search for: magrathea

...948] TCP: cubic registered [ 0.276208] Initializing XFRM netlink socket [ 0.276789] NET: Registered protocol family 10 [ 0.277534] mip6: Mobile IPv6 [ 0.277725] NET: Registered protocol family 17 [ 0.278531] Loading compiled-in X.509 certificates [ 0.279646] Loaded X.509 cert 'Magrathea: Glacier signing key: 3eebbbadc3f26af1e0e934f81a18bf4c578bd268' [ 0.280317] registered taskstats version 1 [ 0.281101] Magic number: 14:106:979 [ 0.281406] rtc_cmos rtc_cmos: setting system clock to 2018-08-24 08:57:50 UTC (1535101070) [ 0.281982] Warning: unable to open an initia...

...now to contact the person concerned] This message is for Stephen Karrington - it appears that you have over-agressive 'spam' filters and we can no longer email you. Please rectify this if we are to have meaningful conversation! The original message was received from Linus Surguy <linus@magrathea-telecom.co.uk> ----- This message has been blocked by our spam filter. ----- ----- If this has been a mistake, please contact ----- ----- the recipient through other means. ----- Undelivered message for: <"Stephen Karrington" <sk@dreamtime.net>...

We have an Asterisk server that speaks IAX2 to Magrathea to get to the PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s all configured for SIP with silence-suppression disabled. Everything is configured to use a-law encoding. The version is: sip*CLI> show version Asterisk CVS-05/06/04-18:45:57 built by root@sip on a i686 running...

...to recognize my dtmf there either. I have to presume that either oob->inband is broken for -> rtp or it is broken w/o a zap timing source.... -JimC -- James H. Cloos, Jr. <cloos@jhcloos.com> <http://jhcloos.com> --__--__-- Message: 6 From: "Linus Surguy" <linus@magrathea-telecom.co.uk> To: <asterisk-users@lists.digium.com> Date: Mon, 12 Apr 2004 20:30:40 +0100 Subject: [Asterisk-Users] OT appologies to list Reply-To: asterisk-users@lists.digium.com [I'm sorry to trouble the list with this, but this is the only way I know to contact the person concerne...

Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK

I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US

Hi, I allow only TLS/SSL connection to dovecot. The managesieve doesn't work with TLS but it works without it. gladiac at magrathea:~> gnutls-cli -p 2000 -s --insecure -V milliways.cynapses.org Resolving 'milliways.cynapses.org'... Connecting to '213.239.217.124:2000'... - Simple Client Mode: - Received[194]: "IMPLEMENTATION" "dovecot" "SASL" "PLAIN" "SIEVE"...

...D subscriber number - within a couple of seconds you should be connected. We can also terminate UK 0800/0808 numbers for SIP/IAX -> PSTN calls, at the moment we don't have an FWD number setup for this, but simply use the username/password of guest/guest and point your connection at voip-gw1.magrathea-telecom.co.uk and send the number as 0800xxxxxxx Of course, this is all a trial at the moment, so no commerical warranties are available! Finally, if you wanted to trial your own personal 0870 number pointing to FWD, drop me an e-mail and I'll tell you how to do it. Linus Magrathea

Good time of day all! 1) I am trying to use as5300 and asterisk. As5300 sends calls to me. I get the following in * console: -- IAX2/magrathea/6 is making progress passing it to OH323/R27464 Sep 8 10:57:59 NOTICE[1140046640]: chan_oh323.c:1159 oh323_indicate: Ignoring PROGRESS indication. As5300 user does not hear anything, just silense instead of dial tones. My config is oh323.conf default config. 2) * logs CDR records with NO Answe...

I have several of incoming numbers on IAX from voiptalk and magrathea but have a problem with IAX.conf. If I follow the example from voiptalk [VoIPTalk Incoming Number] type=friend username=VoIPTalk Incoming Number context=[XXXXXXXX] and make incoming entries in IAX.conf for the numbers like below with a different entry for each number pointing to a different con...

Hi, Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? Ta. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040421/3d91c7f6/attachment.htm

Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a "Service Unavailable"

...srcversion: D0FCF627599F97CE5A4C810 alias: pci:v000012D2d*sv*sd*bc03sc*i* alias: pci:v000010DEd*sv*sd*bc03sc*i* depends: drm,drm_kms_helper,ttm,mxm-wmi,wmi,video,i2c-algo-bit intree: Y vermagic: 3.16.0-38-generic SMP mod_unload modversions signer: Magrathea: Glacier signing key sig_key: C2:84:ED:AC:CF:0B:47:36:52:C3:4D:23:BE:C3:56:94:42:36:E6:3B sig_hashalgo: sha512 parm: tv_norm:Default TV norm. Supported: PAL, PAL-M, PAL-N, PAL-Nc, NTSC-M, NTSC-J, hd480i, hd480p, hd576i, hd576p, hd720p, hd1080i. Default: PAL *NOTE* Ignored for car...

...3-805-... via: Axialys Interactive Germany: * +49-800-... +49-130-... via: [netzquadrat] GmbH Netherlands:* +31-800-... via: ag-projects.com United Kingdom:* +44-800-... +44-808-... +44-500-... via: Magrathea Systems Ltd. via: Telecomplete Ltd. United States:* +1-800-... +1-888-... +1-877-... +1-866-... via: Telesthetic/Local Exchange Carriers of Michigan Universal Toll Free:* +800-... via: [netzquadrat] GmbH via: Magrathea...

HI all, Having a mental block today - can someone confirm which direction the echo cancelation applies to for the Zap PRI channels? ie. is it removing traces of the transmitted data to the PSTN from the received data, *or* is it removing traces of the data transmitted to Asterisk from the data received back from Asterisk? Got a configuration that is based on a call made from the PSTN to a SIP

Hi All, Is there anyone providing UK geographic numbers that can be terminated on Asterisk using IAX ? It must be a geographic number (eg. Start 01 or 02, not 08xx). I've tried the sipcall.co.uk service and it looks very good when using X-Lite but it will not work with Asterisk. Switching to IAX should also resolve issues around NAT - hurray! -Nathan

DID's from Allstream (AT&T) are $2 Cdn/month but I think they have a rule that it has to terminate on their network somewhere... -----Original Message----- From: Linus Surguy [mailto:linus@magrathea-telecom.co.uk] Sent: Monday, June 14, 2004 6:53 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Canadian DID Can anyone point me in the direction of a wholesaler of Canadian DID numbers? If they'd be interested in trading them for UK numbering that would be even better! Lin...

If anyone can help me with this I'll be soooo grateful :) We have a isdn30 line, with a DDI range. We also have 2 business units that have separate 0870 numbers that are mapped onto 2 DDI numbers. I would like to be able to present these 0870 numbers from the business units so that the correct number is displayed on a callerid, or when 1471 is dialled. BT claim that I can only have a

On 10/12/2014 10:19 PM, Jd wrote: [please don't top-post on technical lists] > Thanks Kashyap, > The command line examples makes thing quite clear :) > rsync for image (large) files create a new file for every little > change, that was the reason I started looking in to using dirty bitmaps. Upstream qemu still doesn't have persistent dirty bitmap tracking working yet; once

Hello All, Here's the problem, we have happily set up several Asterisk servers to offer commercial service in the UK, our wholesale SIP termination partner (Magrathea - use SER/CiscoGW to provide us the service on a public IP address) - till now we have used Asterisk to connect clients on private IP's with Asterisk doing the required conversion for SIP/IAX between public and private IP's. The current issue is that we have recently agreed to support Shor...