Re: [asterisk-users] Inbound DAHDI Error
Hi, it seems, that the caller hangs up immediatly after calling. Try to reproduce it by yourself. Dial the number (to reach your asterisk server) and hangup after ~ 0.5 sec (or whatever). Best regards, -Thorsten- Am 30.04.2014 01:11, schrieb Bryce Lowe: Hello, I am trying to diagnose an intermittent error when a call comes in over our PRI lines. The problem appears random, however I have feeling it has something to do with the call volume, as the frequency increases with more calls on the system. I am not an expert when it comes to reading the PRI Span Debug statements but here is a call that had a problem and I bolded, italicized, and underlined the part of the debug statement that looks odd (listed under PRI Debug Output (failed call)). Any help is appreciated. Thanks, Bryce *Version(s):* ** Asterisk 11.8.1, installed from the Digium YUM Repositories DAHDI Version: 2.9.0 Digium Card: Wildcard TE235 (VPMOCT064) OS: CentOS 6.5 *My Observations:* ** When I have the problem, the only way I see that Asterisk received a signal on my PRI lines was through the pri debug statements, I don't see anything being hit in the dialplan (for instance the NoOp at the start of my sub-dial-cudatel-extension sub context). Is there another tool I should be using to debug this issue? *PRI Debug Output (failed call):* ** PRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=73 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 23832/0x5D18) (Sent from originator) PRI Span: 1 Message Type: SETUP (5) PRI Span: 1 [04 03 80 90 a2] PRI Span: 1 Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) PRI Span: 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) PRI Span: 1 User information layer 1: u-Law (34) PRI Span: 1 [18 03 a1 83 81] PRI Span: 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Preferred Dchan: 0 PRI Span: 1 ChanSel: As indicated in following octets PRI Span: 1 Ext: 1 Coding: 0 Number Specified Channel Type: 3 PRI Span: 1 Ext: 1 Channel: 1 Type: CPE] PRI Span: 1 [1c 1d 9f 8b 01 00 a1 17 02 01 01 02 01 00 80 0f 4f 4d 41 58 20 43 4f 52 50 20 4e 20 47 53 4d] PRI Span: 1 Facility (len=31, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0F, 'source_caller_name' ] PRI Span: 1 [6c 0c 21 83 32 35 33 33 38 30 35 35 39 31] PRI Span: 1 Calling Party Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) PRI Span: 1 Presentation: Presentation allowed, Network provided (3) 'calling_caller_id' ] PRI Span: 1 [70 0b a1 32 35 33 38 37 32 32 33 30 30] PRI Span: 1 Called Party Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'dest_number' ] PRI Span: 1 -- Making new call for cref 23832 PRI Span: 1 Received message for call 0x7f7a900012f0 on link 0x1a3cf70 TEI/SAPI 0/0 PRI Span: 1 -- Processing Q.931 Call Setup PRI Span: 1 -- Processing IE 4 (cs0, Bearer Capability) PRI Span: 1 -- Processing IE 24 (cs0, Channel ID) PRI Span: 1 -- Processing IE 28 (cs0, Facility) PRI Span: 1 -- Processing IE 108 (cs0, Calling Party Number) PRI Span: 1 -- Processing IE 112 (cs0, Called Party Number) PRI Span: 1 -- Delayed processing IE 28 (cs0, Facility) PRI Span: 1 ASN.1 dump PRI Span: 1 Context Specific [11 0x0B] 8B Len:1 01 PRI Span: 1 00 - ~ PRI Span: 1 Context Specific/C [1 0x01] A1 Len:23 17 PRI Span: 1 Integer(2 0x02) 02 Len:1 01 PRI Span: 1 01 - ~ PRI Span: 1 Integer(2 0x02) 02 Len:1 01 PRI Span: 1 00 - ~ PRI Span: 1 Context Specific [0 0x00] 80 Len:15 0F PRI Span: 1 4F 4D 41 58 20 43 4F 52-50 20 4E 20 47 53 4D - source_caller_name PRI Span: 1 ASN.1 end PRI Span: 1 interpretation Context Specific [11 0x0B] = 0 0x PRI Span: 1 INVOKE Component Context Specific/C [1 0x01] PRI Span: 1 invokeId Integer(2 0x02) = 1 0x0001 PRI Span: 1 operationValue Integer(2 0x02) = 0 0x PRI Span: 1 operationValue = ROSE_QSIG_CallingName PRI Span: 1 callingName Name PRI Span: 1 namePresentationAllowedSimple Context Specific [0 0x00] = PRI Span: 1 4F 4D 41 58 20 43 4F 52-50 20 4E 20 47 53 4D - source_caller_name PRI Span: 1 q931.c:8646 post_handle_q931_message: Call 23832 enters state 6 (Call Present). Hold state: Idle Span 1: Processing event PRI_EVENT_RING(5) */_PRI Span: 1 q931.c:7135 q931_hangup: Hangup other cref:23832_/* */_PRI Span: 1 q931.c:6892 __q931_hangup: ourstate Call Present, peerstate Call Initiated, hold-state Idle_/* */_PRI Span: 1 q931.c:6081 q931_disconnect: Call 23832 enters state 11 (Disconnect Request). Hold state: Idle_/* PRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=73 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 23832/0x5D18) (Sent from originator) PRI Span: 1 Message Type: SETUP (5) PRI Span: 1 [04 03 80 90 a2] PRI Span: 1 Bearer
