search for: _1xxx

...sername=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend context=phones host=dynamic [root at localhost asterisk]# cat extensions.conf [globals] [general] autofallthrough=yes [default] [incoming_calls] [phones] exten => _1XXX,1,NoOp() exten => _1XXX,n,Dial(SIP/${EXTEN},30) exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail) exten => _1XXX,n,Hangup() PS: My sip server and softphones are in the same network subnet. There are not any firewall or iptables rules. I tried the "nat=yes&q...

We are using the following to record conversations. exten => _1XXX.,1,SetVar(CALLFILENAME=call_to_${EXTEN:1}_dated_${TIMESTAMP}) exten => _1XXX.,2,Monitor(wav,${CALLFILENAME},m) exten => _1XXX.,3,Dial(IAX2/4506:zj5S3A5a@nl.voipgate.nl/${EXTEN:1}) exten => _1XXX.,4,Congestion exten => _1XXX.,104,Congestion This was working previously to record both sid...

...part of the dialplan is exten => _2XXX,1,Verbose(1|Extension 2xxx) exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN}) exten => _2XXX,n,Hangup() exten => _3XXX,1,Verbose(1|Extension 3xxx) exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN}) exten => _3xxx,n,Hangup() For B: exten => _1XXX,1,NoOp() exten => _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN}) exten => _1XXX,n,Hangup() exten => _3xxx,1,NoOp() exten => _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN}) exten => _3xxx,n,Hangup() For C: exten => _2XXX,1,Verbose(1|Extension 2xxx) exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTE...

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Mateo, Dialing the extension to your softphone is the same as any hardware extension. Exten => 1000,1,Dial,(SIP/1000,20,trf) pretty exten => 1000,2,Macro(vmessage,1000) exten => 1000,3,Hangup Change [mateo01] to [1000] in your sip and you will be saying that ext. 1000 is registered with the specifics you are using. Update the settings in your softphone to register the name and

Anyone know if - it is possible to limit 1 agent per extension where the last agent to log in overrides any previous agents or - a Command/application to clear all agents logged in on extension Does this look like it would require a custom mod to do it? J __________________________________ Discover Yahoo! Get on-the-go sports scores, stock quotes, news and more. Check it out!

...s like that in the server 1 with card sip.conf [asterisk1] type=freind host=ipadresseserver2 context=internal insecure=invite allow=all [2000] type=friend host=dynamic context=internal allow=all extensions.conf [internal] exten => 2000,1,Dial(SIP/2000) exten => 2000,n,Hangup() exten => _1XXX,1,Dial(SIP/${EXTEN}@asterisk1) exten => _1XXX,n,Hangup() in the unit 2 without card sip.conf [asterisk2] type=freind host=ipadresseserver1 context=internal insecure=invite allow=all [1000] type=friend host=dynamic context=internal allow=all extensions.conf [internal] exten => 1000,1,Dial(S...

Hi all, Right now our queue system is working quite well. One thing that I would like to have the option of doing would be to play a "ring" for the customer when they are connected to an agent. Right now they go from music on hold to an agent with no indication that their call is "going through." Our agents are played around a five second message when they pick up a call,

Hello all! I am currently making a script which is supposed to set up a call on request from a user, say, through a web page, for support issues etc. I am new into both asterisk and php, but I am working my way through the path as good as I can. Basically, what I would want to do, is to give the user the possibility to initiate a call by clicking a button. I?ve seen a cgi-alternative for this,

Hi People, I work in a company that are using asterisk as pbx. I need a way to identify what client my employees are calling. For example: - For each call that an employee of my company make to a customer, must identify the client name in the CDR table. - Is there a way of my employee enter a code to identify the client and then enter a phone number to make the call? I would like to identify

Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the

Hi All, I just noticed something interesting. When a sip device registers and regcontext is setup in sip.conf, a NoOp priority 1 extension is dynamically created in the dialplan within the specified regcontext. Works great. If for some reason, modification is made to the extension.conf and a >reload extension is performed, those dynamically created extensions in the regcontext vanish. Now

...ten => 30182849,103,Voicemail,b550 [te405p-in] exten => _2XX,1,Dial(Zap/g4/${EXTEN},60,r) exten => _2XX,2,Hangup exten => _73816592XX,1,Dial(Zap/g4/${EXTEN:-3},60,r) exten => _73816592XX,2,Hangup exten => _7XX,1,Dial(Zap/g4/${EXTEN},60,r) exten => _7XX,2,Hangup exten => _1XXX,1,Dial(Zap/g4/${EXTEN},60,r) exten => _1XXX,2,Hangup include => sip include => parkedcalls include => te405p-outgoing include => transfer-record [te405p-ext] exten => s,1,SetMusicOnHold(random) exten => s,2,Dial(SIP/bt-pavilion,45,t) exten => s,4,VoiceMail,u500 exten =&gt...

