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Hi, I am fairly new at working with Asterisk. I am having a call quality issue that I really need to get
ironed out before we go to rollout the system in a week. Any help would be greatly appreciated!!! Even if it is just
pointing me in the right direction. My current setup: I have Asterisk Setup on a Dell Server. It has 2 T100P
cards. One will be for out T1 PRI from the Phone Company (We don’t have
this installed yet) The other T100P connects to a VINA T1 IAD (Channel Bank) I also have a Cisco 7960 SIP Phone attached and registered. The Server is connected to a broadband connection. My issue: When I call the IAX Demo from the SIP Phone, the call is
perfect. Asterisk Voice is 100%, and the Voice from the Digium Test server is
almost 100% (an occasional stutter)..but very usable. When I call the IAX Demo from a Phone connected to the VINA
Channel Bank, the Asterisk Voice is 100%, but once it connects to the test
server it is extremely choppy. You can kind of understand what is being said,
but it is very very poor quality and quite unusable. When I between the Channel Bank and the SIP phone, the
quality is 100% no problems at all. So..why does the VINA Channel Bank connection not seem to
like the IAX side of things, When I know that the IAX side is functioning great
when used from a SIP Phone? I don’t know what details would be pertinent to this,
but here is what the Asterisk Console Displays: -- Executing
Playback("SIP/200-ad26", "demo-abouttotry") in new stack -- Playing 'demo-abouttotry' (language
'en') -- Executing
Dial("SIP/200-ad26",
"IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]") in new stack -- Called
[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 216.207.245.8 (format
gsm) -- Format for call is gsm -- IAX2/216.207.245.8:4569-1 is ringing -- IAX2/216.207.245.8:4569-1 answered
SIP/200-ad26 -- Hungup 'IAX2/216.207.245.8:4569-1' == Spawn extension (from-sip, 861, 2) exited non-zero
on 'SIP/200-ad26' asterisk1*CLI> asterisk1*CLI> -- Starting simple switch on 'Zap/25-1' -- Executing
Playback("Zap/25-1", "demo-abouttotry") in new stack -- Playing 'demo-abouttotry' (language
'en') -- Executing Dial("Zap/25-1",
"IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]") in new stack -- Called
[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 216.207.245.8 (format
gsm) -- Format for call is gsm -- IAX2/216.207.245.8:4569-2 is ringing -- IAX2/216.207.245.8:4569-2 answered
Zap/25-1
<---------- THIS IS WHERE THE AUDIO BECOMES ALL CHOPPED UP. -- Hungup 'IAX2/216.207.245.8:4569-2' == Spawn extension (chan_bank, 861, 2) exited
non-zero on 'Zap/25-1' -- Hungup 'Zap/25-1' asterisk1*CLI> |
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