[asterisk-users] AGI GET DATA behavior
Hello all, I have a strange problem with a very simple AGI script, using the GET DATA command. When using this command, Asterisk often returns 0 as a result after a GET DATA beep 5000 command, without even waiting for input from the calling party. It is quite random : sometimes Asterisk behaves exactly as documented, and sometimes it gives 200 result=0 without any reason. Do you have an idea of what might be happening ? I'm using version 11.6.0. Hoggins! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Hi, after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like sip reload or iax2 reload does nothing. We opened bug 23683 but it was immediately closed by Matt Jordan, telling that he can't reproduce it. But we can. Example: - switching back to 11.8.1 respectively 1.8.26.1 does the job working again (We just run a make install from within this directory) - cleaning 11.8.0 source directory -make clean ./configure make make install- all is good - cleaning 11.9.0 source directory -make clean ./configure make make install- problem appears again - switching back to 11.8.0 does the job working again (We just run a make install from within this directory) The first installation of latest version was done by patching the previous version, we downloaded the source tar.gz and compile = problem stays Does anybody else face this problem with latest version? If it was a server problem, earlier version should have same behaviour after compiling but they don't. Server OS is Debian Wheezy 3.2.0-4 amd64 in KVM virtual machine Thanks for any hint Regards -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Le 30/04/2014 12:15, Administrator TOOTAI a écrit : Hi, after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like sip reload or iax2 reload does nothing. We opened bug 23683 but it was immediately closed by Matt Jordan, telling that he can't reproduce it. But we can. Example: - switching back to 11.8.1 respectively 1.8.26.1 does the job working again (We just run a make install from within this directory) - cleaning 11.8.0 source directory -make clean ./configure make make install- all is good - cleaning 11.9.0 source directory -make clean ./configure make make install- problem appears again - switching back to 11.8.0 does the job working again (We just run a make install from within this directory) The first installation of latest version was done by patching the previous version, we downloaded the source tar.gz and compile = problem stays Does anybody else face this problem with latest version? If it was a server problem, earlier version should have same behaviour after compiling but they don't. Server OS is Debian Wheezy 3.2.0-4 amd64 in KVM virtual machine Thanks for any hint Regards We checked on a customer installation made one week ago: they have the same problem! It's a Debian Squeeze 2.6.32-5-amd64 on a real server. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMR installation error
make gives this: codec_amr.c: In function 'amrtolin_sample': codec_amr.c:227: error: 'AST_FORMAT_AMRNB' undeclared (first use in this function) codec_amr.c:227: error: (Each undeclared identifier is reported only once codec_amr.c:227: error: for each function it appears in.) codec_amr.c: In function 'lintoamr_frameout': codec_amr.c:345: warning: unused variable 'byte_count' codec_amr.c: At top level: codec_amr.c:409: error: 'AST_FORMAT_AMRNB' undeclared here (not in a function) make[1]: *** [codec_amr.o] Error 1 make: *** [codecs] Error 2 Any ideas how to fix it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Hi, some more information could be usefull. On Wed, Apr 30, 2014 at 12:15:03PM +0200, Administrator