similar to: unable to disconnect a bridged channel

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)

----- 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 >

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

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

Thanks Paradise, this seems to have worked a treat!!! I commented out the: exten => 110,hint,SIP/110 lines which were in extensions_additional.conf for each sip extension I had. This seems to have stopped the crashes which were previously 3-5 times a day, now: System uptime: 1 day, 18 hours, 10 minutes, 3 seconds Interestingly it had the knock on effect of fixing another problem I had

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

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

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

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

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

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 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

Does any one knows of an Windows based SIP video phone???... Thanks... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, January 11, 2005 9:27 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 6, Issue 142 Send Asterisk-Users mailing

hi, it seems that spandsp-0.0.2pre22 is not functioning right. downgrading to pre21 makes it work again. debug messages: Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Changed from phase 1 to 4 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW ???: Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Real-time Internet fax (T.38) Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW V.8 capable Jan 18 11:54:29

Hi, I'm having a problem getting my Windows machines to access shares in Samba. When they browse to the Samba box it sometimes gives them an error saying that they don't have permission or that the server is unavailable. However this doesn't always happen and other times it lists the shares. When I try to access the shares it just prompts for the username/password over and over.

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

Sorry, meant this to go to the list. Timo Sirainen put forth on 3/27/2010 12:08 AM: > On 27.3.2010, at 6.49, Stan Hoeppner wrote: > >>> Tail just doesn't make much sense to me. Also it's not completely free >>> of pornographic associations either. :) >> >> No, no, totally free of pornographic connotation. Dovetail is a type of >> woodworking

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