Displaying 5 results from an estimated 5 matches for "siprout".
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sipout
2015 May 20
2
reduce delay in fax detection
hello everybody
i want to send fax via asterisk in pass through mode. everything is ok if
enable fax detection in ooh323 and write fax extension in extensions.conf
file. just one problem: delay. i have to wait 5 seconds in order to fax
detection done. it is too long for me when i have voice call and no fax. my
phone rings after five seconds. is there any way to omit or reduce this
time? i test
2015 May 21
1
reduce delay in fax detection
...me crazy. How I solved it was by doing a ?core show channels
> <concise|verbose>? and detect if there was a fax transmission going on.
> Doing it this way shows up instantaneously without any delay. Like so:
>
>
>
> mte6*CLI> core show channels concise
>
>
> SIP/SIPRoutes-00000054!faxserver-tx!fax!11!Up!SendFAX!/var/spool/asterisk/fax/documents/faxadmin/default.tif,dfz!!voipbusiness!!3!4!(None)!1432132315.97
>
>
>
> mte6*CLI> core show channels verbose
>
> Channel Context Extension Prio State
> Application...
2015 May 20
0
reduce delay in fax detection
...delay with our fax server and it drove me crazy. How I solved it was by doing a ?core show channels <concise|verbose>? and detect if there was a fax transmission going on. Doing it this way shows up instantaneously without any delay. Like so:
mte6*CLI> core show channels concise
SIP/SIPRoutes-00000054!faxserver-tx!fax!11!Up!SendFAX!/var/spool/asterisk/fax/documents/faxadmin/default.tif,dfz!!voipbusiness!!3!4!(None)!1432132315.97
mte6*CLI> core show channels verbose
Channel Context Extension Prio State Application Data C...
2006 Oct 10
3
Understanding NAT Traversal
Quick question re. NAT traversal.
I understand how sitting behind a NAT could cause problems for a SIP UA.
The SIP UA would create SIP mesages using IP addresses from inside the
network (i.e. 192.#.#.# or 10.#.#.#) and these IP addresses are of course
unnavigable for the recipient.
What I don't get is why don't web browsers suffer the same problem?
A web brower behind a NAT sends an
2014 Jan 15
2
No compatible codecs, not accepting this offer!
Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
---
username=5x5x7x9x0x3
type=friend
secret=CRcxn7sqwm
qualify=yes
port=5060
insecure=port,invite
host=sip.txtxlxoxp.it
fromuser=5x5x7x9x0x3
fromdomain=sip.txtxlxoxp.it
disallow=all
context=from-trunk
allow=alaw
---
A typical invite from my