search for: _0xxxxxxxxx

Hi, I have small confusion in my configuration which is I had some DID's like 044578900-04457999. I was configured dial plan below mention. exten => _0XXXXXXXXX,1,NoOp(Int exten:${CALLERID(num)}) exten => _0XXXXXXXXX,2,Set(outgoing_ident=0445789${CALLERID(num):-2}) exten => _0XXXXXXXXX,3,NoOp(Ext ident:${outgoing_ident}) exten => _0XXXXXXXXX,4,Set(CALLERID(name)=${outgoing_ident}) exten => _0XXXXXXXXX,5,AGI(agi://127.0.0.1:4577/call_log) exten...

Hi List, I am making outgoing call from asterisk to GSM network with the help of VoIP trunk(SIP trunk) then I am not geting any caller ID at destination end. Is this the asterisk issue or VoIP trunk issue? Is this is due to asterisk then how we solve it? I already user Set(CALLERID(num)=XXXXXXXXXXX) in dialplan. ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer

...nikotel] secret=my_nikotel_pass username=mvanbaak fromuser=mvanbaak type=peer context=from-nikotel host=calamar0.nikotel.com canreinvite=no nat=yes ...some more entries for sip phones/softphones follow, they all work... the dial statement in my extensions.conf [outgoing-budgetphone] exten => _0XXXXXXXXX,1,SetAccount(outgoing-budgetphone) exten => _0XXXXXXXXX,2,SetCallerID(31557110304) exten => _0XXXXXXXXX,3,Dial(SIP/31557110304/${EXTEN}) exten => _0XXXXXXXXX,4,Congestion exten => _0XXXXXXXXX,104,Busy And this is wat I get on the CLI when I call my cellphone: -- Executing SetAccou...

Hello, I have a problem with Asterisk, on 10% of incoming calls the IP Phone ring but I don't hear the caller and the caller doesn't hear me (all IP Phones have the same problem). This problem appear also if the call is directly send to the second E1 of the digium card who is connected to an IVR. It does not depand on the charge of the server (I have the problem with only one call).

...pan 4: B4/0/4 "B4XXP (PCI) Card 0 Span 4" (MASTER) AMI/CCS RED span=4,4,0,ccs,ami # termtype: te bchan=10-11 dchan=12 echocanceller=mg2,10-11 # Global data loadzone = us defaultzone = us [root at go asterisk]# extensions.conf TRUNK=Dahdi/g0 [default] exten => _0XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _0XXXXXXXXX,2,MixMonitor(/var/spool/asterisk/astrec/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0)) exten => _0XXXXXXXXX,3,Dial(${TRUNK}/${EXTEN},,tTo) exten => _0XXXXXXXXX,4,Hangup exten => _XXXXXX...

...capi language=en type=autodetect stripmsd=0 dialtype=tone mode=answer group=1 ; group=1,2,3,9-12 msn=xxxxxxxxx incomingmsn=* device => /dev/ttyI0 device => /dev/ttyI1 device => /dev/ttyI2 device => /dev/ttyI3 The lines in extensions.conf that handle outgoing calls exten => _0XXXXXXXXX,1,Dial(${TRUNK}:${EXTEN},50,Ttr) exten => _0XXXXXXXXX,2,congestion exten => _00X.,1,Dial(${TRUNK}:${EXTEN},50,Ttr) exten => _00X.,2,congestion exten => _4XXXXX,1,Dial(${TRUNK}:0342${EXTEN},50,Ttr) exten => 112,1,Dial(${TRUNK}:112,50,Ttr) exten => _0[89]00X.,1,Dial(${TRUNK}:${EXTEN...

...ng the information I need to understand the issue and create a work around. Your assistance would be greatly appreciated. Asterisk v1.2.9.1 Linux v2.6.15-1.1830_FC4smp (FC4) SNOM v6.2.1 <---SNIP---> [from-sip] exten => _[2345]XX, 1,Macro(dial_localextension, ${EXTEN}) exten => _0XXXXXXXXX/_[2345]XX,1,Macro(outbound-calling,${EXTEN}) <----------> Successful call diversion on the SNOM phone to a local extension. ----------------------------------------------------------------- 2006-06-26 13:34:22 VERBOSE[18902] logger.c: -- Executing NoOp ("SIP/225-cdf5", "...

[outgoing-fax] exten => _0XXXXXXXXX,1,SetVar(NumberCalled=${EXTEN}) exten => _0XXXXXXXXX,2,Wait(10) exten => fax,1,SetCallerid(${FAX_CALLERID}) exten => fax,2,Dial(Zap/g1/${NumberCalled},60) exten => fax,3,Hangup exten => t,1,Busy exten => i,1,Busy -----Oorspronkelijk bericht----- Van: asterisk-users-bounces@lists....

...hannels] busydetect=yes callprogress=yes language=fr group=1 context=sortant signalling=fxo_ks channel => 1,2 group=2 context=entrant signalling=fxs_ks channel => 3,4 -------------------------------- /etc/asterisk/extensions.conf : ----------------------------------- [sortant] exten => _0XXXXXXXXX,1,Dial(Zap/g2/${EXTEN}) [entrant] exten => s,1,Answer exten => s,2,Wait(1) exten => s,3,DigitTimeout,2 exten => s,4,ResponseTimeout,3 ; menu d'accueil et de premier choix de langue exten => s,5,Background(fr_bienvenue) exten => s,6,Background(fr_continuer) exten => s,7,Bac...

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Any suggestions? Here is the console log: