similar to: Force SIP peers to Re-Autheticate

Hello everybody, I would like to implement "callback" function. When I call a person and his extension is busy I can press, for example, 5 and get a callback when his phone is not busy anymore. When I create a call file and copy it to spool call folder asterisk makes a call. One problem is that when extension is still busy my phone rings and I get busy tone of the person who I am

does * support call return? i want when the operator transfers a call if the transferee is busy or doesn't answer the call the call return back to operator again... this feature may be called: call return on busy call return on no answer Paradise Dove

Dec 1 12:08:43 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! what does this

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i get this error on dmesg: zaptel: Unknown symbol __stack_smash_handler zaptel: Unknown symbol __guard paradise dove

hi, i've just upgraded to latest trunk. everything compiles fine but when starting this message appears and fails to start. WARNING[3990] loader.c: module chan_zap.so error /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call thanks, paradise dove

can anybody help me to choose the most reliable fax solution for * . after googling the net i found that there are at least two solutions for this, app_rxfax+spandsp and iaxmodem+hylafax. - what's the differences between these two? - which one's better? why? thanks

Hi all, I'm trying to sort out my dial plan. What I'm wanting is something like the following - a bit simplified but hopefully you'll get the idea. 1) match internal extensions: dial them 2) anything else: send out zap 1 is easy :) it's 2 that's giving me problems. I had hoped that the 'i' extension would act as a catchall extension but it seems to only do that from

I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? How would one extend this calculation to varying bandwidth circuits and various VOIP protocols (MGCP,

Some of you might already know that we are releasing a new phone, snom 360. To make the phone well-known and stable, we have made a soft phone version out of it and offer it for trial or private use for free (for more details, see the license conditions). There are only few limitations to the phone. First of all, the audio subsystem will work only work with an acceptable quality if you are using

Hi, I've created a test at "extensions.conf" like this with 3 steps: ; When dial 5555, get the first available channel and dial do 482343400 exten => 5555,1,Dial(Zap/g1/482343400,5,rt) ; When dial 5555, get the channel 20 and dial do 482343400 exten => 5555,2,Dial(Zap/20/482343400) ; Go to Voicemail 1234 exten => 5555,3,Voicemail(u1234) I've tried using just the

I get seg. fault with my * box. at the crash time i had about 35 Bridged Channel. i have: - dual xeon box (3.2Ghz) - 2Gb of memory - E7501 chipset motherboard. - U320 scsi disks - intel Gb ethernet device. - i only use sip for clients (no fxs in box) - TE405P for fxo (with 4 atran TA750). - ulaw is used as codec and echo cancellationo is enabled. but the core dump file has nothing to show with

hi, just got an strange crash, and don't know what could cause this type of crashs - hardware failure - memory - cpu ....? i have 1xTE405P installed with 4xTA750. using fresh kernel 2.6.9 (no patch). * version is latest CVS HEAD. thanks Program terminated with signal 11, Segmentation fault. Cannot access memory at address 0xb80014bc #0 0xb7fbbce4 in ?? () (gdb) bt #0 0xb7fbbce4 in ?? ()

Hi guys, I 've tried asterisk 1.2.2. It work flawlessly for about 3 days then at the third days I activated setting jitterbuffer=yes and suddenly there is no voice when the call is picked up. It's really weird as if asterisk stops sending rtp packet. I've checked asterisk log and found nothing suspicious. Just weird :S. I tried it in 3 asterisk server and all of them are having

hi, is there anyway to Answer() the caller channel after the called number pickedup the phone. when an outside caller calls * system just continue ringing and not pick up the line and just dial an extension and then answer the caller channel after the called extension picked up the phone. is this possible in *? something like this: [incoming] exten => s,1,NoOp() exten => s,n,Dial(SIP/120)

Hi, I'm running FC3 with kernel 2.6.11. All the binary files and zaptel kernel modules is not available to system at boot time. They are extracted in a ram disk at system startup and then zaptel modules are loaded manually and so on. I have no problem with this boot routine and i've been tested all digium cards (expect tdm24) and they work fine till now. The problem appeared when i

Hi, I'm having a problem accessing my samba server from Windows. The problem occurs every time I try to browse to it on either XP or 2003 Server. The error message I get is: "\\Zeus is not accessible. You might not have permission to use this network resource. Contact the administrator of this server to find out if you have access permissions. The account is not authorized to log in

Hi I am trying to develop a night divert. Caller dials in after hours on Zap and it gets divert to a mobile number via a second Zap. The call bridges but will not hangup the channels when the parties finish. Is there something I am missing or an dial option that I should be using. I am using latest CVS. [night] exten => s,1,Answer exten => s,2,Wait,1 exten =>

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, December 01, 2004 7:07 AM Subject: Asterisk-Users Digest, Vol 5, Issue 6 > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Has anyone started work on a ''top''-like tool that would show the current CPU and memory usage of domains? I''m thinking of writing something really simple in python, starting from listdoms.py, but want to make sure that someone else isn''t already there. Steve -- Stephen G. Traugott (KG6HDQ) UNIX/Linux Infrastructure Architect, TerraLuna LLC