similar to: Can I start recording channel in the middle ofconversation ?

As far as I can remember I only opened sip and tftp ports for the phone. For some reason (didn't look into it too much) the call stays with sip and doesn't use RTP. The problem you describe (the call doesn't even ring on the other side) is something I had and was solved by upgrading the firmware. Checkpoint's tracker explicitly said what connection attempts were blocked and why.

print to standard error output in your perl script: print STDERR "This is how perl-AGI prints to Asterisk CLI output\n"; MATT--- -----Original Message----- From: Robert Rozman [mailto:rozman@fri.uni-lj.si] Sent: Tuesday, August 24, 2004 8:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Perl AGI - no output from agi script to Asterisk Hi,

> Date: Thu, 23 Jun 2005 08:50:50 +0200 > From: "Robert Rozman" <rozman@fri.uni-lj.si> > Subject: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- * > - Euroisdn Italy > > I'm pulling my hair down and getting bold :-) ..... I have Asterisk between > Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff > Asterisk).... Plenty of

Check your FW-1 tracker and see if any sip packets are dropped during call initiation. I had this problem and it went away when I upgraded the BT's firmware to the latest (16). Beware, though, that people on the list claim that this firmware breaks functionality of the message button and autoanswer. I haven't checked this yet, cause I can't afford to go back a version. I prefer a

Hi, I'd kindly ask for simple example if this is possible ? Is any key press encountered during conversation and action taken in dialplan ? Thanks, regards, Rob.

>-----Original Message----- >From: Larry Shields [mailto:LJ.Shields@Verizon.net] >Sent: Friday, August 27, 2004 12:20 PM >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe() >If I assign the DID to ring extension SIP/2000 and then after time-out send >it to MeetMe() or Playback() it works and the caller

> -----Original Message----- > From: Dale [mailto:dale-list@lightwavetech.com] > Sent: Tuesday, January 18, 2005 1:45 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] QoS tagging - can Asterisk do this, > if not,what do you recommend? > > So my question is, does Asterisk offer the > ability to mark the voice data with the

Hi, we're testing Asterisk 1 RC 2 behind ordinary router and NAT. Since we're sharing network with web server it seems like voip packets are not coming through fast enough (Digium demo dies after few seconds...). It's the same if I make direct calls (passing Asterisk) so we conclude it's network problem - it also work normally outside our router... I wonder what solutions can we

Alright, folks. I just deployed * into full production at my office. We have around 50 7905's, 5 7940's, and a handful of soft clients. We run a call center with around 15 agents. I also have a queue set up for the receptionists so that they don't get bombarded with calls. Everything seems to be working with a very few minor glitches. I firmly believe that the few problems we are

Hi, thank you for last info. we've tried to use te110p but failed. We're quite surprised that cable wasn't included with the card as any documentation, at least on HW setup and installation, yet cable pinout for connection to PRI interfaces.... 1. We have followed instructions on your site and from Beronet guide, but card just keeps blinking and nothing happens (also no useful

Hi, I have following problem. Asterisk is connected to ISDN router on BRI interface. ISDN PBX is connected to another channel of BRI interface. Now I'd like to route all incoming calls first to Asterisk and then if caller wants to talk to extension on ISDN PBX then I'd like to route call to another capi channel but free the current one. Is this possible at all or do I need to take 2 capi

Hi, I'd like to organize my Asterisk to properly handle incoming SIP/IAX/H323 callerids so they can be called back if needed. I have three incoming contexts for sip, iax and h323 calls. To each incoming call I'd like to prepend certain number that will be catched with pattern matching on output calls. For instance for iax I have: [from-iax] exten => s,1,NoOp(IAX call from outside

I think you will find the functionality you are looking for is in * already. Here is an excerpt from the sample extensions.conf file that is included with the source: exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten

[Cross-posted to the asterisk forums, so if it seems familiar...] My company is looking to implement 100% recording of inbound and outbound calls. We're pondering using Asterisk as a trunk-side monitoring system sitting between our Avaya Definity switch and our mixed bag of 6 T1 and PRI circuits. I've read with interest Matt Roth's thread on the Asterisk mailing list, however, since

Sirs, I have recently been interested in your Maxstat. I have computed with my own programme the ranks (by using the Kaplan-Meier method and the log-rank test) with the formula (Observed-Expected)/(SQR Var). The results are similar but not exact to the M value obtained with the Maxstat. I would like to know whether you are using some correction or adjustment in computing the different ranks. Thank

For the last few days I can not resolve voip-info.org from many DNS servers. It does resolve with some DNS servers but I suspect it may be related more to caching. Using the host command: host -a voip-info.org 130.179.16.23 Trying "voip-info.org" Using domain server: Name: 130.179.16.23 Address: 130.179.16.23#53 Aliases: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR,

Does Asterisk need a sound card or functional Console/dsp to answer inbound DID number from PRI and playback .gsm files? I can call from any of the SIP extensions on Asterisk and hear audio from Playback(), MeetMe(), or MOH. The problem I am having with calls from my PRI is as follows: I have an Asterisk (CVS-HEAD-08/25/04-20:28:51) currently interfacing a NEAX 2400 IPX with PRI. I have a

Hello, I believe these descriptors are specific to flang, not to LLVM. You should probably ask your question on flang-dev list. Thank you, --Eugene From: llvm-dev [mailto:llvm-dev-bounces at lists.llvm.org] On Behalf Of Venkataramanan Kumar via llvm-dev Sent: Wednesday, April 25, 2018 8:44 AM To: llvm-dev at lists.llvm.org Subject: [llvm-dev] Help on understanding assume shape array

Hi, I am trying to understand how assume shaped arrays are received and processed in LLVM IR. I am using "flang" for my front end. There seems to be an array descriptor received as implicit argument for every assume shaped array. For my test routine: ---snip-- SUBROUTINE test(a,b,Li,Lj,Istr,Iend,Jstr,Jend) INTEGER, INTENT(IN) :: Li,Lj INTEGER, INTENT(IN) :: Istr, Iend,

Duncan, you are a king! Thanks a lot for this cookie. It really helped me. Thanks for the code as well as detailed explanation at the end. >Hi Gregor. > >Here is a function that will collect all of the nodes in the >XML document whose names are in the vector elementNames > >getElements = >function(elementNames) >{ > els = list() > > startElement = function(node,