search for: zap3

ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and archive w/o good results. Thks in advance for any help, Dave sip.conf -------- [general] port =

...p_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 zaptel mo...

...ansfer. 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: Dropping frame since I'm still dialing...

Hello, I've spent a couple days searching and posted into the forum with no luck, apologies to anyone who reads the Digium forums for the cross-post. I'm having a problem with an asterisk set up where I have a TDM402B connected to a POTS line. Also connected to the POTS line are plain telephones, non SIP, just plain old telephones. When one of the normal handsets goes off-hook,

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

...ax message seems to fail in the training 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, Ste...

...'/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]: creating device node '/udev/zap/3' Sep 26 22:39:09 voip udev[3629]: configured rule in '/etc/udev/rules.d//50-udev.rules' at line 34 applied, 'zap4' becomes 'zap/%n' Sep 26 22:39:09 voip udev[3629]: 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...

...ax message seems to fail in the training 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;...

...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[2786]: either wait_for_sysfs (udev 039)needs an update to handle the device '/class/zaptel/zap3' 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[2788]: either wait_for_sysfs (udev 039)needs an update to handle the device '/class...

...oblem. 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: extensions: [callme] ... exten...