[asterisk-users] why not forwarding to this number?
exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,SayAlpha(495256) exten = s,n,Wait(2) exten = s,n,Dial(SIP/222) exten = s,n,Hangup -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why not forwarding to this number?
hi resolved. added: include=outgoing cheers On Sat, Aug 9, 2014 at 7:34 AM, Administrator TOOTAI ad...@tootai.net wrote: Le 09/08/2014 12:23, Thomas Perron a écrit : exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,SayAlpha(495256) exten = s,n,Wait(2) exten = s,n,Dial(SIP/222) exten = s,n,Hangup Hi, you could at least tell us why you think the number is not forwarded (error ?) and what you have in logs I would at first check if the peer is registered assuming you have a congestion or unavailable error. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Basic Dial Plan Why is this plan not engaging the line exten = 105,n,Dial(SIP/voipvoip.com/1703501) and dialing the 703 number? The logs and debug dont show any problems [incoming] exten = 44,1,Answer() exten = 44,n,Wait(1) exten = 44,n,Playback(beep) exten = 44,n,Goto(105,105,1) ; ; [105] exten = 105,1,Wait(2) exten = 105,n,Playback(hello-world) exten = 105,n,Dial(SIP/voipvoip.com/1703501) exten = 105,n,Hangup() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect to an outbound channel and dial a phone number??
This seems basic but something is missing. I dial from my cell phone to my DID and enter the context in extensions.conf I am hoping to cascade through the plan and successfully automatically dial the 1444 number listed. But it fails. And, I dpon't know why? Should I removed the Hangup application? Syntax issue somewhere? I have a good SIP registration with the vendor, voipvoip. Thanks in advance for any feedback... [incoming] exten = 5552530146,1,Answer() exten = 5552530146,n,Wait(1) exten = 5552530146,n,Playback(beep) exten = 5552530146,n,Goto(105,105,1) ; ; [105] exten = 105,1,Wait(2) exten = 105,n,Playback(hello-world) exten = 105,n,Dial(SIP/voipvoip/1444514) exten = 105,n,Hangup() console output ... -- Executing [5552530146@incoming:1] Answer(SIP/voipvoip.com-000f, ) in new stack -- Executing [5552530146@incoming:2] Wait(SIP/voipvoip.com-000f, 1) in new stack -- Executing [5552530146@incoming:3] Playback(SIP/voipvoip.com-000f, beep) in new stack -- SIP/voipvoip.com-000f Playing 'beep.alaw' (language 'en') -- Executing [5552530146@incoming:4] Goto(SIP/voipvoip.com-000f, 105,105,1) in new stack -- Goto (105,105,1) -- Executing [105@105:1] Wait(SIP/voipvoip.com-000f, 2) in new stack -- Executing [105@105:2] Playback(SIP/voipvoip.com-000f, hello-world) in new stack -- SIP/voipvoip.com-000f Playing 'hello-world.alaw' (language 'en') -- Executing [105@105:3] Dial(SIP/voipvoip.com-000f, SIP/ sip3.voipvoip.com/17037171624) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/sip3.voipvoip.com/1444514 [Apr 9 16:07:11] WARNING[994]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission 4dd167154ea52bd26d63a95a56aa9526@192.168.1.10:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Apr 9 16:07:11] WARNING[994]: chan_sip.c:4198 retrans_pkt: Hanging up call 4dd167154ea52bd26d63a95a56aa9526@192.168.1.10:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). -- SIP/sip3.voipvoip.com-0010 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [105@105:4] Hangup(SIP/voipvoip.com-000f, ) in new stack == Spawn extension (105, 105, 4) exited non-zero on 'SIP/voipvoip.com-000f' Asterisk*CLI -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extensions.conf / test DID
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session registered:
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI sip show registry
Hostdnsmgr Username Refresh
StateReg.Time
sip3.voipvoip.com:5060 N 1112530146 105
Registered Mon, 08 Apr 2013 06:02:09
1 SIP registrations.
Asterisk*CLI
Here is the dial plan:
[incoming]
exten = 17036361355,1,Playback(beep)
exten = 17036361355,2,SayDigits(${EXTEN})
exten = 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle mobile phone.
;exten = s,1,Answer()
;exten = s,n,Dial(SIP/17037171234,150,r,t,)
[general]
register =1112530146:albany!@#1...@sip3.voipvoip.com/1112530146
registertimeout=20
context=incoming
allowoverlap=no
bindport=5060
bindaddr=192.168.1.10
srvlookup=no
;context=incoming
; The SIP provider
[voipvoip.com]
canreinvite=no
username=1112530146
fromuser=1112530146
secret=albany!@#123
context=incoming
type=friend
fromdomain=s...@voipvoip.com
host=69.90.209.57
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
nat=force_rport
insecure=port,invite
Thoughts please?I think something to do w/ incoming is incorrect.
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Re: [asterisk-users] sip registration
Got it... Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 954) Asterisk*CLI sip show registry Hostdnsmgr Username Refresh State Reg.Time sip3.voipvoip.com:5060 N 444222146 105 Registered Sun, 07 Apr 2013 09:42:31 1 SIP registrations. Asterisk*CLI Next hurdle is extensions.conf I must need to establish / correlate my DID number to something. When I dial my DID I get you have reached a non working number On Sat, Apr 6, 2013 at 5:36 PM, Steve Edwards asterisk@sedwards.comwrote: A better subject will yield better replies. On Sat, 6 Apr 2013, Thomas Perron wrote: Shouldnt I be able to at least ping the SIP provider IP? Not if they don't allow it. They don't. sip3.voipvoip.com registers fine for me with your credentials. Did you put the registration statement in the [general] section? I use the 'append' format so I can group all the cruft for a provider together. Like: ; voipvoip.com [general](+) register= nn:xx@sip3.** voipvoip.com/nnhttp://nn:xxx...@sip3.voipvoip.com/nn [outgoing] secret = xx username= nn ... I have not configured anything other then entries in the sip.conf I used your credentials and successfully placed a call to all of my Caribbean premium numbers*. Please change your password. Maybe your issue lies elsewhere. Does enabling SIP debugging on the console yield any clues? *) just kidding. -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip registration
I have a very lite layout and attempting to get the SIP configuration set up initially before proceeding into other areas. VMware is running my Asterisk 11 on Ubuntu 12. Shouldnt I be able to at least ping the SIP provider IP? I run command sip show registry and do not see it set up. I run sip show peers and I do see an entry. I have not configured anything other then entries in the sip.conf results are: Name/username HostDyn Forcerport ACL Port Status Description outgoing/5552530146 (your 69.90.209.57 5060 OK (85 ms) 1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline] Asterisk*CLI sip show registry Hostdnsmgr Username Refresh StateReg.Time 0 SIP registrations. Asterisk*CLI my config is this: [outgoing] username=5552530146 (your VoIP VoIP account assigned while signing up) type=peer qualify=yes secret=iblockedthis (your VoIP VoIP password) nat=auto insecure=invite,port host=sip3.voipvoip.com fromuser=5552530146 (your VoIP VoIP account assigned while signing up) fromdomain=sip3.voipvoip.com dtmfmode=rfc2833 disallow=all allow=g729 allow=ilbc allow=ulaw allow=alaw ; ; ; ; ; ;register = 5552530146:7036361399@69.90.209.57/5552530146 register=5552530146:boston!@#1...@sip3.voipvoip.com/5552530146 ; Please send input or guidance... Thanks Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rookie / sip and extensions
Sorry for blasting another desperate note but I am trying! I have changed the username and password and IP to protect my system. But, the logic is unchanged. It is does not work! I simply want to dial the telephone number provided to me for my DID which corresponds with my SIP info. And, then it should connect and hit the incoming context and simply dial the 617 number. I am close but no cigar. Now I get a fast busy tone only. What is missing or what is needed please? extensions.conf [globals] ; ; [incoming] ; ;exten= s,1,Goto(125010155_incoming) ; ;[125010155_incoming] exten = s,1,Answer exten = s,n,Dial(SIP/16175551212) sip.conf [general] ;register = 125010155:funnyti...@sip3.voipvoip.com/125010155 register = 125010...@sip3.voipvoip.com:funnytiger@69.90.209.11 ; [incoming] username=125010155 type=peer secret=funnytiger nat=auto insecure=invite,port host=69.90.209.11 fromdomain=69.90.209.11 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc srvlookup=yes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip and extensions
I am new. Here is the code that I am playing with on CentOS 6.x When I dial the number that corresponds w/ my SIP account I get a recording: reached a non-working number I built Asterisk a few times last year and am now back working on a similar project. In my view, there is something wrong in sip.conf I don't remember using a file that long to get a basic call set up. The format was provided to me by voipvoip.com (the SIP provider). Does anyone have any comments please? I just want a very simple config to get my machine to recognize a call to the SIP provider. Here is results of sip show registry: Hostdnsmgr Username Refresh StateReg.Time sip3.voipvoip.com:5060 N 5552530146 285 Registered Thu, 05 Jul 2012 21:39:56 1 SIP registrations. Here is sip and extensions.conf sip.conf [general] register = 5552530146:funnytiger...@sip3.voipvoip.com ; [sip3.voipvoip.com] [outgoing] username=5552530146 type=peer qualify=yes secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromuser=5552530146 fromdomain=69.90.209.57 dtmfmode=rfc2833 allow=g729 allow=ilbc allow=ulaw allow=alaw disallow=all srvlookup=no [incoming] username=5552530146 type=user secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromdomain=69.90.209.57 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc disallow=all srvlookup=no extensions.conf [general] ; ; [incoming] ;first creating extensions for your local users exten= s,1,Dial(SIP/1703717) exten= s,2,Hangup() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip and extensions
Hi, I changed these codes to not coincide with actual account info. Thanks On Thu, Jul 5, 2012 at 5:48 PM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - I am new. Here is the code that I am playing with on CentOS 6.x register = 5552530146:funnytiger...@sip3.voipvoip.com [outgoing] username=5552530146 type=peer qualify=yes secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromuser=5552530146 fromdomain=69.90.209.57 dtmfmode=rfc2833 allow=g729 allow=ilbc allow=ulaw allow=alaw disallow=all srvlookup=no [incoming] username=5552530146 type=user secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromdomain=69.90.209.57 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc disallow=all srvlookup=no *PLEASE* if that is your real username/password with your VoIP provider change it immediately. Just FYI, you've broadcast it to (tens or hundreds of) thousands of list readers. I have to believe some are of the nefarious type that would love to use your account for free calling at your expense. :/ --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] basic sip quesiton
What am I missing please? sip show registry shows that I am registered. [general] register = 5552530146:tam...@sip3.voipvoip.com ; ; [sip3.voipvoip.com] bindport=5060 ;you can use different port if the default is blocked bindaddr=0.0.0.0 ;binds to all ;this is for codec negotiation between the useragent and asterisk disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm context=incoming ;default context where incoming calls are passed. this should be the context where your local user.s extensions reside [outbound-trunk] ;this is the second section of you sip.conf file. Here you can create your trunk through which you will throw your outgoing calls to axvoice. host=sip3.voipvoip.com type=peer dtmfmode=rfc2833 canreinvite=yes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call files .vbs
This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses but I want to know in any case! Can a vb script run somehow on a Linux machine or does it only work on Windows? If I were to build a call file script (described in this link http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out ) then how does it work if my Asterisk machine is running on Centos 5.5? I simply want to execute a script that helps me automate the voice broadcasting/IVR of up to 1 phone numbers. Thank you Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files .vbs
Hi Doug, Yes. I have sorted that part out. Also, it seems like the pscp function is the way that I can tie together the vb script with the logic of the Asterisk call files learning curve!! Thanks On Sun, May 22, 2011 at 8:37 PM, Doug Lytle supp...@drdos.info wrote: Thomas Perron wrote: Can a vb script run somehow on a Linux machine or does it only work on Windows? Visual Basic is Windows specific. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DHCP / DNS
Are there any internal DHCP or DNS services built-in to the Asterisk code? -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IP Address Management / Open Source / IPAM
Does anyone have a recommendation for an Open Source IP Address Management solution please? There are several commercial players such as BlueCat, BT Diamond, InfoBlox, VitalQIP. And, Solarwinds makes a module that focuses on IPAM. Most vendors tie logic into DNS and DHCP into IPAM designs. In any case, does anyone have awareness of an Open Source solution? Thank you Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] console debugging
I used the command asterisk -vc to see console messages and it works fine. Now, I want to turn off this feature. How please? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] console debugging
OK. That is what I needed to know. Thannks On Sat, Jan 29, 2011 at 12:34 AM, Warren Selby wcse...@selbytech.com wrote: On Fri, Jan 28, 2011 at 8:59 PM, Thomas Perron thomas.per...@gmail.com wrote: I used the command asterisk -vc to see console messages and it works fine. Now, I want to turn off this feature. How please? Please elaborate what you would like to do? If you've started asterisk with the -c switch, you need to stop it using the command core stop now. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Basic Sip.conf and extensions.conf
Thanks. I fixed that. Still does not work. On Mon, Jan 17, 2011 at 12:53 AM, Jeroen Eeuwes jeroeneeu...@gmail.com wrote: Hi Thomas, register = 999:999...@sip.callwithus.comi Perhaps this should be .com instead of .comi ? Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Basic Sip.conf and extensions.conf
Does anyone see any issues here? I cannot get it to work. Passwords are not real! [general] ;register = 999:999...@carrier.callwithus.com register = 999:999...@sip.callwithus.comi context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes ; enable DNS SRV server [joesipshow] type=friend host=sip.callwithus.com authuser=999 username=999 secret=999222 qualify=no insecure=very context=default bindport=5060 fromdomain=sip.callwithus.com qualify=3600 nat=no ; or yes if you are behind NAT [default] exten = s,1,Answer() exten = s,n,Wait(1) exten = s,n,Dial(SIP/callwithus/1703111) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Basic Sip.conf and extensions.conf
OK. I set up the logger.conf via the steps provided. Now, how do I get the results. I reproduced the scenario. On Sun, Jan 16, 2011 at 4:02 PM, Paul Belanger pabelan...@digium.com wrote: On 11-01-16 03:58 PM, Thomas Perron wrote: Does anyone see any issues here? I cannot get it to work. Passwords are not real! No, however you did not provide any debug logs [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming
Is it possible to have Calls incoming to different DIDs? I want an AA that handles 100s of businesses. [Incoming-pizza] Exten = 4045551212,1,Goto(pizza,s,1) [Incoming-hvac] Exten = 8085551212,1,Goto(hvac,s,1) [Incoming-gutter] Exten = 6175551212,1,Goto(gutter,s,1) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming
Cool. So, one Asterisk machine handling up to 100 DID numbers, correct? Yes. I will have unique IVR flows/plans for each. I assume that the DID mumbers dialed would be the exaxt match needed to start the respective context. Correct? On 1/3/11, Rick Hall r...@readywire.com wrote: Yes, I don't see why not. You just need to setup an IVR for each business and then assign each individual DID to the appropriate IVR. This may help: http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu Cheers! Rick -- Rick Hall Senior Vice President ReadyWire Multimedia Solutions Affordable Website Reseller Hosting http://www.readywire.com/ (312) 278-4446 x5446 Technical Support: 24 hours a day / 7 days a week Customer Login...: https://secure.readywire.com/ Server Notices.: http://status.readywire.com/ Support Center: https://secure.readywire.com/ Twitter.: http://twitter.com/readywire Blog: http://blog.readywire.com/ This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. ReadyWire Multimedia Solutions. PO BOX 811061, Chicago, IL, USA, 60681. www.readywire.com. On Sun, Jan 2, 2011 at 11:50 AM, Thomas Perron thomas.per...@gmail.comwrote: Is it possible to have Calls incoming to different DIDs? I want an AA that handles 100s of businesses. [Incoming-pizza] Exten = 4045551212,1,Goto(pizza,s,1) [Incoming-hvac] Exten = 8085551212,1,Goto(hvac,s,1) [Incoming-gutter] Exten = 6175551212,1,Goto(gutter,s,1) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mail Integration
Does anyone have a super simple cookbook describing the steps to integrate Mail into an Asterisk Dial Plan. I have googled but have a lot of choppy results. I am running RH and Asterisk 1.8 Cheers Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mail Integration
Email integrated to voicemail. Thanks for the nice humor My bust. How do I set up an Exchange or other Mail MX server to interoperate with VoiceMail? On Mon, Dec 13, 2010 at 9:25 AM, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron Sent: Monday, December 13, 2010 5:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Mail Integration Does anyone have a super simple cookbook describing the steps to integrate Mail into an Asterisk Dial Plan. I have googled but have a lot of choppy results. I am running RH and Asterisk 1.8 Cheers Tom Mail should have a better definition. Do you mean voicemail, e-mail or (god forbid) postal mail or do you mean some mail product like outlook? Anyway, this might be a good starting point http://www.calliflower.com/2007/02/09/3121/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] debug audio or channel
Does anyone have any short answers on how I can fix this problem: A calls B. B rings Says connected. But the call is not bridged and therefor no audio passes. very simple dial plan. Frustrated. v 1.8 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi issue with Asterisk 1.8.0
Do you have any issues with getting audio to bridge? I am using 1.8 also. On Tue, Dec 7, 2010 at 12:38 PM, Timothy Legge timle...@gmail.com wrote: Hi I was using the delivered Ubuntu 1.6.x packages but I wanted to look at gtalk integration so I downloaded, compiled and installed the source (after removing the Ubuntu packages) have installed the following: asterisk-1.8.0 dahdi-linux-complete-2.4.0+2.4.0 libpri-1.4.11.5 I copied my config back into place and most seems to work, but I cannot get my phone that is plugged into the Wildcard TDM400P REV E/F card that I have to work. Basically, I don't hear the dial tone and Asterisk does not register off hook events. I have spent time reviewing my config but I don't see what the issue is. Is there anything I am missing, or can you suggest some additional things to look at? Tim chan_dahdi.conf grep -v ^; /etc/asterisk/chan_dahdi.conf | grep -v ^$ [trunkgroups] [channels] language=en context=phones signalling=fxo_ks usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no group=1 callgroup=1 pickupgroup=1 dahdi-channels.conf: ; Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) ;;; line=1 WCTDM/4/0 FXOKS signalling=fxo_ks callerid=Channel 1 4001 mailbox=4001 group=5 context=phones channel = 1 callerid= mailbox= group= context=default ;;; line=2 WCTDM/4/1 FXSKS signalling=fxs_ks callerid=asreceived group=0 context=incoming-local channel = 2 callerid= group= context=default -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no audio on end-point when call is connected/bridged via PBX
I am trying to dial through my asterisk machine from phone A to phone B. My DID is registered properly with the SIP provider. When I dial from A to B it looks fine so far. A rings B and B can pick up and the call is bridged. However, I don't hear any audio so therefor it is not working. I am running Asterisk 1.8 on a cloud server. I have had the same configuration running on a physical machine with a similar configuration. Thoughts? I know I posted this yesterday but was hoping for some more creative comments! Zip*CLI sip show registry Hostdnsmgr Username Refresh StateReg.Time sip.callwithus.com:5060 N 105 Registered Tue, 07 Dec 2010 02:36:43 1 SIP registrations. my sip.conf [general] context=default allowoverlap=no ;bindport=5060 port=5060 bindaddr=0.0.0.0 canreinvite=no ;if your asterisk box is behind a NAT ro ;register = :3...@carrier.callwithus.com register = :3...@sip.callwithus.com [callwithus] type=friend host=sip.callwithus.com username= secret=31 qualify=no insecure=invite my extensions.conf [general] [globals] CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus [default] exten = s,1,Answer() exten = s,n,Dial(SIP/callwithus/122) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no audio
Any reason why I don't get audio on the channel after it rings and the end user picks up. Here are my files. CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus [default] include = stdexten exten = s,1,Answer() exten = s,n,Wait(1) exten = s,n,Dial(SIP/callwithus/44,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() my sip.conf file [general] context=default allowoverlap=no bindport=5060 port=5060 bindaddr=0.0.0.0 canreinvite=no ;if your asterisk box is behind a NAT ro ;register = xxx:y...@carrier.callwithus.com register = xxx:y...@sip.callwithus.com [callwithus] type=friend host=sip.callwithus.com username=xxx secret=yyy qualify=no insecure=invite -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no audio
