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Hello! Client is using ulaw, however server sometimes fills the log with following: [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)

Hello! I have just upgraded to asterisk 1.8.2.1 and see some weird messages in log when client tries to register: [2011-01-19 00:52:47] WARNING[25624] chan_sip.c: Failed to parse contact info [2011-01-19 00:52:50] NOTICE[25624] chan_sip.c: Peer '0010101' is now UNREACHABLE! Last qualify: 105 [2011-01-19 00:53:03] VERBOSE[25624] chan_sip.c: -- Registered SIP '0010101' at

Hello using asterisk 1.8, compiled res_fax.so and res_fax_spandsp.so - both loaded successfuly in sip.conf set t38pt_udptl=yes but faxes still don't work even in passthru mode. if i do a 'sip show channel' on the channel via which i am sending fax it shows: T.38 support Yes however if i do sip show channel of my channel (from other server) it shows T.38 support

...4 sip_write: Asked to transmit frame type ulaw, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) [2011-03-28 18:22:27] WARNING[22502]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) -- Nick Ustinov -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110328/0ba14c76/attachment.htm>

Hi, my log is full of errors from this mobile user: -- Registered SIP '0010106' at 212.93.97.135:7759 [2011-05-17 17:44:05] NOTICE[21456]: chan_sip.c:19804 handle_response_peerpoke: Peer '0010106' is now Reachable. (381ms / 10000ms) [2011-05-17 17:44:06] ERROR[21456]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("212.93.97.135:7759", "7759", ...): Name

Any help would be much appreciated: $refcheck = defined(File["/home/$name/.ssh"]) if $refcheck {} else { file { "/home/${name}/.ssh": ensure => directory, owner => $name, group => $name, mode => 700 } } puppetd says: err: Syntax error at ''File'' at

dd if=/dev/zero of=sl1 bs=512 count=256000 dd if=/dev/zero of=sl2 bs=512 count=256000 dd if=/dev/zero of=sl3 bs=512 count=256000 dd if=/dev/zero of=sl4 bs=512 count=256000 zpool create -m /export/test1 test1 raidz /export/sl1 /export/sl2 /export/sl3 zpool add -f test1 /export/sl4 dd if=/dev/zero of=sl4 bs=512 count=256000 zpool scrub test1 panic. and message like on image. This message posted

Hello We have recently upgraded to Realtime engine (sip buddies and extensions) and now have problems with calling local SIP users. I have rtcachefriends=yes but tried with 'no' and it's even worse. (asterisk 1.8.1.1 + realtime mysql) Here's an example: User 1000 registers successfully and can then be called with Dial(SIP/1000,30) successfully After some time when I try to call

Hi. Show you how to install oVirt in FC11/FC12? I acted on instructions http://ovirt.et.redhat.com/install-instructions.html, but there were problems with the versions of the module locale (requires 2.0.4 and installed 2.0.5). WBR, Fyodor.

Yesterday, I switched our node structure from using node basenode, node nodetype inherits basenode, to using node classes per the changes discussed recently on this list and as defined in the wiki:GlossaryOfTerms. In the manifest, we have a syslog module which sets up all the components of syslog-ng we''re using. The ''syslog'' class is used by default on clients and then

...dir instead of datadir for a sound file (Reported by Tzafrir Cohen) * ASTERISK-26739 - voicemail API test: confuses expected and actual values (Reported by Tzafrir Cohen) * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every sorcery memory cache populate (Reported by Ustinov Artem) * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0 (Reported by Aaron An) * ASTERISK-26672 - Crash when setting remote address on RTP instance (Reported by Richard Mudgett) * ASTERISK-26670 - [...

...dir instead of datadir for a sound file (Reported by Tzafrir Cohen) * ASTERISK-26739 - voicemail API test: confuses expected and actual values (Reported by Tzafrir Cohen) * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every sorcery memory cache populate (Reported by Ustinov Artem) * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0 (Reported by Aaron An) * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP (Reported by Alexander Traud) * ASTERISK-26691 - Remember...

In the normal override method, you can change the value of a parameter, but can you unset a parameter? file { "/etc/somefile": mode => 644, owner => "dude" } File ["/etc/somefile"] { mode => unset } ??

...of datadir for a sound file (Reported by Tzafrir Cohen) * ASTERISK-26739 - voicemail API test: confuses expected and actual values (Reported by Tzafrir Cohen) * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every sorcery memory cache populate (Reported by Ustinov Artem) * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0 (Reported by Aaron An) * ASTERISK-26672 - Crash when setting remote address on RTP instance (Reported by Richard Mudgett) *...

Since I had my type development epiphany a couple of days ago, I''ve decided to write down my understanding of developing simple types, at http://reductivelabs.com/trac/puppet/wiki/PracticalTypes. I''d appreciate comments from people who already know how to develop types as to correctness, and also comments from people who are new to type development about whether it''s a

There''s definitely enough complexity in some of the modules out there to warrant solid test coverage, especially if people start extending a module to support more distributions and OSes, while trying to keep the existing support working. That''s even before you start thinking about functions, facts, and native types. They''re *really* in need of solid testing, being all

...dd support for SRV (Reported by Joshua Colp) * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level does not work. (Reported by Richard Mudgett) * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every sorcery memory cache populate (Reported by Ustinov Artem) * ASTERISK-26739 - voicemail API test: confuses expected and actual values (Reported by Tzafrir Cohen) * ASTERISK-26740 - voicemail API test: uses varlibdir instead of datadir for a sound file (Reported by Tzafrir Cohen) * ASTERISK-26665 - app_queue: Agent ri...

...dd support for SRV (Reported by Joshua Colp) * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level does not work. (Reported by Richard Mudgett) * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every sorcery memory cache populate (Reported by Ustinov Artem) * ASTERISK-26739 - voicemail API test: confuses expected and actual values (Reported by Tzafrir Cohen) * ASTERISK-26740 - voicemail API test: uses varlibdir instead of datadir for a sound file (Reported by Tzafrir Cohen) * ASTERISK-26665 - app_queue: Agent ri...

...rowse/ASTERISK-26743>] - PJPROJECT: Detecting compiled max log level does not work. (Reported by Richard Mudgett) - [ASTERISK-26731 <https://issues.asterisk.org/jira/browse/ASTERISK-26731>] - res_sorcery_memory_cache: memory leak on every sorcery memory cache populate (Reported by Ustinov Artem) - [ASTERISK-26739 <https://issues.asterisk.org/jira/browse/ASTERISK-26739>] - voicemail API test: confuses expected and actual values (Reported by Tzafrir Cohen) - [ASTERISK-26740 <https://issues.asterisk.org/jira/browse/ASTERISK-26740>] - voicemail API test: uses...

Hello ! My asterisk log is full of messages like this: [2011-03-06 19:01:15] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06 19:01:20] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06 19:01:25] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06