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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

does TDM2400 work on 3.3v pci slot? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070211/ac7d0c79/attachment.htm

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

...ists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 4 > Date: Wed, 1 Dec 2004 09:34:22 -0500 > From: mattf <mattf@vicimarketing.com> > Subject: RE: [Asterisk-Users] Avoided deadlock > To: 'Paradise Dove' <pardove@gmail.com>, 'Asterisk Users Mailing List > - Non-Commercial Discussion' <asterisk-users@lists.digium.com> > Message-ID: > <DB43F516702AAF4392AA45573F18181950250B@vicimail.vicimarketinggroup.com> > > Content-Type: text/plain; charset="iso-8859-1" >...

...sk-users > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050111/c996dd af/attachment-0001.htm ------------------------------ Message: 5 Date: Tue, 11 Jan 2005 16:38:25 +0330 From: Paradise Dove <pardove@gmail.com> Subject: Re: [Asterisk-Users] Analogue RAS Server To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <6f1184cc050111050847817dce@mail.gmail.com> Content-Type: text/plain; charset=US-ASCII > I don't think it...

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 all, is there any way to force all sip peers to re-authenticate themselves? thanks, Paradise Dove

Hi, i've just faced with some bridged calls which could not be hungup just killing the asterisk process solved the problem: Zap/63-1 (incoming s 1 ) Up Bridged Call SIP/2035-e9cb logs say: Jul 22 14:54:12 NOTICE[17161] chan_sip.c: Disconnecting call 'SIP/2035-e9cb' for lack of RTP activity in 6785 seconds Jul 22 14:54:13 NOTICE[17161] chan_sip.c: Disconnecting

hi all, when i start * all zap channels get ring signal so i get a huge number of incoming dummy calls when starting *. i'm using TE105P with 4 TA750 full fxo with latest CVS HEAD: zaptel.conf: span=1,0,0,esf,b8zs fxsks=1-24 span=2,0,0,esf,b8zs fxsks=25-48 span=3,0,0,esf,b8zs fxsks=49-72 span=4,0,0,esf,b8zs fxsks=73-96 loadzone=us defaultzone=us zapata.conf: [channels] context=incoming

hi, what does this mean?: www*CLI> show channels Channel (Context Extension Pri ) State Appl. Data 0 active channels 1 active call after some searchs got this: www*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Last Msg 172.22.22.27 2399 20830697669 00101/3343865 ulaw Rx: ACK 1 active SIP channel(s)

i've tried it on both snom190 and eyeBeam none of them work. nothing is changed in configs. is there any success in making snom LEDs work on CVS HEAD? thanks, paradise dove

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

hi, is there anyway to make * to detect callerid before first ring. i know that it seems silly; but here i have a case that Telco sends the caller-id before first ring. this issue is detected by installing a callerid detection device on the line. it shows callerid just before the first ring. so * can't detect the callerid. thanks, 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

i have a full fxo TDM24 and i have problem with installing it. when i run modprobe wctdm24xxp dmesg shows the following messages. and it waits for ever and nothing will happen. i'm sure that: - the power is plugged into tdm24 board - udev is configured and is working with other tdm cards. - zaptel.conf is configured. Zapata Telephony Interface Registered on major 196 Zaptel Version:

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 all, i have problem with dtmf detection on wctdm24xxp with full fxo and vpm module. after pushing dtmf tones on my phone for several times the card just detects one or two digits randomly.so now i can't use any voice menu on my box with this card. i have tried the following scenarios: - the card with / without vpm module has the same dtmf detection problem. - relaxdtmf=yes/no didn't

hi guys, i have a problem with 1.8 branch no matter which release of 1.8 i'm using. i can't make any sip calls, this is the error message i get on each call: [Jan 18 19:02:15] ERROR[1698] rtp_engine.c: No RTP engine was found. Do you have one loaded? [Jan 18 19:02:15] ERROR[1698] chan_sip.c: Got SDP but have no RTP session allocated. i'm sure that the rtp engine is loaded this is the