Steve, thanks for your note negative. no joy. removed the line to make is very basic. see below. [globals] CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus ;[general] [default] include = stdexten exten = s,1,Answer() exten = s,n,Wait(1) exten = s,n,Dial(SIP/callwithus/44) exten = s,n,Wait(2) exten = s,n,Hangup() ~ On Sun, Dec 5, 2010 at 8:34 PM, Steve Edwards asterisk@sedwards.com wrote: On Sun, 5 Dec 2010, Thomas Perron wrote: Any reason why I don't get audio on the channel after it rings and the end user picks up. exten = s,n,Dial(SIP/callwithus/44,120,A,(demo-thanks)) Re-read 'core show application dial' Where is your prompt to option 'A' ? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial plan and sip
thank you i will try it. On Mon, Nov 15, 2010 at 4:52 PM, Chad Wallace cwall...@lodgingcompany.com wrote: On Sat, 13 Nov 2010 20:38:30 -0500 Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? [...] exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Dial(SIP/jazzey/1703111,120,A(demo-thanks)) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dial plan and sip
Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial plan and sip
Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] upgrade
i am running 1.4.37 and am hosted on Rackspace. I feel like a took a step back by using the Cloud server service since I am having a little trouble proving that my basic configuration is working. Nevertheless, I want to upgrade to 1.8. I use Centos 5.5 Anyone know of a good link that can help please? I searched Google and got confused by the options. Upgrade to 1.8. How please? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial plan and sip
How do I see the error message? the phone call seemed to get through but I did not see anything on my 1.4 console. i used 1.6.x before. having trouble with this for some reason. older stuff. i have one session open at the prompt but nothing shows up. On Sat, Nov 13, 2010 at 9:53 PM, Brett Woollum br...@woollum.com wrote: What is the error message? Sent from my iPhone On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote: Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial plan and sip
Jim, Thanks. But, no joy. I set to 3, then 5. I don't think I am getting registered somewhere. The console shows nothing. The call to the DID drops after 5 seconds or so. It does not ring. I know. Basic stuff. I really think the version of this code is not robust enough. My sip.conf and extensions.conf is very simple. On Sat, Nov 13, 2010 at 10:13 PM, Jim Dickenson dicken...@cfmc.com wrote: You get into asterisk by saying asterisk -r. You then up the verbosity by saying core set verbose 3 or some such number. You the call your number and you should see the steps of your dialplan execute. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 13, 2010, at 7:02 PM, Thomas Perron wrote: How do I see the error message? the phone call seemed to get through but I did not see anything on my 1.4 console. i used 1.6.x before. having trouble with this for some reason. older stuff. i have one session open at the prompt but nothing shows up. On Sat, Nov 13, 2010 at 9:53 PM, Brett Woollum br...@woollum.com wrote: What is the error message? Sent from my iPhone On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote: Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] install
I have installed Asterisk before w/ no issues but while trying today (1.6.2.13 and centors 5.4) I receive the following at the CLI: The configure script must be executed before running 'make'. Please run ./configure. Any tricks on getting through this? I did not select to libpri or zapata. only asterisk as i am building a voip only design on rackspace.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registering and initiating a SIP call without a SIP client
Yes. Send your code. Consider using call files. Here is a part of what works for me. [-system] exten = s,1,Answer exten = s,n,Wait(2) exten = s,n,Playback(pa-welcome) please record your broadcast after the beep ;exten = s,n,Playback(beep) exten = s,n,Wait(1) exten = s,n,Record(/var/lib/asterisk/sounds/en/record713.gsm) ;exten = s,n,Record(LINDA_RISTIG_linda005) ; record this: this welcome to dial a restaurant ??? ;exten = s,n,Wait(1) exten = s,n,Background(pa-confirm) ; press 1 to send or zero to hangup exten = s,n,WaitExten(10) ;exten = s,n,Hangup() exten = 1,1,System(cp /etc/asterisk/pizza/*.call /tmp/) exten = 1,n,System(mv /tmp/*.call /var/spool/asterisk/outgoing/) exten = 0,1, Hangup() ;; ;; [pizza] exten = 13,1,Answer() exten = 13,n,Wait(1) exten = 13,n,Playback(record713) ;exten = 13,n,Playback(LINDA_RISTIG_IVR) ;exten = 13,n,Playback(calleveryone) ;exten = 13,n,WaitExten(5) exten = 13,n,Goto(13,1) On Sun, Sep 5, 2010 at 6:20 PM, Gautam Desai gdesai...@yahoo.com wrote: Can I generate SIP registration and call from Asterisk without a SIP client? I need to initiate a call from asterisk and play a recorded message. Gautam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fast busy out?
why does this not work? i simply want to hear the recorded message exten = s,1,Answer() ;exten = s,n,Record(zipcodegutter1.gsm) ;zcg1 exten = s,n,Playback(zipcodegutter1) exten = s,n,Dial(SIP/c01s/159,120,A,(demo-thanks)) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fast busy out?
Thank for your the tip Ondrej. Here is what worked on my CentOS box.
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Record(zipcodegutter%d:gsm)
exten = s,n,Wait(2)
exten = s,n,Playback(${RECORDED_FILE})
exten = s,n,Wait(2)
exten = s,n,Hangup()
2010/9/4 Ondrej Škopek skopekond...@gmail.com:
no I am not sorry, and please reply to this list, and not to me directly..
On Sat, Sep 4, 2010 at 6:16 PM, Thomas Perron thomas.per...@gmail.com
wrote:
thank you for your note on the Asterisk users group list
Are you in Scandanavia somewhere?
Cheers
Tom
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[asterisk-users] append CID label
I want a call to connect via my DID to my dialplan. Then, I want to attach a label to the incoming call call arrives starts to dive through the dial plan then rings a trunk/channel via SIP (see below) Question: before answering my 1212111 endpoint I want to see a flag CID that correlates to the DID number that was called. And, then change it to something like the characters blue How??? please. exten = s,1,Answer() exten = s,n,Dial(SIP/callwithus/1212111,120,A,(test)) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] append CID label
ok
thank you
i will try
On Sat, Jun 26, 2010 at 10:31 PM, C F shma...@gmail.com wrote:
exten = s,n,Set(CALLERID(name)=label${CALLERID(name)})
put this before the dial command.
On Sat, Jun 26, 2010 at 10:09 PM, Thomas Perron thomas.per...@gmail.com
wrote:
I want a call to connect via my DID to my dialplan.
Then, I want to attach a label to the incoming call
call arrives
starts to dive through the dial plan
then rings a trunk/channel via SIP (see below)
Question: before answering my 1212111 endpoint I want to see a
flag CID that correlates to the DID number that was called. And, then
change it to something like the characters blue
How??? please.
exten = s,1,Answer()
exten = s,n,Dial(SIP/callwithus/1212111,120,A,(test))
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Re: [asterisk-users] MeetMe problem
try using confbridge in lastest asterisk version On Sat, Jun 12, 2010 at 11:30 AM, Daniel Knoll dan...@danielknoll.de wrote: Hi Guys, sometimes if one caller or many callers are in a meetme Room and a new one join the room, then he or another caller into the same room where kickt from the room. It's very strange for me and in logs (full) I can't see anything. is it possible to log more from meetme.c ? can anyone help me and maybe someone has also the problem as i and have an solution. I use: asterisk-1.6.2.7 dahdi-linux-complete-2.3.0+2.3.0 asterisk-addons-1.6.2.1 Thanx a lot for any answers that helps me. Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] text
Does anyone know how to send a text message from Asterisk? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] text
thanks do i need to have an smtp server somewhere. i tried directly from my dialplan but no joy! i know you know that i am not a star with this but any help would be cool here is my config: exten = 600,1,Answer() exten = 600,n,Wait(1) exten = 600,n,system(echo foo | mail -s bar 2224441...@txt.att.net) On Fri, May 7, 2010 at 8:32 PM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 7 May 2010, Thomas Perron wrote: Does anyone know how to send a text message from Asterisk? Carrier specific, but how about: system(echo foo | mail -s bar 551...@txt.att.net) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] meetme / upgrade to 1.6.2.6
I read that I need to run 1.6.2.6 (at least 1.6) to use the MeetMe application since I don't have a zdummy timing driver. Anyway, I want to upgrade my machine to 1.6.2.6. Does anyone have the exact steps? I see a lot of references on the web but any other links from our community may be preferred! Thank you Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] migration
My client wants to use my service that I will host. It is an IVR application. I have the solution worked out on the server side. I will port his current POTS line phone number to a DID service where I can control it via SIP. Question relates to his current phones. Forgive me as I am new. Does he need his current phones? How will they ring if I port the number? Should I simply have him remove the phones and I can send the calls to his cell phones? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play music to caller after answer, before dial
Does this help? The A near the end calls the audio file ginr3 exten = 551,1,Answer() exten = 551,n,Dial(SIP/callwithus/17025551212,120,A(ginr3)) On Mon, Mar 22, 2010 at 6:41 PM, Michelle Dupuis mdup...@ocg.ca wrote: I think I forgot some important information... I'm actually running an AGI script after the answer (and before the dial). I would like to play MOH while the AGI script is running, and then perform the dial (ending the MOH). This is where I'm stuck Thanks! Michelle From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, March 22, 2010 5:22 PM To: Asterisk Users List Subject: Re: [asterisk-users] Play music to caller after answer, before dial This might be your answer: Exten = s,1,answer Exten = s,n,wait(10,m) Exten = s,n,Dial… This would wait 10 seconds playing MOH before dialing. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Monday, March 22, 2010 3:58 PM To: 'Asterisk Users List' Subject: [asterisk-users] Play music to caller after answer, before dial I would like to play music to an inbound caller, AFTER asterisk answers the call, but before the call is bridged by DIAL. Is there a simple way to achieve this? MD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue MOH
I want callers to enter a queue and then hear music on hold. does anyone have notes on how to integrate queuing to a dial plan that uses moh? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DID forwarding ?