TOOTAI wrote: after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like sip reload or iax2 reload does nothing. Is Asterisk fully booted? There should be such a message for each AMI connection. Is it possible to unload chan_sip.so and to load it again? Are there error messages related to sip and iax modules? -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Le 30/04/2014 12:39, Administrator TOOTAI a écrit : Le 30/04/2014 12:15, Administrator TOOTAI a écrit : Hi, after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like sip reload or iax2 reload does nothing. We opened bug 23683 but it was immediately closed by Matt Jordan, telling that he can't reproduce it. But we can. Example: - switching back to 11.8.1 respectively 1.8.26.1 does the job working again (We just run a make install from within this directory) - cleaning 11.8.0 source directory -make clean ./configure make make install- all is good - cleaning 11.9.0 source directory -make clean ./configure make make install- problem appears again - switching back to 11.8.0 does the job working again (We just run a make install from within this directory) The first installation of latest version was done by patching the previous version, we downloaded the source tar.gz and compile = problem stays Does anybody else face this problem with latest version? If it was a server problem, earlier version should have same behaviour after compiling but they don't. Server OS is Debian Wheezy 3.2.0-4 amd64 in KVM virtual machine Thanks for any hint Regards We checked on a customer installation made one week ago: they have the same problem! It's a Debian Squeeze 2.6.32-5-amd64 on a real server. And finally the explanation: if you modify sip.conf file, the reload is taken in account, all is good. But if the sip.conf contains includes and you modify one of those includes *without modifying* sip.conf, no reload. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMR installation error
On Wednesday 30 Apr 2014, [Digital^Dude] ® wrote: make gives this: codec_amr.c: In function 'amrtolin_sample': codec_amr.c:227: error: 'AST_FORMAT_AMRNB' undeclared (first use in this function) codec_amr.c:227: error: (Each undeclared identifier is reported only once codec_amr.c:227: error: for each function it appears in.) codec_amr.c: In function 'lintoamr_frameout': codec_amr.c:345: warning: unused variable 'byte_count' codec_amr.c: At top level: codec_amr.c:409: error: 'AST_FORMAT_AMRNB' undeclared here (not in a function) make[1]: *** [codec_amr.o] Error 1 make: *** [codecs] Error 2 Any ideas how to fix it? Your question reads like Hello, is that the vet? One of my animals is poorly and is liable to get the same sort of answer. Which version of Asterisk are you trying to build, on what OS and for what architecture? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Does a reload (not a sip reload) reload everything or does it also require the sip.conf file to be modified? On Wed, Apr 30, 2014 at 5:00 AM, Administrator TOOTAI ad...@tootai.netwrote: Le 30/04/2014 12:39, Administrator TOOTAI a écrit : Le 30/04/2014 12:15, Administrator TOOTAI a écrit : Hi, after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like sip reload or iax2 reload does nothing. We opened bug 23683 but it was immediately closed by Matt Jordan, telling that he can't reproduce it. But we can. Example: - switching back to 11.8.1 respectively 1.8.26.1 does the job working again (We just run a make install from within this directory) - cleaning 11.8.0 source directory -make clean ./configure make make install- all is good - cleaning 11.9.0 source directory -make clean ./configure make make install- problem appears again - switching back to 11.8.0 does the job working again (We just run a make install from within this directory) The first installation of latest version was done by patching the previous version, we downloaded the source tar.gz and compile = problem stays Does anybody else face this problem with latest version? If it was a server problem, earlier version should have same behaviour after compiling but they don't. Server OS is Debian Wheezy 3.2.0-4 amd64 in KVM virtual machine Thanks for any hint Regards We checked on a customer installation made one week ago: they have the same problem! It's a Debian Squeeze 2.6.32-5-amd64 on a real server. And finally the explanation: if you modify sip.conf file, the reload is taken in account, all is good. But if the sip.conf contains includes and you modify one of those includes *without modifying* sip.conf, no reload. