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Hello, Is there or can anyone provide a comprehensive guide (designed for Linux/Asterisk novices) to installing/setting up Asterisk in order to support Fax2Mail service? In my case, I would like Asterisk to receive fax calls to predefined numbers (ranges) and to associate each of these numbers to email addresses. Thank you in advance. David --------------------------------- Yahoo!

Hi I have a problem with Asterisk-1.6.0.3-rc1 and manager API. I want to dial out from manager's console and with Asterisk 1.4.X this settings were OK. Action: Originate Channel: SIP/384 Context: main Exten: 102 Priority: 1 Callerid: 384 I could dial out, but with asterisk 1.6 I get this error. Response: Error Message: Channel not specified I have originate and system privilege in

Hi, Did anyone try to set up alarmreceiver application over IP network? Which ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck. Maybe I did something wrong with alarmreceiver.conf (I tried diverse settings, but nothing worked). Sometimes alarmreceiver is able to get some events but sometimes not. I think Linksys PAP-2 has a problem with recognizing digits in appropriate

Hi, I have to build Asterisk server for about 30 user (30 concurrent calls). I decided to buy this box: -- motherboard Intel E7210 + Hence Rapids -- processor P4 3.0 GHz -- RAM 2x512 MB DDR ECC -- network interface Intel 82541 GI Is this configuration enough to handle 30 users at the same time. I am not planning to use any transcoding (everything will be alaw). Cheers Andrew --------------

Hi, Few days ago I installed Email2fax application on my Asterisk box. I works but not in 100 %. Sometimes (to be certain quite often) I don't receive the whole fax. My fax machine cuts off transmission in 1/2 or 1/3 of the page. I read about it on a wiki and some user lists and people say that this behaviour could be cause be the Ghostscript and the conversion to the tiff format, but when I

Hi I am keep getting this warning message when doing attendant transfer: WARNING[6027] channel.c: Codec mismatch on channel Local/XX at Inside-1f32;1 setting write format to slin from alaw native formats (alaw) What can I do to lose it. I am using asterisk 10.1.2 Best regards

Hi I have a question if there was a major change in CDR? Few days ago I have upgraded to 1.4.12.1 from 1.4.4 and something bizarre happened. After the upgrade I have no call details in the cdr table when the call did not go through because of for example: Unable to create the channel of type Sip - no route to destination. In such situation the call does not exist in the cdr table while it was

Hi, Does anyone know what is the amount of max concurrent calls that can be made in one Asterisk box? I heard that it is 256 and it doesn't depend on how good your machine is. It is the program constraint. What can I do when I need to have more calls than that. I read about connecting Asterisk boxes with IAX. Is it a good solution? Does anyone have other proposals? Cheers Andrew

...mply don't work: attach (emails sent if there is something in the email field) maxsilence (docs say the default is 0/off, but the default is 10s) maxmessage minmessage maxlogins (how hard can this be?) pbxskip Has anyone got any idea on this? Doug. -----Original Message----- From: Andrew Nowrot [mailto:andrew.nowrot@gmail.com] Sent: Tuesday, June 27, 2006 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call length limitation On 6/27/06, William Piper < william.piper@gmail.com> wrote: Well, It was worth a shot. Perhaps doing a so...

Hi I am using asterisk-1.6.0.6 and I have noticed strange behaviour lately. When a user ends his call asterisk executes twice the h extensions (in my case this is the AGI script) and writes this to the logs: cdr.c: CDR on channel 'SIP/xxxxxx-b6623038' already posted. and after that it crashes immediately. This had happened twice so far. Does anyone know what is causing this.? Cheers

Hi I am having a problem with extension matching in asterisk (I am using asterisk 1.6.0.6). Is there a difference between extensions matching in realtime architecture and extensions matching in extensions.conf file. For example when I have these two extensions: -- _0699[134]XXXXX -- _06[069]XXXXXXX that are in the database and number 0699123123 comes in asterisk will always choose exten

Hi Is there a way to disable blind and attended transfer during a call. I am trying this configuration but unfortunately with no luck: - in features.conf [applicationmap] disabletransfer => 9*9,self,GoSub(disabletransfer,s,1) - in extensions.conf [incoming] exten => 99,1,Set(__DYNAMIC_FEATURES=disabletransfer) exten => 99,n,Dial(Sip/alice,120,tT) exten => 99,n,Hangup()

Hi, I have built another Asterisk box using one ISDN HFC-S card and Bristuff-0.2.0-RC8p. But this time it behaves very strangely. Asterisk simply hangs and in logs I receive something like this: --NOTICE [1197]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 --NOTICE [1197]: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 And when I want to call a ZAP channel I get

Hi, I have small issue with Asterisk. My customers complaining that sometimes (not always) the outgoing voice (the voice which can be heard by the user a the other end) quality is very low (stutter and sudden clicks). The problem exist in only-IP configuration and in IPtoTDM connections as well. I use alaw codecs. I know that they consume a lot of bandwidth, but the upload and download stream is

Hi I am trying to build reliable fax solution with asterisk, iaxmodem and hylafax. I am attempting to do this on Compaq DL-360 with 2 pentium 3 1.2GHz (512 cache) and 2GB of RAM. I am using a Sangoma A101. After installing the newest zaptel and wanpipe-3.1.0 beta I did zttest and it didn't give me good results: 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586%

Hi I am having problems with Asterisk 1.4.18 and realtime architecture. I use Postgresql-8.3 as the database. Everything works OK; all sip phones (their configs are in the database) are able to register to the server and I can make calls (dialplan is in the database), but each time Asterisk reads the information from the database it shows me this on the console: [Feb 17 12:32:50] WARNING[620]:

The USB core was completely rewritten in 2.6, and as such the functions that ztdummy depends on do not exist in 2.6. I get the feeling that these changes are too much to easily fix ztdummy, so I don't expect to see it working on 2.6 any time soon (if ever) I made some small changes to zaprtc to work on 2.6 and I have MoH and Meetme functions working fine in my lab. For production I would

Hi, In my dialplan I'm using a WaitExten() command. It works only with Zap phones. When I dial this command with Sip phone asterisk do nothing. Should I put extra definition in sip.conf to make this work with Sip phones? Thanks in advance Cheers

Hi, Can anyone tell me if Asterisk sets some variable when doing a call parking (when someone presses an exten set in features.conf). In can't find this information on a wiki. Cheers ;) Andrew

Hi, I am still struggling with the E1 card!!!! Does anyone has some experience with sangoma E1 card? I have this card in soekris net 4801. First I was runnig it with deactivated DMA and I was receiving overruns (even with no channels in use). Then I enabled the DMA. Now I have the overruns only during a call. When I'm using more than one channel, the voice is stuttering and the delay is very