similar to: * -> Lucent PBX

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

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

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

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

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

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

This one is for the archives. I got a call today that the * at one of my clients was not working. The switchboard is set up to ring for a while and then the rest of the phones start up if the switchboard doesn't pick up. This was not happening. Instead the mobile phone of one of the people there was ringing and after the delay the internals started ringing. When I connected to the web

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>

I have SIP registrations working correctly for FWD and Sipphone, but it is impossible to connect to Sipcall or Nikotel, I saw that someone on the list has problems with ICH. To try and sort out the problem I tried to register to Sipcall with Linphone and sent the dialogs to tech support of the equipment provider. Here is their answer:- The reason the registration fails is because not

Up until yesterday I've had a lot of high pitched noise when connecting a BT101 to the PSTN via the X100P. I was using an Asus A7V133 with raid motherboard and an 850 AMD Duron. Over the weekend I thought I'd try another machine. I had an HP Vectra 400 Mhz PII MMX with 128 Mb RAM available, today no noise at all. Now I must see if the Vectra is up to the job. -- Dave Cotton

With the current CVS-HEAD line 88 of app_rxfax.c causes an error. #if (ASTERISK_VERSION_NUM <= 010300) chan->callerid, app_rxfax.c:88: error: 'struct ast_channel' has no member named 'callerid' Commenting out the if else combination of course gives a clean compile. -- Dave Cotton <dcotton@linuxautrement.com>