similar to: sip:EXTEN;phone-context in asterisk dial plan

Hi,all I have a POP3 problem about multi users from one IP address concurrent. [Issue description] The network architecture is as following. I have setup a mail server(sendmail/dovecot) in PlaceA with IP_A. And about 6 users are receving mail from the same IP_B( NATed public IP) in PlaceB. Their mails are many( about 1Mb/hour each) and ususally set fetch time as 10min/fetch. Then they

I''ve tried with iproute and it works now ! commands for A: ip tunnel add mode ipip tunl1 local IP_A remote IP_B ip addr add dev net1 local virtual-IP_A remote virtual-IP_B ip link set net1 up Same on B. And then from A i''m able to ping virtual-IP_B. It seems one can not use tunl0 as a tunnel name, is it true ? I think it will be a good thing to add to the section 5.2 of the

Hi Everyone, I am running multiple instances of Asterisk in Proxmox and so far I had one central Asterisk feeding all others with trunks from one provider. Now, I want to connect each Asterisk server directly to the provider. Based on my understanding, each connection made to the provider port 5060 would be on a port that is unique to that server. And so other connections made to the same

Hi, I have setup a dovecot pop3 server on IP_A and using mbox format I have 6 users behind a NATed network, the public IP is IP_B. And one mail of one user seems contain format error. He always cannot receive the 5th mail from anywhere. Then when he receive mail in the intranet, all the TCP/IP connection REST between IP_A and IP_B. http/ssh/pop/smtp etc. all NG. After I delete the 5th mail

Hello, I am working on a automation project, which uses ssh for connection to the server. once the ssh prompt is displayed there is a ' ' space by default between the ssh username and the prompt. This space is causing my automation script to stop working. Can I get some pointers in verification and modification of the ssh prompt after providing username & password. Thanks alot in

Hi all. I experienced a strange behaviour in the "arp" command when I use the "pub" (publishing) flag. This is the scenario (I know it''s an unusual one, but it''s just for a test): __________ | | | host C | | | |__________| | __________|________________________________

Greetings all- I have a handful of Asterisk 1.4.x installations where users dial 'outbound calls' to the PSTN even though the destination is on the same Asterisk box or on another Asterisk box on the same network. Instead of paying twice for the call to go out to the PSTN on one channel and back in on another channel, I'd like the ability to lookup the destination number in a MySQL

On Sat, Mar 11, 2017, at 11:50 AM, Kseniya Blashchuk wrote: > Hey guys, any thoughts on that? Probably a bug or is it a default > behavior? I'd suggest providing the configuration to make sure it is correct. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org

Hi, I have been out of touch with asterisk for quit some time and needed some recommendations. I am looking for SIP hardphone that works well with asterisk server. Pls suggest. cheers /ag -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110212/ccd9d985/attachment.htm>

p 176 has exten => 1NXXNXXXXXXX,1,Dial(SIP/${EXTEN}@myprovider) how is "exten" distinct from "EXTEN"? What is this line of code doing? https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables says that EXTEN is the current extension. In ruby, you this: H = Hash["a" => 100, "b" => 200] The => is a mapping, or at least

Hi All; Why we use the agent password when we configure the agent in the agents.conf if the agent login by dialing the number configured in the extensions.conf? example: exten => 28, 1, AgentLogin(1001) I know that agent username is used to assign the agent to the queue, but when we use the agent password? Regards Bilal

Hello List! I have 2 asterisk servers, The Admin(.20), and the Call Center(.21). The Admin server contains the 1XXX extension and the Call Center hosts the 2XXX extensions. I would like for our Admin folks to be able to call the Call Center folks (and vice versa). The call will go over the server fine, but when the Call Center server answer, the CLI returns: "NOTICE[4296]: chan_iax2.c:7398

Hi, I have VoiceMailMain on extention 1001 so it would be nice to get that sent to the phone instead of asterisk@[IP Address] when there is a message waiting.. Is it possible to change the extention that is sent to the phone when MWI is lit up?? Thanks.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by

Hi friends, I have the entry exten => h,Hangup in my extensions.conf, and I am trying to record the call details for billing. From the wiki i found out that the use of "exten=>h,..." is not suggested for the CDRs. What impact will the use of 'h' make on CDRs? Also, what is the advantage of using ResetCDR with exten=>h? Regards... Girish

Hi to all, I have configured in my * a avm fritz card and in my isdn connection I have only 1 extensions. I read from documentation tht is possible to use the exten,fax to redirect an incoming fax. Is it possible also with avm card or only with digium card. I used in this way but with no success: exten => fax,1,Goto(fax,2202,1) 'Configured for spandsp exten =>

Ok so here's one Has anyone managed to get the 'i' extension to work. I have included within each context the following exten => i,1,Goto(wrong-number,s,1) then in [wrong-number] exten => s,1,GotoIf($[${EXTEN:0:2} = 43}]?10:2) exten => s,2,GotoIf($[${EXTEN:0:2} = 62}]?11:99) exten => s,10,Goto(main-office,${EXTEN},1) exten => s,11,Goto(remote-office,${EXTEN},1) exten

Hi, I've got a strange behavior on one of my new extentions. This is what I got: extensions.conf with to #includes: extensions-manual.conf & extensions-db.conf I have around 50 users and 100 contexts in these files. I reacently included one: (created from db) [marcela] include => Common include => VoiceMail include => marcela-extensions [marcela-extensions] ;; =======

For make outgoing call, i setup 0. However 0 is write in the cdr dst field. Is there a way to remove it when asterisk send it to cdr_mysql ? exten => _0X.,1,Dial,SIP/${EXTEN:1}@mygateway I just want have in cdr dst = ${EXTEN:1} This don't work : exten => _0X.,1,SetVar(EXTEN=${EXTEN:1}) exten => _0X.,2,Dial,SIP/${EXTEN}@mygateway Use another variable still record ${EXTEN} --

How I can retrieve the original ${EXTEN} value when falling into the exten => i,1,whatever "context"?? I'm trying to implement this extension rule: exten => i,1,NoOp(${EXTEN}) exten => i,2,Wait(1) exten => i,3,Playback(vm-extension) exten => i,4,SayDigits(${EXTEN}) exten => i,5,Playback(is-invalid) exten => i,6,Playback(pls-try-again) exten => i,7,Goto(s,12)

Hi, I tried set up a global var for an extension, like this [globals] IPPHONES=_3XX [sip] exten=>${IPPHONES},1,Answer What I would like to do is to make a dialplan without fixed extension numbers to change the entire dialplan according to the customer requests: what exten number do you want for your IP Phones ? let me change a variable and we are set! It seems that this is not supported,