DID number A. I have a DID (a regular line from Verizon). number A. Can I have A ported to my SIP provider? Then, interface the A DID to my system so that I can build a solution. I want to write an IVR for the A number and allow callers dialing A to interact with my Asterisk machine. I need to keep number A. Kindly advise -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IVR Demo / Create file / Move file / Demo all
Do you see any syntax errors?
Positive comments welcomed.
The short version of the logic is as follows:
create a file based on the NUMBER
create a an audio file
move to a new extension (label) and play the results
exten = 621,1,Answer()
exten = 621,n,Read(NUMBER,enteryournumberstartingwithaone,12,,5)
; create a variable from a DTMF entry / 12 characters long
exten = 621,n,System{(/tmp touch($NUMBER)}
; create the file based on the variable entered
exten = 621,n,Set(audioscript=$[${NUMBER} + 1])
; set a channel variable in advance of recording to it
exten = 621,n,SayDigits(${NUMBER})
; say the NUMBER that was entered
exten = 621,n,System{/tmp touch($(audioscript)}
; create a file
exten = 621,n,Record(${audioscript})
; record a file based on the NUMBER + 1
exten = 621,n,Playback(audioscript)
; listen to the recording - it changes for each
Demo
exten = 621,n,System(mv audioscript /var/lib/asterisk/sounds/en)
; move the recording to the sounds directory
exten = 621,n,Goto(${audioscript},1)
; goto the label/alias to hear it all together
exten = audioscript,1,Answer()
; Nothing
exten = audioscript,n,Playback(audioscript)
; plays audioscript
exten = audioscript,n,Playback(staticIVR_sample)
; adds some boring IVR lingo
exten = audioscript,n,Hangup()
; drops the call
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Re: [asterisk-users] IVR Demo / Create file / Move file / Demo all
what is OP please? can you just simply comment on the technical work please? On Mon, Feb 8, 2010 at 12:24 PM, Danny Nicholas da...@debsinc.com wrote: Steve, if he had that kind of power, he wouldn't have made the OP. BTW, I doubt it took you 5 min to actually figure out that the syntax wasn't correct. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Monday, February 08, 2010 11:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR Demo / Create file / Move file / Demo all On Mon, 8 Feb 2010, Thomas Perron wrote: Do you see any syntax errors? Yes. Lots. Can I please have the last 5 minutes of my life back? Positive comments welcomed. Please don't bother the list to syntax check your code if you are too lazy to type it into your dialplan and see if Asterisk likes it. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR Demo / Create file / Move file / Demo all
this solution works. thanks for the helpful comments.
exten = 621,n,Read(NUMBER,snowday,12,,10)
; create a variable from a DTMF entry / 12 characters long
;exten = 621,n,System{(/tmp touch($NUMBER)}
; create the file based on the variable entered
exten = 621,n,Set(audioscript=$[${NUMBER} + 1])
; set a channel variable in advance of recording to it
exten = 621,n,SayDigits(${NUMBER})
; say the NUMBER that was entered
exten = 621,n,SendDTMF(${NUMBER})
;exten = 621,n,System{/tmp touch($(audioscript)}
; create a file
exten = 621,n,Record(${audioscript}.gsm)
; record a file based on the NUMBER + 1
exten = 621,n,Playback(${audioscript})
; listen to the recording , etc.
exten = 621,n,System(mv ${audioscript}.gsm
/var/lib/asterisk/sounds/en) ; move the recording to the
sounds directory
exten = 621,n,Playback(dir-welcome)
exten = 621,n,Playback(${audioscript})
exten = 621,n,Playback(snowday2)
exten = 621,n,Goto(s,1)
On Mon, Feb 8, 2010 at 2:00 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Mon, Feb 08, 2010 at 12:36:18PM -0500, Thomas Perron wrote:
what is OP please?
can you just simply comment on the technical work please?
Original Poster. The one who started the thread. In this case it's you.
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[asterisk-users] syntax
I am trying to understand .call files. The logs seems to indicate issues with the audio file that I am trying to have played when the call is connected. Any thoughts? Some sample files and logs to console are shown. zipp-code.call Channel: SIP/callwithus/12023519259 Application: Playback Data: zipp-code.gsm [r...@localhost tmp]# touch zipp-code.call [r...@localhost tmp]# vi zipp-code.call [r...@localhost tmp]# mv zipp-code.call /var/spool/asterisk/outgoing/ -- Attempting call on SIP/callwithus/12023519259 for application Playback(zipp-code.gsm) (Retry 1) == Using SIP RTP CoS mark 5 [Feb 7 18:44:07] WARNING[20197]: file.c:635 ast_openstream_full: File zipp-code.gsm does not exist in any format [Feb 7 18:44:07] WARNING[20197]: file.c:936 ast_streamfile: Unable to open zipp-code.gsm (format 0x2 (gsm)): No such file or directory [Feb 7 18:44:07] WARNING[20197]: app_playback.c:447 playback_exec: ast_streamfile failed on SIP/callwithus-03d98080 for zipp-code.gsm [Feb 7 18:44:07] NOTICE[20197]: pbx_spool.c:357 attempt_thread: Call completed to SIP/callwithus/12023519259 -- Attempting call on SIP/callwithus/12023519259 for application Playback(yvrspecialemail) (Retry 1) == Using SIP RTP CoS mark 5 [Feb 7 18:54:58] WARNING[20228]: file.c:635 ast_openstream_full: File yvrspecialemail does not exist in any format [Feb 7 18:54:58] WARNING[20228]: file.c:936 ast_streamfile: Unable to open yvrspecialemail (format 0x2 (gsm)): No such file or directory [Feb 7 18:54:58] WARNING[20228]: app_playback.c:447 playback_exec: ast_streamfile failed on SIP/callwithus-03d98080 for yvrspecialemail [Feb 7 18:54:58] NOTICE[20228]: pbx_spool.c:357 attempt_thread: Call completed to SIP/callwithus/12023519259 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] syntax
Hi Tommy Thank you works like magic. thank you. I love this list. when you get stumped you can always (almost!) count on some great input! regards, tom On Sun, Feb 7, 2010 at 7:32 PM, Tom Moore tommym2...@gmail.com wrote: Your sound file needs to be in the asterisk sounds directory. Another thing is that you may not have to put the file extension in the name if the file is in the proper place as well. Try that and see what happens. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron Sent: Sunday, February 07, 2010 7:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] syntax I am trying to understand .call files. The logs seems to indicate issues with the audio file that I am trying to have played when the call is connected. Any thoughts? Some sample files and logs to console are shown. zipp-code.call Channel: SIP/callwithus/12023519259 Application: Playback Data: zipp-code.gsm [r...@localhost tmp]# touch zipp-code.call [r...@localhost tmp]# vi zipp-code.call [r...@localhost tmp]# mv zipp-code.call /var/spool/asterisk/outgoing/ -- Attempting call on SIP/callwithus/12023519259 for application Playback(zipp-code.gsm) (Retry 1) == Using SIP RTP CoS mark 5 [Feb 7 18:44:07] WARNING[20197]: file.c:635 ast_openstream_full: File zipp-code.gsm does not exist in any format [Feb 7 18:44:07] WARNING[20197]: file.c:936 ast_streamfile: Unable to open zipp-code.gsm (format 0x2 (gsm)): No such file or directory [Feb 7 18:44:07] WARNING[20197]: app_playback.c:447 playback_exec: ast_streamfile failed on SIP/callwithus-03d98080 for zipp-code.gsm [Feb 7 18:44:07] NOTICE[20197]: pbx_spool.c:357 attempt_thread: Call completed to SIP/callwithus/12023519259 -- Attempting call on SIP/callwithus/12023519259 for application Playback(yvrspecialemail) (Retry 1) == Using SIP RTP CoS mark 5 [Feb 7 18:54:58] WARNING[20228]: file.c:635 ast_openstream_full: File yvrspecialemail does not exist in any format [Feb 7 18:54:58] WARNING[20228]: file.c:936 ast_streamfile: Unable to open yvrspecialemail (format 0x2 (gsm)): No such file or directory [Feb 7 18:54:58] WARNING[20228]: app_playback.c:447 playback_exec: ast_streamfile failed on SIP/callwithus-03d98080 for yvrspecialemail [Feb 7 18:54:58] NOTICE[20228]: pbx_spool.c:357 attempt_thread: Call completed to SIP/callwithus/12023519259 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
My inquiry is to understand how I could configure a system to do it. I have since learned that Asterisk has features in the code to do this (auto dial out, features.conf and .call files.) The 1 example is a bit extreme but it really does not matter what the number is for this. Dialogic has a system that provides notification so I am trying to see how I can build my own. Understanding simultaneous and concurrent call capabilities is important. Karl. Steve. Please don't bother me with you immature insults. On Sat, Feb 6, 2010 at 1:21 AM, Karl Fife karlf...@gmail.com wrote: Nice. :-) Didn't see that, I concede. - Original Message - From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 06, 2010 12:10 AM Subject: Re: [asterisk-users] Dial script On Fri, 5 Feb 2010, Karl Fife wrote: Try this: #rm -rf / Copycat! On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote: Is there any tested script available for this purpose. On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org=20 Sure. Add this to root's crontab: * * * * rm --farce --recursive / -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
