search for: 5063

https://bugs.freedesktop.org/show_bug.cgi?id=99954 Bug ID: 99954 Summary: Errors when using VirtualBox with 3D acceleration: gr: ILLEGAL_CLASS ch 6 [007f7f8000 VirtualBox[5063]] subc 0 class c000 mthd 2390 data 00000000 Product: xorg Version: unspecified Hardware: Other OS: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau Assig...

I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001 at 192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes. I'm not seeing anything in the messages log. I'm sure I'm doing something wrong, just not sure where to look or how to track down...

...mented out, But just incase I also changed it to *7 (We don't use that feature). It appears to be something completely SIP based, As if the call originates from DAHDI, It works fine.. If anyone has any ideas, Please let me know. Thanks! SIP Trace Below <--- SIP read from UDP:208.65.55.170:5063 ---> INVITE sip:*8 at 10.65.6.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576 From: "nicktest" <sip:nicktest at 10.65.6.10>;tag=1470823868 To: <sip:*8 at 10.65.6.10> Call-ID: 695101044 at 172.16.10.101 CSeq: 1 INVITE Contact: <sip:nickt...

...DISTRIB_RELEASE=16.04 DISTRIB_CODENAME=xenial DISTRIB_DESCRIPTION="Ubuntu 16.04.2 LTS" CPU architecture : Linux 4.4.67-1-pve #1 SMP PVE 4.4.67-92 (Fri, 23 Jun 2017 08:22:06 +0200) x86_64 GNU/Linux FIle system : local UID GID Aug 17 11:47:28 azizee dovecot: imap(jra11[*5063*:*5011*]): Debug: Effective uid=5063, gid=5011, home=/var/spool/domaines/vitalnet/jra/ Aug 17 11:47:28 azizee dovecot: imap(jra11[5063:5011]): Debug: Namespace inbox: type=private, prefix=, sep=, inbox=yes, hidden=no, list=yes, subscriptions=yes location=maildir:/var/spool/domaines/vitalnet/jra/...

...han_sip.c:12640 handle_response_invite: Failed to authenticate on INVITE to '<sip:092xx90xx at 85.xx.xx.xx>;tag=as5b139383' And after restarting Asterisk, the CLI is flooded by : [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1: Operation not permitted [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b704e0 (len 543) to my_ip:5062 returned -1: Operation not permitted [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b722b0 (len 543) to my_ip:5061 returned -1...

...ix and with letters in my matrix I can't count. My question. Is it possible to import the file to got 3 columns only with numbers and no letters like x=, y=? Thank's a lot Felix My R Code: ---------- # na.strings = "S=" Measure1 <- matrix(scan("data.dat", n= 5063 * 4, skip = 20, what = character() ), 5063, 3, byrow = TRUE) Measure2 <- matrix(scan("data.dat", n= 5063 * 4, skip = 5220, what = character() ), 5063, 3, byrow = TRUE) My data file: ----------- FILEDATE:02.02.2007 ... START OF HEIGHT DATA S= 0 y=0.0 x=0.00000000 S= 0 y=0.1 x=0.0...

...quot;eric" <sip:10 at 192.168.1.150>") in new stack [Jul 12 14:56:19] -- Executing [10 at from-TEST:3] Dial("SIP/test6-00000094", "SIP/test2") in new stack/ SIP debug : /asterisk*CLI> sip set debug peer test6 SIP Debugging Enabled for IP: 192.168.1.104:5063 [Jul 12 15:02:42] <--- SIP read from 192.168.1.104:5063 ---> INVITE sip:10 at 192.168.1.150 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5063;branch=z9hG4bK-fe158095 From: "test 6" <sip:test6 at 192.168.1.150>;tag=adbbedf0959298ddo3 To: <sip:10 at 192.168.1.150> *Remote-Party-...

Hey all, Does someone want to look over a super-trivial patch I just submitted? It''s just correcting a typo; someone accidentally spelled "group_by_title" as "gruop_by_title" in the category.rb model in the activerecord tests. I just fixed it in the model, and in the two places it''s referenced in the habtm test. Thanks, Ben -- You received this message

...ading multiple layers of the To header, but it still didn't retrieve the newer dialog To headers. I am including the SIP messages reported by Asterisk for the call coming in... *** Phone sends INVITE to Asterisk *** INVITE sip:333 at xxx.xxx.xxx.xxx SIP/2.0^M Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5063;branch=z9hG4bK-18e552c3^M From: "1004" <sip:1004 at xxx.xxx.xxx.xxx>;tag=79e7940882a792ao2^M To: <sip:333 at xxx.xxx.xxx.xxx>^M Call-ID: 3162d378-ea2b2452 at yyy.yyy.yyy.yyy^M CSeq: 102 INVITE^M Max-Forwards: 70^M Authorization: Digest username="1004",realm="ast...

