search for: sp501

Displaying 4 results from an estimated 4 matches for "sp501".

Did you mean: ip501
2006 Oct 23
2
Polycom SP4000 ftp problem
...ot;Updating initial configuration..." screen. when i switch it to tftp, it'll work fine. i though it was bootrom/firmware issue so i upgrade it to bootrom 3.2.2/sip 2.0.1 but it makes no difference. any thoughts? p.s. i'm using debian sarge proftpd 1.2.10 and the setting works fine w/ SP501 with bootrom 3.1.2/sip 1.6.3 -- Edwin Lam <edwin@officegeneral.com> Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
2009 Jul 02
1
AGI Transfer?
I've been trying to get an AGI "transfer" to work for several weeks now. It isn't error-ing out, but it isn't working either. I can't use "dial" in this case due to what I'm trying to accomplish. Does an AGI Transfer actually work? -= Info about application 'Transfer' =- [Synopsis] Transfer caller to remote extension [Description]
2011 Apr 12
0
No subject
...this issue? Thanks, -Ryan On Tue, Aug 2, 2011 at 3:51 PM, Ryan McGuire <rdmcguire01 at gmail.com> wrote: > Running build 1.8.5.0 (compiled from source) I seem to be having an issue > with codec negotiation. I have a Grandstream HT503 FXO port connected to a > pstn line, a Polycom SP501, and a SIP trunk with callwithus. > > What I'm essentially looking to accomplish is for ulaw or g729 (preferably > ulaw) to be used to the Grandstream FXO or any other internal endpoint, and > for g729 only to be used outbound to my SIP trunk. > > Here are the basics of my con...
2011 Aug 02
1
Codec negotiation issue (no audio format found to offer)
Running build 1.8.5.0 (compiled from source) I seem to be having an issue with codec negotiation. I have a Grandstream HT503 FXO port connected to a pstn line, a Polycom SP501, and a SIP trunk with callwithus. What I'm essentially looking to accomplish is for ulaw or g729 (preferably ulaw) to be used to the Grandstream FXO or any other internal endpoint, and for g729 only to be used outbound to my SIP trunk. Here are the basics of my config, showing the codec list...