Re: [asterisk-users] PJSIP Weirdness, or just my weirdness?
On Thursday, September 8, 2016 1:12:36 PM CDT Steve Murphy wrote: > Hello! > > Oh, wise ones, ponder with me over two of the surprises that > populate the universe! > > > I have a phone, that I sometimes cannot reach, connected via pjsip. > It can call other extensions just fine, it can call out over a > trunk to my cell, all is well, but getting a call? Forget it most of the > time. > > Here is all the config relevant to that phone: > > > [murftest12] > type=aor > qualify_frequency=1992 > max_contacts=2 > > [murftest12] > type=auth > auth_type=userpass > username=murftest12 > password=SjU3 > > [transport-udp] > type=transport > protocol=udp > bind=0.0.0.0:57969 > > > [murftest12]; Cisco SPA514G mac=A4:93:4C:FE:1D:A2 > type=endpoint > auth=murftest12 > transport=transport-udp > aors=murftest12 > moh_suggest=default > force_rport=yes > rewrite_contact=yes > rtp_symmetric=yes > dtmf_mode=rfc4733 > disallow=all > allow=ulaw ; from phonetype > allow=g722 ; from phonetype > allow=alaw ; from phonetype > allow=alaw ; from phonetype (G.729 replaced with alaw) > direct_media=no > context=phone > rtp_timeout=120 > set_var=__phoneid=12 > set_var=__contacttypeid=4 > set_var=__phonelineid=78 > callerid="Steve Murphy" <101> > call_group=2 > pickup_group=2 > mailboxes=101@murftest > language=en > send_rpid=yes > send_pai=yes > > OK, that completes the config (I hope). > > Now, when I run "pjsip show endpoints, I get: > > SFO02-HostedPBXPJSip-Dev03*CLI> pjsip show endpoints > > Endpoint: > > I/OAuth: ...> > Aor: > > Contact: > > Transport: > >Identify: ...> > Match: > Channel: > > Exten: CLCID: > === > == > > Endpoint: murftest12/101 Not in > use0 of inf > InAuth: murftest12/murftest12 > Aor: murftest12 2 > Contact: murftest12/sip:murftest12@67.215.23.186:54 171a08228b > Unavail 0.000 > Contact: murftest12/sip:murftest12@67.215.23.186:21 d9a15f4e35 > Avail50.514 > Transport: transport-udp udp 0 0 0.0.0.0:57969 > > Note that there are TWO Contact: entries! one Avail, the other Unavail... > the show endpoints doesn't display all the URL, but the show contacts does: > > Contact: murftest12/sip:murftest12@67.215.23.186:21800 d9a15f4e35 > Avail50.514 > Contact: murftest12/sip:murftest12@67.215.23.186:54004 171a08228b > Unavail 0.000 > > None of my other phones have two contacts listed and this phone, a > cisco-spa-514, has just one sip account... > > The trouble is, when I try to call it sometimes the INVITE is directed > to the "Unavail" entry, and the call never completes. The phone doesn't > even ring then. Any ideas? I tried to get the "Unavail" entry out... I > removed it from the db, I rebooted the phone, restarted asterisk, and it is > still there. > > MYSTERY #2: > > The above cisco-spa, when it calls out over the trunk, all is well, > wonderful 2-way audio. > But when I do the same operation from my yealink phones, I get my cell with > one-way audio. I just resolved a similar issue with a new Yealink phone and PJSIP. It seems that Asterisk (depending on many transcoding parameters and types of calls) may send out a different codec on leg B than it receives on leg A. While less than optimal for the end user, this is allowed by the RFCs. Yealink doesn't seem to handle this well. The firmware referenced in this link fixed the issue for me, as least with my T48G and DAHDI/PJSIP calls. http://forum.yealink.com/forum/showthread.php?tid=8330&pid=39161#pid39161 > It's a classic NAT situation: the phone system is in a droplet at digital > ocean, but my phones are here at home behind a NAT. I see only 3 NAT > related options: > > force_rport > rtp_symmetric > rewrite_contact > > and I set them all to "yes", and they can call each other, but as > explained, in > dialing out thru a trunk, the yealinks get one-way audio... > > Any more NAT options? > > many thanks... > > murf -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6 signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 13 and WebRTC
Hello list, before to lost my time, I'd like know if someone have a WebRTC working configuration on Asterisk 13.11.0 SIP or PJSIP channel. Thank you Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP Weirdness, or just my weirdness?
