similar to: DID Provider

I'm getting: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 00009ms SCall: 00008 DCall: 00002 [66.98.180.77:4569] CAUSE : No authority found Under the DIDx number I'm putting IAX: default@asterisk.crystalstream.net/1567252NNNN(IAX) where N is the rest of the numbers. In iax.conf I have: [default] ; DIDx type=user

I assume most people here know what a joke DIDX is -- but in case you didn't already know, please avoid these people. Basic features of their service don't work, their tech support refuses/drags their feet to fix them for a month and if you post publicly about them, they terminate your service. Instead of investing their effort in reading mailinglists to terminate customers maybe they

Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 | 2 | hangup | | 455 | DID | 14193016880 | 1 | Dial | H323/1169#989181310524@66.152.61.66|60 | didx.org for

Hi Friends, We have created a video based training for Asterisk in English and Urdu. Please check them and let us know how we can improve them for no-voice users. http://www.youtube.com/watch?v=KXq9g8UiGnQ http://www.youtube.com/watch?v=MID2RvgdD7s http://www.youtube.com/watch?v=_LbDUdAGfSY http://www.youtube.com/watch?v=J9Chkrg7E-M http://www.youtube.com/watch?v=MsC12wc9ZnU

I successfully obtained the Asterisk code and extracted them into /usr/src. When I make and install asterisk, zaptel, libpri etc. Are they supposed to move automatically into their respective directories? I cannot find: /etc/asterisk/ /usr/lib/asterisk/modules/ /var/lib/asterisk Do I have to manually create them or this is failed install? Thanks. -------------- next part -------------- An

I am using Asterisk and A2billing Calling Card Platform and after the 6th call the quality starts to degrade. The way it set up is the user calls into the system then dial out so I have 12 channels being used up but 6 active calls. Here are my specs Asterisk SVN-branch-1.4-r79142 on a i686 running Linux Fedora 6, Pentium 4 Hyper-Threading, 64 bit, 1GB of RAM, 80 GB Sata Drive, bandwidth 4 Mbps

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. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 exten => 777,4,Wait,2 exten => 777,5,Hangup I am using 777 as the calling card

Don't know if anyone's got experiences on this they'd be able to share... I'm trying to obtain numbers for Kenya/Nigeria, but I'm struggling to find a company selling them. There's one for Nigeria listed on didx.com, but $22/month seems a little steep. Has anyone had any luck getting DIDs for countries in Africa, and are there any companies out there selling them? Thanks

Hi Friends, Can you please tell me who is the best VoIP Service Provider using Asterisk (With trail version for sometime) . Waiting for your quick response. Thank you. Regards, Chandra. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML

I'm using SixTel as a test (Opened account w/ $10) and am happy with them so far... In their basic service package, they don't charge a monthly fee, and it's outbound only, and you get charged for every minute. I paid for a DID, which is $1.50 or so per month, and it lets me receive inbound calls, which I also pay for by the minute.. I don't mind this for a service like this

Rather than requiring the scopes to be the first members of their respective structures (so that casts can be used to switch between the different views), properly use types and container_of(). Signed-off-by: Jan Beulich <jbeulich@suse.com> --- a/xen/drivers/passthrough/vtd/dmar.c +++ b/xen/drivers/passthrough/vtd/dmar.c @@ -304,13 +304,15 @@ static int __init scope_device_count(con

I figured it out. The ftp site was not named well and corrected. The other problem I have it after the extraction and make; it was suppose to go under /etc but that did not happen. I am trying to figure out why. On 12/28/07, broadband Voice <broadbandvoice at gmail.com> wrote: > > I successfully downloaded the Asterisk package from Digium but not able > tar zxvf zaptel-*.tar.gz.

Could not find your post for 4 months ago. -------------- Original message -------------- From: "Anton Krall" <akrall-lists@intruder.com.mx> > 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: asterisk-users-bounces@lists.digium.com >

On this weeks VUC call we will collectively be our own guests. That is, we'd like to know what was the big issue that impacted YOU in 2010? All opinions welcome. Here are a few things to get you thinking in advance: - Apple's Antenna-gate - Asterisk 1.8 Launches - Amazon EC2 as a DOS platform - Cisco launched UMI video conference device - More HDVoice capable phones - Skype Outage - VoIP

I am receiving this message on my asterisk server and I have commented out 5748150837 in my sip.conf file but it keeps showing this message on the server. Dec 10 07:59:31 NOTICE[30448]: chan_sip.c:5267 sip_reg_timeout: -- Registration for '5748150837@69.25.143.141' timed out, trying again (Attempt #1546) any ideas? -------------- next part -------------- An HTML attachment was

Digium has a compatibility list of servers, however, it has not been updated since 2006. One of the servers on the list has since been taken out of production by Dell. Here are the remaining servers on the list: HP Proliant DL360IBM x206IBM x346 Does anyone has a most recent list and I will be adding the digium cards for T1 the 220 series with echo cancellation? -------------- next part

hi guy, i have a problem, i have an sellvoip account and i want configure asterisk for outbound calls. this is my sip.conf register => XXXXX0000000000:PassWord@70.42.34.200 ; this is one of the sellvoip server [sellvoip_out] type=friend secret=PassWord username=XXXXXX0000000000 host=70.42.34.200 dtmfmode=rfc2833 context=testing disallow=all allow=ulaw extensions.conf this is a semplified

We have two servers but looks like G729 issues. Works fine on the old server and not sure if it is T1 related. See SIP Debug. Any experiences to share. Thanks --- Newark1*CLI> <--- SIP read from 194.xx.Xx.Xx:5060 ---> SIP/2.0 183 Session progress Via: SIP/2.0/UDP 76.xx.xx.xx:5060;branch=xxxxK784d2637;rport From: "Cell Phone DC" <sip:202xxxxxxx at

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 project. It won't be a comercial project, it would be free and it's target would be spanish talking asterisk enthusiasts. I'm wrinting these for the sake of finding contributors, people who want to help me