search for: nomax

Displaying 19 results from an estimated 19 matches for "nomax".

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2007 May 17
2
Quadbri Cellular Issue
...P/200-09fc1698", "1?report") in new stack -- Goto (macro-outbound-callerid,s,22) -- Executing NoOp("SIP/200-09fc1698", "CallerID set to "CITY LOGO" <965245915>") in new stack -- Executing GotoIf("SIP/200-09fc1698", "0?nomax") in new stack -- Executing GotoIf("SIP/200-09fc1698", "0?chanfull") in new stack -- Executing DeadAGI("SIP/200-09fc1698", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fi...
2008 Jan 15
0
busy/congestion random
...("SIP/206-090a7dd8", "1?report") in new stack -- Goto (macro-outbound-callerid,s,22) -- Executing NoOp("SIP/206-090a7dd8", "CallerID set to "Centralino" <206>") in new stack -- Executing GotoIf("SIP/206-090a7dd8", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,17) -- Executing AGI("SIP/206-090a7dd8", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set(&qu...
2009 May 08
2
Configuring SIP 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 -- Executing [s at macro-dialout-trunk:10] Set("SIP/2022-083c53f0", "DIAL_TRUNK_OPTIONS=") in...
2009 Oct 09
0
calls ansowered for 1 second or less
..."SIP/100-b609f9c0", "DIAL_TRUNK_OPTIONS=trf") in new stack -- Executing [s at macro-dialout-trunk:6] Set("SIP/100-b609f9c0", "OUTBOUND_GROUP=OUT_12") in new stack -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/100-b609f9c0", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/100-b609f9c0", "0?skipoutcid") in new stack -- Executing [s at macro-dialout-trunk:10] Set("SIP/100-b609f9c0", "DIAL_TRUNK_OPTIONS=") in...
2010 Mar 26
1
SIP/2.0 403 Forbidden
...t;, > "DIAL_TRUNK_OPTIONS=Ttr") in new stack > -- Executing [s at macro-dialout-trunk:6] Set("SIP/75002-b7705298", > "OUTBOUND_GROUP=OUT_7") in new stack > -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/75002-b7705298", > "1?nomax") in new stack > -- Goto (macro-dialout-trunk,s,9) > -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/75002-b7705298", > "0?skipoutcid") in new stack > -- Executing [s at macro-dialout-trunk:10] Set("SIP/75002-b7705298", > "DIA...
2010 Mar 26
1
send a call from A to B use sip trunk prablem
...IAL_TRUNK_OPTIONS=Ttr") in new stack > -- Executing [s at macro-dialout-trunk:6] > Set("SIP/192.168.0.151-088e7938", "OUTBOUND_GROUP=OUT_1") in new stack > -- Executing [s at macro-dialout-trunk:7] > GotoIf("SIP/192.168.0.151-088e7938", "1?nomax") in new stack > -- Goto (macro-dialout-trunk,s,9) > -- Executing [s at macro-dialout-trunk:9] > GotoIf("SIP/192.168.0.151-088e7938", "0?skipoutcid") in new stack > -- Executing [s at macro-dialout-trunk:10] > Set("SIP/192.168.0.151-088e7938&...
2010 Nov 03
1
Ring back problem on SIP calls. Order of 183 Session Progress and 180 Ringing
Hello everyone! I've had this problem for a while and cant figure it out. When an outside caller calls an extension on my asterisk system, they do not hear any sort of ringing. Inside extensions calling other extensions do hear ringing. We have 3 other asterisk systems that are configured the same way, but do not have this problem. We think it has something to do with asterisk 1.6. The other
2013 Feb 16
1
Dial failed due to trunk reporting BUSY - giving up
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:
2010 May 05
0
T38 trunk configuration for relay appears to affect default trunks for voip
...ack [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: -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/21-00000058", "0?skipoutcid") in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c:...
