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When I reload the asterisk I get the following warnings: Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring switchtype Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring signalling Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring rxwink Everything works fine as far as I know, I can dial and complete calls. So why am I getting this

Hi, Is there a way I can stop logging this specific messages: Dec 18 13:24:52 ERROR[6913] chan_sip.c: Call to user 'remi' rejected due to usage limit of 1 Without having to completely stop logging all error messages in my log files. Thanks, Remi

Hi, Has anyone ever had any problem with the TE220B card with it showing up as four ports instead of two. I RMA'd the first one with the retailer (Digium tech advice), but I just got another brand new card and it is coming up as four ports again. The card identifier is showing 0420 when I do lspci. Has this happened to anyone and if so is there a fix? Remi

...high CPU load. But when I install the x86_64 Kernel the high CPU load problem disappears. It appears that there must be a kernel setting that is causing the CPU to spike up higher than normal for Asterisk 1.4. Does anyone have an idea on what could be causing this high CPU load? Thanks, Remi Quezada

Hi, I'm getting crazy about ztdummy. I have to replicate a PBX where ztdummy is working fine but for some reason I cannot. The two machines have the same kernel, motherboard, the same gcc version and the same zaptel 1.4.8. On the second machine zaptel compiles without errors and ztdummy.ko is generated but when I modprobe it I get the following error in messages: rtc: lost some interrupts

Hi, I have a Asterisk that connects to the PSTN via a PRI. After Asterisk sends the setup message it immediately sends a 183 Session Progress. Is there a way I can change it so that it sends a 100 Trying instead? Because I am having some issues with a equipment where it does not play a busy tone as a result of sending a 183 Session Progress then the 486 Busy. Thanks Remi

...information: http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html * Significant fixes and improvements to parking lots. (Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430, ASTERISK-17452, ASTERISK-17452, ASTERISK-15792. Reported by: David Cabrejos, Remi Quezada, Philippe Lindheimer, David Woolley, Mat Murdock. Patched by: rmudgett) * Numerous issues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to. A change to Asterisk adds some checks to make sure tha...

...information: http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html * Significant fixes and improvements to parking lots. (Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430, ASTERISK-17452, ASTERISK-17452, ASTERISK-15792. Reported by: David Cabrejos, Remi Quezada, Philippe Lindheimer, David Woolley, Mat Murdock. Patched by: rmudgett) * Numerous issues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to. A change to Asterisk adds some checks to make sure tha...

Hi, I have two asterisk servers one is connected to the PSTN and the other one is connected to SIP users. The two servers connect with each other using IAX. When I have an incoming call from PSTN to the asterisk servers and have a forward to go back out to the PSTN the two IAX channel bridge together. Now every time I dial a DTMF digit, the asterisk is sending two DTMF digits. I enable

When dtmfmode is set to inband for SIP, and i originate a call from sip out to the PSTN, I can hear the DTMF digit twice in the audio stream. Once very briefly and once for normal duration. Our Theory: While Asterisk is parsing the DTMF, for a fraction of a second, while the end user generated DTMF is being detected, the DTMF is passed inband. Once the DTMF is detected Asterisk silences it

Hi, Could anyone tell me how to read the values in the "zonedata.c" file? I am looking at the "zt_tone_ringtone" field mainly. Thank you. Jad Wauthier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070512/4c0387be/attachment.htm

Hello, I am using smbspool and I seem to be getting an error that I can't find a solution to. I checked Google, Usenet and the samba archives. I seem to find people with the problem, but no answer. When I try to use the smbspool program, I get the following error: > root@HomeLinux:/home/steve# smbspool smb://192.168.1.4/LaserPrinter 0 0 0 0 0 README > ERROR: cli_session_request()