search for: 4005

...ble yes AutomaticCallTargetAddress 4000 (which is obviously the problem...) Any help appreciated Thanks robbie Sip.conf <snip> [inbound] type=friend host=192.168.0.253 context=from-pstn canreinvite=no allow=ulaw allow=alaw <snip> asterisk1*CLI> -- Executing Macro("SIP/4005-9d61", "dialout-trunk|7|mymobilenumber|") in new stack -- Executing GotoIf("SIP/4005-9d61", "1?3:2)") in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro("SIP/4005-9d61", "record-enable|4005|OUT") in new stack -- E...

...SIP Debug. Looks to me like the phones are expecting the server to do the conference mixing, which I guess it would do in CallManager? <--- SIP read from TCP:xxx.xxx.xxx.xxx:50604 ---> REFER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/TCP xxx.xxx.xxx.xxx:50604;branch=z9hG4bK48c7492c From: "4005" <sip:4005 at xxx.xxx.xxx.xxx>;tag=203a07fceb4b00eff1377deb-da93e2ee To: <sip:4004 at xxx.xxx.xxx.xxx> Call-ID: OutOfDialog--001e-67a906f5-5333c2b8 at xxx.xxx.xxx.xxx Max-Forwards: 70 Date: Tue, 20 Jan 2015 17:10:19 GMT CSeq: 101 REFER User-Agent: Cisco-CP7945G/9.4.2 Contact: <s...

...t; phones are expecting the server to do the conference mixing, which I guess > it would do in CallManager? > > <--- SIP read from TCP:xxx.xxx.xxx.xxx:50604 ---> > REFER sip:xxx.xxx.xxx.xxx SIP/2.0 > Via: SIP/2.0/TCP xxx.xxx.xxx.xxx:50604;branch=z9hG4bK48c7492c > From: "4005" <sip:4005 at xxx.xxx.xxx.xxx > >;tag=203a07fceb4b00eff1377deb-da93e2ee > To: <sip:4004 at xxx.xxx.xxx.xxx> > Call-ID: OutOfDialog--001e-67a906f5-5333c2b8 at xxx.xxx.xxx.xxx > Max-Forwards: 70 > Date: Tue, 20 Jan 2015 17:10:19 GMT > CSeq: 101 REFER > User-Agent...

We were using G722 - I thought similarly and tried a call with alaw. Same problem occurred, any other ideas? > I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can > only do a single G729 channel, and if you require G729 for the second leg of a > conference, it will fail. This message may be private and confidential. If you have received this message in

...Severity: enhancement Priority: P2 Component: ssh AssignedTo: openssh-unix-dev at mindrot.org ReportedBy: foomzilla at fuhm.net I have two SSH servers running on the same machine. One is running on port 22 for standard logins; the other is running port 4005, and is not run by root. They both must have different host keys, as the sshd on port 4005 cannot (and should not) be able to read the host keys for the standard sshd on port 22. This all works fine, except that every time you connect, it complains that the host key is wrong. I see that there is...

...usbhid-ups vendorid = 09ae port = auto When I run upsdrvctl -DD start, I get: Network UPS Tools - UPS driver controller 2.4.1 Starting UPS: test exec: /lib/nut/usbhid-ups -a test Network UPS Tools - Generic HID driver 0.34 (2.4.1) USB communication driver 0.31 This Tripp Lite device (09ae:4005) is not (or perhaps not yet) supported by usbhid-ups. Please make sure you have an up-to-date version of NUT. If this does not fix the problem, try running the driver with the '-x productid=4005' option. Please report your results to the NUT user's mailing list <nut-upsuser at lists....

