search for: zap2

...xt for both lines. With the setup the way I want it, last two lines of channel 1 configuration and whole channel 2 configuration in zapata.conf would be uncommented and there would be no fax detection in from-analog-zap context - as per the comments in the config. As it is now, from- analog-zap2 may not be giving enough time for fax detection, but I've tried variations. Generally, all it will do is keep on ringing and ringing and not detecting the fax tone. I've tried turning off echo cancelation and few other things with no luck. Is this possible bug in chan_zap or somethin...

...setup_zap: Unable to register channel '1' Mar 13 20:43:35 WARNING[5779]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 ] As a matter of fact this is what is listed in /dev/: aaberga@epsilon-stargate:~/sources/voip/zaptel-1.0.6> ls /dev [....] z2ram zap1 zap2 zap3 zap4 zapchannel zapctl zappseudo zaptimer zero zkshim zqft0 I edited the list to avoid a huge message. As said I am not a Linux low level expert ... still it's striking that Asterisk does not find a pesudo file /zap/channel and there is something similar, ie zapchannel. Anyways the zapt...

...t SCEV* srcSCEV = (srcIdx != srcEnd) ? SE->getSCEV(*srcIdx) :* * GetZeroSCEV(SE);* * srcOpds.push_back(srcSCEV);* * }* But in my various test cases, I never see the loop iterate more than once. That is, there always seems to be exactly 1 index. Here's a typical test case: *i**nt zap2(long int n, long int A[][n]) {* * long int sum = 0;* * for (long int i = 0; i < n; i++)* * for (long int j = i; j < n; j++)* * sum += A[i][j];* * return sum;* *}* Any ideas about what I'm doing wrong? Could it be that I'm not running certain important passes? Here's...

...After doing some test on my asterisk box I can successfully receive calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network Dial out from a sip phone is also not an issue, all calls connect and terminate normally. If I call the Asterisk PBX say from PSTN in Zap1-1 and out through ZAP2-2 back to the PSTN (after entering the correct pin off course) the card does not appear to detect the hang-up, I then have to issues a soft hang-up to close the call, I presume this indicates the card is configured to receive the correct hangup signal I have tried enabling callprogress, busydetec...

...aptel and tried to load the wctdm module but this failed with the following message: Notice: Configuration file is /etc/zaptel.conf line 206: Unable to open master device '/dev/zap/ctl' I checked /dev/ and there is no /zap directory but, instead, some devices that look relevant like zap1, zap2 etc. Obviously, if boot with the older kernel, wctdm will not load either. Any help is welcome Thanks in advance, dionisis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050912/c688e9ea/attachment.htm

...day I had a call and could not get it to transfer. The logs show asterisk getting the DTMF #, then a bunch of stuff about a zap channel not being answered yet. I see two #'s coming in withing the time configured in features.conf but being ignored. It looks like the original call came in on Zap2-1 and asterisk called my cell phone on Zap3-1 and bridged the calls but still thinks Zap2-1 is still dialing and ignores my dtmf even though I am on Zap3-1. Here are a few lines from the log: Jan 25 11:29:20 DEBUG[1131] chan_zap.c: DTMF digit: # on Zap/3-1 Jan 25 11:29:20 DEBUG[1131] chan_zap.c:...

Hi, Here is a preliminary (monolithic) version you can comment on. This is still buggy, however, and I'll be testing for and fixing bugs over the next few days. I've used your version of the strong siv test. Thanks! -- Sanjoy Das. http://playingwithpointers.com -------------- next part -------------- A non-text attachment was scrubbed... Name: patch.diff Type: application/octet-stream

I'm having a major problem at one of my branch offices with "Phantom Rings" on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and

Hi all How can I make * ring one phone then if no answer Go to a different extension ?? Any help always appreciated Regards Mick

I have a TDM400 running telco lines on ZAP2-4 My after hours config is supposed to receive the incoming call then divert it to my home phone by calling out one of the other zap channels available. console output as such... Starting simple switch on 'Zap/3-1' -- Executing Dial("Zap/3-1", "ZAP/G1/0823274210&qu...

...terisk determine that the call is coming in from 5611, > and will it understand the G3's caller ID? > Only if you can configure the G3 to pass caller id info, our G3 admin couldn't. the 4 lines you want to pass to Asterisk will connect via their own ZAP channel. 5610=ZAP1, 5611, ZAP2 and so on. You can use that in your dial-plan to pass incoming to certain extensions on Asterisk. Since this is only a proof of concept, this should do very well. But, keep in mind that the X100P cards have limits on the number of cards within 1 PC. They generate LARGE amounts of interrupts....

