similar to: What has happened to my asterisk/PRI ?

Hi, I evaluated asttapi, it fits our needs to dial with asterisk out of windows applications. But we are working with rdesktop on a windows 2000 terminal-server, 3 Users on 1 Machine Whenever a user Sets up asttapi via outlook the configuration changes for all users. For example when I enter sip/340 as my extension, the other users also have sip/340 in their configuration. Do you know a trick

Hi There, I have an asterisk an a Digium 4 Port E1 Card On E1 Port No. 1 I have the Telekom PRI On E1 Port No. 2 I have an Alcatel PBX that cannot be changed So I have setup my asterisk between Alcatel and Telekom In extension.conf i configured: [telekom] exten => _9149.,1,Dial,ZAP/g2/${EXTEN}; exten => _9149.,2,Hangup This works great, all incoming calls are directly routed to alcatel

Hi there, I want to add 4000 Callerids and Callernames to my asterisk-db. (/var/lib/asterisk/astdb) I do not want an external database or an sql-database because I do not want asterisk to depend on external processes. However, when I do 4000 "database put number name" via a shellscript and "asterisk -rx" I only have 600 entries later in my asterisk- database. The asterisk

Hi there, i have a Problem with dialtone and a TE401P Card. I swear I surfed the wiki, the mailing list and google for 4 hours and did not find the solution, can you help me ? In Germany I have an E1-Line and an Alcatel 4200 PRO PBX. Without using asterisk I dial the "0" on an Alcatel Phone and have the local exchange dialtone, then I can dial. Most users do not dial en block, they

I've got an interesting scenario where transfers while getting an invite seem to break. Here is the scenario: You have a receptionist who has a 6 line phone (in this case, a polycom ip600 - also tested with a Cisco 7960) the receptionist has all six lines available for use (in the case of the cisco I tried registering all lines as one number as well as registering multiple lines and

Hello! My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on Fedora Core 4. I am making automatic call-out campaign with this setup on 4 PRI. The scripts for this: ==================================================================== caller php script write this to outgoung folder: fwrite($outfile,"Channel: Zap/g1/$phonenumber\n"); fwrite($outfile,"MaxRetries:

Good afternoon gentlemen (and ladies). A costumer of mine has many servers and each one maps their SIP extensions to the others via DUNDi. It works like a charm. SIP extensions can only register at one server, the one they "belong" to. In case one extension wants to call other that is registered in another server, DUNDi takes care of that by calling the other server using IAX2 and G.729

hi is there any way to build 4.8 release with this fstack protection? or atleast some ports is there any good info on this? the only page i found was that ibm page but it seemed outdated. //martin

I use the following macro for my extensions. It only works with Zap channels and assumes that any Caller*ID number that is 4 digits is an internal call and all other calls are external calls. Use like this: exten => 1234,1,Macro(std-exten,Zap/4,20) [macro-std-exten] ; ; Caller*ID is 4 digits (internal call) ; exten => s/_XXXX,1,Dial(${ARG1}r2,${ARG2}) exten =>

Hello! I have following connections over isdn pri: avaya definity---pri--asterisk--pri-panasonic 500 Just because panasonic 500 can't send user's names. I also want to have reverse callerid for avaya users. But if there is no answer in dial plan: exten => _XXXX,1,Set(CONNECTEDLINE(name)=${DB(names/${EXTEN})}) ;exten => _XXXX,n,Answer exten => _XXXX,n,Dial(DAHDI/g4/${EXTEN})

Auto Callback on Busy Register on Busy I have implemented it as 1- I store Caller and Called party numbers in database when Called part is busy 2- I retrieve it from database and Caller is called by called party when Called party hangs up It is working fine with all kind of SIP phones I have with me basic configuration for extensions.conf is given and can be accommodated according to

2008/9/29 Alex Balashov <abalashov at evaristesys.com> > Try this: > > exten => _XXXX,1,Set(THISTECH=${CUT(CHANNEL,/,1)}) > exten => _XXXX,n,NoOp(Technology is ${THISTECH}) > exten => _XXXX,n,Set(THISCHANNEL=${CUT(CHANNEL,/,2)}) > exten => _XXXX,n,NoOp(Channel is ${THISCHANNEL}) Hi, I don't have any spare zaptel enabled system I could try this on, but I

> I'm having an issue deploying softphones into my DUNDi/regcontext > setup. My current design is that all SIP users get registered into a > sipregistration context in the sip.conf. I then have a dialplan > function that includes that and does the dial: > > include => sipregistration > exten => _XXXX,2,Answer() > exten => _XXXX,3,Wait(1) > exten =>

hi folks. i'm experimenting with iaxmodem + hylafax using DID to determine where to send the fax to it's final destination. however i have difficulties passing the DID information from iaxmodem to hylafax. in extensions.conf: exten => _XXXX,1,Dial(IAX2/iaxmodem0/${EXTEN}|20|r) exten => _XXXX,n,Dial(IAX2/iaxmodem1/${EXTEN}|20|r) exten => _XXXX,n,Busy exten => _XXXX,n,Hangup

I've lurked for a while, but I think this is one of my first "pleas" for help. I'm having issues where a parked call using the macro below is getting "stuck". Users park the call via a blfxfer key on an Aastra phone. If the call is a blind transfer, it tries to park the call. If it isn't a blind transfer, it tries to unpark the call. Only 2 extensions (2759 and

Hi there, I have a customer with domain a at tld and domain b at tld. b.tld is an alias for a.tld The Server is using virtual users based on mysql, dovecot for delivery and Maildir for Mail-Storage. If someone writes an e-mail to a.tld with a cc: to b.tld the customer receives the E-Mail twice. Is there a way to deliver the same Mail in the same Mailbox only once ? -- Christoph Adomeit

Hello, I would like to known how to use DUNDi with a Lua dialplan ? In extensions.conf, we should do like these: |[lookupdundi] switch => DUNDi/priv [internal] include => dundiextens include => lookupdundi exten => _XXXX,2,NoOp(calling ${EXTEN}) exten => _XXXX,n,Dial(SIP/${EXTEN}) exten => _XXXX,n,Hangup()| priority 1 is either defined in dundiextens (local registered

I'm trying to get this zap dial to work. I want to send DNIS and ANI to other system (ZAP/g2) at answer, while the caller hears ring (RBT). I hear the RBT, but callerid and exten is not sent to other T1 - The ZAP/g2 T1 is standard D4, SF, E&M Wink Start. - At ZAP/g2 wink, asterisk should send DTMF "*ANI*DNIS*" exten => _XXXX,1,NoOp,${CALLERID} exten =>

All, I'm having an issue deploying softphones into my DUNDi/regcontext setup. My current design is that all SIP users get registered into a sipregistration context in the sip.conf. I then have a dialplan function that includes that and does the dial: include => sipregistration exten => _XXXX,2,Answer() exten => _XXXX,3,Wait(1) exten => _XXXX,4,NoOp(sipregistration call - Name:

Hi, Running 1.4.15. I've a SIP user as below. My default context in sip.conf is [incomming_pstn] I'm having trouble with inbound calls going to the wrong context. [test-ubi] username=test-ubi type=friend secret=XXXXXXX host=dynamic canreinvite=no context=testinbound nat=yes allow=ulaw allow=gsm allow=alaw qualify=no the testinbound context includes the code to