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...so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. Re: Sipura SP3000 question (Roshan Sembacuttiaratchy) 2. Re: Sipura SP3000 question (Gonzalo Servat) 3. Re: PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3 (broadbandvoice@comcast.net) 4. Re: Sipura SP3000 question (Rich Adamson) 5. How can I get a recording from a CD to my asterisk digital assistant (Davi-Ann) 6. Asterisk on FreeBSD + Passive ISDN BRI (Cian Hughes) 7. Re: How can I get a recording from a CD to my asterisk digital assistant (A...

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

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

...nd calls, which I also pay for by the minute.. I don't mind this for a service like this that doesn't charge me $10 or $25/month... I'm not using it as my primary for outbound or inbound, but it makes a nice backup, and seems to work well... ________________________________ From: broadbandvoice@comcast.net [mailto:broadbandvoice@comcast.net] Sent: Tuesday, April 18, 2006 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] re: Sixtel Services there 2 types of inbound metered and unmetered. unmetered is unlimited inbound and metered charges...

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 am not familiar with that phone. Is it single pair? -----Original Message----- From: broadbandvoice@comcast.net [mailto:broadbandvoice@comcast.net] Sent: Sat 4/22/2006 12:13 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone IncompatibleWith Asterisk 1.2.3 I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with...

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. See log below. Thanks. > > > [root at fc7234153 src]# wget --passive-ftp http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/as...

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

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 >

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

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

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

Hi, Can anyone assist me in resolving this problem? I installed the G729 on a 32 and just found out that the server is 64. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071128/599591ac/attachment.htm

Hi Folks, I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that can handle high volumes. Thanks John. --------------------------------- Yahoo!

I am trying to use the GotoIfTime function and get a busy signal. What I am trying to accomplish is to have the system tell callers that we are closed after 5:00pm. Here is the code below. ; If we're open, then go to the open context ; We're open from 9am to 6pm Monday through Friday exten => 3200,1,GotoIfTime(09:00-17:59,mon-fri,*,*?open,3200,1) ; ; We're also late on Tuesday and

I read the thread about what IP phone is best for business deployment with great interest. Our need is slightly different however. We are deploying VoiP as a value-add with our high speed internet service and are having trouble finding the right SIP analog terminal adapter. In order to support people's existing phones and wiring we need to use an ATA. 1) The first priority is we want

Has anyone had any compatibility issues with a TE110P card installed on a Dell Poweredge 1950? I noted the following error on the LCD display of the Dell Poweredge 1950: E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. The Dell hardware owners manual states that it means the system BIOS has reported a PCI parity error on a component that resides in PCI configuration space at bus 0,

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 set up. [general] context=default ; Default context for incoming calls