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Hi, I can't compile Asterisk on a Debian machine. gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o auto...

Excellent work! Thanks a lot ------------------------------------- Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use

> -- Format for call is ulaw Try the slin codec, I didn't have good results until I used slin.

...any* embedded NIC unless it was eepro100 or 3com (not very common these days) 'cause embedded these days is pretty much crapola. >One additional question -- are VoIP lines generally easier to get going >w/ good sound quality than POTS lines? Yes and no. They are different animals. Asterisk bridges the two, but the kind of latency/ echos / bad call quality etc issues are on the same order of magnitude for PSTN and VoIP. They just require different methodologies and training to troubleshoot. It's easy to set up a crappy VoIP link. >Another thing I was wondering is >whether...

I'm having trouble connecting a asterisk server to a SIP Express router. Inbound calls to my asterisk server works just fine, but when i try to make outbound calls I get the following error message: Nov 27 22:40:48 NOTICE[4687]: chan_sip2.c:7967 handle_response: Failed to WWW-authenticate on INVITE to '"username" <sip:u...

>Please I have combed the Archive to no avail on this problem protocol >control problem in oh323. >I'm receiving calls from CISCO AS5300 -> Asterisk -> Zap Channel. The >calls clears the remote location but drops on my own end. Please what >could be >wrong. I have included the oh323.conf and log files. I have tried >various configuration and I thought I should let some other peer of eyes >look >at it. Any help would...

Hi , I am currently experiencing problems with DTMF detection. My sip-phone using INFO to transfer DTMF, and I see that the version downloaded from asterisk ftp ( 0.4.0) do not support the d= field. I do a clean cvs from a W2K machine and see that in new chan_sip.c , this field has been recognized. But compiling under Linux give me following error. The error come from generating the dependency files. All the Makefile(s) was leave untouched. I w...

Hi, all I am trying to set up voicemail. I've done it to the point where I can leave messages. How do I retrieve them? Actually I have few questions: 1. I want voice mail to be available at certain extension, say 100. How do I set it up so all users can call this number and get to their respective mailboxes. 2. How do I let users to create their own voicemail passwords from the phone? 3.

...oject, than a dead one. Otherwise who is going to patch vulnerab= ilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What is the most stable version of asterisk? On Wed, Mar 23, 2011 at 12:58 PM, Douglas Mortensen <doug at impalanetworks.co= m> wrote: 1.2? 1.4? 1.6? 1.8? Thanks, - Doug Mortensen Network Consultant Impala Network...

<snip> Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102 <snip> This Replaces header refers to RFC3891 which is not yet supported in Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA) This INVITE fails with : <snip> chan_sip.c: Trying to pick up 7792 at subs <snip> app_directed_pickup.c: No target channel found for 7792. If I'm dialing *87792 instead of using BLF, then I'm entering the dialplan part in...

What is required to get the mwi to work? Is it more of a phone subject or *? I have the mailbox= line in sip.conf, but only one extension is named, and in some of the examples, I have seen that there are two... What is that all about and how does it affect the extensions.conf and voicemail.conf? Thanks again..... Just some background as you start seeing my lists, I just started my own business

-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = --_221841ce-c665-475b-9d69-2fe2ae10aa2b_ Content-Type: tex...

...miliar with exactly how it would work in this situation. Anyway, that's it. As for some background, we initially were using ring gro= ups, but realized that these phones do NOT have the ability to handle a 2nd= ringing call. So in the event that 2 inbound calls rang within a few secon= ds, asterisk would send the first to all phones, and then when tyring to se= nd the 2nd, would receive a BUSY message from the phones (because they were= busy processing a ring for the first caller), and the 2nd caller would win= d up going straight to the unavilable destination for the ring group, inste= ad o...