search for: _nxxxxxx

...from something else to ZAP, but I can't get it to "hunt" to the other context if the zapata channel (or group) is used first. Can anyone help? Here is my extensions.conf, and the error message I get. [trunklocal] ; ; Local area dialing accessed through trunk interface ; exten => _NXXXXXX,1,Dial(ZAP/26/952${EXTEN}) exten => _NXXXXXX,2,Congestion ; exten => _612NXXXXXX,1,Dial(ZAP/26/${EXTEN}) exten => _612NXXXXXX,2,Congestion ; exten => _651NXXXXXX,1,Dial(ZAP/26/${EXTEN}) exten => _651NXXXXXX,2,Congestion ; exten => _763NXXXXXX,1,Dial(ZAP/26/${EXTEN}) exten => _7...

...te=no context=from_test ;codecs disallow=all allow=ulaw My From Context has changed several times here is some of the iterations I've tried. inf=DAHDI/g2 bell=DAHDI/G3 [from_test] ; noted but not repaired. exten=> _NXXXXXX,1,Dial(${belltd}/409${EXTEN}) exten=> 9819213,1,Dial(${inf}/409${EXTEN} [from_test] ; noted but not repaired. exten=> _NXXXXXX,1,Set(CALLERID(name)=James Shigley) exten=> _NXXXXXX,2,Set(CALLERID(number)=4099819213) exten=> _NXXXXXX,3,Set(CALLERID(all)=${CALLERID(name)}<${CALL...

...XXX,1,Set(CALLERID(num)=XXXXXXXXXX) ;exten => _1NXXNXXXXXX,2,Set(CALLERID(name)=XXXXXXXXXX) exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _1NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@vitel-outbound) exten => _NXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _NXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _NXXXXXX,n,Dial(SIP/1904${EXTEN}@vitel-outbound) exten => _NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten =&...

In my context I have: exten => _NXXXXXX,1,Set(CHANNEL(musicclass)=default) exten => _NXXXXXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav) exten => _NXXXXXX,n,MixMonitor(${recordfilename},b) but the recorded conversation ended in 10min so it = 600sec I was looking in asterisk configu...

In my dial plan I have these two lines: exten => _NXXXXXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav) exten => _NXXXXXX,n,MixMonitor(${recordfilename},b) How to add "if" statement to execute these line only after let say 5pm. To record conversation only after 5pm. -- Joseph

Could someone please comment on the current state of chan_capi, chan_misdn and chan_modem channel drivers in terms of functionality and stability. Specifically, which channel driver would be best for a passive PCI HFC or W6692 ISDN card. The chan_misdn wiki claims that chan_capi distinguishes between junghanns and non-junghans cards, and that chan_misdn is better suited for general misdn

...o no avail, and I am assuming that it is some sort of timing problem. Are there any parameters I can tweak to try and rectify this? zapata.conf context=hardwire group=3 signalling=fxo_ks mailbox=8765 callerid="Acme" <8765> channel=32 extensions.conf [hardwire] ; exten => _NXXXXXX,1,SetCallerID(0141411${CALLERIDNUM}) exten => _NXXXXXX,2,CallingPres(3) exten => _NXXXXXX,3,Dial(Zap/g1/0141${EXTEN}) exten => _0.,1,SetCallerID(0141411${CALLERIDNUM}) exten => _0.,2,CallingPres(3) exten => _0.,3,Dial(Zap/g1/${EXTEN}) exten => t,1,Hangup ; If...

...,Set(CALLERID(num)=6162997590) exten => _1NXXNXXXXXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN}) exten => _1NXXNXXXXXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|500) exten => _1NXXNXXXXXX,500,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/${EXTEN}) Is there a way I can create a _NXXXXXX extension and insert 1 and areacode when dialing? Any help appreciated, Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061221/3a4d6628/attachment.htm

...his what you need? Basically, the local trunk and the Nufone trunk >fail over to each other. So, if you have a forward set up and transfer to >a non-local extension, the call will go out even if the original incoming >call was made on the PSTN line. > >[trunklocal] >exten => _NXXXXXX,1,Dial(${TRUNK}/${EXTEN}) >exten => _NXXXXXX,102,Dial(${NUFONE}/1${AREACODE}${EXTEN}) >exten => _NXXXXXX,203,Congestion() > >[iaxprovider] >exten => _1NXXNXXXXXX,1,Dial(${NUFONE}/${EXTEN}) >exten => _1NXXNXXXXXX,102,Dial(${TRUNK}) >exten => _1NXXNXXXXXX,203,Conge...

...ocal seven-digit dialing accessed through trunk interface ; exten => _9NXXXXXX,1,StripMSD,1 exten => _NXXXXXX,1,Dial,Phone/phone0/BYEXTENSION [local] ; ; Master context for local, toll-free, and iaxtel calls only...