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I took the degree of Doctor of Philosophy in 1903. The meaning of this degree is that the recipient of instruction is examined for the last time in his life, and is pronounced completely full. After this, no new ideas can be imparted to him. - Stephen Leacock -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI GET DATA behavior
Is your script really so simple? Enable agi debugging (agi set debug on) and take look at it when this happens. -Thorsten- Am 30.04.2014 11:47, schrieb Hoggins!: Hello all, I have a strange problem with a very simple AGI script, using the GET DATA command. When using this command, Asterisk often returns 0 as a result after a GET DATA beep 5000 command, without even waiting for input from the calling party. It is quite random : sometimes Asterisk behaves exactly as documented, and sometimes it gives 200 result=0 without any reason. Do you have an idea of what might be happening ? I'm using version 11.6.0. Hoggins! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Le 30/04/2014 13:04, Derek Andrew a écrit : Does a reload (not a sip reload) reload everything or does it also require the sip.conf file to be modified? reload as well as module reload chan_sip.so does nothing. Only way i had till I found why was to restart asterisk :-( On Wed, Apr 30, 2014 at 5:00 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: Le 30/04/2014 12:39, Administrator TOOTAI a écrit : Le 30/04/2014 12:15, Administrator TOOTAI a écrit : Hi, after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like sip reload or iax2 reload does nothing. We opened bug 23683 but it was immediately closed by Matt Jordan, telling that he can't reproduce it. But we can. Example: - switching back to 11.8.1 respectively 1.8.26.1 does the job working again (We just run a make install from within this directory) - cleaning 11.8.0 source directory -make clean ./configure make make install- all is good - cleaning 11.9.0 source directory -make clean ./configure make make install- problem appears again - switching back to 11.8.0 does the job working again (We just run a make install from within this directory) The first installation of latest version was done by patching the previous version, we downloaded the source tar.gz and compile = problem stays Does anybody else face this problem with latest version? If it was a server problem, earlier version should have same behaviour after compiling but they don't. Server OS is Debian Wheezy 3.2.0-4 amd64 in KVM virtual machine Thanks for any hint Regards We checked on a customer installation made one week ago: they have the same problem! It's a Debian Squeeze 2.6.32-5-amd64 on a real server. And finally the explanation: if you modify sip.conf file, the reload is taken in account, all is good. But if the sip.conf contains includes and you modify one of those includes *without modifying* sip.conf, no reload. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I took the degree of Doctor of Philosophy in 1903. The meaning of this degree is that the recipient of instruction is examined for the last time in his life, and is pronounced completely full. After this, no new ideas can be imparted to him. - Stephen Leacock -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Q.850 Cause
Hello, I'm trying to fetch outbound SIP PROGRESS Reason cause code in the dialplan, Asterisk 1.8.26.1 sip show settings: Q.850 Reason header:Yes Store SIP_CAUSE:Yes However, i'm not getting any value in the dialplan variables, any successful users of this feature? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
On Tue, Apr 29, 2014 at 6:03 PM, Richard Kenner ken...@gnat.com wrote:
If the channel still hangs around after the conference is destroyed
then there is a problem.