Karl, You interpretation and assumption about my interest in a technical solution is simply wrong. I wanted positive feedback on the site. Your response was clearly negative and loaded with an insulting tone. I will continue to try and understand the internal protocols and technoligies that make this (and other) solutions work. My advise to you: Drop your ego at the front door. You are wrong. I am trying to learn. Your correct, Google has a lot of information. The purpose of the community is to support one another. Not to instigate like you have done. Later pal. Yes. I did give you an F.. You have no respect for others that are not as smart as you. On Sat, Feb 6, 2010 at 10:54 AM, Karl Fife karlf...@gmail.com wrote: If Mr. Perron's request were truly academic, he probably would not have sent me an email off-list telling me to go fu▎ myself. - Original Message - From: Thomas Perron thomas.per...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 06, 2010 4:56 AM Subject: Re: [asterisk-users] Dial script My inquiry is to understand how I could configure a system to do it. I have since learned that Asterisk has features in the code to do this (auto dial out, features.conf and .call files.) The 1 example is a bit extreme but it really does not matter what the number is for this. Dialogic has a system that provides notification so I am trying to see how I can build my own. Understanding simultaneous and concurrent call capabilities is important. Karl. Steve. Please don't bother me with you immature insults. On Sat, Feb 6, 2010 at 1:21 AM, Karl Fife karlf...@gmail.com wrote: Nice. :-) Didn't see that, I concede. - Original Message - From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 06, 2010 12:10 AM Subject: Re: [asterisk-users] Dial script On Fri, 5 Feb 2010, Karl Fife wrote: Try this: #rm -rf / Copycat! On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote: Is there any tested script available for this purpose. On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org=20 Sure. Add this to root's crontab: * * * * rm --farce --recursive / -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
Thank you for your interesting comments. On Sat, Feb 6, 2010 at 4:14 PM, Erik de Wild: Tripple-o i...@tripple-o.nl wrote: Thomas, Yes you can do this. I actually have done this and run it as a service under the name Meetmecall. I use MSN as the user interface to record the message, create phone lists of the numbers that has to be called and to actually schedule and perform the delivery. It is possible to use it for spam but the customers I have use it to notify, remember, offer or let the callee know about something relevant, but always as part of an already existing relation. With some extra parameters used, you can start a groupcall and use all the other Asterisk magic available like doing a questionarry using a smart IVR etc. etc. I can think about a long list of useful use of this service. I have no idea about the rules and legislation in other countries but in Holland you will end up with serious trouble and extreme high penalties to pay if you actually use it for spamming. I will not send you a copy of the solution but it is based on the use of call files pointing to local channels/extensions where the Asterisk magic is combined in a working (and I think clever) way. The CDR isn't perfect but disable and enable CDR at the proper points in the dial plan and clever use of the USERFIELD variable will result in useable data for billing the users. The CDR shows that most callees, listen to the message until it ends and yes, sometime there are complaints about the use but that is very rare. About the scheduling of the calls to make. It is not Asterisk that limits you. Far before reaching the limits of Asterisk it will be the bandwidth available and the SIP trunk provider that normally doesn't allow an endless number of concurrent calls. When I started this I was working for a Norwegian company offering the dial tone on the internet and I had a server almost directly connected to the backbone of internet with more or less endless bandwidth. I did some stress testing of a call center solution and 80 concurrent calls wasn't a problem and my guess is that you can far beyond 80 calls. It is wise to move the call files one after the other or one batch after the other. Moving large numbers of call files into /var/spool/asterisk/ outgoing might sometimes result in unexpected and not intended results. There are other scenarios but this was my choice. 10.000 calls will take some time but with a 30 seconds message, 20 concurrent calls and 10 seconds average to dial after around 5,5 hours the last phone call will be dialed. If the message is just 15 seconds it will take around 3,5 hours. If you want to deliver in short time, like 10 minutes, you really have to scale up to 420 concurrent calls which doesn't sound doable unless you have real serious budgets. If you put everything in place at your side you will probably run into constraints of the SIP provider and the interconnection to the pstn. btw: Asterisk has the potential to build lots of evil features and lots of standard features can be used in an evil way. Personally I think it is kind of strange that, if a question is asked, one has to explain why the answer is not mend for evil use. We don't have to help someone out and we can refuse because of lots of reasons: no time, not an interesting question, not a single sign of any effort by the one asking the question, not willing to give something away that costs lots of time and energy, the feeling that it will be used in an evil way etc. etc. I think the tone and the content of this discussion harms the Asterisk community as a whole. with friendly regards, Erik de Wild Tripple-o: your asterisk migration partner the Netherlands On 6 feb 2010, at 03:54, Thomas Perron wrote: Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial script
Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
karl, does it make you feel good ? wow. pathetic. On Fri, Feb 5, 2010 at 11:00 PM, Karl Fife karlf...@gmail.com wrote: Try this: #rm -rf / - Original Message - From: Thomas Perron thomas.per...@gmail.com To: asterisk-users@lists.digium.com Sent: Friday, February 05, 2010 8:54 PM Subject: [asterisk-users] Dial script Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MATH
I want to allow users to dial my DID
Then, hear my ginger3 intro
Then, depending on the number that they press, provide a total via MATH.
Comments. Will this work?
exten = 866,1,Goto(tommath,s,1)
[tommath]
exten = s,1,Read(NUMBER,ginger3,2,skip,5)
exten = s,n,Gotoif($[${NUMBER} = 14]?onefour)
exten = s,n,Gotoif($[${NUMBER} = 24]?twofour)
exten = s,n,Gotoif($[${NUMBER} = 34]?threefour)
exten = s,n,Gotoif($[${NUMBER} = 20]?done)
exten = s,playback(system) - error message
exten = s,n,Set(TOTAL=0)
exten = s,n(onefour),Set(TOTAL1=${MATH(${TOTAL}+500,int)})
exten = s,n,Goto(tommath,s,1)
exten = s,n(twofour),Set(TOTAL2=${MATH(${TOTAL+TOTAL1}+200,int)})
exten = s,n,Goto(tommath,s,1)
exten = s,n(threefour),Set(TOTAL3=${MATH(${TOTAL+TOTAL1+TOTAL2}+300,int)})
exten = s,n,Goto(tommath,s,1)
exten = s,n(done),SayNumber(${TOTAL=TOTAL1+TOTAL2+TOTAL3})
exten = s,n,playback(vm-goodbye)
exten = s,n,hangup
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Re: [asterisk-users] MATH
hi Steve, I am trying it and I am using the feedback from the group. In my view, that is the purpose; try, test, talk. Thanks for your interest. On Tue, Feb 2, 2010 at 7:15 PM, Steve Edwards asterisk@sedwards.com wrote: On Tue, 2 Feb 2010, Thomas Perron wrote: I want to allow users to dial my DID Then, hear my ginger3 intro Then, depending on the number that they press, provide a total via MATH. Comments. Will this work? [snip] You've been asking this and related questions for days. Wouldn't it be faster to try it yourself? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MATH
Thank you.
I was also thinking of using the READ application to store dtmp variabes.
Then total them up at the end.
More to follow.
P
On Mon, Feb 1, 2010 at 9:20 AM, Danny Nicholas da...@debsinc.com wrote:
There's nothing wrong with this per se; it just needs to be in a context;
try it this way;
- exten = 8284,1,Goto(domath,s,1)
[domath]
Exten = s,1,play(to-call-num-press)
- exten = 4,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
- exten = 4,n,WaitExten(3)
- exten = 4,n,Goto(domath,s,1)
- exten = 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
- exten = 2,n,Waitexten(3)
- exten = 2,n,Goto(domath,s,1)
- exten = 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
- exten = 3,n,WaitExten(3)
- exten = 3,n,Goto(domath,s,1)
- exten = 9,1,SayNumber(${TOTAL})
- exten = 9,n,Play(vm-goodbye)
- exten = 9,n,hangup
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Sunday, January 31, 2010 5:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MATH
On 31/01/10 6:27 PM, Thomas Perron wrote:
what is wrong with this please:
;exten = 4,1,WaitExten(3)
exten = 4,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten = 4,n,WaitExten(3)
exten = 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
exten = 2,n,Waitexten(3)
exten = 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
exten = 3,n,WaitExten(3)
exten = 9,1,SayNumber(${TOTAL})
Heh, you might need to say what you're expecting and what you're getting :D
Straight off, all I can see is that 2 does 200, 3 does 300 and 4 does 500.
