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...ond Nokia phone under the Asterisk "realm" with a different userid and PW? Thanks Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candrews at sayersmedia.com <mailto:brett at voipsupply.com> Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers <mailto:bsayers at voipsupply.com> , CEO NOTICE: The information contained in this email and any document attached hereto is intended only...

Hello, If anyone is using a TE121B card and it works reliably (i.e. no "HDLC Bad FCS" or similar errors), could you pass along the make, model, and basic configuration of your Asterisk server? We tried upgrading our old Dell PowerEdge server to a SuperMicro system, but that didn't help. I would like a solid recommendation before I suggest another purchase. Thanks. -- Kevin

...experience implementing OpenSBC (opensourcesip.org)? Have a possible consulting gig referral. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candrews at sayersmedia.com <mailto:brett at voipsupply.com> Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers <mailto:bsayers at voipsupply.com> , CEO NOTICE: The information contained in this email and any document attached hereto is intended only...

Hello I am going to try the new Digium Fax for Asterisk product. I'm planning to connect fax machines to Asterisk (currently 1.6.0.9) via T.38 ATAs. I'm looking at Grandstream HT502 or Linksys SPA2102 ATAs. If anyone has any experience with these devices, or other recommendations, I would be grateful if you could share your experiences. Regards Ian

I used to use IDEFISK, but since it was taken over/renamed into Zoiper it's been really hard work - now I'm told that they won't support my chosen distribution - Debian Etch - the current stable version of Debian I prefer. So, looking to dump Zoiper and go with something else - I want something light-weigh (So that rules out Ekiga - and Zoiper was going down the bloatware route

While I myself am not a MagicJack user, I'm curious as to whether anyone here has managed to heist their MagicJack account's sip credentials, and use them to terminate calls using asterisk. Apparently it's as simple as sniffing the SIP credentials. If so, said person would enjoy unlimited termination for $20 year while retaining the flexibility of setting their CallerID to a

We are currently moving away from a wide-spread Cisco CallManager deployment to Asterisk. For many of our small sites we have the routers configured for what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP registrar. We are converting to SIP, and from what I can tell Cisco wants a license for each router to run SRST over SIP... So my question to the group is: What are

As part of a project to move a clients Cisco phones to SIP to work with an Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk setup. Now, I've gotten the firmware files from the site, the latest version is 8.4 which contains the following files: apps41.8-4-3-16.sbn cnu41.8-4-3-16.sbn cvm41sip.8-4-3-16.sbn dsp41.8-4-3-16.sbn jar41sip.8-4-3-16.sbn

I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com

When I make a regular call from my SIP phone connected to my Asterisk server I have no issues, however when I make a call using Originate : 'Channel'=>"SIP/1XXXXXXXXXX@sip.broadvoice.com", 'Context'=>'mycontext', 'Exten'=>'899', 'Priority'=>1, 'Callerid'=>'whatever')); It creates a screech sound when the

Is there any built in functionality when using Originate to retry a call based on the DIALSTATUS? Similar to the .call file where you can set max retries and time between them? I've tried putting the logic in an outbound context/macro, but it just times out if the time between retries is too long...I suppose your not supposed to leave calls in limbo too long. Is there any other method

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to