Hi all, I'd like to setup a really lean Asterisk installation that essentially has a full ISDN PRI (AT&T, T1, 23 B-chans, 1 D-chan, BZ8S, 5ESS, National dialplan) on a Digium TE207P adapter that all it does is convert the ISDN channels to SIP/IAX channels. Then I would add this Asterisk 'gateway' as a provider on one (or many) Asterisk systems on the back. With such a config I

...=g729 qualify=yes requirecalltoken=no host=10.X.X.141 language=es callerid=asreceived Server 2 (Panama) [mexico] type=peer context=oficina trunk=yes disallow=all allow=g729 qualify=yes requirecalltoken=no host=10.Y.Y.5 language=es callerid=asreceived So from Panama to Mexico we use: exten => _1XXX,1,Dial(IAX2/mexico/${EXTEN}) Call comes in and is answered but there is no CID in CDR or in the phone display.  Other trunks to other servers have no problem sending CID from one server to the other (all using IAX).  Any pointers? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos C...

...isallow=all allow=alaw [1002] ; copy of 1001 type=friend host=dynamic canreinvite=no disallow=all allow=alaw So far, so good. But if I would like to deploy e.g. 100 sip phones, I would have to add 100 sections? What I would like to do, is group 'm like: extensions.conf: [default] exten => _1XXX,1,Dial(SIP/norm,20) and in the sip.conf: [norm] type=friend host=dynamic canreinvite=no disallow=all allow=alaw But this doesn't seem to work. Any suggestions? Thanks Martin.

Hi, i'm currently using Originate command on AMI, i can call a certain channel like a SIP user SIP/1000 then once 1000 is answered it dials out to amobile or landline. Would just like to know if i can use AMI to dialout to a mobile or landline first (instead of SIP user) and once answered, dial another mobile or landline again. If not is it possible to call a macro from the AMI? i think

...dial-peer voice 8 voip destination-pattern 2... session target ipv4:<asterisk ip> codec g711alaw no vad h323.conf [general] port = 1720 bindaddr = 0.0.0.0 ;tos=lowdelay ; disallow=all allow=alaw allow=ulaw allow=gsm context=from-internal extension.conf [from-internal] exten => _1XXX,1,Dial(SIP/${EXTEN}@<cme ip>) exten => 2000,1,Dial(SIP/2000) I'm able from Asterisk to call ip phone connected to cme but from cme to asterisk the phones ring but go in hangup immediatly. My debug: --- localhosAnswering call ip$192.168.99.2:53716/21 localhos-- Transmitting RFC2833...

Hi guys I'm using asterisk 1.4 and going on to Snom phones. I'm trying to add a sip header to make the Snom phone use a different ring tone on one particular incoming number. I have added the following to the dial plan of the incoming context +------+------------------+-------+----------+--------------+-------------------------+ | id | context | exten | priority | app

...k(invalid) exten => i,2,Hangup exten => t,1,Hangup [te405p-frombp250] include => to-sip include => te405p-outtelstra [te405p-tobp250] exten => _2XX,1,Dial(Zap/g4/${EXTEN},60,r) exten => _4XX,1,Dial(Zap/g4/${EXTEN},60,r) exten => _7XX,1,Dial(Zap/g4/${EXTEN},60,r) exten => _1XXX,1,Dial(Zap/g4/${EXTEN},60,r) exten => _73816592XX,1,Dial(Zap/g4/${EXTEN:7},60,r) [te405p-intelstra] exten => s,1,SetMusicOnHold(record) exten => s,2,Dial(SIP/bt-pavilion,45,t) exten => s,4,VoiceMail,u500 exten => s,5,Hangup exten => 38166400,1,SetMusicOnHold(random) exten =>...