On Thu, Sep 8, 2016 at 1:12 PM, Steve Murphy wrote: > Hello! > > Oh, wise ones, ponder with me over two of the surprises that > populate the universe! > > > I have a phone, that I sometimes cannot reach, connected via pjsip. > It can call other extensions just fine, it can call out over a > trunk to my cell, all is well, but getting a call? Forget it most of the > time. > > Here is all the config relevant to that phone: > > > [murftest12] > type=aor > qualify_frequency=1992 > max_contacts=2 > *max_contacts = 1* *remove_existing = yes* *This should take care of mystery 1.* > > > MYSTERY #2: > > The above cisco-spa, when it calls out over the trunk, all is well, > wonderful 2-way audio. > But when I do the same operation from my yealink phones, I get my cell > with one-way audio. > It's a classic NAT situation: the phone system is in a droplet at digital > ocean, but my phones are here at home behind a NAT. I see only 3 NAT > related options: > > force_rport > rtp_symmetric > rewrite_contact > > and I set them all to "yes", and they can call each other, but as > explained, in > dialing out thru a trunk, the yealinks get one-way audio... > > Any more NAT options? > Do the phones have settings for NAT? If so, turn them on. Does your home router have a setting for SIP-ALG? If so *TURN IT OFF!!* Give that a shot. > > many thanks... > > murf > -- > > Steve Murphy > > > ✉ murf at parsetree dot com > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 > http://www.asterisk.org/community/astricon-user-conference > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2016-007: RTP Resource Exhaustion
Asterisk Project Security Advisory - AST-2016-007 ProductAsterisk SummaryRTP Resource Exhaustion Nature of Advisory Denial of Service SusceptibilityRemote Authenticated Sessions Severity Moderate Exploits KnownNo Reported On August 5, 2016 Reported By Etienne Lessard Posted On Last Updated OnSeptember 8, 2016 Advisory Contact Joshua Colp CVE Name Description The overlap dialing feature in chan_sip allows chan_sip to report to a device that the number that has been dialed is incomplete and more digits are required. If this functionality is used with a device that has performed username/password authentication RTP resources are leaked. This occurs because the code fails to release the old RTP resources before allocating new ones in this scenario. If all resources are used then RTP port exhaustion will occur and no RTP sessions are able to be set up. Resolution If overlap dialing support is not needed the âallowoverlapâ option can be set to no. This will stop any usage of the scenario which causes the resource exhaustion. If overlap dialing support is needed a change has been made so that existing RTP resources are destroyed in this scenario before allocating new resources. Affected Versions Product Release Series Asterisk Open Source 11.xAll Versions Asterisk Open Source 13.xAll Versions Certified Asterisk 11.6All Versions Certified Asterisk 13.8All Versions Corrected In Product Release Asterisk Open Source 11.23.1, 13.11.1 Certified Asterisk11.6-cert15, 13.8-cert3 Patches SVN URL Revision Links https://issues.asterisk.org/jira/browse/ASTERISK-26272 Asterisk Project Security Advisories are posted at http://www.asterisk.org/security This document may be superseded by later versions; if so, the latest version will be posted at http://downloads.digium.com/pub/security/AST-2016-007.pdf and http://downloads.digium.com/pub/security/AST-2016-007.html Revision History Date Editor Revisions Made August 23, 2016 Joshua Colp Initial creation Asterisk Project Security Advisory - AST-2016-007 Copyright © 2016 Digium, Inc. All Rights Reserved. Permission is hereby granted to distribute and publish this advisory in its original, unaltered form. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2016-006: Crash on ACK from unknown endpoint
