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Hi, I'm curious if there is any other option beside relaxdtmf in zapata , or any where else to tune dtmf detection on TDM400 fxo boards. in one of our sites provider is giving us 4 analog lines out of Adtran router and Asterisk often recognize DTMF wrong. Obviously playing with relaxdtmf was not helpfull. What do we know anout 1.2 and 1.4 DTMF handling diffrences? At this time i'm using

Hi List, I need to change my provider, at this time Asterisk box is on VOIP trunk. I have two options, T1 or 15 analog lines. I have some experience with analog and I have had two main issues with it. first is echo (I have not tried HPEC yet) and second unpredictable volume. The question is, if I use TE100 with PRI , will I have same issues? I would appreciate any comments and sample zaptel.conf

Hi List, any good comparison between TDM404B and TDM2401B . i'm not very happy with TDM404B voice quality, low volume and sometimes echo. I was wondering if any of you guys have good experience with TDM2401B. thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070116/921051c9/attachment-0001.htm

Hello list, i'm sure this is not a new issue, i'm having DTMF recognition issues with TDM404. I've already tried relaxdtmf=on/off and that did not do any good. i was wondering if there is any where else in zaptel/zapata to play with and have it fine tuning. Or maybe this card is not handeling DTMF very well?? peace! -------------- next part -------------- An HTML attachment was

I'm planning to upgrade my asterisk 1.4.4 to 1.4.6. usually for asterisk upgrade i delete modules directory and include, then compile the new version. Since i have couple of G729 Licenses on this server installed, would i need to call Digium to reactivate these Licenses? Is there any better and faster way of upgrade asterisk? Possibly without losing G729 License? Thanks!

Hello list, I was wondering if any of you guys have had any luck with polycom in remote offices, I'm facing a weird issue, polycom phones work fine in the main office, in remote office it says, Registration from '<sip:202@10.0.1.190>' failed for '70.59.21.112' - Wrong password the odd thing is Linksys phone works without any issue!! polycom wont register but its able to

nat is equal to yes, and server definition in ftp provisioning server is correct. i followed packets between phone and asterisk, it seems for some reason asterisk is not happy about challenge response its getting from polycom. why its not happening in the same LAN, beats me! and also NAT device is cisco SIP aware and works flawlessly with Linksys. the only close definition of issue i have found is

Hello list, I have always though codec translation table is dircetly connected to system speed, utill i came across this: in my lab, i have 2 boxes, First box is an Intel Celeron 1.7 GHZ with 256M RAM: show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw