search for: frompstn

...di-linux-2.1.0.4 linux-2.6.28-gentoo-r5 wanpipe-3.3.16 ###chan_dahdi.conf ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2009-02-21 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=fromPSTN usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=no ;callwaitingcallerid=yes callwaitingcallerid=no threewaycalling=no transfer=yes canpark=yes cancallforward=yes callreturn=no echocancel=no echocancelwhenbridged=no relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1...

...FWDNUMBER=223611 ; your calling number FWDCIDNAME="Robert Webb"; your caller id FWDPASSWORD=password ; your password FWDRINGS=IAX2/rwebb ; the phone to ring FWDVMBOX=2002 ; the VM box for this user ANALOGPHONE=zap/2 OFFICEPHONE=SIP/2002 LAPTOPPHONE=IAX2/rwebb VMBOX=2000 PSTNOUT=zap/5 [fromPSTN] exten => s,1,LookupBlacklist exten => s,2,DigitTimeout(3) exten => s,3,ResponseTimeout(5) exten => s,4,Wait(1) exten => s,5,Background(custom/pls-wait) exten => s,102,Goto(blacklisted,s,1) exten => t,1,Dial(${ANALOGPHONE}&${OFFICEPHONE}&${LAPTOPPHONE},15,m) exten =&g...

Hi How do i set up so that everyone can dial, for example *21* to forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions. I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113

I have managed to get spandsp working, and if I dial a specific extension I can receive faxes. WhooHoo. However, I was wanting to use the "fax detect" option in order to allow individuals to receive faxes, but can't get that to work. Given the following extensions (mainly copied from examples on the wiki), why is the call simply passed onto the sip device rather than being

...erisk is connected in the middle the management connection wont work any more - getting back isdn cause code 100. I have already tried dial options d und c (make it digital - clear channel) - no success. Here is the isdn debug [Nov 5 08:49:22] VERBOSE[1815] logger.c: -- Executing [99 at fromPSTN:4] Dial("Zap/19-1", "ZAP/g1/99") in new stack [Nov 5 08:49:22] VERBOSE[1815] logger.c: -- Making new call for cr 33240 [Nov 5 08:49:22] VERBOSE[1815] logger.c: -- Requested transfer capability: 0x08 - DIGITAL [Nov 5 08:49:22] VERBOSE[1815] logger.c: > Protocol Discrim...

...gt; 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party hears the please wait while I connect your call, but does not hear any ringing. I tried inserting exten => 1,1,Ringing but that does not work either. The same call flow from the pstn DOES generate ringback: [fromPSTN] exten => s,1,DigitTimeout(2) exten => s,2,ResponseTimeout(10) exten => s,3,Wait(1) exten => s,4,Background(custom/ivr-greeting) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},15,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup...

...Span 1" (MASTER) span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Global data loadzone = nl defaultzone = nl /etc/asterisk/chan_dahdi.conf [trunkgroups] [channels] ; Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) group=1 context=frompstn switchtype = euroisdn signalling = pri_cpe channel => 1-15,17-31 context = default cat /proc/dahdi/1 Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS/CRC4 RED 1 TE4/0/1/1 Clear (In use) RED(SWEC: MG2) 2 TE4/0/1/2 Clear (In use) RED(SWEC: MG2)...

...04 in state 4 Oct 20 17:18:26 VERBOSE[2663] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Oct 20 17:18:26 VERBOSE[2663] logger.c: -- Executing Hangup("CAPI/ISDN2/32xx11-8ab8", "") in new stack Oct 20 17:18:26 VERBOSE[2663] logger.c: == Spawn extension (frompstn, 32xx, 2) exited non-zero on 'CAPI/ISDN2/32xx11-8ab8' Oct 20 17:18:26 VERBOSE[2663] logger.c: == ISDN2#02: CAPI Hangingup for PLCI=0x202 in state 7 Oct 20 17:18:26 VERBOSE[19772] logger.c: > ISDN2#02: CAPI INFO 0x34e6: Recovery on timer expiry What does the timer expiry exa...

...iver w/PRI) I'm not sure why it says PRI though. Here's my configuration: zaptel.conf =========== span=1,1,2,ccs,ami bchan=1-2 dchan=3 defaultzone=se loadzone=se zapata.conf =========== signalling=bri_cpe echocancel=yes echocancelwhenbridged=yes group=1 callerid=asreceived context=frompstn prilocaldialplan=unknown pridialplan = unknown channel => 1-2 Thanks, Johan.

...de" > around and that seems to be what makes it see a digit vs see nothing. > When I call from an internal line (fromzap context) I'm able to dial > the extensions just fine. Can anyone tell me why this isn't working? > > TIA, > Ben > > extensions.conf: > [frompstn] > exten => s,1,Answer > exten => s,2,Wait(1) > exten => s,3,ResponseTimeout(2) > exten => s,4,DigitTimeout(6) > exten => s,5,Background(bgw-ThanksForCallingBgw) > exten => s,6,Background(bgw-IfYouKnowExt) > exten => s,7,Background(bgw-OtherwiseHold) > i...

Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me