Asterisk Project Security Advisory - AST-2016-006 ProductAsterisk SummaryCrash on ACK from unknown endpoint Nature of Advisory Remote Crash SusceptibilityRemote unauthenticated sessions Severity Critical Exploits KnownNo Reported On August 3, 2016 Reported By Nappsoft Posted On Last Updated OnAugust 31, 2016 Advisory Contact mark DOT michelson AT digium DOT com CVE Name Description Asterisk can be crashed remotely by sending an ACK to it from an endpoint username that Asterisk does not recognize. Most SIP request types result in an "artificial" endpoint being looked up, but ACKs bypass this lookup. The resulting NULL pointer results in a crash when attempting to determine if ACLs should be applied. This issue was introduced in the Asterisk 13.10 release and only affects that release. This issue only affects users using the PJSIP stack with Asterisk. Those users that use chan_sip are unaffected. Resolution ACKs now result in an artificial endpoint being looked up just like other SIP request types. Affected Versions ProductRelease Series Asterisk Open Source 11.xUnaffected Asterisk Open Source 13.x13.10.0 Certified Asterisk11.6Unaffected Certified Asterisk13.8Unaffected Corrected In Product Release Asterisk Open Source13.11.1 Patches SVN URL Revision Links Asterisk Project Security Advisories are posted at http://www.asterisk.org/security This document may be superseded by later versions; if so, the latest version will be posted at http://downloads.digium.com/pub/security/AST-2016-006.pdf and http://downloads.digium.com/pub/security/AST-2016-006.html Revision History DateEditor Revisions Made August 16, 2016 Mark Michelson Initial draft of Advisory Asterisk Project Security Advisory - AST-2016-006 Copyright (c) 2016 Digium, Inc. All Rights Reserved. Permission is hereby granted to distribute and publish this advisory in its original, unaltered form. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11.6-cert15, 11.23.1, 13.8-cert3, 13.11.1 Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified Asterisk 11.6, Asterisk 11, Certified Asterisk 13.8 and Asterisk 13. The available security releases are released as versions 11.6-cert15, 11.23.1, 13.8-cert3 and 13.11.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these versions resolves the following security vulnerabilities: * AST-2016-006: Crash on ACK from unknown endpoint Asterisk can be crashed remotely by sending an ACK to it from an endpoint username that Asterisk does not recognize. Most SIP request types result in an "artificial" endpoint being looked up, but ACKs bypass this lookup. The resulting NULL pointer results in a crash when attempting to determine if ACLs should be applied. This issue was introduced in the Asterisk 13.10 release and only affects that release and later releases. This issue only affects users using the PJSIP stack with Asterisk. Those users that use chan_sip are unaffected. * AST-2016-007: RTP Resource Exhaustion The overlap dialing feature in chan_sip allows chan_sip to report to a device that the number that has been dialed is incomplete and more digits are required. If this functionality is used with a device that has performed username/password authentication RTP resources are leaked. This occurs because the code fails to release the old RTP resources before allocating new ones in this scenario. If all resources are used then RTP port exhaustion will occur and no RTP sessions are able to be set up. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/certified- asterisk/releases/ChangeLog-certified-11.6-cert15 http://downloads.asterisk.org/pub/telephony/asterisk/ releases/ChangeLog-11.23.1 http://downloads.asterisk.org/pub/telephony/certified- asterisk/releases/ChangeLog-certified-13.8-cert3 http://downloads.asterisk.org/pub/telephony/asterisk/ releases/ChangeLog-13.11.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2016-006.pdf * http://downloads.asterisk.org/pub/security/AST-2016-007.pdf Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP Weirdness, or just my weirdness?