Am I missing something obvious: I'm looking in the confbridge_exec
function. I see a conference = NULL line, but no attempt to free
that structure, which is what I understand will destroy the playback
channel. So where it is freed?
Conferences are reference counted objects. When the reference count reaches
0 on the conference object, its destructor is automatically called. The
destructor, in this case, is destroy_conference_bridge. That is where the
CBAnn channel should be hung up.
/* Try to allocate memory for a new conference bridge, if we
fail... this won't end well. */
if (!(conference = ao2_alloc(sizeof(*conference),
destroy_conference_bridge))) {
ao2_unlock(conference_bridges);
ast_log(LOG_ERROR, Conference '%s' could not be created.\n,
conference_name);
return NULL;
}
If the reference count on the bridge is off, you should see the conference
bridge 'hanging around' after the last participant has left. If so, please
file a bug report. We'll need a REF_DEBUG log to figure out who the guilty
party is in holding onto a reference. The easiest way to get that is to
reproduce the problem using the latest from the 12 branch (as we made
reference count debugging easier just recently). Enable REF_DEBUG in
menuselect under Compiler Flags, make/make install, and re-run the scenario
that reproduces the result. A refs file will be created in your Asterisk
log directory - attach that to the issue along with DEBUG log.
Thanks!
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Please, people from Digium, Matt again closed the new bug ASTERISK-23689 I opened (clone from 23683) telling that it's not a bug. Did he carefully read the comments on the new bug? If not, please forward him this email, *it's* a bug or you have to explain me why it is not! Le 30/04/2014 13:00, Administrator TOOTAI a écrit : Le 30/04/2014 12:39, Administrator TOOTAI a écrit : Le 30/04/2014 12:15, Administrator TOOTAI a écrit : Hi, after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like sip reload or iax2 reload does nothing. We opened bug 23683 but it was immediately closed by Matt Jordan, telling that he can't reproduce it. But we can. Example: - switching back to 11.8.1 respectively 1.8.26.1 does the job working again (We just run a make install from within this directory) - cleaning 11.8.0 source directory -make clean ./configure make make install- all is good - cleaning 11.9.0 source directory -make clean ./configure make make install- problem appears again - switching back to 11.8.0 does the job working again (We just run a make install from within this directory) The first installation of latest version was done by patching the previous version, we downloaded the source tar.gz and compile = problem stays Does anybody else face this problem with latest version? If it was a server problem, earlier version should have same behaviour after compiling but they don't. Server OS is Debian Wheezy 3.2.0-4 amd64 in KVM virtual machine Thanks for any hint Regards We checked on a customer installation made one week ago: they have the same problem! It's a Debian Squeeze 2.6.32-5-amd64 on a real server. And finally the explanation: if you modify sip.conf file, the reload is taken in account, all is good. But if the sip.conf contains includes and you modify one of those includes *without modifying* sip.conf, no reload. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
did you try rebooting after installing 11.9? -Original Message- From: Administrator TOOTAI ad...@tootai.net Sender: asterisk-users-bounces@lists.digium.comDate: Wed, 30 Apr 2014 15:13:59 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ? Please, people from Digium, Matt again closed the new bug ASTERISK-23689 I opened (clone from 23683) telling that it's not a bug. Did he carefully read the comments on the new bug? If not, please forward him this email, *it's* a bug or you have to explain me why it is not! Le 30/04/2014 13:00, Administrator TOOTAI a écrit : Le 30/04/2014 12:39, Administrator TOOTAI a écrit : Le 30/04/2014 12:15, Administrator TOOTAI a écrit : Hi, after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like sip reload or iax2 reload does nothing. We opened bug 23683 but it was immediately closed by Matt Jordan, telling that he can't reproduce it. But we can. Example: - switching back to 11.8.1 respectively 1.8.26.1 does the job working again (We just run a make install from within this directory) - cleaning 11.8.0 source directory -make clean ./configure make make install- all is good - cleaning 11.9.0 source directory -make clean ./configure make make install- problem appears again - switching back to 11.8.0 does the job working again (We just run a make install from within this directory) The first installation of latest version was done by patching the previous version, we downloaded the source tar.gz and compile = problem stays Does anybody else face this problem with latest version? If it was a server problem, earlier version should have same behaviour after compiling but they don't. Server OS is Debian Wheezy 3.2.0-4 amd64 in KVM virtual machine Thanks for any hint Regards We checked on a customer installation made one week ago: they have the same problem! It's a Debian Squeeze 2.6.32-5-amd64 on a real server. And finally the explanation: if you modify sip.conf file, the reload is taken in account, all is good. But if the sip.conf contains includes and you modify one of those includes *without modifying* sip.conf, no reload. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
