similar to: DTMF Issue?

Jan 19 10:00:29 VERBOSE [7177] logger.c: -- Executing [1000 at ext-meetme:7] Read("DAHDI/2-1", "PIN|enter-conf-pin-number||||") in new stack Jan 19 10:00:29 VERBOSE [7177] logger.c: -- Playing 'enter-conf-pin-number' (language 'en') Jan 19 10:00:43 VERBOSE [7177] logger.c: -- USER ENTERED 'THE PIN NUMBER' Jan 19 10:00:43 VERBOSE [7177] logger.c: --

Hi! Trying to figure out what I am doing wrong... 1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256) 1 Cell phone 00733025975 attached through H323. extensions.conf exten => 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1) exten => 975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 975-INUSE,2,Hangup() exten =>

Hi! Have probably not understand how fax is working in Asterisk 1.6. I did install: ptlib-v1_12_0 h323plus-v1_19_7 dahdi-linux-complete-2.2.0.2+2.2.0 spandsp-0.0.5 asterisk-1.6.2 asterisk-addons-1.6.2 make menuselect in asterisk-1.6.2 source directory shows: [*] app_fax But "core show applications" doesnt show me any "fax applications" and when I try to receive a fax:

Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized

Hi! Part of extensions.conf: exten => 985,1,Dial(SIP/0317998985&H323/00702221448 at Avaya,20) exten => 985,2,Goto(985-${DIALSTATUS},1) exten => 985-BUSY,1,VoiceMail(0317998985 at inputinterior.se,b) exten => 985-BUSY,2,PlayBack(vm-goodbye) exten => 985-BUSY,3,HangUp() exten => 985-NOANSWER,1,VoiceMail(0317998985 at inputinterior.se,u) exten =>

Hi, I am looking for a way to check the status of a cell phone. Found one way that worked for me and would like to have some feedback or suggestion of improvments. Below example is for a ?Swedish? cell phone, dont know if it works in the same way for other countries. I could define ?redirecting? numbers for 3 traffic cases when u dial my mobile (073-302 59 75): NOT_INUSE call forward to A INUSE

Hi! Trying to figure out how to rewrite calling number of an incoming call... A cell phone (0733025975) dials a X-Lite (977). X-Lite "shows" 733025975 at the display, but I want it to be 0317998975. I thought i could do something like: exten => 977/733025975,1,Set(CALLERID(number)=0317998975) exten => 977,n,Dial(SIP/0317998977) [Dec 14 19:07:43] NOTICE[20731]: chan_h323.c:2272

Hi! Did play around with queues and need some help. I thought that MEMBERINTERFACE and MEMBERNAME should be set to the ?device? the call was queued to not the device that called the queue, or do i miss something? Running: Asterisk 1.8.2.3 built by root @ sip on a i686 running Linux on 2011-01-31 13:38:23 UTC 0317998985 calls Kinna (0320209030) Tomas Ekman (SIP/0317998972) receives the call but

Hi! I try to get rid of some part of CALLERID(name) but I cant realy figure out a way to do it. For example: CALLERID(name) = "Martela (fax)" I am just looking for the part before ? (? in my case ?Martela?. I can?t serch for ? ?, could be many ? ?, but only one ? (?, thought i could do something like: exten => 0424449631,n,NoOp(${CUT(CALLERID(name),\(,1):0:-1}) But that gave me

Gentlemen, Did tryout "CONNECTEDLINE" function, was exactly what I have been looking for. But there are at least one thing I cant figure out. Did a very simple and "stupid" extension 0317998955 and ran a test. My phone (0317998975) dials 955, the display on my phone changes from "955" to "Connected Line 955" when my call is answered, shouldn't the

Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what.

OOH323 Debugging Enabled -- Executing Answer("SIP/3513-090f7d40", "") in new stack -- Executing Wait("SIP/3513-090f7d40", "1") in new stack -- Executing DeadAGI("SIP/3513-090f7d40", "a2billing.php|1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: line:58 - IDCONFIG : 1

Anyone know how to override the 99 message limit in voicemail? (yeah, we have a public VM that gets that many a day). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050812/1bff12e4/attachment.htm

Hi all! I am trying to figure out how DEVICE_STATE is working, no luck so far. sip.conf [0317998975] type=friend regexten=0317998975 secret=???? username=0317998975 callerid="Magnus Benngard" mailbox=0317998975 at inputinterior.se host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes disallow=all allow=alaw extensions.conf exten => 0317998975,hint,SIP/0317998975 exten =>

I have tried everything. to register with this gatekeeper to make and receive calls These are the details I received from the voip provider: protocol H.323 Gatekeeper Address - AVS@210.21.118.XXX Port - 1719 RAS - 53 Q931 - 80 h245 - 1722 RTP - 1722 Username - H323 I have 2 phone number/accounts with this gatekeeper that I need to register to. ID - HMA0200.10szxn-xxxx e.164 - 22xx2912

Hi all! First of all i would like to say that i am not a Samba4 guru so my question may be "stupid". I have 2 Samba4 servers up and runnning: Server 1: netbios name = PDC workgroup = GBG realm = GBG.INPUTINTERIOR.SE server role = domain controller Server 2: netbios name = PDC workgroup = MLM realm = MLM.INPUTINTERIOR.SE server role = domain controller Here comes my

Morning, Have some questions regarding receiving and sending faxes... 1:st example: exten => 101,1,Answer() exten => 101,2,Wait(3) exten => 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten => 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf) exten => 101,5,System(mutt -s 'New FAX for you sir' -a

I am having a slight issue. I am trying to register 2 asterisk boxes with GNUGK and when I try to add the 2nd it gets denied cause of it saying its a duplicate. How do I change the configs to allow more than one asterisk box register to the same GK? brian This email was scanned by: Mcafee GroupShield ---------------- CONFIDENTIAL DISCLAMER ---------------- All information provided in this

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38

Hi, I try to fit a non linear regression by minimising the sum of the sum of squares. The model is number[2]-(x/number[1])^number[3] Number [2] and number [1] change as the data changes but for all the set of data number[3] must be identical. I have 3 set of data (x1,y1), (x2,y2), (x3,y3). x_a<-c(0,0.5,1,1.5,2,3,4,6) y_a<-c(5.4,5,4.84,4.3,4,2,1.56,1.3)