ipcb.net, so you got live support 24 hours a day every day. The quality is very good and the reliability is near
You might want to repost it with a subject or you miss a lot of people seeing or opening it up.
-- Original message -- From: "Scott Pinhorne" [EMAIL PROTECTED]
Hi All
I would greatly appreciate some advice or some direction as to where to go next.
I have a provider
I use Fedora Core and it works fine. I'm not connected to call manager though. which version of Asterisk are you using?
-- Original message -- From: "Ward, Bill" [EMAIL PROTECTED]
I would like to setup Asterisk for voicemail with CallManager 3.3(5). I would like to know
Need to add context in the exten files, to differentiate between company A and company B
-- Original message -- From: [EMAIL PROTECTED] It's the 'o' extension in your context that hits the voicemail. (thats a lower case o not a zero) PaulH - Original Message
Grandstreams are totally useless, I had to switch all my phones to Linksys. Grandstream will not even support you and their router side do not work for the 486 or 496.
-- Original message -- From: "Andre Rodrigues (Cheyenne)" [EMAIL PROTECTED] I have more than 20 ATA
Any know of any working smart open source billing? Something smart that can do prepay/postpay and disconnect customers when they owe or a disconnect a call in progress for low balance.
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I'm running 1.2.3 and that seems to be the most stable version, had problems with other versions too.
-- Original message -- From: "Brian Roy" [EMAIL PROTECTED]
On 3/3/06, Gary Richardson [EMAIL PROTECTED] wrote:
I'm running 1.2.4 and just about every call is cut
Try 1.2.3, it works fine.
-- Original message -- From: "James Sturges" [EMAIL PROTECTED] I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of the site. It is sending CRC errors )to Telsta, drops all calls once a day for 1 second, calls getting
How's tou're service with Sellvoip, I was not able to intergrate them into my system and they had no phone support. I'm using Gafachi now but prefer the rates Sellvoip provide.
-- Original message -- From: "AR Tarzi" [EMAIL PROTECTED]
SellVoIP appears to follow a US
there 2 types of inbound metered and unmetered. unmetered is unlimited inbound and metered charges per the minutes.
-- Original message -- From: "Steve Totaro" [EMAIL PROTECTED] Inbound should be free as far as I am concerned unless you have a toll free number.
They're for inbound only though some of them provide termination services
-- Original message -- From: "VIC IP Communications" [EMAIL PROTECTED]
Hi,Companies like DIDx and Sixtel, when they state DIDs at $XX.XX per month and $XX.XX per minute/monthly,do these
Does Sixtel provide E911 service? have tried it out.
-- Original message -- From: "Kaleb L. Kunzler" [EMAIL PROTECTED] Being a sixTel customer I can tell you how sixTel bills. They charge $X.XX per month for a DID, they also charge per minute inbound (a certain rate)
Did you get an answer to this? I am interested in SIP to SIP calls on other networks thereby by-passing the pstn.
-- Original message -- From: Nick Hoffman [EMAIL PROTECTED] Hi guys. Without having a FWD account, can Asterisk redirect calls to FWD? For instance, an
I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with it. It works fine when you pick up the handset. Anyone experinced this problem before, the speaker works fine with Verizon line. The phone is behind a Linsys router RT31P2.
You got me there, it's at a customer's premise. I will have to find out from them, if it a single pair.
-- Original message -- From: "Steve Totaro" [EMAIL PROTECTED] I am not familiar with that phone. Is it single pair?-Original Message- From: [EMAIL
Thanks for the response, I'll ask the client to change batteries, though it is a new phone less than two weeks. is there any reason why the Lanline(Verizon) work and not the Asterisk? The only differences is the Asterisk, Linksys router and the DSL modem. One of these 3 should be interfering.
Gafachi can, I've been using them with you problems.
-- Original message -- From: [EMAIL PROTECTED]
First of all, try sending it to the asterisk-biz list.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John RichSent: Monday, April 17, 2006
disable three-way calling, restric channels to one per call.
-- Original message -- From: "Bill Gibbs" [EMAIL PROTECTED] I say just bill the user at extension 333 it's his responsibility to keep the login info private. If he disputes it, refund the first time then
I had the same problem, I reloaded Asterisk 1.2.3 and set the dtmf 2833 that fixed it.
-- Original message -- From: "Mark Edwards" [EMAIL PROTECTED]
Try dtmfmode=info and see if that works.
Mark
-Original Message-From: Dovid Bender [mailto:[EMAIL PROTECTED]
Could not find your post for 4 months ago.