On Wed, Apr 30, 2014 at 8:13 AM, Administrator TOOTAI ad...@tootai.netwrote: Please, people from Digium, Matt again closed the new bug ASTERISK-23689 I opened (clone from 23683) telling that it's not a bug. Did he carefully read the comments on the new bug? If not, please forward him this email, *it's* a bug or you have to explain me why it is not! I asked you not to clone and issues and to take your issue to the mailing list (which you did, thank-you). Cloning issues makes a mess of the issue tracker, and causes information to get lost. If your issue is deemed to be a bug, the original issue will get re-opened. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
If the reference count on the bridge is off, you should see the conference bridge 'hanging around' after the last participant has left. And how would I be sure this is the case? I did core set debug 1 and didn't see the debug line about destroying the conference, but it doesn't show up in confbridge list. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Le 30/04/2014 15:19, Matthew Jordan a écrit : On Wed, Apr 30, 2014 at 8:13 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: Please, people from Digium, Matt again closed the new bug ASTERISK-23689 I opened (clone from 23683) telling that it's not a bug. Did he carefully read the comments on the new bug? If not, please forward him this email, *it's* a bug or you have to explain me why it is not! I asked you not to clone and issues and to take your issue to the mailing list (which you did, thank-you). Cloning issues makes a mess of the issue tracker, and causes information to get lost. If your issue is deemed to be a bug, the original issue will get re-opened. I cloned the issue as it is a bug and I could explain how to reproduce it. If I shouldn't clone the bug, please explain me how to do to inform developpers about new informations concerning a closed bug. That say, sorry for inconvenience. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
On Wed, Apr 30, 2014 at 8:20 AM, Richard Kenner ken...@gnat.com wrote: If the reference count on the bridge is off, you should see the conference bridge 'hanging around' after the last participant has left. And how would I be sure this is the case? I did core set debug 1 and didn't see the debug line about destroying the conference, but it doesn't show up in confbridge list. That's not terribly surprising, as the bridge is typically removed from the list of active conferences prior to the destructor being called (the destructor being the thing that releases memory, not the thing that notifies everyone that this thing *should* be destroyed). It may show up in 'bridge show all' - but I'd actually expect it not to show up there either. Really, I think we're pretty positive there's a ref leak (since otherwise, the CBAnn channel would be long gone). If you can get a ref debug log and the standard Asterisk DEBUG log showing the problem, that would help a lot in finding out what is going on. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound DAHDI Error
Thank you for your feedback. The call that I attached was actually from me. I get a message on my ATT phone that says the wireless customer you...message 1 WA01ML. It is pretty hit or miss. On Wed, Apr 30, 2014 at 1:45 AM, Thorsten Göllner t...@ovm-group.com wrote: Hi, it seems, that the caller hangs up immediatly after calling. Try to reproduce it by yourself. Dial the number (to reach your asterisk server) and hangup after ~ 0.5 sec (or whatever). Best regards, -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMR installation error
On 30-04-14 12:50, [Digital^Dude] ® wrote: make gives this: IIRC Digium's policy is that there's no support on this list for patented technologies like AMR which are possibly not officially licensed. Obviously to prevent any legal liability. HTH, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk's internal database
