search for: out_2

...-- Executing GotoIf("SIP/700-8d41", "1?6") in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing NoOp("SIP/700-8d41", "CallerID set to "2002-ATA" <700>") in new stack -- Executing SetGroup("SIP/700-8d41", "OUT_2") in new stack -- Executing CheckGroup("SIP/700-8d41", "1") in new stack -- Executing SetVar("SIP/700-8d41", "DIAL_NUMBER=9******") in new stack -- Executing SetVar("SIP/700-8d41", "DIAL_TRUNK=2") in new stack -- Execu...

...";, "1?7") in new stack -- Goto (macro-dialout-trunk,s,7) -- Executing GotoIf("IAX2/200@200/2";, "0?9") in new stack -- Executing SetCallerID("IAX2/200@200/2";, "0872596100") in new stack -- Executing SetGroup("IAX2/200@200/2";, "OUT_2") in new stack -- Executing CheckGroup("IAX2/200@200/2";, "") in new stack -- Executing SetVar("IAX2/200@200/2";, "DIAL_NUMBER=0639006374") in new stack -- Executing SetVar("IAX2/200@200/2";, "DIAL_TRUNK=2") in new stack -- Executing...

...canreinvite=no reinvite=no context=frombandwidth nat=no [bandwidth.com_outbound] host=216.82.224.202 port=5060 type=peer disallow=all allow=ulaw dtmfmode=rfc2833 nat=no fromuser=11234567890 extensions.conf [globals] ;?irrelevant stuff trunk_1 = Dahdi/g1 trunk_2 = SIP/trunk_2 OUT_2 = SIP/bandwidth.com_outbound ;Took out the Set(GROUP()) because I moved it elsewhere to try and fix it added all the phones when Asterisk calls agents on a Queue. [frombandwidth] ;exten = _+1.,1,Set(GROUP()=SIPGROUP) exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)}) exten = _+1.,n,Set(...

...ontext 'from-internal' May 26 07:35:23 NOTICE[2761]: pbx.c:1738 pbx_extension_helper: Cannot find extension context 'from-internal' my extension.conf is :- [globals] VM_PREFIX = * RINGTIMER = 15 REGTIME = 7:55-17:05 REGDAYS = mon-fri RECORDEXTEN = "" PARKNOTIFY = SIP/200 OUT_2 = IAX2/fwd OUT_1 = ZAP/g0 OUTPREFIX_2 = OUTMAXCHANS_2 = 1 OUTCID_2 = mithunafila<672648> OPERATOR = NULL = "" IN_OVERRIDE = forcereghours INCOMING = GRP-1 FAX_RX_EMAIL = fax@mydomain.com FAX_RX = system FAX = Eabhijit = SIP E9002 = SIP E9001 = SIP E8002 = SIP E8001 = SIP DIRECTORY_O...

...33:19 VERBOSE[3239] logger.c: -- Goto (macro-outbound-callerid,s,6) Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing NoOp("IAX2/206-3", "CallerID set to K2 Systems Inc.") in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing SetGroup("IAX2/206-3", "OUT_2") in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing CheckGroup("IAX2/206-3", "2") in new stack Feb 22 14:33:19 VERBOSE[3239] logger.c: -- Executing SetVar("IAX2/206-3", "DIAL_NUMBER=5149635279") in new stack Feb 22 14:33:19 VERBOSE[3239]...

I have installed asterisk @ home 2.6. I am using a Telasip VOIP account. When I make outbound or inbound calls the calls seem to connect and then get hung up. I was wondering if there was something that I am misisng. I have tried several different sip.conf configurations. Here is what they are currently. telasip-gw canreinvite=yes context=telasip-in dtmfmode=rfc2833 fromuser=jrasxxx

...re connected - they were disconnected when I ran this command) zap show channels: sterisk2*CLI> zap show ch channel channels asterisk2*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo from-pstn en /extension_additional.conf says: OUT_2 = ZAP/2 OUT_1 = ZAP/1 (changing these to Zap/1 or zap/1 makes no difference) Anyone have any thoughts on what I am missing? This must be something simple. Mark Buckaway mabuckaway@mac.com http://homepage.mac.com/mark.buckaway Blog: http://homepage.mac.com/mark.buckaway/blog --- Positive think...

