search for: ontent

...--rsh="/usr/bin/ssh -i \ /Users/kurt/.ssh/id_rsa" root at home.thisisnotajoke.com:/home \ /Volumes/Backup/.sync/server/ root at home.thisisnotajoke.com's password: overflow: xflags=0x7a l1=255 l2=784 lastname=/home/cpanelBackup/cpbackup/daily/haspixie/homedir/public_html/wp-c ontent/cache/supercache/icanhaspixiedust.com/category/characters-and-stars/m ickey-mouse/http%3A/icanhaspixiedust.com/category/characters-and-stars/micke y-mouse/&r=http%3A/images.google.com/imgres%3Fimgurl%3Dhttp%3A/icanhaspixied ust.wordpress.com/files/2009/04/waltalittlehelp.jpg%26imgrefurl%3Dhttp%...

...lt;sip:13033372500 at xxx.yyy.34.195>;tag=as4fe079a5 Call-ID: call-71EEDB19-F102-2A10-0B0C-1B at xxx.yyy.34.210 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:13033372500 at xxx.yyy.34.195> ontent-Type: application/sdp Content-Length: 261 v=0 o=root 2235 2235 IN IP4 xxx.yyy.34.195 s=session c=IN IP4 xxx.yyy.34.195 t=0 0 m=audio 10484 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20...

...-4d6ef889@10.253.4.50 Contact: <sip:eden-1000a@10.253.4.50> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50 s=Polycom IP Phone c=IN IP4 10.253.4.50 t=0 0 m=audio 2228 RTP/AVP 0 18 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 11 li...

...nsport=udp>;expires=600000;+g.oma.sip-im;language="en,fr";+g.3gpp.smsip;+g.oma.sip-im.large-message;audio;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-application.ims.iari.gsma-vs";+g.3gpp.cs-voice > Call-ID: 3e130372-8de8-a6b4-6250-36307973a88f > CSeq: 2010551627 REGISTER > ontent-Length: 0 > Max-Forwards: 70 > Authorization: Digest username="imsdroid",realm="asterisk",nonce="39f59329",uri="sip:asterisk",response="2ba48ca360d592ca183ba6706e6feae9",algorithm=MD5 > Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOT...

...SUBSCRIB 0200 45 2c 20 49 4e 46 4f 0d 0a 55 73 65 72 2d 41 67 E, INFO. .User-Ag 0210 65 6e 74 3a 20 58 2d 4c 69 74 65 20 72 65 6c 65 ent: X-L ite rele 0220 61 73 65 20 31 31 30 33 6b 20 73 74 61 6d 70 20 ase 1103 k stamp 0230 35 33 36 32 31 0d 0a 43 6f 6e 74 65 6e 74 2d 4c 53621..C ontent-L 0240 65 6e 67 74 68 3a 20 30 0d 0a 0d 0a ength: 0 ....* *0000 00 17 c4 59 b6 56 00 0a e6 23 92 6b 08 00 45 00 ...Y.V.. .#.k..E. 0010 01 ea 0b 80 00 00 40 11 e9 5a c0 a8 01 64 c0 a8 ...... at . .Z...d.. 0020 01 74 13 c4 d3 22 01 d6 10 53 53 49 50 2f 32 2e .t...".....

...asterisk box being associated with the "entryunauthorized" context, not the "entryinternal" context? (See below) 3. Why is the contact "<sip:s@123.183.86.231:5060>" in our SIP messages, why s@ anything? Thanks MD ------------------------------------------ Contents of sip.conf at ITSP: [999] context=entryinternal ; I know this context exists! This is the right context. type=friend username=999 secret=1111 callerid="Test" <999> host=dynamic nat=no canreinvite=no allow=ulaw allow=a...

...o: <sip:6615xxxxx at 130.117.xxx.xxx;user=phone>;tag=as2e12c791 Call-ID: 2117388659-5062-4 at 82.158.83.xxx CSeq: 31 INVITE Server: VoIPSwitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:6615xxxxx at 130.117.xxx.xxx> Content-Type: application/sdp Content-Length: 235 v=0 o=root 1698171141 1698171142 IN IP4 130.117.xxx.xxx s=Asterisk PBX 1.6.1.18 c=IN IP4 130.117.xxx.xxx t=0 0 m=audio 39124 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- P...

...2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4a2529b8 From: <sip:andy@192.168.1.2>;tag=as72c0d7da To: <sip:andy@192.168.1.2> Call-ID: 3d1b58ba507ed7ab2eb141f241b71efb@127.0.0.1 CSeq: 105 REGISTER User-Agent: Asterisk PBX Expires: 3600 Contact: <sip:1000@192.168.1.2> Event: registration Content-Length: 0 (no NAT) to 192.168.1.2:5060 Sip read: REGISTER sip:sip.nixhelp.org SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4a2529b8 From: <sip:andy@192.168.1.2>;tag=as72c0d7da To: <sip:andy@192.168.1.2> Call-ID: 3d1b58ba507ed7ab2eb141f241b71efb@127.0.0.1 CSeq: 105 REGISTE...

Skipped content of type multipart/alternative-------------- next part -------------- _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

...ith a VOIP provider. I was wondering whether anybody else has come across this kind of problem before. Thanks for your anticipated help. Cheers, Richard Here is a snippet from the debug log; asterisk1*CLI> <-- SIP read from 82.35.xxx.23:33344: REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0 Content-Length: 0 Via: SIP/2.0/UDP 82.35.xxx.23:33344;branch=z9hG4bK21852074 To: "RN SYSTEMS LTD" <sip:119@ pbx.sytes.net > From: "RN SYSTEMS LTD" <sip:119@pbx.sytes.net>;tag=22d32002 Call-ID: 0a4132dc-44a3276f-8d5e4f44-bc07bcd5@192.168.1.3 CSeq: 2230 REGISTER Expires:...

(using samba-2.0.7 on Debian Linux 2.2.15) Hi, I'm wondering how do I get around the problem when you smbmount a share on a win32 machine and that machine reboots, the mount becomes a blackhole. If you attempt an 'ls', 'df' or anything that would access that mount point the program then goes into a state of void and cannot be killed, leaving the proc open until reboot

...s3606b893 To: <sip:448715046363@iptel.tgfslp.dalmany.co.uk> Contact: <sip:Unavailable@213.166.5.129> Call-ID: 1f9465ed1482e9804b089a351a4174a4@213.166.5.129 CSeq: 102 INVITE User-Agent: MSS VoIP Gateway Date: Sat, 14 May 2005 01:18:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 265 v=0 o=root 31661 31661 IN IP4 213.166.5.129 s=session c=IN IP4 213.166.5.129 t=0 0 m=audio 14474 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --...

Hi All First time poster, long time reader. I just ran into something really bizarre. I've enabled videosupport and have been testing sip calls with Xten Eyebeam software, it works (mostly) However, when I have Videosupport=yes In the [general] section of my sip.conf, broadvoice calls fail w/ "We're sorry your call cannot be completed at this time" So... I've