Re: [asterisk-dev] [Asterisk 16.x / pjsip TLS] Memory leak with core reload
On Thu, Oct 3, 2019 at 3:58 PM Michael Maier wrote: > On 03.10.19 at 15:52 Michael Maier wrote: > > On 02.10.19 at 22:45 Sean Bright wrote: > >> On 10/2/2019 4:02 PM, Michael Maier wrote: > >>> I found one more problem regarding the configuration options, provided > by FreePBX, which should be supported by asterisk. > >>> I'm referring to the possibility, to add additional options not > supported by FreePBX using special config files like > >>> pjsip.registration_custom_post.conf or pjsip.aor_custom_post.conf and > pjsip.endpoint_custom_post.conf or pjsip.transports_custom_post.conf. > >>> The last two files are working pretty fine as expected, but the first > two just don't work. > >>> > >>> I'm configuring in pjsip.registration_custom_post.conf for example: > >>> > >>> [extName](+) > >>> key=value > >>> > >>> Asterisk reads it (asterisk complains if it doesn't know the key), but > asterisk doesn't apply the provided value for a known key - it's always the > default value. That's > >>> strange, too. > >> > >> Please file an issue[1] with a configuration that exhibits this. > > > > https://issues.asterisk.org/jira/browse/ASTERISK-28563 > > Thanks to all clarifying the correct way to configure it with FreePBX: > > [extName](+type=registration) > key=value > > eg. > > This really works as expected! > > I'm now wondering if I missed some documentation about this feature, > because I searched the whole web and only found others having the same > problem but never a solution! > > I'm sorry for the noise! > Actually, you caused the documentation [1] of the functionality to be improved. I don't think it was actually documented anywhere except in the code and the issue that added the functionality. Richard [1] https://wiki.asterisk.org/wiki/display/AST/Adding+to+an+existing+section -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Asterisk 16.x / pjsip TLS] Memory leak with core reload
On 03.10.19 at 15:52 Michael Maier wrote: > On 02.10.19 at 22:45 Sean Bright wrote: >> On 10/2/2019 4:02 PM, Michael Maier wrote: >>> I found one more problem regarding the configuration options, provided by >>> FreePBX, which should be supported by asterisk. >>> I'm referring to the possibility, to add additional options not supported >>> by FreePBX using special config files like >>> pjsip.registration_custom_post.conf or pjsip.aor_custom_post.conf and >>> pjsip.endpoint_custom_post.conf or pjsip.transports_custom_post.conf. >>> The last two files are working pretty fine as expected, but the first two >>> just don't work. >>> >>> I'm configuring in pjsip.registration_custom_post.conf for example: >>> >>> [extName](+) >>> key=value >>> >>> Asterisk reads it (asterisk complains if it doesn't know the key), but >>> asterisk doesn't apply the provided value for a known key - it's always the >>> default value. That's >>> strange, too. >> >> Please file an issue[1] with a configuration that exhibits this. > > https://issues.asterisk.org/jira/browse/ASTERISK-28563 Thanks to all clarifying the correct way to configure it with FreePBX: [extName](+type=registration) key=value eg. This really works as expected! I'm now wondering if I missed some documentation about this feature, because I searched the whole web and only found others having the same problem but never a solution! I'm sorry for the noise! Thanks Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Asterisk 16.x / pjsip TLS] Memory leak with core reload
On 10/3/2019 10:17 AM, Michael Maier wrote: This is not a FreePBX issue OK. Thank you for clarifying. provided by asterisk (take a look at the source code) That's a good idea. I'll take a look, thanks. Kind regards, Sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Asterisk 16.x / pjsip TLS] Memory leak with core reload