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Re: [asterisk-users] MATH
hi
i don't claim to be a star at this but there must be some obvious part missing;
my dial plan is below. out put from cli follows.
exten = 3011,1,Answer()
exten = 3011,n,Set(TOTAL=0)
exten = 3011,n,Set(TOTAL=${Math(${TOTAL}+300,int)})
exten = 3011,n,WaitExten(3)
exten = 988,1,SayNumber(${TOTAL})
[Jan 31 10:21:35] ERROR[1318]: pbx.c:2770 ast_func_read: Function Math
not registered
-- Executing [3...@default:3] Set(SIP/64.85.162.137-c0132f50,
TOTAL=) in new stack
-- Executing [3...@default:4]
WaitExten(SIP/64.85.162.137-c0132f50, 3) in new stack
[Jan 31 10:21:38] WARNING[1318]: pbx.c:7855 pbx_builtin_waitexten:
Timeout but no rule 't' in context 'default'
== Spawn extension (default, 3011, 4) exited non-zero on
'SIP/64.85.162.137-c0132f50'
localhost*CLI Function Math not registered
No such command ' Function Math not registered' (type 'help Function
Math' for other possible commands)
2010/1/31 Håkon Nessjøen haa...@avelia.no:
You probably have to do a
exten = s,1,n,Set(TOTAL=0)
in the start of the call, to initialize the TOTAL variable
On Sun, Jan 31, 2010 at 4:29 AM, Thomas Perron thomas.per...@gmail.com
wrote:
thanks for the response.
I tried to simplify and am now tuning the following, but it is not
responding with anything.
something wrong with timing?
here is what I have:
exten = 1625,1,Answer()
exten = 1625,n,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten = 1625,n,WaitExten(3)
exten = 9625,1,Answer()
exten = 9625,n,SayNumber(${TOTAL})
output from the console
[Jan 30 22:25:16] WARNING[22987]: func_math.c:194 math: '' is not a valid
number
-- Executing [1...@default:2] Set(SIP/64.85.162.137-c00d10e0,
TOTAL=) in new stack
-- Executing [1...@default:3]
WaitExten(SIP/64.85.162.137-c00d10e0, 3) in new stack
[Jan 30 22:25:19] WARNING[22987]: pbx.c:7855 pbx_builtin_waitexten:
Timeout but no rule 't' in context 'default'
== Spawn extension (default, 1625, 3) exited non-zero on
'SIP/64.85.162.137-c00d10e0'
2010/1/30 Håkon Nessjøen haa...@avelia.no:
Try something like:
exten = 1,1,WaitExten(3)
exten = 1,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten = 1,n,WaitExten(3)
exten = 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
exten = 2,n,WaitExten(3)
exten = 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
exten = 3,n,WaitExten(3)
exten = 9,1,SayNumber(${TOTAL})
Or something. Never used either math or saynumber before, but according
to
the documentation, something like this should work..
On Sat, Jan 30, 2010 at 3:06 PM, Thomas Perron thomas.per...@gmail.com
wrote:
total up for current call.
then read back the number
2010/1/30 Håkon Nessjøen haa...@avelia.no:
For all calls combined, or for the current call?
On Sat, Jan 30, 2010 at 2:48 PM, Thomas Perron
thomas.per...@gmail.com
wrote:
I want to create a script for IVR that compiles responses,
aggregates
them to a total number.
Then, run an equation based on the result.
Press 1 for X (X is a positive number 500)
Press 2 for Y (Y is a positive number 200)
Press 3 for Z (Z is a positive number 300)
Press 20 to calculate the results
= 500+200+300 =1000
then,
exten = s,n,Read(NUMBER,,1000)
exten = s,n,SayDigits(${NUMBER})
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Re: [asterisk-users] MATH
ok.
that worked
thanks!!
On Sun, Jan 31, 2010 at 10:50 AM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
On Sun, Jan 31, 2010 at 10:37:29AM -0500, Thomas Perron wrote:
hi
i don't claim to be a star at this but there must be some obvious part
missing;
my dial plan is below. out put from cli follows.
exten = 3011,1,Answer()
exten = 3011,n,Set(TOTAL=0)
exten = 3011,n,Set(TOTAL=${Math(${TOTAL}+300,int)})
exten = 3011,n,WaitExten(3)
exten = 988,1,SayNumber(${TOTAL})
[Jan 31 10:21:35] ERROR[1318]: pbx.c:2770 ast_func_read: Function Math
not registered
Function names are CaSe SenSitive, and are normally ALL CAPS. You should
use 'MATH' instead of 'Math'.
/me is done shouting for today, hopefully.
--
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icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
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Re: [asterisk-users] MATH
does dtmf any any variable that i can capture and use w/ some logic
like in the case of a gotoif
so, if caller enters a certain number then gotoif matches XX
otherwise go to YY.
On Sun, Jan 31, 2010 at 10:58 AM, Thomas Perron thomas.per...@gmail.com wrote:
ok.
that worked
thanks!!
On Sun, Jan 31, 2010 at 10:50 AM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
On Sun, Jan 31, 2010 at 10:37:29AM -0500, Thomas Perron wrote:
hi
i don't claim to be a star at this but there must be some obvious part
missing;
my dial plan is below. out put from cli follows.
exten = 3011,1,Answer()
exten = 3011,n,Set(TOTAL=0)
exten = 3011,n,Set(TOTAL=${Math(${TOTAL}+300,int)})
exten = 3011,n,WaitExten(3)
exten = 988,1,SayNumber(${TOTAL})
[Jan 31 10:21:35] ERROR[1318]: pbx.c:2770 ast_func_read: Function Math
not registered
Function names are CaSe SenSitive, and are normally ALL CAPS. You should
use 'MATH' instead of 'Math'.
/me is done shouting for today, hopefully.
--
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icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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[asterisk-users] MATH
I want to create a script for IVR that compiles responses, aggregates
them to a total number.
Then, run an equation based on the result.
Press 1 for X (X is a positive number 500)
Press 2 for Y (Y is a positive number 200)
Press 3 for Z (Z is a positive number 300)
Press 20 to calculate the results
= 500+200+300 =1000
then,
exten = s,n,Read(NUMBER,,1000)
exten = s,n,SayDigits(${NUMBER})
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Re: [asterisk-users] MATH
total up for current call.
then read back the number
2010/1/30 Håkon Nessjøen haa...@avelia.no:
For all calls combined, or for the current call?
On Sat, Jan 30, 2010 at 2:48 PM, Thomas Perron thomas.per...@gmail.com
wrote:
I want to create a script for IVR that compiles responses, aggregates
them to a total number.
Then, run an equation based on the result.
Press 1 for X (X is a positive number 500)
Press 2 for Y (Y is a positive number 200)
Press 3 for Z (Z is a positive number 300)
Press 20 to calculate the results
= 500+200+300 =1000
then,
exten = s,n,Read(NUMBER,,1000)
exten = s,n,SayDigits(${NUMBER})
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Re: [asterisk-users] MATH
thanks for the response.
I tried to simplify and am now tuning the following, but it is not
responding with anything.
something wrong with timing?
here is what I have:
exten = 1625,1,Answer()
exten = 1625,n,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten = 1625,n,WaitExten(3)
exten = 9625,1,Answer()
exten = 9625,n,SayNumber(${TOTAL})
output from the console
[Jan 30 22:25:16] WARNING[22987]: func_math.c:194 math: '' is not a valid number
-- Executing [1...@default:2] Set(SIP/64.85.162.137-c00d10e0,
TOTAL=) in new stack
-- Executing [1...@default:3]
WaitExten(SIP/64.85.162.137-c00d10e0, 3) in new stack
[Jan 30 22:25:19] WARNING[22987]: pbx.c:7855 pbx_builtin_waitexten:
Timeout but no rule 't' in context 'default'
== Spawn extension (default, 1625, 3) exited non-zero on
'SIP/64.85.162.137-c00d10e0'
2010/1/30 Håkon Nessjøen haa...@avelia.no:
Try something like:
exten = 1,1,WaitExten(3)
exten = 1,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten = 1,n,WaitExten(3)
exten = 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
exten = 2,n,WaitExten(3)
exten = 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
exten = 3,n,WaitExten(3)
exten = 9,1,SayNumber(${TOTAL})
Or something. Never used either math or saynumber before, but according to
the documentation, something like this should work..
On Sat, Jan 30, 2010 at 3:06 PM, Thomas Perron thomas.per...@gmail.com
wrote:
total up for current call.
then read back the number
2010/1/30 Håkon Nessjøen haa...@avelia.no:
For all calls combined, or for the current call?
On Sat, Jan 30, 2010 at 2:48 PM, Thomas Perron thomas.per...@gmail.com
wrote:
I want to create a script for IVR that compiles responses, aggregates
them to a total number.
Then, run an equation based on the result.
Press 1 for X (X is a positive number 500)
Press 2 for Y (Y is a positive number 200)
Press 3 for Z (Z is a positive number 300)
Press 20 to calculate the results
= 500+200+300 =1000
then,
exten = s,n,Read(NUMBER,,1000)
exten = s,n,SayDigits(${NUMBER})
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[asterisk-users] MATH
what is wrong with this please:
;exten = 4,1,WaitExten(3)
exten = 4,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten = 4,n,WaitExten(3)
exten = 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
exten = 2,n,Waitexten(3)
exten = 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
exten = 3,n,WaitExten(3)
exten = 9,1,SayNumber(${TOTAL})
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[asterisk-users] receive text
Is there any code that I can cut/paste that will allow me to receive an SMS text on Asterisk? and, where can I capture the incoming text? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial String command after audio background
exten = s,1,Answer()
exten = s,n,Background(astcc-please-enter-your)
exten = s,n,Background(zip-code)
exten = s,n,WaitExten(5)
exten = s,n,Read(NUMBER,,5)
exten = s,n,SayDigits(${NUMBER})
exten = 22042,n,Dial(SIP/sipvendor/111,120,A(ginger3))
exten = 22601,n,Dial(SIP/sipvendor/111,120,A(ginger3)) ;
x/ winchester
exten = 21230,n,Dial(SIP/sipvendor/111,120,A(ginger3)) ;
Mobile/Baltimore
I want to background to play please enter your zip code
Then say the digits pressed (5 digits)
Then map the five digits to an extension as shown to engage a Dial string
Examples above are not working.
Do I need an Answer() entry first for each zip code (extension)?
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Re: [asterisk-users] Dial String command after audio background
veilen danke timm
cheers
tom
On Sun, Jan 17, 2010 at 2:10 PM, Timm Korte
korte-ast-us...@easycrypt.de wrote:
Am 17.01.2010 18:39, schrieb Thomas Perron:
exten = s,1,Answer()
exten = s,n,Background(astcc-please-enter-your)
exten = s,n,Background(zip-code)
exten = s,n,WaitExten(5)
exten = s,n,Read(NUMBER,,5)
exten = s,n,SayDigits(${NUMBER})
you might want to add a GoTo(${NUMBER},1)
as well as start your other extensions with
exten = 22042,1,Dial(SIP/sipvendor/111,120,A(ginger3))
then
exten = 22042,n,Dial(SIP/sipvendor/111,120,A(ginger3))
I want to background to play please enter your zip code
Then say the digits pressed (5 digits)
Then map the five digits to an extension as shown to engage a Dial string
Examples above are not working.
Because your're staying in the s extension - you need to switch to another
extension by using (for example, since there are other ways...)
goto.
Do I need an Answer() entry first for each zip code (extension)?
Nope - just give each a real id or label (instead of n) so you can address
them via goto.
Timm
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[asterisk-users] Music / Background
I want to play soft music in the background while the IVR passes
through various contexts.
In short, I need to mix the script with music and my pre-staged .gsm
or .wav audio.
What tools to I need to use in Asterisk to make this happen please?
exten = s,1,Answer()
;exten = s,n,system(echo ${DATETIME} - ${CALLERID(all)} -
${CHANNEL} /var/log/asterisk/calls)
;exten = s,n,System(echo body of message | mail -s subject line
${the_caller_...@txt.att.net)
exten = s,n,Background(dir-welcome)
exten =
s,n,Background(/var/lib/asterisk/sounds/en/baseline_introduction_script.gsm)
exten = s,n,Background(astcc-please-enter-your)
exten = s,n,Background(zip-code)
exten = s,n,Read(NUMBER,,5)
so, play music behind (dir-welcome),
/var/lib/asterisk/sounds/en/baseline_introduction_script.gsm,
(astcc-please-enter-your) and (zip-code).
Send solutions please.
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[asterisk-users] pattern matching
I want to ensure that only this extension is executed. But, I have others that are similar. I want: exten = 34101,1,Answer() exten = 34101,n,Record(34101:gsm) ; 34101 test zip code exten = 34101,n,Playback(34101) exten = 34101,n,Hangup Is this correct or do I need to have each of the four statements lead with an underscore (_) to make an exact match? Other code looks similar so I don't want the 102 to connect when I am dialing 101 exten = 34102,1,Answer() exten = 34102,,n,Record(34102:gsm) ; 34102 test zip code exten = 34102,n,Playback(34102) exten = 34102,n,Hangup ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] script
I want to have Asterisk Dial individual numbers and play a recording if each answers. If they don't answer then the code rolls to the next number. Should I set up a spreadsheet somewhere and load with the numbers? Or, an AGI script? 1. Dial number 1 2. If connect, then play message 3. If connect, finish message and move to next number 4. Dial 1 - 10,000 in succession ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sendmail
Anyone have a cookbook on configuring sendmail to work with Asterisk? Or,a few config examples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.12 Now Available
How does Fax for Asterisk work? On Fri, Dec 18, 2009 at 7:51 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: Warren Selby wrote: Is the new Fax For Asterisk being released in conjunction with this release? If it's not already available, then it will be available very early next week. Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto Attendant / Receptionist system
Does anyone have a script that performs Auto Attendant / Receptionist system If so, please send. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto Attendant / Receptionist system
I want to list 100 indiviual businesses. and do an ivr for them specifically some use databases so i need an agi script in .pl or php. On Sat, Dec 12, 2009 at 7:26 PM, Doug Lytle supp...@drdos.info wrote: Thomas Perron wrote: Does anyone have a script that performs Auto Attendant / Receptionist system If so, please send. You need to be more specific. What are you looking for this to do? Your question is too generic. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to Email
Interesting response but I am not that saavy to follow it!
Thank you
On Sun, Dec 6, 2009 at 2:40 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Sat, Dec 05, 2009 at 08:25:33PM -0500, Thomas Perron wrote:
And, then send an email to the party. Example
3035551...@tmobile.net
Summary
1. Capture the CID number.
2. Prepend his number to his service provider SMTP address
3. Email it to his phone
System(echo body of message | mail -s subject line
${the_caller_...@tmobile.net)
Note the usage of '|' here. IIRC it needs to be escaped on Asterisk
1.4.x and below.
I assume I need to install SendMail and play around with CID stuff.
Sendmail, postfix, exim, qmail - any program that provides a local
sendmail interface.
I personally prefer postfix.
--
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icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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[asterisk-users] sequential dialing preferences
I am trying to use a simple tool in the Dial plan so that if the first number does not connect the logic will go to the second and/or third. Basically, I want the call to ring and connect to the first number Then, if it is not answered I want another number to try to get connected Then, if second number does not answer I want the third to be tried i only list the scenario for the first two numbers Here is what I have now which works fine for the one and only number... exten = s,n,Dial(SIP/callwithus/12135551212,120,A(ginger3)) ; Service line so, will this work ... .. exten = s,n,Dial(SIP/callwithus/12135551212[SIP/callwithus/12145551212],120,A(ginger3)) ; Service line Please send comments to make this work. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to Email
I am reading a lot of the material but need your input to help me
understand what you mean.
System(echo body of message | mail -s subject line
${the_caller_...@tmobile.net)
I understand the System application generally
echo body of message .?
mail -s --what does this do please?
subject line .comes from where?
${the_caller_...@tmobile.net) i understand this part.
thank you
On Sun, Dec 6, 2009 at 2:40 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Sat, Dec 05, 2009 at 08:25:33PM -0500, Thomas Perron wrote:
And, then send an email to the party. Example
3035551...@tmobile.net
Summary
1. Capture the CID number.
2. Prepend his number to his service provider SMTP address
3. Email it to his phone
System(echo body of message | mail -s subject line
${the_caller_...@tmobile.net)
Note the usage of '|' here. IIRC it needs to be escaped on Asterisk
1.4.x and below.
I assume I need to install SendMail and play around with CID stuff.
Sendmail, postfix, exim, qmail - any program that provides a local
sendmail interface.
I personally prefer postfix.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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[asterisk-users] Asterisk to Email
How can this scenario be implemented please? THIS IS NOT A SEND TEXT application. A call arrives on the IVR. After hearing several vectors to guide the person through information I want to confirm a transaction via email to his cell phone. Specifically, I want to use his phone number and then append the SMTP suffix from his service provider. Press 1 if you use Verizon, 2 if you use ATT, 3 if you use Sprint, 4 for T-Mobile, etc. And, then send an email to the party. Example 3035551...@tmobile.net Summary 1. Capture the CID number. 2. Prepend his number to his service provider SMTP address 3. Email it to his phone I assume I need to install SendMail and play around with CID stuff. Any hints? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI
I am trying to find an AGI script that runs via PHP and performs the send text application. Does anyone have any tools or scripts set up for this please? If so, kindly send some info or the code that performs this action. Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI stuff
How do I get to this prompt? #!/usr/bin/php -q ?php ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI stuff
Hallo Philipp, Wei Gehts ist Einen. Danke. I am in USA. Thanks. On Sun, Nov 29, 2009 at 8:49 PM, Philipp Kempgen philipp.kemp...@amooma.de wrote: Thomas Perron schrieb: How do I get to this prompt? #!/usr/bin/php -q ?php http://en.wikipedia.org/wiki/Shebang_%28Unix%29 Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DIDs
I have two DID numbers. I want to configurate my IVR to initiate a context 1 if I dial DID 1. If DID2 is dialed then start with context 2. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDs
Hi Alex,
Thank you
Tom
On Sat, Nov 21, 2009 at 10:24 AM, Alex Balashov
abalas...@evaristesys.comwrote:
Thomas,
Thomas Perron wrote:
I have two DID numbers. I want to configurate my IVR to initiate a
context 1 if I dial DID 1.
If DID2 is dialed then start with context 2.
Assuming that the DID originator sends you the number in the Request
URI, you can just treat them like extensions in Asterisk. Example:
if you have DID 6789540670 and 6789540671:
exten = 6789540670,1,Goto(context_1,${EXTEN},1)
exten = 6789540671,1,Goto(context_2,${EXTEN},1)
[context_1]
; IVR
exten = 6789540670,1,Answer
exten = 6789540670,n,Playback(hello-world)
exten = 6789540670,n,Hangup
[context_2]
exten = 6789540671,1,Dial(SIP/abalashov,30,r)
exten = 6789540671,n,Congestion
-- Alex
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Re: [asterisk-users] DIDs
thanks
On Sat, Nov 21, 2009 at 12:26 PM, Steve Edwards
asterisk@sedwards.comwrote:
Thomas Perron wrote:
I have two DID numbers. I want to configurate my IVR to initiate a
context 1 if I dial DID 1. If DID2 is dialed then start with context 2.
If the DIDs are from different providers, you can specify different
contexts in [iax|sip].conf.
On Sat, 21 Nov 2009, Alex Balashov wrote:
Assuming that the DID originator sends you the number in the Request
URI, you can just treat them like extensions in Asterisk. Example: if
you have DID 6789540670 and 6789540671:
exten = 6789540670,1,Goto(context_1,${EXTEN},1)
exten = 6789540671,1,Goto(context_2,${EXTEN},1)
I prefer to save EXTEN (after any pattern matching nonsense) in a more
meaningful variable like DNIS and then use the s extension from then
on. I find it cleaner and more maintainable. For example (typing off the
top of my head):
[incoming-from-xyz]
exten = _678954067x,1, set(DNIS=${EXTEN})
exten = 6789540670,2, goto(home,s,1)
exten = 6789540671,2, goto(work,s,1)
[home]
exten = s,1,dial(sip/home-phone)
[work]
exten = s,1,dial(sip/work-phone)
If I get another work number, I just add another line to
incoming-from-xyz. If I change a number, I just change that single line.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Verification number / code
I want to distribute a random number to each of the first 100 callers to my IVR. This random number will be matched to their telephone number. Where in Asterisk can I do this? And, how? Logic. Call arrives. Context for announcement begins. You will receive a authentication code at the end of the message. Then, if they press a certain digit to confirm then I simply pass a code to them. These codes are distributed to the first 100. The 101st call does not get a code. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verification number / code
that is a bit heavy for me. how about some simple way to announce a random
number. using RAND. and saydigit
exten = s,1,Set(junky=${RAND(1,8)})
On Sat, Nov 21, 2009 at 7:20 PM, Steve Edwards asterisk@sedwards.comwrote:
On Sat, 21 Nov 2009, Thomas Perron wrote:
I want to distribute a random number to each of the first 100 callers to
my IVR. This random number will be matched to their telephone number.
Where in Asterisk can I do this? And, how?
Logic.
Call arrives.
Context for announcement begins.
You will receive a authentication code at the end of the message.
Then, if they press a certain digit to confirm then I simply pass a code
to
them.
These codes are distributed to the first 100.
The 101st call does not get a code.
I'm guessing you really don't want a random number since a random number
generator can generate duplicates.
Matching the number to their ANI also has issues. What if my ANI is
blocked? What if I spoof my ANI? What if I call from a SIP phone?
I would pre-compute the random numbers and store them in a database.
When a call arrives, I would invoke an AGI that would lock the table, read
the first value with a null ANI, update the row with the caller's ANI, and
unlock the table.
You could do it in dialplan, but I find database access in dialplan ugly.
Alternatively, you could mung UNIQUEID (number of seconds since
Epoch.number of channels created by this instance of Asterisk) to
appear to the caller as random and then store that and their ANI in a
database.
What happens if Asterisk is restarted in the middle of your campaign?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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Re: [asterisk-users] SendText
OK. Thanks On Thu, Nov 12, 2009 at 4:33 AM, Tarek Sawah tareksa...@hotmail.com wrote: i have my own SMS provider as we sell SMS .. so i have setup my call center with SMS sending for several services and alerts like a Missed Call when i'm not registered it will send me an sms to alert me. it's pretty the same as Matt discribed.. you call an AGI which may use cURL to hit the HTTP API -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 -- Date: Mon, 9 Nov 2009 22:19:08 -0500 From: thomas.per...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SendText Will text messages work to non-SIP enpoints using your logic/code? thank you On Mon, Nov 9, 2009 at 8:59 PM, Matt Riddell li...@venturevoip.comwrote: On 10/11/09 12:58 PM, Thomas Perron wrote: Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). We use clickatel. Basically we use the PHP API and call it via an AGI which sends texts. Therefore the extensions.conf is pretty sparse: exten = s,1,Read(destination) exten = s,2,AGI(agi://127.0.0.1/send_sms.php) Pseudo code for send_sms is: 1. Read AGI variables 2. Get destination variable 3. Include clickatel API file 4. call send_sms function We also provide an API from our telephone exchanges, but to be fair you're likely better off just using clickatel yourself :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Windows 7: Unclutter your desktop. Learn more.http://go.microsoft.com/?linkid=9690331ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen:112009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SendText
Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText
Will text messages work to non-SIP enpoints using your logic/code? thank you On Mon, Nov 9, 2009 at 8:59 PM, Matt Riddell li...@venturevoip.com wrote: On 10/11/09 12:58 PM, Thomas Perron wrote: Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). We use clickatel. Basically we use the PHP API and call it via an AGI which sends texts. Therefore the extensions.conf is pretty sparse: exten = s,1,Read(destination) exten = s,2,AGI(agi://127.0.0.1/send_sms.php) Pseudo code for send_sms is: 1. Read AGI variables 2. Get destination variable 3. Include clickatel API file 4. call send_sms function We also provide an API from our telephone exchanges, but to be fair you're likely better off just using clickatel yourself :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Text messaging
IVR question: Users dial my DID numbers and get connected to macros and other vectors that guide them to the appropriate context. Once connected to a specific context I would like to send a text message to their phone. Do I need a PERL script or is there something native in Asterisk 1.6 that can trigger a text to the endpoint? Thank you [default] ;include = stdexten include = big10-IVR include = cleveland-IVR exten = _1703XXX,1,Goto(big10-IVR,s,1) exten = _1517XXX,1,Goto(cleveland-IVR,s,1) [big10-IVR] exten = s,1,Answer() exten = s,n,Background(dir-welcome) ;exten = s,n,WaitExten(1) ;exten = s,n,Background(astcc-please-enter-your) ;exten = s,n,Background(zip-code) ;exten = s,n,Wait(7) exten = s,n,Background(washington-dc) ;exten = s,n,Authenticate(,a) ;exten = s,n,Background(pin-number-accepted) exten = s,n,Playback(queue-thankyou) exten = s,n,Background(ginger110109) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tutorial for SIP user
I am having the same issue. Please assist. On Sun, Nov 1, 2009 at 1:27 PM, giancarlo lombardo gianclomba...@gmail.comwrote: Dear all, I'm trying to setup a SIP call with XLITE using my asterisk PBX, but I have trouble, I see on XLITE console: Registration Error: 503 - Service unavailable. Someone have a tutorial or a step by step description how to do that ? Thanks in advance -- Giancarlo Lombardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pattern matching DID
I have two DID numbers. I want callers who dial 1 703 to get placed in a specific part of IVR I want other callers who dial 1 567 to get placed in a different area. How do I do this please? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pattern matching DID
Thank you. I am trying it shortly. This is a lot of fun. I am trying to find places where I can get customers with IVR or anything relating to Asterisk. Any ideas? Cheers Tom On 11/1/09, Aggio Alberto alberto.ag...@loquendo.com wrote: Hi, it's quite straightforward: you can do your dialplan like this (default is the default context answered when inbound calls happen) - remember the underscores! - [default] exten = _1703,1,Goto(place-IVR,s,1) exten = _1567 ,1,Goto(place-other,s,1) [place-IVR] exten = s,1,Answer exten = s,2,Background(menu-file) exten = 1,1,Goto(submenu,1) exten = 2,1,Goto(submenu,2) (...) [place-other] exten = s,1,Answer exten = s,n,... (...) exten = s,n,Hangup If you want to jump into a specific part of context, you should put a label near the 'n' priority where you want to jump to (eg. exten = s,n(jumphere),application/function) then specify that label into Goto() application. Cheers, //Al. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron Sent: domenica 1 novembre 2009 21.46 To: asterisk-users@lists.digium.com Subject: [asterisk-users] pattern matching DID I have two DID numbers. I want callers who dial 1 703 to get placed in a specific part of IVR I want other callers who dial 1 567 to get placed in a different area. How do I do this please? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pattern matching DID
Where is everyone located? I am in Virginia, USA On 11/1/09, Aggio Alberto alberto.ag...@loquendo.com wrote: Hi, it's quite straightforward: you can do your dialplan like this (default is the default context answered when inbound calls happen) - remember the underscores! - [default] exten = _1703,1,Goto(place-IVR,s,1) exten = _1567 ,1,Goto(place-other,s,1) [place-IVR] exten = s,1,Answer exten = s,2,Background(menu-file) exten = 1,1,Goto(submenu,1) exten = 2,1,Goto(submenu,2) (...) [place-other] exten = s,1,Answer exten = s,n,... (...) exten = s,n,Hangup If you want to jump into a specific part of context, you should put a label near the 'n' priority where you want to jump to (eg. exten = s,n(jumphere),application/function) then specify that label into Goto() application. Cheers, //Al. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron Sent: domenica 1 novembre 2009 21.46 To: asterisk-users@lists.digium.com Subject: [asterisk-users] pattern matching DID I have two DID numbers. I want callers who dial 1 703 to get placed in a specific part of IVR I want other callers who dial 1 567 to get placed in a different area. How do I do this please? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pattern matching DID
Does anyone know of a low price SIP termination service to Nepal? For VoIP calling card solutIon On 11/1/09, Aggio Alberto alberto.ag...@loquendo.com wrote: Hi, it's quite straightforward: you can do your dialplan like this (default is the default context answered when inbound calls happen) - remember the underscores! - [default] exten = _1703,1,Goto(place-IVR,s,1) exten = _1567 ,1,Goto(place-other,s,1) [place-IVR] exten = s,1,Answer exten = s,2,Background(menu-file) exten = 1,1,Goto(submenu,1) exten = 2,1,Goto(submenu,2) (...) [place-other] exten = s,1,Answer exten = s,n,... (...) exten = s,n,Hangup If you want to jump into a specific part of context, you should put a label near the 'n' priority where you want to jump to (eg. exten = s,n(jumphere),application/function) then specify that label into Goto() application. Cheers, //Al. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron Sent: domenica 1 novembre 2009 21.46 To: asterisk-users@lists.digium.com Subject: [asterisk-users] pattern matching DID I have two DID numbers. I want callers who dial 1 703 to get placed in a specific part of IVR I want other callers who dial 1 567 to get placed in a different area. How do I do this please? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
