similar to: BRI detection

Hi, Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:saigop at gtalk2voip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110401/051f68d3/attachment.htm>

Sir, i am using goautodial server , bri card is showing ok but when i try to call that showing below , This configuration is in doing in dubai , so kindly help me how can connet the call from this , what is my mistake is in this :::chan-dahdi.conf [channels] #include dahdi-channels.conf language=en context=default usecallerid=yes hidecallerid=yes callwaiting=yes usecallingpres=yes

Hi, THIS IS IN DUBAI. I am having PRI line with 100 DID's (00-99) and when we call to any landline or mobile number then it shows us our board number or pilot number (i.e 4663000 means 00).. As i give all the extensions a particular DID, so people from outside world can call them. The problem is the CALLERID ... When we call from any of other extension PSTN line carries out our pilot number

Hi all, I have n no. of extensions in my dialer. from 456 to 556 extensions. I was created 2 other extensions 667 and 668 I need to allow only STD calls to go from this extensions. These all extensions are same context . I need to define the STD dialplan for only this 2 extensions. how I can ? Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI |

Hello List, I have found that asterisk supports only forwards in-dialog MESSAGE method. That is, if the MESSAGE method is sent within an active call. But according our requirement we need to send MESSAGE method to the other leg without being in a call (general stateless proxy forward). Is it possible to do this in asterisk using some tricks? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran

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

Hi , In vicidial dialer I need small Dialplan require. when i call from hardphone , in that has 1to9 no.s i want define the dipositions like when i press the 1 it will goes NotIntrest, press 2 for NotAvailable. How can i configure for this. -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75

Hi all, After running fine for a few months now asterisk seems to hang frequently , still functioning but the DAHDI channels seem busy (users report a busy signal when calling or being called) A reboot will allow it to run for another day or maybe 2 or 3 till the problem occurs again. running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel 2.6.32-5-686 i get the following

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

Hello list i have configured astersik 1.4 with sip i have a question when i put in dial plan.conf exten => _0678922645.,1,Set(CALLERID(number)=520460587) exten => _0678922645 .,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten => _0678922645 .,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)) exten => _067892264*5*,2,Hangup() i can not call my

Hi all, I have 100 DID's which is 4578900 to 4578999 , and i have 5001 to 5099 extensions. when incomming call come to this DID no. (4578901) that time 5001 extestinsion should ring. below my dial plan is not getting any result , inthat has any mistakes. please help. exten => _45789XX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _45789XX,1,Set(Dest=2{EXTEN:-2}) exten =>

Alex, Thank you so much for your response. I've been so consumed with other business that I only just now getting back to this issue. We have implemented your suggestion which is perfect. Thank you again. I've never asked a question of the community before and I'm extremely happy with the rapid response I received. Somewhat related to this initial problem I have an additional

i DID SOME VOICE MAIL CONFIGURATION. SO HOW CAN YOU RETRIVE THAT VOICEMAIL. WHEN I RETRIVE THE VOCIE MAIL ITS NOT GETTING ANY MSG Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax

Hi, I am using asterisk1.2 In this, my dialplan is going large , so i need to configure this small pieces for this, i did in my extensions.conf when I dial the 123 its not going , means that file is not reading. is there any parameters to add any where ? please tell me this #include is not working ... extensions.conf [general] [global] trunk=zap/g0 #include exten-internal.conf [default] exten

Hi, We have upgraded the openssh 4.6P1 on Solaris 8 servers. After upgrade we get the below error message whenever we execute the remote commands using ssh. Please let me know what the fix is for this. Apr 8 03:03:43 dvsrv10 sshd[25379]: [ID 800047 auth.info] Accepted publickey for osteam from 10.0.93.31 port 35856 ssh2 Apr 8 03:03:50 dvsrv10 sshd[25381]: [ID 800047 auth.error] error:

Hi sir, I was installed Goautodial server and I have b410p BRI card. BRI card showing OK with dahdi_tool, this NT mode. whenever I am dialing from server i am not able to connect the call . in Cli below mention warning is comming . please what is the mistake with me . help me Executing [0559566768 at default:1] AGI("Console/dsp", "agi:// 127.0.0.1:4577/call_log") in new

Hello, using 1.8.4. using a very simple local AGI script in bash which has only one line in it: echo -e 'STREAM FILE welcome 123 \n' dialplan: exten => 5150,1,Answer() same => n,Set(CHANNEL(language)=en_AU) same => n,AGI(testagi.sh) same => n,Hangup console output: -- Executing [5150 at AllPhones:1] Answer("SIP/PBX-00000024", "") in new stack

Dear All, We have a customer who is opening a new office in Dubai and we know that VoIP is blocked over there. Has anyone a solution to getting VoIP back out (we want interoffice calls back to the UK)? We we're thinking of IAX trunking, but not sure if that is blocked or just SIP etc. A VPN works, but is not great. We have seen: http://www.speed-voip.com/voiceguard.html At the moment it

I think we need this feature at least for debugging purpose, right? Or, do we already have this feature? Thanks, Neo -- I would remember that if researchers were not ambitious probably today we haven''t the technology we are using! _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com http://lists.xensource.com/xen-devel

Hello, I am using cups-1.1.19final-1.deb and samba-2.2.3a. When I was working with cups-1.1.15 before, cupsaddsmb worked fine, but now with cups-1.1.19 it gives me: Running command: smbclient //localhost/print\$ -N -U'root%secret?' -c 'mkdir WIN40;put /var/spool/cups/tmp/3eef829eea1b8 WIN40/ps948esb.PPD;put /usr/share/cups/drivers/ADFONTS.MFM WIN40/ADFONTS.MFM;put