similar to: QoS on Linksys SWR208P?

Found my answer for those who would like to know: Profile Rule: [--key $A]http://your.addre.ss/$B/$MAC.cfg GPP A: urtopsecretultrasecureaesencryptionkey GPP B: OddBallDirectory123098 Hope that helps someone! Curt -----Original Message----- From: Curt Shaffer [mailto:cshaffer@gmail.com] Sent: Wednesday, February 07, 2007 11:35 AM To: 'Asterisk Users Mailing List - Non-Commercial

I am having an issue with doing FTP auto provisioning of Polycom 501's when they are behind a NAT. If I put the phone on the same subnet as the provision server it loads the configs and changes fine but as soon as I put in behind a NAT it comes up with cannot contact boot server. I have tried behind and replicated this behind a PIX 501, a Linksys SOHO router and a Motorolla SOHO router.

Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using

Thanks Curt, that's "too cool for school", any idea on when this is coming to the MS SBS platform? I use SBS for myself at home and would love that level of functionality included. Does Asterisk therefore handoff voicemail storage etc to Exchange for this level of integration? Cheers, Dean ________________________________ From: Curt Shaffer [mailto:cshaffer at

Hello list, I need to set Voice QoS and SIP QoS for YeaLink. The possible values are 0 ~ 63. With Grandstream I can fill in DiffServ 46, which is EF. That's what I want. With Snom I fill in 184, which corresponds to EF or DSCP 46 (according to their wiki) But what value do I want to fill in with this YeaLink ??? This is a conversion table :

https://bugzilla.mindrot.org/show_bug.cgi?id=1964 Bug #: 1964 Summary: QoS/DSCP names false translated to ToS hex value Classification: Unclassified Product: Portable OpenSSH Version: 5.9p1 Platform: amd64 OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: ssh

Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt

What is everyone out there doing in an all IP termination environment to change trunks when quality drops to a certain provider automatically? Thanks Curt

I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 10000-20000. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some

Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091001/abf55045/attachment.htm

I wanted to see if there was anything reasonable in price out there yet that performed an RF to IP bridge via asterisk. What I mean by this is callers from PSTN can be patched to a UHF/VHF radio and vis-?-vis. I know there is an option available for the Avaya systems but it?s a little out of the price range I?m looking for (~$200/channel). Has anyone out there found a stable way to do this?

Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don't really know where to start on measuring jitter

I am having the same issue as below. Has this issue been solved or does anyone know an answer? This error recently began and we have multiple phones out of commission. PLEASE HELP!! http://lists.digium.com/pipermail/asterisk-users/2006-August/162841.html How did you find out about 468*??? It's sure as poop not documented in the Polycom Admin Guide anywhere. -----Original Message-----

I have decided to move on from Asterisk@Home and start compiling asterisk myself now. I got a dedicated box and put my X100P in it. I installed the server version of CentOS 4.2 (2.6.9-22.EL kernel) barebones. The box is a dell GX270 workstation with 1GB of RAM. I got a fresh copy of O'Reilly's Asterisk the future of technology and begun. I downloaded the zaptel-1.2.4.tar.gz, libpri-1.0.9,

I have read different posts over the months wondering who Vonage uses for their VoIP technologies. I stumbled across this article (although it's from 2002, I think) that suggests strongly that they use Cisco. There is no telling what they might use in conjunction with this but this should clear some of the conjecture.

I have been working with a couple companies who are interested in integrating Cisco VoIP (mostly call manager express) but utilizing Asterisk for AA, VM, failover trunks etc. I have found some materials and guidance out there but I think a list and/or wiki for general asterisk integration with other vendors would be great and feel that it is enough off topic that it deserves its own space. Just

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I am trying to perform a "rollover" when the primary number is busy. This is coming from a POTS line. Apparently I need call waiting on the POTS line as I get immediate busy from the FXS if I don't have it. So my question is this. I have an Aastra 480I CT. The call forward when busy here seems pretty straight forward. Choose the mode as busy enter the extension in the forward number

I am looking for an ATA that has had very reliable results when passing FAX over IP. I was thinking of testing the Cisco (not Linksys) ATA 186 I1, ATA 186 I2, ATA 188 I1. This is what I'm looking for: FAX -> PTSN -> through Asterisk -> ATA -> Fax Machine. I have QoS from PSTN entry to ATA on the network so I can assure precedence. What has everyone out there been using

I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP