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https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=567 ------- Additional Comments From kaber@trash.net 2007-05-14 14:28 MET ------- There are two len fields, caplen and len. Which one is wrong? -- Configure bugmail: https://bugzilla.netfilter.org/bugzilla/userprefs.cgi?tab=email ------- You are receiving this mail because: ------- You are the assignee for the bug, or are watching the

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context...

Am 13.06.2020 um 18:06 schrieb Michael Keuter: > So the call used Alaw as Codec. Yes, so seems it to be... It should has the better quality... But the calls done using my mobile phone in VoIP with the Asterisk have better quality as the calls done using the normal VoIP-telefon... I'm really puzzled... Luca Bertoncello (lucabert at lu...

What is the best asterisk platform to use? What are you guys using? I am looking for something to host either in our data center or at the customer prem where I have the control over the unit and not through a contractor. I dont mind paying a license fee for a front end interface but still would rather not have to pay. Thanks, --Eric -------------- next part -------------- An HTML attachment

On Saturday 13 June 2020 at 17:23:14, Luca Bertoncello wrote: > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > > Try "sip show peer <peername>" for a phone. > bpi*CLI> sip show peer 0049177xxxxxxx > Codecs : > (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin| > slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|s...

On Saturday 13 June 2020 at 18:26:53, Luca Bertoncello wrote: > Am 13.06.2020 um 18:06 schrieb Michael Keuter: > > So the call used Alaw as Codec. > > Yes, so seems it to be... > It should has the better quality... But the calls done using my mobile > phone in VoIP with the Asterisk have better quality as the calls done > using the normal VoIP-telefon... You are *assuming* that it...

...tion. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.2 Thank you for your continued support of Asterisk!

...tion. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.2 Thank you for your continued support of Asterisk!

...destination when 'sendrpid=yes' (in proxy environment) (Reported by Karsten Wemheuer) * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT (Reported by Kristian H??gh) * ASTERISK-20744 - [patch] Security event logging does not work over syslog (Reported by Michael Keuter) * ASTERISK-23850 - Park Application does not respect Return Context Priority (Reported by Andrew Nagy) * ASTERISK-23991 - [patch]asterisk.pc file contains a small error in the CFlags returned (Reported by Diederik de Groot) * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail...

...destination when 'sendrpid=yes' (in proxy environment) (Reported by Karsten Wemheuer) * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT (Reported by Kristian H??gh) * ASTERISK-20744 - [patch] Security event logging does not work over syslog (Reported by Michael Keuter) * ASTERISK-23850 - Park Application does not respect Return Context Priority (Reported by Andrew Nagy) * ASTERISK-23991 - [patch]asterisk.pc file contains a small error in the CFlags returned (Reported by Diederik de Groot) * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail...

...25610 - Asterisk crash during "sip reload" (Reported by Dud??s J??zsef) * ASTERISK-25498 - Asterisk crashes when negotiating g729 without that module installed (Reported by Ben Langfeld) * ASTERISK-25476 - chan_sip loses registrations after a while (Reported by Michael Keuter) * ASTERISK-25593 - fastagi: record file closed after sending result (Reported by Kevin Harwell) * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but it's assumed to (Reported by Walter Doekes) * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by Jo...

...channels when using the newer chan_dahdi.conf sections for channel configuration. (Closes issue ASTERISK-18496. Reported by Sean Darcy. Patched by Richard Mudgett) * Remove unnecessary libpri dependency checks in the configure script. (Closes issue ASTERISK-18535. Reported by Michael Keuter. Patched by Richard Mudgett) * Update get_ilbc_source.sh script to work again. (Closes issue ASTERISK-18412) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.7.0 Thank you for your continued support o...

...channels when using the newer chan_dahdi.conf sections for channel configuration. (Closes issue ASTERISK-18496. Reported by Sean Darcy. Patched by Richard Mudgett) * Remove unnecessary libpri dependency checks in the configure script. (Closes issue ASTERISK-18535. Reported by Michael Keuter. Patched by Richard Mudgett) * Update get_ilbc_source.sh script to work again. (Closes issue ASTERISK-18412) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.7.0 Thank you for your continued support o...

...Joseph) * ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10 (Reported by Jonathan Harris) * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) * ASTERISK-26468 - ari: Bridge events stop working after this sequence of ARI calls (Reported by Daniele Pallastrelli) * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types. (Reported by Alexander Traud) * ASTERISK-26549 - app_dial: When PickupChan() is use...

...tch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no (Reported by Alexei Gradinari) * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) Improvements made in this release: ----------------------------------- * ASTERISK-23828 - pjsip - Need a command to list active SIP subscriptions (Reported by Rusty Newton) * ASTE...

...Joseph) * ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10 (Reported by Jonathan Harris) * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) * ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state (Reported by Joshua Colp) * ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used (Reported by Frankie Chin) * ASTERI...

...STERISK-25498 - Asterisk crashes when negotiating g729 without that module installed (Reported by Ben Langfeld) * ASTERISK-25595 - Unescaped : in messge sent to statsd (Reported by Niklas Larsson) * ASTERISK-25476 - chan_sip loses registrations after a while (Reported by Michael Keuter) * ASTERISK-25598 - res_pjsip: Contact status messages are printing a hash instead of the uri (Reported by George Joseph) * ASTERISK-25600 - bridging: Inconsistency in BRIDGEPEER (Reported by Jonathan Rose) * ASTERISK-25582 - Testsuite: Reactor timeout error in tests/fax/pjsip...

...24474 - sip_to_pjsip.py lacks documentation and does not function (Reported by John Kiniston) * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT (Reported by Kristian H??gh) * ASTERISK-20744 - [patch] Security event logging does not work over syslog (Reported by Michael Keuter) * ASTERISK-24665 - Configure check required for pjsip_get_dest_info() (Reported by Mark Michelson) * ASTERISK-23850 - Park Application does not respect Return Context Priority (Reported by Andrew Nagy) * ASTERISK-23991 - [patch]asterisk.pc file contains a small error in the CF...

...24474 - sip_to_pjsip.py lacks documentation and does not function (Reported by John Kiniston) * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT (Reported by Kristian H??gh) * ASTERISK-20744 - [patch] Security event logging does not work over syslog (Reported by Michael Keuter) * ASTERISK-24665 - Configure check required for pjsip_get_dest_info() (Reported by Mark Michelson) * ASTERISK-23850 - Park Application does not respect Return Context Priority (Reported by Andrew Nagy) * ASTERISK-23991 - [patch]asterisk.pc file contains a small error in the CF...

Am 13.06.2020 um 22:56 schrieb Antony Stone: Hi again, > 2b. Take your Thomson telephone to some other location with Internet access, > let it register to your home Asterisk server, and them make a call to the same > number yet again. I'm sure you can get the Thomson to connect to Asterisk via > some external network, since you say you can do this from your Android phone.