search for: 123456789

...als "999" it is intercepted and sent via the zap channels. If no zap channel is free, a call on channel 1 is ended and the number re-dialled. This makes sure that emergency calls can always be placed on a landline. Any ideas would be appreciated! Phil [softoption-zap] exten => _0[123456789].,1,NoOp(${EXTEN}) exten => _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j) exten => _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,) exten => _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,) exten => _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,) ; The below section will allow for 3...

...gunas combinaciones de esas variables. Unas combinaciones tienen que pasar a ser los sujetos (son sujetos, de hecho) y las otras combinaciones tienen que pasar a ser las variables dicotomicas. Para intentar ejemplificarlo, propongo una matriz original como esta: caracteristica valor sitio edad 123456789 tiza acá joven 123456789 blanco acá viejo 123456789 blanco allá joven 123456789 blanco allá viejo 234567891 rojo acá joven 234567891 colorado acá viejo 234567891 azul allá joven 234567891 azul allá viejo 234567891 azul acá viejo 345678912 teja acá joven 3...

...c_ctx) > setting sec ctx (5227, 513) - sec_ctx_stack_ndx = 0 > [2017/01/31 10:08:43.323612, 5] ../libcli/security/security_token.c:63(security_token_debug) > Security token SIDs (45): > SID[ 0]: S-1-22-1-5227 > SID[ 1]: S-1-22-2-513 > SID[ 2]: S-1-5-21-12345678-123456789-868425949-35723 > SID[ 3]: S-1-5-32-551 > SID[ 4]: S-1-5-21-12345678-123456789-868425949-54195 > SID[ 5]: S-1-22-2-5923 > SID[ 6]: S-1-22-2-512 > SID[ 7]: S-1-22-2-1074 > SID[ 8]: S-1-5-21-12345678-123456789-868425949-1427 > SID[ 9]: S-1-5-21-...

...ed but it just can not set/change anything (except password and some user flags). When i try to change group membership and add user to "Domain Admins" group is says "The Group name can not be found" users is the output of my groupmap list. Domain Admins (S-1-5-21-4276035903-123456789-2739979074-512) -> Domain Admins Domain Users (S-1-5-21-4276035903-123456789-2739979074-513) -> Domain Users Domain Guests (S-1-5-21-4276035903-123456789-2739979074-514) -> Domain Guests Administrators (S-1-5-21-4276035903-123456789-2739979074-544) -> Administrators users (S-1-5-21-4...

Hi, I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now having a problem with Originate and chan_local. I'm using the following Manager API action to originate a call: Action: originate Priority: 1 Context: trunk Callerid: 100 Channel: Local/100 at callback/n Exten: 123456789 Variable: USERFIELD=127.0.0.1|USEREXT=123456789 WaitTime: 30 This is intended to first call extension 100 in the callback context, and then when that is answered, call 123456789 in the trunk context. I have the following in the callback context: exten => 100,1,Answer exten => 100,2,Wait(2)...

hi! anyone has a idea how to send caller id (incoming number) to shell script like: exten => s,1,wait,2 exten => s,2,System(/bin/script callerid-number) exten =>s,3,answer i need to run script in shell with command line like "/bin/script -a 123456789 " this 123456789 must be callerid (incoming call) number. tnx, Thomas

Hi, Try this: lines1<- readLines(textConnection("gene1 or1|1234 or3|56 or4|793 gene4 or2|347 gene5 or3|23 or7|123456789")) lines2<-readLines(textConnection(">or1|1234 ATCGGATTCAGG >or2|347 GAACCTATCGGGGGGGGAATTTATATATTTTA >or3|56 ATCGGAGATATAACCAATC >or3|23 AAAATTAACAAGAGAATAGACAAAAAAA >or4|793 ATCTCTCTCCTCTCTCTCTAAAAA >or7|123456789 ACGTGTGTACCCCC"))? lines2New<-unlist(lap...

Back to the RE-provision workbench. :-)

Hello all, I would like to use Asterisk to add/modify SIP headers in the INVITE message, to include Privacy information, if the INVITE includes a *67 prefix (or another predefined prefix). That's an example of the INVITE I get: /INVITE sip:*6700112233445 at 192.168.1.100 SIP/2.0 From: "123456789"<sip:*123456789*@192.168.1.100>;tag=333333333 To: <sip:*6700112233445 at 192.168.1.100> / These are the sip headers I need to add to the INVITE: /P-Asserted-Identity: <sip:*123456789*@192.168.1.100:5060> Remote-Party-ID: <sip:*123456789*@192.168.1.100:5060>;party=calli...

