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Hello all, ? I have some issues with the MeetMe application. ? The working topology is as follows. The Asterisk (1.4.22-rc5) is connected through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco Voice Gateway. The Gateway is connected to PSTN through a PRI. The calls are forwarded to Asterisk by the CM. ? The problem is that some users who are calling in from PSTN are getting

I yesterday brought up to date the version of * the CVS and now I have a problem. I cannot effect the RELOAD that * it breaks. Somebody can help or say as to load new users without stopping * ? Thank?s Excuse my English Joao Carlos Moura

Hello list I don't get to compile h323. I have the mistake: asteriskaudio.cxx: In destructor `virtual PAsteriskSoundChannel::~PAsteriskSoundChannel()': asteriskaudio.cxx:167: `baseChannel' undeclared (first use this function) asteriskaudio.cxx:167: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: ** [asteriskaudio.o] Erro 1 make[1]:

My CDR is displaying wildly inaccurate results. When I make a call the CDR records the time between connecting into the server and hanging up, instead of recording the time between dialling from the server to the PSTN destination via VOIP termination. It is alright to log the duration of the connection to the server, but why it does not log calls for termination via voip provider is the main

I have been trying to connect via sip and things don't seem to work. What do messages like this mean? Oct 9 00:33:57 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81ab834 (len 361) to 216.127.66.119 returned -1: Invalid argument Oct 9 00:33:58 WARNING[22849]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 000638cf3adb579455c0d20b2051ba1d@127.0.0.1 for seqno 102

Is there a key sequence to stop a call as its ringing, before the call is answered? The * key stops a call after it is answered, but I'd like a way to cancel the call during the ringing phase. /Obelix ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because you current number it is not being answered, and you don't want to hangup before dialling again. /Obelix

Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because the current number it is not being answered, and you don't want to hangup before dialling another number. /Obelix

Which Actions and events to the read/write options in manager.conf give access to, ie the options below. read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Are they documented somewhere? /Obelix

We need incoming calls to simultaneously ring SIP phones, and a cell phone which is called via a SIP or IAX trunk. When the cell phone answers we'd like a brief prompt played (e.g. "press # to accept call") and if # is pressed connect the incoming call to the cell phone. ZAP trunks have some of this functionality with the call confirmation option, but we must use SIP or IAX trunks.

I am running a prepaid application with Asterisk. When authentication has to be done by DTMF everything works fine. However when the user is authenticated directly from the sip phone, the channel variables seems to disappear. Trying to retrieve the channel status always returns -1 instead of the 6 that happens normally. It also seems to affected the DIALSTATUS and ANSWEREDTIME variables. The

Hello all, This is my first message to the mailing list. I am seeking advice as to how to proceed/what to get for my current situation. I want to use asterisk to make a system that does pure outbound calls and plays a message upon a live answer or answering machine. Basically it needs to make several calls at once. This is non-profit and is to help out a question we have on the local

asterisk*CLI> dahdi show status Description Alarms IRQ bpviol CRC4 T2XXP (PCI) Card 0 Span 1 OK 0 0 0 T2XXP (PCI) Card 0 Span 2 RED 0 0 0 On Thu, Apr 2, 2009 at 9:40 PM, Martin <asterisklist at callthem.info> wrote: > That's very strange ... the code when is compiling checks whether > zaptel is present and then > the #define HAVE_ZAPTEL is set. > > Since your error says No "ZAP" channel ... > > and the code says > > ast_log(LOG_WARNING, &quo...

Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look at your install and they said we are having a NAT problem but didn'ttell me if it was with the asterisk conf or the Cisco ASA. _________________________________________________________________ Hotmail: Trusted email with Microsoft?s powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/

Hi Guys, I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: [Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy, avgtone: 75, avgsilence 135 [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing [h at macro-dialout-trunk:1] Macro("SIP/2111-b6a400b0", "hangupcall|") in new stack [Jul

...rations (broadbandvoice@comcast.net) 16. RE: No DTMF (broadbandvoice@comcast.net) 17. RE: Don't see my post (Steven Totaro) ---------------------------------------------------------------------- Message: 1 Date: Sat, 22 Apr 2006 19:17:52 +0000 From: Roshan Sembacuttiaratchy <rns.asterisklist.n.semba@xoxy.net> Subject: Re: [Asterisk-Users] Sipura SP3000 question To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <20060422191752.GR24768@roshan.info> Content-Type: text/plain; charset=us-ascii On Sat, Apr 22, 2006 at 1...

This is a global message to all to announce our callcenter / dialer / predictive dialer / vicidial program is now open. Codecs: G711, GSM, G729, G723 Protocols: SIP Duration Rate : 30/6 (6/6 with monthly minutes over 100,000) Channels : 100 to start with , more on demand. We are predictive dialer friendly , your account will not be shut off. Contact us to do a test run. Mike

...usic on hold is played and again the announce message is played. somehow the music on lod doesn start. What am I doing wrong? I run version CVS-12/01/03-14:50:57 Thanks -- Anton Yurchenko<phila@dg.net.ua> Digital Generation --__--__-- Message: 5 From: "Asterisk List" <asterisklist@hotmail.com> To: asterisk-users@lists.digium.com Date: Mon, 26 Jan 2004 13:02:47 +0000 Subject: [Asterisk-Users] Know if a call is answered Reply-To: asterisk-users@lists.digium.com Hello: I have an asterisk server answering SIP calls. Whenever a call comes, asterisk answers, plays a gsm file...

Hello: I need to prepare some detailed stats from asterisk, and I'm asked to show data I don't know how to obtain it: It's the 'final' number (don't know what's its name) In the stats I have to show the caller_id (I have it), the called_id (I have it) and the final number that actually accepted the call. In extensions.conf file, I try to pass the call to several

Hello: I have an asterisk server answering SIP calls. Whenever a call comes, asterisk answers, plays a gsm file (information) and dials to another SIP phone. Using asterisk Master.csv file I only have one record, and don't know if the second call is answered. I only know this if: - The called phone is busy - The called phone doesn't answer in X seconds (the parameter in Dial) But I