-- Original message -- From: "Anton Krall" [EMAIL PROTECTED] Yes, check a post that I made about 4 months ago, I posted the cofig for setting the speaker, handset and ring volumes .. |-Original Message- |From: [EMAIL
Try Termilink. www.termilink.net
-- Original message -- From: "Carlos Chavez" [EMAIL PROTECTED] Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México
Termilink, at www.termilink.net
-- Original message -- From: "C F" [EMAIL PROTECTED] Define best. On 5/23/06, Crazy Boy <[EMAIL PROTECTED]>wrote: Hi Friends, Can you please tell me who is the best VoIP Service Provider using Asterisk (With trail version for
Are the phones behind a NAT? What is the processory memory size? Are the E1 channelized?
-- Original message -- From: [EMAIL PROTECTED] I should add that thease 25 calls where SIP (internal) to Zap (PSTN) calls. Mvh, Jan -Ursprungligt meddelande- Från:
try setting your dial plan in sip.conf using dtmf = rfc2833
-- Original message -- From: El Flynn [EMAIL PROTECTED] Rizwan Hisham wrote: Hi, i need to set the dtmf mode on my quintum tenor a400 gateway. You might want to check the a400 manual on how to do that.
Do you have tetheral network analyser installed on server, that can be a good start, look at the analyses of the graphs. Also try pinging the CPE's and see if there is any latency. Do you also have the abilty to check the upstreams signals?
-- Original message -- From:
I could have told you that. Ihave 4 handy tones wasting in my basement.
-- Original message -- From: "calvis" [EMAIL PROTECTED] Polycom 501 Grandstreams are junk. (I have had bad experiences with them) -Original Message- From: [EMAIL PROTECTED]
I was getting this message throughout yesterday in repitition, anyone experienced this before and what is the best solution?
Jul 10 12:19:03 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Last qualify: 124Jul 10 12:19:03 NOTICE[13020]: chan_sip.c:11364
Anyone know how to use dial plan to play messages as soon as a phone is picked up. Like when a user picks up a phone, get a message to contact administrator instead of a dial tone?
___
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Thanks, I'll try that in a few hours and share the experience.
-- Original message -- From: "brandon kruz" [EMAIL PROTECTED] youll have to decide what context this goes in either [internal] or [incoming] but i hope you can figure this out yourself here is an idea
I'm using SIP channel in Sip.conf and hand the calls over to a termination vendor.
-- Original message -- From: Russell Bryant [EMAIL PROTECTED] On Sat, 2006-07-22 at 19:38 -0500, brandon kruz wrote: [internal] exten = s,1,Answer() exten = s,n,Playback(custom)
It did not work, how can I put in some user intervention so that any numbers they dial will send them to a message? Restrict their outbound calls and a get a message to contact administrator instead of a busy signal.
-- Original message -- From: "brandon kruz" [EMAIL
Thanks for the response, its looks logical, for some reason the authentication is not working for me, I'm using a Linksys Phone adapter and here is a sample dial plan in extensions.conf and also SIP channels.
exten = 8407,1,Dial(SIP/8407,80,rt) ; permit transferexten = 8407,n,Authenticate(9461)
That was a typo its corrected to [8407] but problem still persist with original questions though.
-- Original message -- From: "Eric "ManxPower" Wieling" [EMAIL PROTECTED] "[9507]" is the incoming User ID. "user=8407" is the outgoing User ID. Do you really want them to
MySql password for root:
Domain (realm) for the default user 'admin': localhost.localdomain
creating database openser ...
ERROR 1045 (28000): Access denied for user 'root'@'localhost' (using password:
Y ES)
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Does anyone know where to get infomation on the number of minutes used from US
to another country? I tried the FCC but the infomation was not good enough. Who
keeps those statitics?___
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I am using DIDx.net as my DID provider but they don't seem to get their act
together. A lot of times the phone numbers don't work. How can provide my own
DID, my asterisk server is being hosted at a Data center and has a reliable
vendor that does my termination and do SIP to SIP and have no T1
Thanks Alex, I'll try the rapidvox also. I regret ever using didx.net.
-- Original message --
From: Alex [EMAIL PROTECTED]
I have the same problem. Also, the web interface is really awkward, they don't
have DIDs in the countries where I need them (Chile, for example),
Any ideas or tutorial on creating your own DIDs without buying them bulk from a
Telco. I have the Asterisk server being hosted in a data center in California.
I guess I can order PRI through them but how can get DID from other states onto
their system.
-- Original message
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