[asterisk-users] Is existing CDR in Asterisk is enough for complete billing

Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on thi

[asterisk-users] Quintum AFT800 on Asterisk 1.4.29

Hi All, Is it possible to use Quintum AFT800 on Asterisk 1.4.29 as Trunk for Analog (like Digium Analog Card) ? And if it's possible, could any one please give me the reference how to configure it on Asterisk 1.4.29. Thanks Regards, Zoel Hairi -- __

Re: [asterisk-users] Early audio (long distance codes) not working after upgrading to 1.8?

I know about the Progress command, but isn't that only for *inbound* channels? It's only outbound calls that I have an issue with. My two test chases are: SIP Phone -> Asterisk -> PRI ...and... Channel Bank -> Asterisk -> PRI -Jon - Original Message - From: "Paul Belanger" To: "Aster

Re: [asterisk-users] Early audio (long distance codes) not working after upgrading to 1.8?

On 10-11-21 09:41 PM, Jonathan C. Bailey wrote: > Does anyone know what changed between 1.4 and 1.8 in regards to early audio > (both hearing it and interacting with it)? > Read UPGRADE.txt and CHANGES *CLI> core show application Progress -- Paul Belanger Digium, Inc. | Software Developer twitt

[asterisk-users] Early audio (long distance codes) not working after upgrading to 1.8?

Hello, We recently upgraded to Asterisk 1.8/DAHDI 2.4/WANPipe 3.5.16. This system is connected to a PRI where the provider requires long distance codes. Normally, you dial, see progress and hear a tone (call is still "unanswered" at this point), enter your code, and it starts ringing as a norma

[asterisk-users] cisco 7970 multiple lines with asterisk

Hi I have a problem that I can't pass. I have asterisk and cisco 7970 phones with 8.0.3 sip firmware. I registered two extensions: Line1: 260 Line2: 160 Regardless of which extension I call, always Line 1 on cisco is blinking. This makes impossible to recognize which extension is calling. Also,

[asterisk-users] SIP Extensions and loss of Internet connection

Hi all! A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. I'm using Asterisk 1.4.24.1. I was researching on the Internet and I found that it can be related to a bug of chan_sip, can

[asterisk-users] DAHDI phantom pickup when ringing

Hi, I've been experiencing trouble with my DAHDI channels for some time and have finally decided to try and resolve the issue. Essentially, the problem I am having is that when a call comes in, and my DAHDI phones therefore ring, Asterisk thinks that one of the handsets has picked up to answer th

Re: [asterisk-users] Asterisk 1.8 VM_DUR problems

After all is this a bug ? Anyone who is using asterisk 1.8 has the same problems ? Thanks, Bogdan From: Bogdan Sarandan Sent: Thursday, November 18, 2010 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.8 VM_DUR problems Hi all, We h

[asterisk-users] How to configure a Linksys PAP2T ATA to connect an analog fax machine to Asterisk

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some

Re: [asterisk-users] Please help me in configuring asterisk for the scenario

On Sun, 21 Nov 2010, Teithi-Chen-Akira wrote: When Mobile user attends an incoming call from asterisk then I want to play a recorded voice  and then I want to get four digit request by following voice instructions from the user mobile keypad and that input can be stored.   I want to know techn

[asterisk-users] Please help me in configuring asterisk for the scenario

Hi All, When Mobile user attends an incoming call from asterisk then I want to play a recorded voice and then I want to get four digit request by following voice instructions from the user mobile keypad and that input can be stored. I want to know technically whether this scenario is possible or

Re: [asterisk-users] Asterisk behind D-Link ADSL router with private IP

On Sun, 21 Nov 2010, gmail wrote: > i have this configuration , An Asterisk server connected to my private > LAN 192.168.10.0/24 when i do port forwarding for port 5060 so that i > make a call from Internet into Asterisk wireshark show the message > "destintion port unrechable" > > i configured

[asterisk-users] Please help me in configuring asterisk for the scenario

Hi All Please help me in configuring asterisk for the below scenario: I want to make calls to some mobile users then My query is simple, my agent attends an incoming call from Queue and I want my agent to play a recording and request the user to input 4 digit input from his keypad and that inp

Re: [asterisk-users] FFA (Fax For Asterisk) tif file (size) problem

> Hi Michael, Hi Steve, > Use spandsp. It is more relaxed about the file resolution, to avoid this > exact issue. Files with a resolution within 5% of 204x196 are accepted. How do we install/use it? Is there an online guide for using it with Asterisk 1.6.2.x? > However, if you have really made

Re: [asterisk-users] ConfBridge

Hi, We created the following script and the conference room works: [incoming-conference] exten => s, 1, Set(CHANNEL(language)=en) exten => s, n, Set(CHANNEL(musicclass)=default) exten => s, n, Ringing exten => s, n, Wait(1) exten => s, n, Answer exten => s, n, Ringing exten => s, n, Wait(1) exten