Hello! Oh, wise ones, ponder with me over two of the surprises that populate the universe! I have a phone, that I sometimes cannot reach, connected via pjsip. It can call other extensions just fine, it can call out over a trunk to my cell, all is well, but getting a call? Forget it most of the time. Here is all the config relevant to that phone: [murftest12] type=aor qualify_frequency=1992 max_contacts=2 [murftest12] type=auth auth_type=userpass username=murftest12 password=SjU3 [transport-udp] type=transport protocol=udp bind=0.0.0.0:57969 [murftest12]; Cisco SPA514G mac=A4:93:4C:FE:1D:A2 type=endpoint auth=murftest12 transport=transport-udp aors=murftest12 moh_suggest=default force_rport=yes rewrite_contact=yes rtp_symmetric=yes dtmf_mode=rfc4733 disallow=all allow=ulaw ; from phonetype allow=g722 ; from phonetype allow=alaw ; from phonetype allow=alaw ; from phonetype (G.729 replaced with alaw) direct_media=no context=phone rtp_timeout=120 set_var=__phoneid=12 set_var=__contacttypeid=4 set_var=__phonelineid=78 callerid="Steve Murphy" <101> call_group=2 pickup_group=2 mailboxes=101@murftest language=en send_rpid=yes send_pai=yes OK, that completes the config (I hope). Now, when I run "pjsip show endpoints, I get: SFO02-HostedPBXPJSip-Dev03*CLI> pjsip show endpoints Endpoint: I/OAuth: Aor: Contact: Transport: Identify: Match: Channel: Exten: CLCID: === == Endpoint: murftest12/101 Not in use0 of inf InAuth: murftest12/murftest12 Aor: murftest12 2 Contact: murftest12/sip:murftest12@67.215.23.186:54 171a08228b Unavail 0.000 Contact: murftest12/sip:murftest12@67.215.23.186:21 d9a15f4e35 Avail50.514 Transport: transport-udp udp 0 0 0.0.0.0:57969 Note that there are TWO Contact: entries! one Avail, the other Unavail... the show endpoints doesn't display all the URL, but the show contacts does: Contact: murftest12/sip:murftest12@67.215.23.186:21800 d9a15f4e35 Avail50.514 Contact: murftest12/sip:murftest12@67.215.23.186:54004 171a08228b Unavail 0.000 None of my other phones have two contacts listed and this phone, a cisco-spa-514, has just one sip account... The trouble is, when I try to call it sometimes the INVITE is directed to the "Unavail" entry, and the call never completes. The phone doesn't even ring then. Any ideas? I tried to get the "Unavail" entry out... I removed it from the db, I rebooted the phone, restarted asterisk, and it is still there. MYSTERY #2: The above cisco-spa, when it calls out over the trunk, all is well, wonderful 2-way audio. But when I do the same operation from my yealink phones, I get my cell with one-way audio. It's a classic NAT situation: the phone system is in a droplet at digital ocean, but my phones are here at home behind a NAT. I see only 3 NAT related options: force_rport rtp_symmetric rewrite_contact and I set them all to "yes", and they can call each other, but as explained, in dialing out thru a trunk, the yealinks get one-way audio... Any more NAT options? many thanks... murf -- Steve Murphy ✉ murf at parsetree dot com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] App VoiceMail msg_id is not unique
When multiple mailboxes are passed into the VoiceMail application the records created in the database share the same msg_id. For example: [voicemail] exten => s,1,VoiceMail(2520@my-VOICEMAIL&252@my-VOICEMAIL,s) Will create these two records in the voicemessages table: mysql> SELECT msg_id, msgnum, dir, context, macrocontext, callerid, origtime,duration, mailboxuser, mailboxcontext, flag FROM `voicemessages` WHERE msg_id='1473347378-0005'; +-++---+--+--+++--+-++--+ | msg_id | msgnum | dir | context | macrocontext | callerid | origtime | duration | mailboxuser | mailboxcontext | flag | +-++---+--+--+++--+-++--+ | 1473347378-0005 | 4 | /var/spool/asterisk/voicemail/my-VOICEMAIL/2520/INBOX | voicemail_detect_fax | | "Steven Wheeler" <252> | 1473347378 | 3| 2520| my-VOICEMAIL | | | 1473347378-0005 | 2 | /var/spool/asterisk/voicemail/my-VOICEMAIL/252/INBOX | voicemail_detect_fax | | "Steven Wheeler" <252> | 1473347378 | 3| 252 | my-VOICEMAIL | | +-++---+--+--+++--+-++--+ 2 rows in set (0.00 sec) This test was done with Asterisk 11.6-cert11 running on CentOS release 6.7 (Final) Linux 2.6.32-504.8.1.el6.x86_64 Is this expected behavior? Steven Wheeler -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED] Re: Feature Request: what about "core stop panic" ?
