search for: minusw

...Ultra 5 1 Sparcstation 5 I also have a box full of Sun keyboards and mice. Contact me offline if you want them. I've had many good years of development on them and it kills me to just toss them, but the office is just too damn cluttered. thanks, bk... -- Bob Knight [-w] the work option bk@minusw.com 925-449-9163

...640 root root } /var/log/asterisk/*log /var/log/asterisk/messages { missingok rotate 5 weekly create 0640 root root sharedscripts postrotate /usr/sbin/asterisk -rx 'logger reload' > /dev/null 2> /dev/null endscript } -- Bob Knight [-w] the work option bk@minusw.com 925-449-9163

...g in one connection from the voice lan and one from the data lan. I have found quite a few 24 or 48 port switches that will do this, but I really do not need anything that big. There are already switches in place. Any recommendations please? thanks, bk... -- Bob Knight [-w] the work option bk@minusw.com 925-449-9163

pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw

...fine. I connect 2 GS and place one on hold. The GS that is receiving MOH from * is working great because the GS keeps sending back RTP packets. IAX connections work fine. I call an extension on another * box and place it on hold. MOH over IAX/IAX2 is great. -- Bob Knight [-w] the work option bk@minusw.com 925-449-9163

...d now know the new IP. I was thinking of letting the static box keep track off all the dynamics and host an IAX name server. Before I go off and so something really silly and a waste of time, is there an easy way to do this? Has anyone done this with DDNS? -- Bob Knight [-w] the work option bk@minusw.com 925-449-9163

Just received a Mediatrix 1204 fxo sip gateway and playing with the initial config's, etc. It's working, but have a ways to go before it could be considered usable. The box was not designed to "register" like sip phones do. The incoming pstn line is an ordinary 2-wire analog US pots line, and I'm using canreinvite=no to forcably keep * in the middle for now. Questions: 1.

I'm having a brain fart.... What's the proper syntax for dialing out via a sip g/w (Mediatrix)? Been trying stuff similar to: exten => _6X.,1,Dial(SIP/3091@205.22.93.1/${EXTEN-1}) where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did even try the IP. Rich

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Hello guys, I need your help related to a mediatrix 1204 configuration. I read some of the messages that you posted in the asterisk users mailing list about the mediatrix 1204 and decided to contact you. I know that the community is not related to Mediatrix devices, but so far I have not found any other group that has work as much as you with them. I bought the mediatrix in Mexico and my provider

...ld be in my dhcpd.conf file that will map to some snmp mib variable? I have peeked at the mediatrix mibs and docs and can not seem to find what I am looking for. I am guessing the dhcp client in the gateway is parsing dhcp packets, looking for option names. -- Bob Knight [-w] the work option bk@minusw.com 925-449-9163

As mentioned on the devel list earlier today, I'm interested in writing an IAX2 plugin for Ethereal to make debugging IAX protocol implementation and simultaneous calls on normal networks easier. Anyway, I started work on it this evening, so it's not complete yet, but it's starting to look quite sensible: - http://raq626.uk2net.com/~al/ethereal.png A couple of people have

Hi List, Here is the config ext 2601 is a GS BT-101 phone ext 2062 is a SNOM 200 latest public firmware on both asterisk is Asterisk CVS-11/14/03-22:55:45 Make a call from 2601 -> 2602 life good, call works have 2602 place call on hold. The music on 2601 IS NOT my music on hold. It seems its a MOH server SNOM has. take call off of hold on 2602 and 2601 still trys to play parts

Sure would like to hear experiences using various FXO to VoIP gateways with *. It seems that any thread that has anything to do with problematic FXO interfaces goes on forever with speculation about everything under the sun. Unless there is someone out there with the engineering experience to build a better one it is a waste of time, let Digium deal with it. If the TDM400P can ever be made 99.99%

Hi, I have received a new card TDM400P revision E, from digium. When I tried to modprobe wcfxs it gave me the following errors: Freshmaker version: 63 Freshmaker passed register test ProSLIC on module 0 insane (1) 255 should be 2 Module 0: Not installed ProSLIC on module 1 insane (1) 255 should be 2 Module 1: Not installed ProSLIC on module 2 insane (1) 255 should be 2 Module 2: Not installed

Hi, I am running * with E100P board. At least every our I got an Error 500 message and ISDN-PRI restarts: Jan 10 12:53:02 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 failed: Unknown error 500 Jan 10 12:54:27 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 failed: Unknown error 500 Jan 10 13:33:01 WARNING[98311]: File chan_zap.c, Line

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040915/65a2736a/attachment.htm -------------- next part -------------- Hi friends, I tried to dial 111 from CLI without any hard/soft phones. I used the following config when i called 111 from CLI by CLI> dial 111 I got these errors -- Executing Dial("OSS/dsp",

Hello - I'm using ATA-186 devices, with RFC2833 DTMF encoding. I am having problems with routines that input long strings of numbers, in that I am getting more than a small number of double digit entries. As an example, I have a section that asks for the user to enter a call forwarding number, and then puts that number into a database. Almost always, there are double digits when the

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Product Review Mediatrix 1204 4-Port SIP FXO Gateway Firmware: v2.4.10.69 - US Version US Retail: ~$750, Street Price: ~$450. The Mediatrix 1204 SIP FXO gateway is equipped with four RJ11 pstn jacks and one RJ45 Ethernet jack on its rear panel. It terminates the four pstn lines in either Loop Start or Ground Start mode, handles incoming CallerID, and generates either Dial Tone (back towards the