similar to: Asterisk uses 3 seconds to send ACK after OK

A long formula which is converted using as.character, looses its last part: ``diagonal = 1e-12)'' Shorter formula is ok though. Best, H??vard ************ Browse[2]> formula.str y ~ -1 + b1 + b2 + b3 + b4 + b5 + b6 + b7 + b8 + b9 + b10 + b11 + b12 + b13 + b14 + b15 + b16 + b17 + b18 + b19 + b20 + b21 + b22 + b23 + b24 + b25 + b26 + b27 + b28 + b29 + b30 + b31 + b32 +

I started sharing out zfs filesystems via NFS last week using sharenfs=on. That seems to work fine until I reboot. Turned out the NFS server wasn''t enabled - I had to enable nfs/server, nfs/lockmgr and nfs/status manually. This is a stock SXCR b49 (ZFS root) install - don''t think I''d changed anything much. Shouldn''t a ZFS share be permanently enabling NFS?

We've a Samba 4 domain (no AD, just DC) with LDAP backend on Ubuntu 14.04. This server has been migrated from files backend to LDAP by the previous maintainer, I know the version is pretty old but we cannot update at the moment. The domain works fine with some W7 and W10 (updated from 7) computers, but we have purchased a new Lenovo laptop with Win10 which joined the domain seamlessly but

I have a Dell 500w UPS and it works great with Ubuntu 12.04 LTS and Network UPS tools 2.6.3 via usb. This powers a standalone server for the purposes of clean shutdowns. This was installed using "sudo apt-get install nut" and there are no issues with the Server > nut > ups chain. I am using the usbhid-ups driver and haven't had any issues. I am trying to also add an NAS to the

I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100

Hi All, I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6, But when making A Call from SIP Client, I got cli Warning ... and no call has been made. My Sip Client is using lib java peers client http://peers.sourceforge.net/ with standard codec PCMU/PCMA [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8

Hi, I created an IAX2 trunk between my old Asterisk 1.4 server (A) and my new one with v. 16.10.0 (B). The trunk seems to be up, and the calls are initiated, eg. an extension from A can dial an extension in B which rings. However, as soon as the extension in B answers, the call is terminated. This is what I see in the console of B: -- Called PJSIP/4053 -- PJSIP/4053-00000002 is ringing

Please help find the cause of strange behavior res_pjsip. Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up! Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4d No diagnostic messages why asterisk suddenly decided to hangup i don't found :( There are suggestions or strong belief

OK, I need a bit of help here. I'm configuring a new Asterisk 11 system and I accidentally let my firewall rules drop for a day or so. When I logged in today, I found messages like the ones below on my asterisk console. Obviously somebody was trying to take advantage of my carelessness. So can someone explain what would cause these types of messages to show up on my console? I understand

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

I am currently running Asterisk 13.1.0-1 I have a chan_sip configuration that works fine with a 3rd party. Third party does not use authentication or registration, it's ip based authentication... When I try switching to PJSIP.conf, I seeing 488 responses from the Asterisk side. What has me really baffled is the debugging indicates [Jul 14 17:28:24] DEBUG[3620] pjsip: sip_endpoint.c

3rd attempt to post it to the list, please ignore if it is duplicate I have the following problem When trying to setup asterisk 13.1 with PJSIP to connect to my IPV6 capable SIP provider the registration fails. [code][Dec 22 19:24:24] DEBUG[25247] pjsip: tsx0x110736c .Transaction created for Request msg REGISTER/cseq=36181 (tdta0x721d90) [Dec 22 19:24:24] DEBUG[25247] pjsip:

I am having an issue that prevents WebSockets over SSL/TLS (or any kind of encrypted HTTP traffic to Asterisk) from working after an openssl library update. My setup is CentOS 6 x86_64, and initially, with openssl[-devel]-1.0.0-20.el6_2.5.x86_64 . With this openssl versions, https over TCP port 8089 initializes correctly with asterisk-11.7.0. After an upgrade to

Hi List, we're calling a sip phone from our Asterisk Server, and try to add logic depending on the dialstatus Stripped down example; exten = 494XXXXXXXXX,n,Dial(SIP/4120089,15,w) exten = 494XXXXXXXXX,n,Goto(98-${DIALSTATUS},1) exten = 494XXXXXXXXX,n,Hangup() ..... exten = 98-BUSY,1,NoOp(Busy) exten = 98-BUSY,n,ExecIf($["${Voicemail}" =

Hello! Ater several successful SRTP-enabled calls with SRTP set to Mandatory, asterisk starts to give the following warnings in Log: WARNING[13714] res_srtp.c: SRTP unprotect: authentication failure (continiously) and client hears no sound. After i restart the client program it works fine again for a while. Then the same warning again. Asterisk 1.8.1.1, RealTime engine, sip peer has

Hi, After upgrading to v2.2.7 yesterday, I am starting to get a larger number of bugs occurring -- unfortunately I hadn't configured things to save core dumps (now done). But I am seeing things like: dovecot: imap(user at example.com): Fatal: master: service(imap): child 27931 killed with signal 11 (core dumped) kernel: [151706.763475] imap[4878]: segfault at 7fff53b0aff8 ip

Hi, Have just been testing asterisk 1.8.0, 1.8.0-rc5 and a trunk version from about half an hour ago. IAX Friends (Zoiper Softphones) don't stay registered for more than a few seconds they start out with status unknown and quickly become unreachable, I am using realtime with postgresql, with tables and configuration that have worked fine for IAX in 1.6 and a trunk release from a few months

Hello folks, since I started with asterisk 1.8.2 I got this messages in my console when finish a call. -- Executing [1610 at from-e1:1] Dial("SIP/xxx-00000027", "SIP/1610,60") in new stack == Using SIP RTP CoS mark 5 -- Called 1610 -- SIP/1610-00000028 is ringing -- SIP/1610-00000028 answered SIP/xxx-00000027 -- Locally bridging SIP/xxx-00000027 and

Hello I have a weird behaviour with our local GSM (3G) provider -- several SIP clients crash on the android phone, when switching to 3G network, and in asterisks logs it looks like this - client registers on server successfull and then crashesh immediately. Here's suspicious part of asterisk log: [2011-08-22 19:38:12] ERROR[28605]: netsock2.c:263 ast_sockaddr_resolve:

Am getting netsock error like this when using IAX2, Connected to Asterisk 11.2.1 currently running on indiaprimaryast01 (pid = 4270) == Using SIP RTP CoS mark 5 -- Executing [2001 at Test:1] Dial("SIP/4090-00000005", "SIP/2001 at IAX2/IND-MAN,30") in new stack [Jun 23 06:31:36] NOTICE[4383][C-00000005]: chan_sip.c:29491 sip_request_call: Conflicting extension values