similar to: small qos switch

I have 4 sparc based sun boxes I am about to pay money so I can get rid of them. They are running older versions of Solaris. You should be able to load Solaris 10 and play around with * on them. Time to clean the office: 3 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

I just bought two new Linksys WRT54G routers. Sveasoft has loaded Linux on this router and included a bunch of Linux tools, one of which is Bandwidth Management. The QoS aspect of this is supposed to be much more granular than the previous solution (Wonder Shaper). I have not been able to find any suggestions for how to impliment QoS (Bandwidth Management) using the web interface of Alchemy.

Hi, We're installing an * box next week (pbxtra from fonality) and I'm trying to come up with a solution for remote users that want a phone in their home. I need VPN and QoS capability, wireless support would be a nice to have. Ethernet handoff is fine, i don't need integrated dsl or cable modem... I've been googling and cruising the list and can find bits and pieces

I have one system that went totally crazy on me. It went into an infinite loop rotating * message and log files. From the asterisk console I kept seeing the message about re-loading logger.conf over and over and it just kept creating more and more files. I baby set many different * boxes all running the same script without this problem. Here is my cron script: /var/log/asterisk/cdr-csv/*csv {

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

I have an * setup with sip phones and sip fxo gateway. When a sip phone places a sip/fxo call on hold, MOH is very choppy. It looks like RTP has a real problem with timing if it is not receiving RTP packets. If the outside call that is placed on hold is not generating any audio, the sip/fxo gateway does not send * RTP packets. Is this valid? Is this a problem with the sip/fxo gateway or a problem

Hi, I would like to make a QoS configuration on a linux based dsl router. It is for a server, so I want to shape outgoing traffic, incoming traffic should not be a problem as long as I have a quite assymetric connection. I would like to achieve the following goals: 1) To have one class (p2p) having all the available bandwith if there is no activity on other classes. 2) If another class (ftp

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

Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ------------------------------------------------------------

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

I've been asked to assess the cost of implementing Asterisk with a single T1 line in one of our offices. I've had plenty of experience w/ TDM400 cards, but T1 is new for me so a couple of questions: 1) Will I need a digital or analogue interface card? I expect digital is the answer, but the Digium web site said something about analogue cards being able to support "provider T1

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",

As seen on my post at: http://www.sveasoft.com/modules/phpBB2/viewtopic.php?p=28112#28112 This works very well... It does NOT work with stable 4.0! sveasoft will be issuing a bug fix for this (4.1) in the near future. Final Rev of working script w/ asterisk support I'm not going to run alchemy on production machines until it is stablish. Remember to set your uplink properly and to set