Hello, I have an internal extension, e.g. 1005 which is being called from an external/public number like 123456789. Now when it comes to the spoken voicemail information it says something like "number 1000 not available", however it should say "number 123456789 not available". How can I configure this? I already googled and I guess this is really easy, but I just couldn't figure out how...

...oesn't see the domain until I manually add an entry for it in /etc/passwd Checking the user entry for two users I can find the following differences. BERENICE is an user imported from the old system and is working fine: dn: uid=berenice,ou=Users,dc=DOMAIN,dc=IT uid: berenice sambaSID: S-1-5-21-1234567890-123456789-123456789-2018 sambaPrimaryGroupSID: S-1-5-21-1234567890-123456789-123456789-513 displayName: berenice sambaLogonTime: 0 sambaLogoffTime: 4294967295 sambaKickoffTime: 4294967295 sambaPwdCanChange: 1161193814 sambaPwdMustChange: 4294967295 sambaLMPassword: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx...

...to present the name of the number called to the operator. Before that went. To identify the sda, I use the assignment of the callerid according to the sda called. Thank's for your help Here what I do: exten => 8489,1,AGI(test.php) exten => 8489,n,Set(CALLERID(all)=${NOM_CLIENT} <123456789>) exten => 8489,n,AGI(test.php) exten => 8489,n,Dial(SIP/7297,,T) ####Presentation of number -- Accepting call from '611134024' to '8489' on channel 0/8, span 1 -- Executing AGI("Zap/8-1", "test.php") in new stack -- Launched AGI Script /v...

...looks like this: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown pendantic=no register=123456789:aaa607ac5aabb25ac037@voipswitch1.voip.xxxxxxx.net [123456789] type=user secret=aaa607ac5aabb25ac037 insecure=very context=from-pstn [xxxxxxxx] username=123456789 type=peer secret=aaa607ac5aabb25ac037 host=voipswitch1.voip.xxxxxxxx.net

How can I make it more readable? Name/username 601/601 123456789/123456789 voipbuster/abcd 601 = hotline 123456789 = Peter Pan only voipbuster/abcd is easy read/understandable! bye Ronald Wiplinger

I am using /set.seed()/ before the /sample/ function. How does the length of the argument of /set.seed()/ and order of the digits affect how the sampling is carried out? Specifically, I have used set.seed(123456789). Will this configuration give me a genuinely random sampling?? Thank you in anticipation. Penny. [[alternative HTML version deleted]]

Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/atta...

On 1 July 2016 at 17:41, Joshua Colp <jcolp at digium.com> wrote: > > >> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) >> same => n,Dial(PJSIP/phone123, 30) >> > > Your exten line has no priority, is that how it is in your dialplan? > Actually no, I stole that line from an earlier email to this list. Mine has a priority. > If not you can isolate things a bit further by trying the...

Hi list! I have a Legacy PBX hooked up to one port of a TE210P. One is connected to I want to use the PBX for incoming and outgoing fax. I managed to get incoming working but I don't understand how to feed calls from the PBX into the FreePBX setup. It seems that when you press zero on the Legacy PBX the PBX just switches over to the PRI to pass each digit nicely one by one as it is

Hello, I want to enable call forwarding for asterisk 1.6.0.6 I couldnt seen any config or option on gui or extensions.conf about it. I found some dialing plans to enable it on web as follows: [apps] ; Unconditional Call Forward exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4}) exten => _*21*X.,2,Hangup exten => #21#,1,DBdel(CFIM/${CALLERIDNUM}) exten => #21#,2,Hangup ;

...ade via the PRI (Zap channel) to an enaged number it simply plays an enaged signal to the caller and never progresses past the Dial priority. I know for a fact the call is not actually being answered, because I get the following onthe console. Executing Dial("SIP/1001-270b", "Zap/g1/123456789") in new stack -- Called g1/123456789 pri debug span 1 gives me: < Protocol Discriminator: Q.931 (8) len=5 < Call Ref: len= 2 (reference 32809/0x8029) (Terminator) < Message type: RELEASE (77) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request >...