...al. Despite the fact that outgoing works some of the time I'm wondering if I have the configuration wrong. Here is the outbound fragment from extensions.conf. [outbound] ; Press * to reach voice mail exten => *,1,VoiceMailMain(${PHONES1VM}) exten => *,2,Hangup ; local calls exten => _NXXXXXX,1,Playback(transfer) ; "Please hold while..." exten => _NXXXXXX,2,Dial(SIP/${EXTEN:0}@phone.gedanken.org:5061,20) exten => _NXXXXXX,3,Congestion ; long distance exten => _1NXXNXXXXXX,1,Playback(transfer) ; "Please hold while..." exten => _1NXXNXXXXXX,2,Dial(SIP/...

If you add "t" to you out-going trunk "Dial" lines: exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@vocal||t) exten => _NXXXXXX,2,Congestion so that you can still use park to park a call or transfer the phones, You have a problem of not being able to use "#" on external IVR systems. Is there any solution to this problem? Other then training hundreds of users no...

...02d1", "") in new stack -- end snip -- Here is how I have my extensions.conf file setup : ; Outbound to 1-nxx-nxx-xxxx goes via PSTN exten => _1NXXNXXXXXX,1,Dial(OH323/${EXTEN:1}@h323gk,20) exten => _1NXXNXXXXXX,2,Congestion ; Outbound to nxx-xxxx goes via PSTN exten => _NXXXXXX,1,Dial(OH323/${EXTEN:1}@h323gk,20) exten => _NXXXXXX,2,Congestion and in my oh323.conf file: gatekeeper=209.xxx.xxx.xx,HCCGK1 ; hid IP gatekeeper=DISCOVER .. .. .. context=heritage-h323 [h323gk] type=h323 host=209.xxx.xxx.xx ; hid IP ;secret=HCCGK1 [register] alias=HCCGW666 -- --...

...he native bridging seems to be the point of failure. The caller (another sip set) gets hung up on and the called pary hears dead air. A sip debug log of the scenario is available here: http://sprackett.com/asterisk.txt extensions.conf snippet: [trunks-local] ; local 7 digit dialing exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@mediatrix-1204) exten => _NXXXXXX,2,Congestion [internal] include => trunks-local sip.conf snippet: [mediatrix-1204] type=peer host=172.20.16.7 mask=255.255.255.255 dtmfmode=inband context=default canreinvite=yes qualify=yes Thanks in advance for any adive you can giv...

...waitingcallerid=no threewaycalling=yes transfer=yes cancallforward=yes callprogress=yes rxgain=0.0 txgain=0.0 group=1 immediate=no context=incoming signalling=fxs_ls channel=1 zaptel.conf fxsls=1 fxoks=2 fxoks=3 loadzone = us defaultzone=us [dialout] exten => _9NXXXXXX,1,StripMSD,1 exten => _NXXXXXX,2,Dial(Zap/1/w${EXTEN}) exten => _NXXXXXX,3,Hangup Thanks Jason Kawakami www.optellabs.com Salt Lake City, UT

I'm looking for a source for 50-pin amphenol cables, the ones used to connect Adtran's to punch down blocks. Preferably, one that's mail order and takes orders over the internet. Thanks.

...I thought I was home free and and everything was just going to work like magic. Wrong. Problem #1) With my previous FXO (HT-488) I was using the following in my dialplan: exten => _NXXXXX,1,Dial(SIP/@2003,60,D(w${EXTEN})) The above doesn't work for the Wellgate. So I tried: exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@2003 This doesn't work either. I am hoping to dial 7 digit numbers directly through the FXO. Problem #2) Something is amiss with the DTMF. I can call in to asterisk from my IAX phones and use Comedian mail fine (tones sent out of band). If I dial the extension of my F...

Hello all, Just starting to setup asterisk v 1.4.11 and need to run three distinct phone systems for three different companies. So far, I have inbound lines going to the appropriate dial plan within the extensions.conf file. I'm using exten => _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10}) to determine which number is being dialed by the caller and then using a gotoif to get to

Hi, I recently got a TDM04B and after installing and getting asterisk up and running I connected a handset to one of the ports. Unfortunately I don't get a dial tone when I lift the handset. What could be the cause of this? Could someone point me in the direction of a proper config for a TDM04B? Thanks.

...XXX,1,Set(CALLERID(num)=XXXXXXXXXX) ;exten => _1NXXNXXXXXX,2,Set(CALLERID(name)=XXXXXXXXXX) exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _1NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@vitel-outbound) exten => _NXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _NXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _NXXXXXX,n,Dial(SIP/1904${EXTEN}@vitel-outbound) exten => _NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten =&...