search for: x1110

...9; -- Hungup 'Zap/99-1' It looks to me as if the Gotoif thinks that [0+1] is greater than or equal to 3 and therefore jumps to hangup. Am I missing something here? -- Steve Woolley IT Manager ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, Florida 32746 Phone: (407)682-6226 x1110 Fax: (407)682-3455 Cell: (321)229-5311 swoolley@adstelecom.com www.adstelecom.com

...built for i386-linux-thread-multi <= I installed ithread support in perl Have not been able to find much on PL_thr_key && undefined symbol on net search Any ideas? -- Steve Woolley IT Manager ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, Florida 32746 Phone: (407)682-6226 x1110 Fax: (407)682-3455 Cell: (321)229-5311 swoolley@adstelecom.com www.adstelecom.com

...ten to the content. There maybe bandwidth issues but this would certainly be an interesting proof of concept. I personally am planning on attending, but I know others may not. -- Steve Woolley IT Manager ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, Florida 32746 Phone: (407)682-6226 x1110 Fax: (407)682-3455 IAXtel: (700)682-6226 x6543 Cell: (321)229-5311 swoolley@adstelecom.com www.adstelecom.com

...de on the Snom's, but this does not seem to be an option under SIP mode. Anyone had any luck using the programmable keys for anything but transfering/calling sip url's? -- Steve Woolley IT Manager ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, Florida 32746 Phone: (407)682-6226 x1110 Fax: (407)682-3455 Cell: (321)229-5311 swoolley@adstelecom.com www.adstelecom.com

...r.com ; Register with a SIP provider ;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as 1234 here. ; But I am a little confused how this should be implemented. -- Steve Woolley ADS Telecom, Inc. 59 Skyline Drive #1250 Lake Mary, FL 32746 Phone: (407) 682-6226 x1110 Fax: (407) 682-3455 swoolley@adstelecom.com

...9 When asterisk is starting up so I belive I am registered. Can I simply not dial my own 700 number from the same asterisk PBX as a test or do I have some real problem? -- Steve Woolley IT Manager ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, Florida 32746 Phone: (407)682-6226 x1110 Fax: (407)682-3455 Cell: (321)229-5311 swoolley@adstelecom.com www.adstelecom.com

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

Friends, Please send all offers for help *off list* to us at info@astricon.net. Do not disturb the list with offers of your services, please. I repeat: Only the Developer's Meeting will be considered for broadcast at this time. In order to enjoy the conference, you will simply have to be there. It's an IRL experience - meeting all the other Asterisk user's from around the globe,

In PHP something like: $pid = pcntl_fork(); if ($pid != 0) { // we are the parent // do parent stuff exit; } // we are the child, detatch from terminal $sid = posix_setsid(); if ($sid < 0) { die; } // do child stuff On 04/19/2019 02:00 PM, Mark Wiater wrote: > On 4/19/2019 1:49 PM, Dovid Bender wrote: >> Mark, >> >> I am using PHP agi and when forking

We have a customer where their VM running Asterisk appears to have crashed. Fortunately, we had some debugging enabled. The asterisk messages file has this... (in notepad+ the blank line in the middle is all [NUL][NUL] [NUL][NUL]....) [08/12 15:30:55.880] VERBOSE[6920] app_mixmonitor.c: Begin MixMonitor Recording CBRec/IS__a37ae004-c780-4c7f-88a9-a04402f0ab4e-0000e70f [08/12 15:30:55.881]

...alled but is needed. However, I can find no complaints on the asterisk console or messages leading me closer to a solution -- even with maximum debugging turned on. Any ideas? -- Steve Woolley IT Manager ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, Florida 32746 Phone: (407)682-6226 x1110 Fax: (407)682-3455 Cell: (321)229-5311 swoolley@adstelecom.com www.adstelecom.com

I have a test SIP account set up with WorldCom and I have been trying to have Asterisk register to the WorldCom server with no luck. It appears that the SIP headers are different coming from Asterisk. I have included a packet capture from a successful login with a Windows Messenger client for reference. I have also copied in the SIP packet I captured with sip debug turned on. In my sip.conf file,

Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio On Thu, Oct 29, 2020 at 20:44 David Cunningham <dcunningham at voisonics.com> wrote: > Hello, > > Does anyone know a way with chan_sip to tell Asterisk to use a specific IP > address for its end of the communication for a specific

...AELSub(pngnpbx-abc-gps,39.739,-104.990,5280ft,${gpsurl}) same = n,AELSub(pngnpbx-abc-sa,144 W Colfax Ave,Denver,CO,80202) same = n,AELSub(pngnpbx-abc-ai,Picnic Shelter 4) same = n,AELSub(pngnpbx-abc-remote,3035557729,static) same = n,Goto(friends) ; Park Gazebo for Software Engineers same = n(x1110),NoOp() same = n(x1104),NoOp() same = n(x1113),NoOp() same = n(x1114),NoOp() same = n,AELSub(pngnpbx-abc-gps,39.739,-104.990,5280ft,${gpsurl}) same = n,AELSub(pngnpbx-abc-sa,144 W Colfax Ave,Denver,CO,80202) same = n,AELSub(pngnpbx-abc-ai,Gazebo) same = n,AELSub(pngnpbx-abc-remote,3035557729...