...e there was no LAN drop? If so what hardware did you use and was it succesful? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245

...:~$ nslookup google.com ;; Got SERVFAIL reply from 192.168.4.5, trying next server Server: 8.8.8.8 Address: 8.8.8.8#53 Non-authoritative answer: Name: google.com Address: 216.58.195.78 ;; Got SERVFAIL reply from 192.168.4.5, trying next server Name: google.com Address: 2607:f8b0:4005:80b::200e

...deo conference device - More HDVoice capable phones - Skype Outage - VoIP on mobile devices - or perhaps something more personal..... Come one, come all. Bring your story. Connect details at http://vuc.me Michael Graves -- Michael Graves mgraves<at>mstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgraves at mstvp.onsip.com skype mjgraves Twitter mjgraves

...can be provisioned in the same way. Are end uers happy with them? Thanks, Michael -- Michael Graves mgraves at pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves at mstvp.com o713-861-4005 c713-201-1262 skype mjgraves fwd 54245

...ut something is going dramatically wrong and I can't see it at the moment. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262

...s being installed are actually HDSL....which would be not a T-1 at all...right? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262

I see a variety of DECT 6 phones available CHEAP at costco. Is there a way to convert these to SIP? I recall someone talking about a Siemens devices that works with all DECT phones, making them SIP (I think).... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081205/fdd14af9/attachment.htm

I am trying to import Indian National Sample Survey Data. It is ASCII flat and an example is below 001460306025001117101011117111201*01*000000211 270104070204093 002460306025001117101011117111201*02*0000008011116104910519572 022 2600 1150 30 55 170 4005 003460306025001117101011117111201*03* 0000111039204112 222 Previous post don't seem to provide much help. I have an document outlining the layout of the data, but don't know how to start. An example of the layout is that the values in bold indicate separate levels.Example of layout is be...

...= default strategy = fewestcalls context = queue-out ; Here we go when the caller presses a single digit, while in the queue timeout = 15 wrapuptime=10 announce-frequency = 30 announce-holdtime = yes joinempty = yes member => Sip/4001 member => Sip/4003 member => Sip/4004 member => Sip/4005 member => Sip/4006 member => Sip/4007 Regards Akhilesh Sent from Samsung Mobile -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140522/8c6012a1/attachment.html>

I will have a small shop with ~4 phones using an HP server with Asterisk on it, it has two NICS and so I planned on plugging one into the cable modem, and the other into the switch. I was going to let this box perform NAT for the company but I am concerned about QOS for the VOIP portion. Anyone got a similar setup and care to share what they successfully implemented? Thanks! jlc

Hi! I have two, not too related questions: - the probably simpler one: if anyone can help me out using a Cisco 7905G with chan_sccp? I did already managed to get it working with a SIP image, I'd just like to see it work with this one as well. It's probably something I screw up with the configuration, as the phone registers, only I don't get any lines with it, although I have it

...r >> Server: 8.8.8.8 >> Address: 8.8.8.8#53 >> >> Non-authoritative answer: >> Name: google.com >> Address: 216.58.195.78 >> ;; Got SERVFAIL reply from 192.168.4.5, trying next server >> Name: google.com >> Address: 2607:f8b0:4005:80b::200e > > Is this on the DC or a client ? > > On my DC I get this: > > adminuser at kdsdc:~$ nslookup google.com <http://google.com/> > Server: 192.168.0.10 > Address: 192.168.0.10#53 > > Non-authoritative answer: > Name: google.com <ht...

For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP range in the U.S. I'm particularly interested in the Gigaset S685 IP. Since it's DECT 6.0, and there's an English (UK) version, I'm thinking it should work just fine, after dealing with the walwart issue (and maybe caller ID signalling). Anyone imported one from the UK and using it in the US? for how

...ad from connections at C level. Is that correct? Alternatively I would be happy to accumulate the message into a string at R level and then parse the message. Unfortunately `readChar` blocks with *server* socket till all nchars have been received. For example: ss <- socketConnection(port = 4005, server = TRUE, open = "r+b") readChar(ss, 2000) blocks on the second line till all 2000 chars were received. The *client* connection doesn't block on read at all: cs <- socketConnection(port = 4005, server = FALSE, open = "r+b") readChar(ss, 2000) quickly re...