2015 Mar 20
3
outbound calls
...Set("SIP/101-00000103", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing [s at macro-dialout-trunk:6] Set("SIP/101-00000103", "OUTBOUND_GROUP=OUT_5") in new stack -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/101-00000103", "0?nomax") in new stack -- Executing [s at macro-dialout-trunk:8] GotoIf("SIP/101-00000103", "0?chanfull") in new stack -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/101-00000103", "0?skipoutcid") in new stack -- Executing [s at macro-dialout-...
2010 Jun 21
1
How to tell if a dropped call is my fault
...alout-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: -- Goto (macro-dialout-trunk,s,9) [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: GotoIf [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/611-b7b9ae38",...
2011 Sep 28
2
PSTN connectivity
Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before
2015 Mar 27
0
call between snom 300 and aastra 6731i
...Set("SIP/300-00000192", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing [s at macro-dialout-trunk:6] Set("SIP/300-00000192", "OUTBOUND_GROUP=OUT_5") in new stack -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/300-00000192", "0?nomax") in new stack -- Executing [s at macro-dialout-trunk:8] GotoIf("SIP/300-00000192", "0?chanfull") in new stack -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/300-00000192", "0?skipoutcid") in new stack -- Executing [s at macro-dialout-...
2015 Mar 20
0
outbound calls
...3", > "DIAL_TRUNK_OPTIONS=tr") in new stack > -- Executing [s at macro-dialout-trunk:6] Set("SIP/101-00000103", > "OUTBOUND_GROUP=OUT_5") in new stack > -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/101-00000103", > "0?nomax") in new stack > -- Executing [s at macro-dialout-trunk:8] GotoIf("SIP/101-00000103", > "0?chanfull") in new stack > -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/101-00000103", > "0?skipoutcid") in new stack > -- Execu...
2015 Mar 27
2
call between snom 300 and aastra 6731i
You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: > please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit >
2015 Mar 20
0
outbound calls
...PTIONS=tr") in new stack >>> -- Executing [s at macro-dialout-trunk:6] Set("SIP/101-00000103", >>> "OUTBOUND_GROUP=OUT_5") in new stack >>> -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/101-00000103", >>> "0?nomax") in new stack >>> -- Executing [s at macro-dialout-trunk:8] GotoIf("SIP/101-00000103", >>> "0?chanfull") in new stack >>> -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/101-00000103", >>> "0?skipoutcid"...
2009 Mar 17
0
Weird issue with outbound calls and MOH
...ialout-trunk:6] Set("SIP/8647-b6f96650", "OUTBOUND_GROUP=OUT_3") in new stack [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: Set [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/8647-b6f96650", "1?nomax") in new stack [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Goto (macro-dialout-trunk,s,9) [Mar 13 13:30:37] DEBUG[28294] app_macro.c: Executed application: GotoIf [Mar 13 13:30:37] VERBOSE[28294] logger.c: -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/8647-b6f96650", &quo...
2008 Apr 28
2
PRI hangup certain outgoing calls
...ro-dialout-trunk:5] Set("IAX2/255-4", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing [s at macro-dialout-trunk:6] Set("IAX2/255-4", "GROUP()=OUT_1") in new stack -- Executing [s at macro-dialout-trunk:7] GotoIf("IAX2/255-4", "0?nomax") in new stack -- Executing [s at macro-dialout-trunk:8] GotoIf("IAX2/255-4", "0?chanfull") in new stack -- Executing [s at macro-dialout-trunk:9] GotoIf("IAX2/255-4", "0?skipoutcid") in new stack -- Executing [s at macro-dialout-trunk:1...
2013 Sep 25
2
users can not hear the audio playback sometimes
...09-25 13:57:33] VERBOSE[9745] pbx.c: -- Executing [s at macro-dialout-trunk:6] Set("SIP/1002-00000292", "OUTBOUND_GROUP=OUT_1") in new stack [2013-09-25 13:57:33] VERBOSE[9745] pbx.c: -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/1002-00000292", "0?nomax") in new stack [2013-09-25 13:57:33] VERBOSE[9745] pbx.c: -- Executing [s at macro-dialout-trunk:8] GotoIf("SIP/1002-00000292", "0?chanfull") in new stack [2013-09-25 13:57:33] VERBOSE[9745] pbx.c: -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/1002-0000...