similar to: Asterisk & RedHat Enterprise

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the

Hi all, I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines. 6 plus 6 is 12, how come it's 24? if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate. thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

From c|net News: "On Monday,Microsoft and nine leading phone manufacturers--Asustek Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and Vitelix--announced the public beta program for Microsoft Office Communications Server 2007 and Microsoft Office Communicator 2007." http://news.com.com/8301-10784_3-9719931-7.html?part=rss&subj=news&tag=2547-1_3-0-20 --

We have a server with a TE120 on a partial PRI trunk that several times a day declares the PRI trunk down and stops handling calls until the asterisk is stopped, the zaptel/te120 modules reloaded, and asterisk started. Just before things go down, the log shows the following error: ERROR[9424] chan_zap.c: Write to 28 failed: Unknown error 500 at which point a "show pri spans"

I was playing with a Sprint Vision phone recently and noticed when viewing the low level ReadyLink configuration screens that there are references to SIP registrars and the like. Does anyone happen to know if Sprint's implementation of ReadyLink truly is SIP based, and if so, managed to get it to interoperate with Asterisk. If so, it would prove to be an interesting paging mechanism and

does anonybody know what is the status of gsm codec in snom phones ? they were some issuses in archives, some problems so i would like to know what is the actual status. best regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ majo at sunteq dot sk

I'm trying to get a Carrier Access Corp. Channel Bank I working with a Digium T100P without success. What is stranger is that the status lights on the channel bank and T100P seem to change almost each time I power cycle the channel bank or reset the T100P. The channel bank has three status lights: T1, Framing, Status. T1 is green, Status is yellow, and Framing is usually red but sometime

Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? What's the biggest dialplan in use right now? If you feel you are a competitor, let me know how many contexts/extensions/priorities you are dealing with. Maybe the context with the most extensions, the extension with the most priorities would be

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'm looking at switching VOIP providers, but want to ensure I move to a company with sufficient capacity. Can any Canadian VOIP users post/email me with feedback on their providers? I'll post the results for all to read...... -------------- next part -------------- An HTML attachment was scrubbed... URL:

I need to accept an incoming call, make a series of outgoing calls, and once I find someone willing to accept the call, bridge the original incoming call to the outgoing call. Using Dial from an AGI script isn't enough because once the Dial'ed number connects, the call is immediately bridged and I need to ask the called party if they will accept the call. I can see a couple of

Greetings, I am trying to get either of the above features to work with *, but can't seem to get it quite right. If anyone has them working, I'd sure appreciate an extract from the relevant config files. Or, maybe I'm tilting at windmills, and * doesn't support them - in which case, the underlying business need is to provide the one incoming call on more than one

Greetings, I've been having a large number of deadlock issues lately on chan_sip occurring only on VIA EPIA ML6000 boards. I'm curious if anyone else is having similar issues. My Config (have multiple systems all running the same hardware with the same problem) VIA EPIA ML6000 1GB RAM 80GB HDD Various Digium Cards (T1 and TDM cards) Trixbox 1.2.2 (though running stock asterisk code)

Having spent the better part of an hour searching the archives and voip-info I hesitantly ask what appears to be an obvious question but one I cannot find an answer for. Using Grandstream phones it seems that the only way to support Call Parking is to enable # transfers (i.e. use T in the dial command) since pressing the TRANSFER button on the BT phone is blind and one does not hear the call

I am looking for some Linksys and GrandStream ATAs in Canada. I am looking for places that ship from Canada so I don't have to deal with the clearing of customs and tax remittance. Any suggestion? -- Thanks

I have noticed that when I try to connect multiple G729 VoIP devices into a MeetMe conference that I can only add up to the number of G729 licenses I have. Now I would think that because all the devices are G729, this wouldn't be the case and the only license that would ever be used would be if a non G729 device or Zap channel was a part of the Meetme conference. This is apparently note the

Any opinions on the best ATA? For example, if someone was having a problem and I wanted to rule out any ATA glitches or firmware issues, what device could I give them that I could count on to always be a trouble free top performer that just plain works?

Just wondering how your experience is with HPEC, Is it just for analog interfaces or we can use it on TE122 as well? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080214/3f0580f1/attachment.htm

Does anyone know how to change the source IP address/Source Interface of RTP packets? Changing the SIP source IP address in sip.conf has no apparent impact on RTP. RTP traffic still uses the address assigned to the outbound interface.

My question pertains to the use of IAE.. I would like to setup 2 Asterisk boxes. One would be located in our office behind the firewall and hooked up to our analog lines. The other would be located in a remote datacenter and used for our remote employees to connect to. I would like to be able to accept calls on the Office Asterisk server and route them to the Datacenter Asterisk server. Is