I have a little confusion about the general settings (other than the register values) in the SIP General area. I understand that for examle in a SIP context like [FWD] or [BROADVOICE] the entries in those areas are ths settings that take effect in any communication woth FWD and/or BROADVOICE. However, I'm confused as to the purpose of the "general" settings -- to what or which

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

...IP/2022-083c53f0", "DIAL_NUMBER=0012127773456") in new stack -- Executing [s at macro-dialout-trunk:5] Set("SIP/2022-083c53f0", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing [s at macro-dialout-trunk:6] Set("SIP/2022-083c53f0", "GROUP()=OUT_2") in new stack -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/2022-083c53f0", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/2022-083c53f0", "0?skipoutcid") in new stack...

...> 666,1,Goto(ext-fax,in_fax,1) exten => h,1,Macro(hangupcall) ---------------------------------------------------------------------------- -------------- Extensions_additional.conf [globals] RINGTIMER = 15 REGTIME = 7:55-17:05 REGDAYS = mon-fri RECORDEXTEN = "" PARKNOTIFY = SIP/200 OUT_2 = ZAP/1 OUT = ${OUT_2} OPERATOR = NULL = "" IN_OVERRIDE = none INCOMING = EXT-202 FAX_RX_EMAIL = fax@mydomain.com FAX_RX = system FAX = E230 = IAX2 E205 = SIP E204 = SIP E203 = SIP E202 = SIP E201 = SIP DIRECTORY = last DIAL_OUT_2 = 9 DIAL_OUT = ${DIAL_OUT_2} DIAL_OPTIONS = tr DIALOUTIDS...

...ack -- Executing Set("SIP/200-09fc1698", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing GotoIf("SIP/200-09fc1698", "1?noauth") in new stack -- Goto (macro-dialout-trunk,s,8) -- Executing Set("SIP/200-09fc1698", "GROUP()=OUT_2") in new stack -- Executing Macro("SIP/200-09fc1698", "user-callerid|SKIPTTL") in new stack -- Executing NoOp("SIP/200-09fc1698", "user-callerid: device 200") in new stack -- Executing GotoIf("SIP/200-09fc1698", "0?report&...

Hi this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I often record radio shows for posterity, and sometimes I have friends who would like to listen to them live over the net. When I am recording for my own purposes, I use a command of the form: "brec | oggenc", (options omitted for clarity) and I send the output to a file, call it radio.ogg. Locally, I use Debian stable. If I want to listen to the show as I record it, I can

...09:34:31] VERBOSE[12649] pbx.c: -- Executing [s at macro-dialout-trunk:5] Set("SIP/21-00000058", "DIAL_TRUNK_OPTIONS=tr") in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [s at macro-dialout-trunk:6] Set("SIP/21-00000058", "OUTBOUND_GROUP=OUT_2") in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/21-00000058", "1?nomax") in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Goto (macro-dialout-trunk,s,9) [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -...

...:5] Set("SIP/611-b7b9ae38", "DIAL_TRUNK_OPTIONS=tr") in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: Set [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [s at macro-dialout-trunk:6] Set("SIP/611-b7b9ae38", "OUTBOUND_GROUP=OUT_2") in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: Set [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/611-b7b9ae38", "1?nomax") in new stack [Jun 21 08:53:29] VERBOSE[21559] logger.c: --...

...new stack -- Executing Set("IAX2/4427-1", "CALLERID(all)=<8xx-6xx-xxxx>") in new stack -- Executing NoOp("IAX2/4427-1", "CallerID set to "" <8006959897>") in new stack -- Executing Set("IAX2/4427-1", "GROUP()=OUT_2") in new stack -- Executing GotoIf("IAX2/4427-1", "0?108") in new stack -- Executing Set("IAX2/4427-1", "DIAL_NUMBER=6167XXXXX") in new stack -- Executing Set("IAX2/4427-1", "DIAL_TRUNK=2") in new stack -- Executing AG...