...13525] [ pid ] uid tgid total_vm rss pgtables_bytes swapents oom_score_adj name Jan 21 23:40:00 GS-CEL-L kernel: [55535.913527] [ 13232] 1000 13232 5030 786 77824 103 0 BackupWindows10 Jan 21 23:40:00 GS-CEL-L kernel: [55535.913528] [ 13267] 1000 13267 5063 567 73728 132 0 BackupWindows10 Jan 21 23:40:00 GS-CEL-L kernel: [55535.913529] [ 13421] 1000 13421 5063 458 73728 132 0 BackupWindows10 Jan 21 23:40:00 GS-CEL-L kernel: [55535.913530] [ 13428] 1000 13428 712847 124686 5586944 523997...

...     rss pgtables_bytes swapents oom_score_adj name >> Jan 21 23:40:00 GS-CEL-L kernel: [55535.913527] [  13232]  1000 >> 13232     5030      786    77824      103             0 BackupWindows10 >> Jan 21 23:40:00 GS-CEL-L kernel: [55535.913528] [  13267]  1000 >> 13267     5063      567    73728      132             0 BackupWindows10 >> Jan 21 23:40:00 GS-CEL-L kernel: [55535.913529] [  13421]  1000 >> 13421     5063      458    73728      132             0 BackupWindows10 >> Jan 21 23:40:00 GS-CEL-L kernel: [55535.913530] [  13428]  1000 13428 >&gt...

It seems that what ever I set in my answer handler does not show up in the hangup handler. In order to do billing I can't rely on the g option where the caller hangs up the call. Looks like I can either use h or a hangup handler along with the shared function. On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote: > Don't use an 'h' extension, use

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Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be> Remote-Party-ID: "VC" <sip:voip2 at sip.domain.be>;screen=yes;party=calling Call-ID: 307124bd-f6881ef at 192.168.1.106 CSeq: 101 INVITE Max-Fo...

...ts withing 192.168.1.0 network. My asterisk is running over NAT. I use linksys router. Now, I am trying to connect from outside to my asterisk server. I use Diax as iax client. For some reason I cannot connect to my server from outside. On my router I forward those ports to my asterisk server. 5060-5063 10000-20000 5036 4569 It works ok with broadvoice, but clinets cannot connect to the server. This is my iax.conf file [general] port=5036 tos=lowdelay jitterbuffer=no disallow=all allow=ulaw allow=ilbc allow=gsm allow=adpcm allow=alaw register => xxx:xxx@gwiax-in-01.voicepulse.com [guest] typ...

...UNREACHABLE 2006/2006 10.1.1.10 5066 UNREACHABLE 2005/2005 10.1.1.10 5065 UNREACHABLE 2004/2004 10.1.1.10 5064 UNREACHABLE 2003/2003 10.1.1.10 5063 UNREACHABLE 2002/2002 10.1.1.10 5062 UNREACHABLE 2001/2001 10.1.1.10 5061 UNREACHABLE Sangoma02/Sangoma02 10.1.1.10 5060 OK (1 ms) 12 sip peers [1 online , 11 offline] Here are my config...

...> total_vm      rss pgtables_bytes swapents oom_score_adj name > Jan 21 23:40:00 GS-CEL-L kernel: [55535.913527] [  13232]  1000 > 13232     5030      786    77824      103             0 BackupWindows10 > Jan 21 23:40:00 GS-CEL-L kernel: [55535.913528] [  13267]  1000 > 13267     5063      567    73728      132             0 BackupWindows10 > Jan 21 23:40:00 GS-CEL-L kernel: [55535.913529] [  13421]  1000 > 13421     5063      458    73728      132             0 BackupWindows10 > Jan 21 23:40:00 GS-CEL-L kernel: [55535.913530] [  13428]  1000 13428 > 712847   124...

...phone rings, I pick up, and the conversation is terminated. Every time. The setup : Grandstream GXP2010 --> SIPproxy (Endian Firewall) --> Asterisk Server --> ITSP Could it be the SIP proxy of my Endian firewall ?? I have 4 accounts on the Grandstream which listen on port 5060 --> 5063. They have a proxy defined namely my Endian firewall. On this SIPproxy I have a port range defined 11500 --> 11600. Default SIP port is 5060. My Asterisk server sees the 4 accounts as follow : > yocangrandstream/yocangra xx.xx.xx.65 D N 5060 OK (50 ms) > VCfa...

...tatic IP (no nat) and 6 Linksys SPA941. All SPA are after a router with NAT: * SPA-1 and SPA-2 are on the same network, we have a pat 5060 => SPA-1 and 5061=> SPA-2 on the internet router * SPA-3, we have a pat 5062 => SPA-3 * SPA-4, we have a pat 5063 => SPA-4 * SPA-5, we have a pat 5064 => SPA-5 * SPA-6, we have a pat 5065 => SPA-6 On the Asterisk Sip conf, we have nat=yes and dynamic host. The problems are SPA-1 and SPA-2 can call to all other SPA except SPA-3 with SPA-3, i speak, it's good, spa-3 ha...

...act" entries which I take to mean that both phones have registered: Endpoint: 101 Not in use 0 of inf InAuth: 101/101 Aor: 101 3 Contact: 101/sip:101 at 192.168.2.193:5063 Avail 178.681 Contact: 101/sip:101 at 192.168.2.197:58086;transport=UDP;r Avail 4.198 Transport: transport-udp udp 0 0 0.0.0.0:5060 I have tried with several phones and have rebooted the Asterisk server and phones several times jus...