IIRC,
the linksys ATAs have a setting that enables 'escaping' the display name
"Escape Display Name" - how appropriate
don't know if the grandstream ones have the same feature
Interesting, not escaping the display name causes audio problem
IIRC,
the linksys ATAs have a setting that enables 'escaping' the display name
"Escape Display Name" - how appropriate
don't know if the grandstream ones have the same feature
Interesting, not escaping the display name causes audio problem
Last edited by voxabox; 05-13-2009 at 01:27 PM. Reason: triggered happy
my wish list:
- Intelligent Call Forwarding that detects the incoming call originated from the "forwarded" phone and rings the original destination instead
Call History that makes use of CallerID/Custom CallerID+Location. Call History only shows Custom CallerId+Loc. No CNAM look up; Albeitly,it's a step in the right direction!
- Scheduled sim. ring with a twist (see wish #1)
Although my symptoms were exactly the same, the cause was most likely not. I say that because I have no idea what a 'line display' is. The day before I started having no audio on answered calls I had two calls do something I'd never seen happen before.
The phone rang, two rings and I answered it, dead air. I hung up, it rang again and the call completed fine. Caller said they called only once, the phone rang at least eight times before I answered.
That said, everything seems to be working fine today.
Russell
Mine only happened the 3 times (that I am aware of)just before noon today.
Dylan said he would look into it. I mentioned that if the issue had not been raised in the forum, I would have totally ignored it. I am aware and understand the fleeting problems with any VOIP service and accept the occasional burp. Unless my service is completely down, I don't get too excited. If a feature has a problem, I can wait until it's fixed as I use only a handful of basics.
I think that many of my trouble reports are to make support aware of a problem that may be brewing. Most of the time it's nothing worth wasting bandwidth on.
I noticed something like this, where I would call my VOIPo one time and it would just go to dead air. The line rang and my wife answered but also had dead air. Then the phone rang probably 4-5 more times even though I wasn't dialling anymore. Something kept retrying over and over. I saw it on my Asterisk box with a message saying "Too Many Hops from" and the central01 IP address.
I think it's good to raise even the little burps in the forum. Until I saw the other posts I was seriously pondering Tim's suggestion about port forwarding, even though my router has been flawless with the PAP2T from ViaTalk and (until these glitches) the VOIPo PAP2T. Seeing that several others are having the issue seems to point that the issue is not at my end. It's also beneficial to the provider - if just one person reports it the inclination would be to "blame" the customer's setup and if the cause is elsewhere such "blame" only causes frustration both to the support person trying to help and the customer.
Russell
Here is my log file from when I have had trouble with incoming calls.
Code:[2009-05-13 14:50:04] VERBOSE[2911] logger.c: -- Incoming call: Got SIP response 483 "Too Many Hops" back from 67.228.251.106 [2009-05-13 15:04:34] NOTICE[2911] chan_sip.c: Peer 'VOIPoIn' is now UNREACHABLE! Last qualify: 37 [2009-05-13 15:04:34] NOTICE[2911] chan_sip.c: Peer 'VOIPo' is now UNREACHABLE! Last qualify: 48 [2009-05-13 15:04:44] NOTICE[2911] chan_sip.c: Peer 'VOIPoIn' is now Reachable. (37ms / 2000ms) [2009-05-13 15:04:44] NOTICE[2911] chan_sip.c: Peer 'VOIPo' is now Reachable. (38ms / 2000ms)
Hey guys -- if anyone else is experiencing any further incoming issues, please ensure that you have a ticket in as we are not seeing any new reports and want to make sure anyone that had seen these characteristics, their account is looked into, thanks!
Just entered a ticket on my line.
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