AS a proof of concept experiment, I want to try and integrate Asterisk with my Lucent Definity G3 switch. I don?t have an available T1 port on the G3 but I can round up 4 analog ports off the G3. What I thought I could do is create a ?Hunt group? on the G3. Let?s say I configure 5610 ? 5613 as a hunt group. I know I would need a 4 port FXO card to install in the Asterisk server. Two configuration

...g phase. I think it's a timing error, however I have no idea about how to rectify the problem. I have included a copy of the log below. I am using a Digium TDM-400P card with 2 x FXO ports and 2 x FXS ports. The fax is connected to one of the FXS ports (Zap3). The outgoing call dials out on Zap2 and is received by SpanDSP on Zap1. I'm not too sure what the Zaptel device is using as a timing source or how to alter it in order to tune the carrier frequency. If anyone has any ideas I'd be very appreciative of your assistance. Many thanks, Stephen. -- Starting simple switch on...

...'/etc/udev/rules.d//50-udev.rules' at line 34 applied, 'zap1' becomes 'zap/%n' Sep 26 22:39:09 voip udev[3608]: creating device node '/udev/zap/1' Sep 26 22:39:09 voip udev[3615]: configured rule in '/etc/udev/rules.d//50-udev.rules' at line 34 applied, 'zap2' becomes 'zap/%n' Sep 26 22:39:09 voip udev[3615]: creating device node '/udev/zap/2' Sep 26 22:39:09 voip udev[3622]: configured rule in '/etc/udev/rules.d//50-udev.rules' at line 34 applied, 'zap3' becomes 'zap/%n' Sep 26 22:39:09 voip udev[3622]: creat...

Hello, I have *@Home 1.5 installed and all is working fine for incoming calls and sometimes outgoing calls. Installed in the box is a digium TDM04B (4xFXO Ports) setup as ZAP1 to ZAP4. I have incoming calls coming in on lines 1-4 in that order and outgoing calls prefering ZAP4 then ZAP3 then ZAP2. When i try to dial out to the PSTN from a SIP phone it sometimes works (normally after a reboot) and a few calls later i get the voiceprompt "All circuits are busy....." I know the PSTN line is on hook and ok to use. Looking in the log file i can see a line saying unable to create...

...g phase. I think it's a timing error, however I have no idea about how to rectify the problem. I have included a copy of the log below. I am using a Digium TDM-400P card with 2 x FXO ports and 2 x FXS ports. The fax is connected to one of the FXS ports (Zap3). The outgoing call dials out on Zap2 and is received by SpanDSP on Zap1. If anyone has any ideas I'd be very appreciative of your assistance. Many thanks, Stephen. [1;30;40m -- [0;37;40mStarting simple switch on 'Zap/3-1' [1;30;40m -- [0;37;40mExecuting [1;36;40mDial [0;37;40m(" [1;35;40mZap/3-1 [0;37;40...

...operly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to <linux-hotplug-devel@lists.sourceforge.net> Jan 5 17:57:59 asterisk wait_for_sysfs[2784]: either wait_for_sysfs (udev 039)needs an update to handle the device '/class/zaptel/zap2' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to <linux-hotplug-devel@lists.sourceforge.net> Jan 5 17:57:59 asterisk wait_for_sysfs[2786]: either wait_for_sysfs (udev 039)needs an update to handle the device '/class...

...has run into this problem. What I'm trying to do is simple. Just make and outbound call using the /var/spool/asterisk/outgoing directory. I copied /usr/src/asterisk/sample.call and only changed the context & extension. I configured my Zap1 to the same context. I have two X100P (Zap1 & Zap2) and one S100U (Zap3). If I use my S100U and dial extension 800, it works. It calls. However when I copy my 1.call file. it says: Unable to create channel of type 'Zap'. Does anyone have any suggestions? or know what am I missing? Thanks, Dante Here's my configuration: extension...

...h asterisk documents. My asterisk system is connected to an ISDN line with HFC card. I use zaphfc module for that. In my extensions I tried several things to dial out and use the *31* but without success. E.g.: 2000,1,Dial(Zap/4/*31*040268000) When I dial 2000 , the verbose logging shows below (Zap2-1 is my internal phone , Zap/4 is connected to outside ISDN line. -- Executing Dial("Zap/2-1", "Zap/4/*31*040268000") in new stack -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called 4/*31*040268000 -- Zap/4-1 is making progress passing it to Zap/2-1...

...y setup is: Analog line ->X100P->asterisk->TDM10B- >phone cord->Dialogic analog port->IVR system. Asterisk should dial the IVR system, which should answer and play back its IVR scenario script to the caller. However, when a call comes in, asterisk answers on Zap1-1 and dials Zap2-1. The IVR system answers the call and begins to play back its scenario. After 5 to 15 seconds, asterisk apparently senses an on-hook condition (exception 17?) and disconnects the connection bridge. The logs on the IVR system shows that it is not initially aware of the hangup, and continues pla...