On 03.10.19 at 16:08 Jared Smith wrote: > On Thu, Oct 3, 2019 at 10:01 AM Sean Bright wrote: > >> In the future, please feel free to skip the mailing list and submit >> issues directly to https://issues.asterisk.org/jira for any Asterisk >> problems. >> >> FreePBX issues like this one can go directly to their issue tracking >> system (I don't know the URL for that off-hand). >> > > The FreePBX issue tracker is at https://issues.freepbx.org This is not a FreePBX issue as the feature [extName](+) is provided by asterisk (take a look at the source code) and not FreePBX - FreePBX e.g. is just using it. Regards Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Asterisk 16.x / pjsip TLS] Memory leak with core reload
On Thu, Oct 3, 2019 at 10:01 AM Sean Bright wrote: > In the future, please feel free to skip the mailing list and submit > issues directly to https://issues.asterisk.org/jira for any Asterisk > problems. > > FreePBX issues like this one can go directly to their issue tracking > system (I don't know the URL for that off-hand). > The FreePBX issue tracker is at https://issues.freepbx.org -Jared -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Asterisk 16.x / pjsip TLS] Memory leak with core reload
On 10/3/2019 9:52 AM, Michael Maier wrote: I think this should be enough - just install FreePBX and you will see it. You don't need any special configuration - it's the default configuration provided by FreePBX. Great. Hopefully someone on the FreePBX team also follows the Asterisk issue tracker. In the future, please feel free to skip the mailing list and submit issues directly to https://issues.asterisk.org/jira for any Asterisk problems. FreePBX issues like this one can go directly to their issue tracking system (I don't know the URL for that off-hand). Kind regards, Sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Asterisk 16.5.x / SIPS / SRTP / pjsip 4.9 - one more memory leak
On 03.10.19 at 15:42 Michael Maier wrote: > there is one more memory leak even in asterisk 16.6.0-rc2, which can't be > seen with pjsip 4.8 instead of 4.9. It can be seen on inbound calls (not sure > if it's > on outbound calls, too) using SIPS and SRTP. https://issues.asterisk.org/jira/browse/ASTERISK-28564 Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Asterisk 16.x / pjsip TLS] Memory leak with core reload
On 02.10.19 at 22:45 Sean Bright wrote: > On 10/2/2019 4:02 PM, Michael Maier wrote: >> I found one more problem regarding the configuration options, provided by >> FreePBX, which should be supported by asterisk. >> I'm referring to the possibility, to add additional options not supported by >> FreePBX using special config files like >> pjsip.registration_custom_post.conf or pjsip.aor_custom_post.conf and >> pjsip.endpoint_custom_post.conf or pjsip.transports_custom_post.conf. >> The last two files are working pretty fine as expected, but the first two >> just don't work. >> >> I'm configuring in pjsip.registration_custom_post.conf for example: >> >> [extName](+) >> key=value >> >> Asterisk reads it (asterisk complains if it doesn't know the key), but >> asterisk doesn't apply the provided value for a known key - it's always the >> default value. That's >> strange, too. > > Please file an issue[1] with a configuration that exhibits this. https://issues.asterisk.org/jira/browse/ASTERISK-28563 I think this should be enough - just install FreePBX and you will see it. You don't need any special configuration - it's the default configuration provided by FreePBX. Thanks Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Asterisk 16.5.x / SIPS / SRTP / pjsip 4.9 - one more memory leak
On 10/3/2019 9:42 AM, Michael Maier wrote: Sorry, but there is one more memory leak even in asterisk 16.6.0-rc2, which can't be seen with pjsip 4.8 instead of 4.9. It can be seen on inbound calls (not sure if it's on outbound calls, too) using SIPS and SRTP. Examples: 1 Call, duration about 1 h: ~ +1,2 MB 5 short calls (< 1 minute): ~ +1 MB Please file an issue[1] with a configuration that exhibits this. [1] https://issues.asterisk.org/jira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Asterisk 16.5.x / SIPS / SRTP / pjsip 4.9 - one more memory leak
Hello again! Sorry, but there is one more memory leak even in asterisk 16.6.0-rc2, which can't be seen with pjsip 4.8 instead of 4.9. It can be seen on inbound calls (not sure if it's on outbound calls, too) using SIPS and SRTP. Examples: 1 Call, duration about 1 h: ~ +1,2 MB 5 short calls (< 1 minute): ~ +1 MB Example for the inbound INVITE and OK package: <--- Received SIP request (2276 bytes) from TLS:217.0.20.195:5061 ---> INVITE sip:+491234567890@12.13.14.15:5061;transport=tcp;line=abcdefg SIP/2.0 Max-Forwards: 49 Via: SIP/2.0/TLS 217.0.20.195:5061;branch=z9hG4bKg3Zqkv7ivdsp3wo1jhdbdvgy5dwsq6jye To: From: ;tag=h7g4Esbg_p65540t1570108521m378032c299263169s1_1621954413-1461120854 Call-ID: p65540t1570108521m378032c299263169s2 CSeq: 1 INVITE Contact: ;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Record-Route: Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" History-Info: ;index=1 Min-Se: 900 P-Asserted-Identity: P-Asserted-Identity: Session-Expires: 1800 Supported: timer Supported: 100rel Supported: histinfo Supported: 199 Supported: uui Supported: norefersub Content-Type: application/sdp Content-Length: 1061 Session-ID: 253f41678c65f936805ef6b071943e64 Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, UPDATE, PRACK, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE v=0 o=- 1011696818 1621954173 IN IP4 217.0.20.195 s=- c=IN IP4 217.0.135.5 t=0 0 m=audio 27888 RTP/SAVP 96 97 9 98 99 100 101 8 102 103 b=AS:84 a=rtpmap:96 AMR-WB/16000 a=fmtp:96 mode-set=0,1,2; mode-change-period=2; mode-change-neighbor=1; max-red=0 a=rtpmap:97 AMR-WB/16000 a=fmtp:97 mode-change-capability=2; max-red=0 a=rtpmap:9 G722/8000 a=rtpmap:98 AMR/8000 a=fmtp:98 mode-set=0,2,4,7; mode-change-period=2; mode-change-neighbor=1; max-red=0 a=rtpmap:99 AMR/8000 a=fmtp:99 mode-set=0,2,4; mode-change-period=2; mode-change-neighbor=1; max-red=0 a=rtpmap:100 AMR/8000 a=fmtp:100 mode-set=0,1,2,3,4,5,6,7; mode-change-period=2; mode-change-neighbor=1; max-red=0 a=rtpmap:101 AMR/8000 a=fmtp:101 mode-set=0,1,2,3,4,5,6,7; max-red=0 a=rtpmap:8 PCMA/8000 a=rtpmap:102 telephone-event/8000 a=rtpmap:103 telephone-event/16000 a=ptime:20 a=maxptime:30 a=3ge2ae:applied a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:HTNhK8lOYS+/1ORuNEbEhnsisXj4PEVIh8FBKmTR a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:6qBJEfKKXxbpJTepS298yUmUl/891GwnlURC3tdn <--- Transmitting SIP response (1178 bytes) to TLS:217.0.20.195:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 217.0.20.195:5061;rport=5061;received=217.0.20.195;branch=z9hG4bKg3Zqkv7ivdsp3wo1jhdbdvgy5dwsq6jye Record-Route: Call-ID: p65540t1570108521m378032c299263169s2 From: ;tag=h7g4Esbg_p65540t1570108521m378032c299263169s1_1621954413-1461120854 To: ;tag=94f22858-9c32-44c5-8a45-76964f62684a CSeq: 1 INVITE Server: FPBX-14.0.11(16.5.1) Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Contact: Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800;refresher=uac Require: timer Content-Type: application/sdp Content-Length: 368 v=0 o=- 1011696818 1621954176 IN IP4 12.13.14.15 s=Asterisk c=IN IP4 12.13.14.15 t=0 0 m=audio 10032 RTP/SAVP 9 8 102 a=3ge2ae:requested a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:OnkHAdHasSl83UnyFNuDSrBx+OsRF8DRZ6c5PnmJ a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:102 telephone-event/8000 a=fmtp:102 0-16 a=ptime:20 a=maxptime:150 a=sendrecv Thanks Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
