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We are using vpn routers to connect home users back to our office network. Basically, shipping a mikrotik router that 'calls home' and establishes a vpn connection for the pc and phone that are connected to the mikrotik... user plugs router in, plugs phone and computer into router, and that traffic is encapsulated back to our office... simple and straighforward. We would like to remove

We, along with a lot of other people, have a phone number that is pretty important to us. Yesterday, our VoIP provider went down... won't call any names VI, but it was pretty bad... Our goal is to create a script within asterisk, that will place a call out one SIP trunk provider (not the one that provides the DID, and have the call come back in on another trunking provider (with a

We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header

Ladies, Gentlemen We are experiencing an unusual problem in our asterisk 1.4.34.. We are attempting to determine if channels are in use before paging to them. This works correctly, as in it pages the phone.. however, we see the error message below on the console... after googling, we discovered limited information regarding the issue... -- Executing [NPANXX7298 at from-pstn:1]

Greetings!! We're getting some strange results using call files.. no matter the technology, DAHDI, SIP, etc., we get a "Call failed to go through, reason (3) Remote end Ringing" message when attempting to originate a call from a call file. Numbers changed to protect the innocent.... using call file.... //------------CALL FILE------------// Channel: DAHDI/g1/918005551212

Good day.. We have what I consider to be a large dialplan (-= 1501 extensions (2559 priorities) in 99 contexts. =-) If we have more than 10 or so channels up (all SIP, no TDM) and issue the "extensions reload" command.. quite often, asterisk will completely freeze up... requiring us to either kill and restart the process or restart the box... I should probably also share that when

In theory ParkAndAnnounce has a lot of usefulness, however, that we've had very little success with application... Our application is similiar to the local Walgreens pharmacy.. Dr. Calls in, selects the "Im a doctor with a prescription option"... call is parked, and announcement overhead is given.. "Doctor holding on ${EXTEN}" exten =>

Has anyone found a way to detect a loop condition in the dialplan.?? We had a condition where this filled up 47 PRI channels in an NFAS group connected to our media gateway... and endless loop if you will.. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/b6bbf116/attachment.htm

Has anyone successfully made this scenario work in 1.4. I found info at http://www.voip-info.org/wiki/view/Asterisk+presence indicating that this does not work with 1.4 implementations. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100903/13ed20ba/attachment.htm

We have a scenario where we wish to present a toll-free caller id, yet have our calls rated based on our billing-telephone-number. Is it possible to present a number in the sip header for billing and another number in the header for jurisdicional call rating? Whereas today, all of our calls are billed at the highest rate (intra-state) because we're presenting a number that isn't in the

In a multi-tenant environment, we are sending various CallerIDs outbound from asterisk based on who the user is. We have an insurance agency who would like to present a toll free callerid. This works.. unless they're calling a toll free number. In that case, occasionally, the call fails. However, should we send a correctly formatted npanxx of a local number, the call completes. We have

Greetings, I'm hoping that an extra pair of eyes might help me to solve a challenge... Anyone have any idea why the following would not work? I'm trying to test for a callerid value that is 4 digits in length.. exten => s,1,NoOp(CLID is ${CALLERID(all)}) exten => s,n,Gotoif($[${LEN(${CALLERID(number)})} != 4]?true) exten => s,n,NoOp(Value is False) exten => s,n,Hangup

Has anyone had success patching chan_sip.c so that Asterisk will recognize the tel: header for RDNIS information? exten = get_in_brackets(tmp); if (!strncasecmp(exten, "sip:", 4)) { exten += 4; } else if (!strncasecmp(exten, "sips:", 5)) { exten += 5; } else { ast_log(LOG_WARNING, "Huh? Not an

In followme , is it be possible to have a third option.... Whereas, takecall=>1 declinecall=>2 proposed option transfercall=>3 or, transferring the call directly from followme isn't really neccessary, if the callee could answer the call, and transfer it someplace, that would work as well.... -------------- next part -------------- An HTML attachment was scrubbed... URL:

We're experiencing an issue where calls disconnect after 15 minutes. It seems to happen just after Asterisk sends an update mesage. RTP is being set up directly. Asterisk is only in the SIP dialog. Has anyone experienced this issue? 4 PRIs inbound, 4 PRIs outbound, asterisk provides switching. SIP/2.0 200 OK Via: SIP/2.0/UDP 38.XXX.XXX.XXX:5060;branch=z9hG4bK1c4b524f From: