similar to: s and i in context not invoked

Hi, I'm trying to compile zttool from the Zaptel lib, but I just can't find the sorcecode for Libnewt. Anyone got a link? Since i'm using LFS, I can't use precompiled packages. -- Med venlig hilsen / Best regards Michael L?jtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 S?borg Tel (+45) 3955 0700 - Fax (+45) 3955 0707

Hi, Just started to get this error after updating to the latest CVS. Asterisk dies if it can't create a channel - not so good. -- Executing SetCallerID("SIP/750-2550", "39660426") in new stack -- Executing Dial("SIP/750-2550", "CAPI/39660426:22179808") in new stack Jun 22 13:52:05 NOTICE[262161]: chan_capi.c:1172 capi_request: didn't find

Hi, I have seen a couple of scripts that should be able to remotely reboot the 79xx phones, but I haven't been able to make it work for my 7940. Anyone able to guide me in the right direction? I am running the SIP 7.1 firmware. -- Med venlig hilsen / Best regards Michael L?jtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 S?borg Tel (+45) 3955 0700 - Fax (+45) 3955

Hi, I have a problem that I just can't figure out how to solve. I start *, dial it using a ISDN phone over PSTM, to a Hisax card installed in * I get the demo-greeting, listen for a few seconds and hang up. I dial it again, but this time the first second is sound from where the previous call ended, then the greeting starts as it should. Right now I have removed all codecs but codec_gsm.so

Hi, If I try to transfer a user (user listens to MOH while I transfer) to eg. a queue, and the transfer occour while the MOH in the queue is playing, the MOH will stop, and the user hears nothing but scilence, but is in the queue. If I make the transfer to the queue, while still listening to the announcement, the user will hear the announcement, and then the MOH will start. Using latest CVS

Hi, I am trying to track down the reason to my problems with sending and reciving fax with my PRI and 2 TDM400P Cards: PSTN <-> PRI (E100P) <-> * <-> TDM400P <-> Fax Machine I have used Zapbarge to listen to the data stream, but I can't say if it really have some time slips - fax kinda noisy in itself. Using the zttool i saw the Sync source for the TDM are

Hi, I am trying to recice a fax with * using SpanDSP - but it doesn't create the output file. (See the bottom of log file). * Loads both app_rxfax.so and app_txfax.so fine. Also I can't make * autodetect an incomming fax call (yes I have enabled faxdetect=both in zapata.conf - though it's not a Zap device) Any ideas are welcome :-) Best Regards Michael L?jtnant System Details:

Guys, Ok - nowhere near as complex as most of the discussions on here ( ex telco engr for 18 years here).. But thought I'd ask for some assistance. Have just set up my first * Pbx - having a play with it and a couple of Cisco 7960 (configured as SIP) phones. The phones are tftp'ing into the server ok, and picking up the configs all ok. Everything _seems_ to be working, but I

Typo in your OS79XX.TXT P00 ? instead of P0S !? -----Original Message----- From: Michael L?jtnant [mailto:ml@zyxel.dk] Sent: 17 August 2004 13:31 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released Hi Shaun, Saw you post, and rushed to their ftp-server and downloaded it :-) But, I can't make my phone (7940) upgrade, so maybe you

Hi I have two small question, if you can help me ;=) Problems with Answer+Music my extension: [Cal-In] exten => _811XXXX20,1,Goto(C-Internal,100,1) exten => _811XXXX21,1,Goto(C-Internal,200,1) [C-Phibee] exten => 100,1,Ringing exten => 100,2,Wait,1 exten => 100,3,Answer exten => 100,4,Dial(SIP/201&SIP/200,30) exten => 100,5,Hangup exten =>

Hi Leo, here's a suggestion: in your dialplan (extensions.conf) send multiple users to the same mailbox (e.g. 999) if they do not pick up within 30 seconds: ; SIP Phone 100, Tom exten => 100,1,Dial(SIP/100,30) exten => 100,2,VoiceMail(999) ; SIP Phone 200, Eric exten => 200,1,Dial(SIP/200,30) exten => 200,2,VoiceMail(999) and in your voicemail.conf you do not specify a single

This 2 line code is doing what I wanted. exten => 200,1,voicemail(200) exten => 200,2,Hangup What I've been told is that they want the 20 year old phone system to light up the message bulb. (yea, a filament bulb, not an LED) To do this you pick up on the line that goes into Asterisk and do a: exten => 200,1,SendDTMF(200w#86) But I don't know the path to take to get that

In LiveIntervals::processImplicitDefs() we have this: for (MachineRegisterInfo::use_iterator UI = mri_->use_begin(Reg), UE = mri_->use_end(); UI != UE; ) { MachineOperand &RMO = UI.getOperand(); MachineInstr *RMI = &*UI; ++UI; MachineBasicBlock *RMBB = RMI->getParent(); if (RMBB == MBB) continue; const

On Jul 17, 2009, at 7:57 AM, David Greene wrote: > In LiveIntervals::processImplicitDefs() we have this: > > for (MachineRegisterInfo::use_iterator UI = mri_->use_begin(Reg), > UE = mri_->use_end(); UI != UE; ) { > MachineOperand &RMO = UI.getOperand(); > MachineInstr *RMI = &*UI; > ++UI; > MachineBasicBlock *RMBB

I can think of many people that use LLVM and maintain out of tree extentions that will likely never be incorporated into mainline. Maintaining out of tree intrnisics takes a fair amount of work and leads to the choice of using a single version of LLVM for a project (and thus have to manually incorporate bug fixes) or risk untimely breakage. I propose that the plugin be extended to allow plugable

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=466 Summary: u.tcp used where u.udp should be, in tftp nat helper Product: netfilter/iptables Version: linux-2.6.x Platform: All OS/Version: All Status: NEW Severity: minor Priority: P2 Component: NAT AssignedTo: laforge@netfilter.org

On Tue, Dec 11, 2012 at 05:30:46PM -0800, Dong Chen wrote: > hi Wei-Ren, > i got it, i have to add a "typedef int uint;" > that's not a big problem Souldn't you use "typedef unsigned uint"? -- Wei-Ren Chen (陳韋任) Computer Systems Lab, Institute of Information Science, Academia Sinica, Taiwan (R.O.C.) Tel:886-2-2788-3799 #1667 Homepage:

Hi, I'm trying to get a Zyxel P2000W (reportedly also sold as WiSIP by Pulver) to work with an asterisk box. The phone connects nicely to an external VoIP company (sipgate.de reportedly using asterisk themselves) but there is a strange problem with my asterisk: - Incoming calls via ISDN (chan_capi) to the Zyxel work perfectly. - Outgoing calls (capi/sip/iax) via asterisk are dialling out

Hi all, I've been using the following method to limit calls to sip clients to 1: exten => 200,1,SetGroup(200) exten => 200,2,CheckGroup(1) exten => 200,3,Dial(SIP/200) exten => 200,103,Busy This works fine for a single extension. However, I also need to dial groups of sip clients. It appears that SetGroup can only be used once per channel. This (useless) example would not

Hello. I am in a strange situation. I have two asterisk. Asterisk "A" makes a call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers it to Openser by SIP. The problem is openser printing this in the screen: ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> . ERROR:parse_from_header: bad from header