I think were getting closer: I did: - I edited /etc/default/asterisk to include : AST_USER="root" AST_GROUP="root" # systemctl daemon-reload # systemctl start asterisk # ps aux | grep asterisk root 3602 7.1 2.5 60332 26012 ?Ssl 16:00 0:03 /usr/sbin/asterisk -U root -G root -g # rasterisk # pkill -SEGV asterisk Then console showed: Segmentation error (core dumped) and a /core file is created I also tried with asterisk run asterisk user and group and looked for any newly created (find / -cmin -2) : nothing looking like a core file, though I might overlooked it. So, it really look like a configuration issue: "how to specify, with asterisk and systemd among others, where core files are dumped ?". I'll read again this thread advices and report back here. 2016-09-07 21:44 GMT+02:00 George Joseph : > > > On Wed, Sep 7, 2016 at 11:15 AM, Olivier wrote: > >> >> >> 2016-09-06 17:48 GMT+02:00 Tzafrir Cohen : >> >>> On Tue, Sep 06, 2016 at 06:37:52AM -0600, George Joseph wrote: >>> > On Tue, Sep 6, 2016 at 1:55 AM, Olivier wrote: >>> >>> > > Where should core file be created when Asterisk is run as a daemon by >>> > > asterisk user and group ? >>> > > Is there a setting I can use to specify the directory used (so that >>> we can >>> > > make sure appropriate ownership is set) ? >>> > > >>> > >>> > "$ sysctl kernel.core_pattern" will show you where core files are >>> written. >>> > For Asterisk to produce the core file, it has to be started with the >>> '-g' >>> > option so make sure your asterisk.service file is adding the option. >>> >> >> My asterisk.service is : >> >> # cat /run/systemd/generator.late/asterisk.service >> # Automatically generated by systemd-sysv-generator >> >> [Unit] >> SourcePath=/etc/init.d/asterisk >> Description=LSB: Asterisk PBX >> Before=runlevel2.target runlevel3.target runlevel4.target >> runlevel5.target shutdown.target >> After=network-online.target systemd-journald-dev-log.socket >> nss-lookup.target local-fs.target remote-fs.target dahdi.service >> misdn.service lcr.service wanrouter.service mysql.service postgresql.service >> Wants=network-online.target >> Conflicts=shutdown.target >> >> [Service] >> Type=forking >> Restart=no >> TimeoutSec=5min >> IgnoreSIGPIPE=no >> KillMode=process >> GuessMainPID=no >> RemainAfterExit=yes >> SysVStartPriority=2 >> ExecStart=/etc/init.d/asterisk start >> ExecStop=/etc/init.d/asterisk stop >> ExecReload=/etc/init.d/asterisk reload >> >> >> My /etc/init.d/asterisk file is sourcing my /etc/default/asterisk which >> includes an (uncommented) line COREDUMP=yes. >> I also have >> # grep core /etc/asterisk/asterisk.conf >> dumpcore = yes >> >> make menuselect shows: >> [*] DONT_OPTIMIZE >> [*] COMPILE_DOUBLE >> [ ] DEBUG_THREADS >> [*] LOADABLE_MODULES >> [ ] DEBUG_FD_LEAKS >> [*] BETTER_BACKTRACES >> [ ] LOTS_OF_SPANS >> [ ] MALLOC_DEBUG >> [ ] DEBUG_CHAOS >> [*] BUILD_NATIVE >> --- Extended --- >> [ ] REF_DEBUG >> [ ] AO2_DEBUG >> [ ] STATIC_BUILD >> XXX REBUILD_PARSERS >> [ ] LOW_MEMORY >> [ ] DISABLE_INLINE >> [*] OPTIONAL_API >> XXX USE_HOARD_ALLOCATOR >> [ ] RADIO_RELAX >> [ ] G711_NEW_ALGORITHM >> < > G711_REDUCED_BRANCHING >> < > TEST_CODING_TABLES >> < > TEST_TANDEM_TRANSCODING >> [ ] ADDRESS_SANITIZER >> [ ] THREAD_SANITIZER >> [ ] LEAK_SANITIZER >> [ ] UNDEFINED_SANITIZER >> [ ] BUSYDETECT_TONEONLY >> [ ] BUSYDETECT_COMPARE_TONE_AND_SILENCE >> [ ] BUSYDETECT_DEBUG >> [ ] INTEGER_CALLERID >> >> >> Is there a way to read Compiler Flags from a running system without >> looking at source file directory ? >> >> If someone forgets to select appropriate Compiler Flags, does it prevent >> Asterisk to produce a core file (even an empty one) ? >> > > If -g is specified a core file should always be produced. See my earlier > reply. I think one is being produced, it's just not where you think it > should be. :) > > > > >> >> >>> Specifically, if the first character of core_pattern is '!', the rest >>> should be an executable, to which the core file is handled. IIRC Centos7 >>> had something of that type installed by default. On Debian Stable you >>> have the package corekeeper (or maybe also systemd-coredump from >>> backports). I haven't tried any of those. >>> >>> -- >>>Tzafrir Cohen >>> icq#16849755 jabber:tzafrir.co...@xorcom.com >>> +972-50-7952406 mailto:tzafrir.co...@xorcom.com >>> http://www.xorcom.com >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 >>> http://www.asterisk.org/community/astricon-user-conference >>> >>> New to Asterisk? Start here: >>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> -- >> __
