similar to: Latest batch of CVS changes

When I enable callprogress I get this error message... (when I call, it will ring forever but asterisk is acting as if it DID pick up the line ... but it never did... ) I'm beginning to think that asterisk can't hangup the line when in voice mail and this, whatever action I'll take... Actually, callprogress doesn't seem to work at all... (in my case) Mar 24 23:37:28

Hello, I've commited the new busydetect routine to CVS. You need to cvs update asterisk of course and then choose it in asterisk/Makefile and recompile asterisk. All you X100P users that had the problems with false hangups or the card not being able to detect the busy tone please check that. In the asterisk/Makefile you need to find a line BUSYDETECT = and uncomment what you want/ comment

Hi, (sorry for my mistake in not deleting the rest of the message just now) The problem seems to be here in zaptel.c (and torisa.c) #ifdef DEFINE_SPINLOCK static DEFINE_SPINLOCK(zaptimerlock); static DEFINE_SPINLOCK(bigzaplock); #else static spinlock_t zaptimerlock = SPIN_LOCK_UNLOCKED; static spinlock_t bigzaplock = SPIN_LOCK_UNLOCKED; #endif If I remark out as follows: //#ifdef

I had a compile problem with the CVS I downloaded on 21 June. I have a Debian box with 2.4.18 kernel (version needed for support of Conexant ADSL). There is a difficulty with Zaptel build regarding HDLC detection. It tries to build it in and then results in unresolved kernel symbols and fails to load. I have had to comment out the entire HDLC defines in zconfig.h to get a driver to install at

Hi, Updating zaptel gives me this during the make. Any ideas, the searches and Wiki gives me no hints. In file included from /usr/src/linux-2.4/include/linux/fs.h:19, from /usr/src/linux-2.4/include/linux/capability.h:17, from /usr/src/linux-2.4/include/linux/binfmts.h:5, from /usr/src/linux-2.4/include/linux/sched.h:9, from

Using CVS update from 11:00 CET today * crashes at this point. == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': Found Sheriff*CLI> Disconnected from Asterisk server -- Dave Cotton <dcotton@linuxautrement.com>

Just idly watching * in console mode and saw that someone from 50.49.54.102 tried to register with my *. whois gives:- OrgName: Internet Assigned Numbers Authority OrgID: IANA Address: 4676 Admiralty Way, Suite 330 City: Marina del Rey StateProv: CA PostalCode: 90292-6695 Country: US NetRange: 50.0.0.0 - 50.255.255.255 CIDR: 50.0.0.0/8 NetName: RESERVED-50

For those who don't wake up at 5.00 am and start reading /. http://searchnetworking.techtarget.com/originalContent/0,289142,sid7_gci935769,00.html -- Dave Cotton <dcotton@linuxautrement.com>

I just got an email from SIPphone advising that there have been problems with the above firmware and advising to reload from their server. This does in fact reload 1.0.4.35 into the phone. And now voicemail has gone AWOL again. -- Dave Cotton <dcotton@linuxautrement.com>

I'm getting this message when I start Asterisk chan_capi.c:2205 capi_handle_msg: Command.Subcommand = 0x5.0x81 but when I try and recompile I get this chan_capi.c:60: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) any help would be greatly appreciated. -- Dave Cotton <dcotton@linuxautrement.com>

I really can't understand what is happening. I have the same set up in two locations:- 2 fritz cards in each with the patches and the same capi.conf except for the MSNs. One installation calls in on controller 1 and starts calls out on controller 2, i.e. perfect. The other calls come in on both controllers at the same time. I have checked the patches etc. Is this possibly a problem caused

I'm getting this error when compiling:- make[1]: Entering directory `/usr/src/asterisk.svn/sounds' --09:22:12-- http://ftp.digium.com/pub/telephony/sounds/releases/asterisk-core-sounds-en-wav-1.4.0.tar.gz => `asterisk-core-sounds-en-wav-1.4.0.tar.gz' Resolving ftp.digium.com... 216.27.40.102, 69.16.138.164 Connecting to ftp.digium.com|216.27.40.102|:80... connected. HTTP

On a different note using Fedora Core 3 I get CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_chan_write': /usr/src/zaptel/zaptel.c:1745: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c: In function `ioctl_load_zone': /usr/src/zaptel/zaptel.c:2392: warning: ignoring return value of

Yeah sorry about that. But I didnt see my message in the list, so I thought it didn't ame through. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Dave Cotton Sendt: 14. september 2005 19:45 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] RxFax problems On Wed,

Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dave Cotton Sent: 16 February 2006 14:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Firmware version 1.3.1 released for AastraIPphones On Thu, 2006-02-16 at 13:28 +0000, Lee Archer wrote: > There is no

I went to the Linux Solutions exhibition in Paris yesterday, visited the well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6 hours later there was only one left. It must say something, also it was the English version. -- Dave Cotton <dcotton@linuxautrement.com>

Since firmware 1.0.3.81, unless I'm imagining things, Voicemail2 seems to be having problems. The Grandstream and sip.conf were set to RFC2833 now with that setting I get extra digits during "Mailbox" and "Password" phases. 222001 instead of 2201 for example. When both are changed to "SIP info" there is no problem. But what is the new setting "DTMF Payload

Firstly, let me just say I am new to asterisk and if anything I've said is covered in an FAQ or in previous posts I apologise but I have tried searching and I've attempted a few of the things I found but they didn't help. Has anybody got any experience using an X100P on an NTL phone line in the UK (I'm in an ex Cable & Wireless area if that makes any difference). The

Some one on another list I subscribe to had a session with an annoying IVR system at their doctor and posted this link. http://www.pendulum.org/humor/humor_psych_hotline.html -- Dave Cotton <dcotton@linuxautrement.com>

Am I the only person in the * world who can't get a sip connection through an iptables firewall? I've got everything else working fine. Xten <-> PSTN, Xten <-> Analog, IAX <-> IAX, but exten => 3733,1,Dial(SIP/fred@somewhere.com) ; evades me, ngrep @ port 5060 says the INVITES go out but how do I get something back? -- Dave Cotton <dcotton@linuxautrement.com>