On Tue, Apr 29, 2014 at 1:31 AM, binary dreamer dreamer.bin...@gmail.com wrote: i would like to read information from a file (txt) There are a few applications and functions that may help you out. In Asterisk 10 or before try the ReadFile application. Otherwise in 11 or beyond I believe you want the FILE function. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ReadFile https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_FILE You could also use the SHELL function to execute a command on the system and capture output https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_SHELL Then you might also look into Asterisk Gateway Interface for more complex tasks and control. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_AGI http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/AGI.html If you are learning Asterisk; the book I linked above contains some great information on AGI and is written by some of our notable community members. The latest edition is available here: http://shop.oreilly.com/product/0636920025894.do -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Create new channel from dialplan
Hi all, I need a command to originate a new channel from dialplan. I should be able to continue execution of the current context after this command. How to do this? Best, Igor -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Create new channel from dialplan
On Wed, Apr 30, 2014 at 4:33 PM, Igor Dvorzhak idm...@gmail.com wrote: Hi all, I need a command to originate a new channel from dialplan. I should be able to continue execution of the current context after this command. How to do this? Look at this application: *CLI core show application Originate Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help troubleshooting Asterisk Auto dial out problem
We've built an alert system at our company so that if our monitoring software notices anything very bad happening, and we don't react to a text message after a few minutes, then it will begin to call our telephones directly. This seems to help a lot with staff who are asleep, or who might not be near enough to the phone to hear text message notifications. We do this by having a folder full of .call files, and programatically moving a copy of the appropriate call file into /var/spool/asterisk/outgoing/ when an outbound shout is required. This has worked well for a couple of years, but recently we are experiencing a problem where sometimes Asterisk will simply ignore this folder. When we notice the logjam of ignored call files in /var/spool/asterisk/outgoing/, then we restart the asterisk daemon and things run smoothly again. When wedged, the Asterisk daemon will function normally in every other way that we use it. The PBX still functions, outbound manual calls still function, inbound calls, voicemail, nothing else appears to be negatively impacted. However new files introduced into /var/spool/asterisk/outgoing/ folder get ignored. No messages spring up on asterisk -rvv console, nothing shows up in the logs, the .call files just get snubbed. We're at a loss to determine what other debugging avenues may be available, and we have googled for every applicable keyword we can think of including asterisk auto dial and pbx_spool to no avail. We are running: Asterisk 1.8.13.1~dfsg1-3+deb7u3 built by pbuilder @ pungenday on a x86_64 running Linux on 2014-01-04 01:03:48 UTC on Debian Linux stable (wheezy). It is possible that this symptom began a few months ago after a system update (apt-get update; apt-get upgrade). Any advise to further track down what is going on would be appreciated, thank you! - - Jesse Thompson Webformix, Bend OR www.webformix.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Create new channel from dialplan
Thanks, it almost what I need. But I can't find a way to pass channel variables to Originate cmd in dialplan. Is it possible at all? On Wed, Apr 30, 2014 at 3:13 PM, Richard Mudgett rmudg...@digium.comwrote: On Wed, Apr 30, 2014 at 4:33 PM, Igor Dvorzhak idm...@gmail.com wrote: Hi all, I need a command to originate a new channel from dialplan. I should be able to continue execution of the current context after this command. How to do this? Look at this application: *CLI core show application Originate Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
Really, I think we're pretty positive there's a ref leak (since otherwise, the CBAnn channel would be long gone). If you can get a ref debug log and the standard Asterisk DEBUG log showing the problem, that would help a lot in finding out what is going on. That can't be done in the 12.2.0 release, just the current SVN, right? Clearly this occurs for me and not in the simple case. So I think what I'll do is see exactly what I have that's causing it and hopefully code inspection of that piece will show the missing ref decrement. I'm away for a few days and so may not be able to get to this until I get back. Thanks for the pointers. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
