Am 14.04.2014 16:19, schrieb Eric Wieling:
So few people use Asteisk on OSX that I doubt anyone will answer.
Look how many answers he got, i got none
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Am 29.03.2014 11:12, schrieb Thomas Rechberger:
Many ITSP are using loadbalancers, so if somebody registers on a sip
peer with specific dns host, an incoming call may be received from a
different ip and the host value in peer section doesnt match, so it will
go to default context.
For example
Many ITSP are using loadbalancers, so if somebody registers on a sip
peer with specific dns host, an incoming call may be received from a
different ip and the host value in peer section doesnt match, so it will
go to default context.
For example Telekom or 1&1, biggest providers in Germany are
meanwhile i got an answer from fprior who was testing patch already.
add this code to chan_sip.c :
/* Allow domain to be overridden */
if (!ast_strlen_zero(p->fromdomain))
d = p->fromdomain;
else /* Save for any further attempts */
ast_stri
Am 20.03.2014 16:07, schrieb Joshua Colp:
That being said... this specific issue was brought up to me by a friend
and I was actually going to look at it over an upcoming weekend when I
have time. Can I guarantee it? No, but I'm going to try. That's how much
I care. Can I do this for every issue?
https://issues.asterisk.org/jira/browse/ASTERISK-20841
The patch was already posted by someone but then was deleted because of
guide lines. Is it really that hard to fix? Since 1.8 there is this
problem but nobody seems to care about. Asterisk isnt only used with
itsp who dont care about fromd
no trunking or bonding involved, so why just everybody calls this a trunk?
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externhost is monitoring for ip changes with an interval of
externrefresh, so far so good.
Wouldnt it be handy if asterisk would do an sip reregister if it detects
an ip change?
My SIP provider has sometimes very high intervals of 1 hour and if ip
changes, the registration doesnt work until it
When deny/permit is used in sip peer, does this only make sense when
host=dynamic is used? What happens if host=ip is set?
And if insecure=invite is set, does this override all above settings?
Whats the relation of those seettings?
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I am using voip with Vodafone as SIP peer for outbound telephony and i
have a huge problem establishing calls to other people. It works like in
1 of 5 tries. The peer is sending SIP 480 temporarily not available.
It took a while to identify this, because on the phone you just hear
busy tone.
On
Hello Matt
Your best option would be to parse out the values in the various channel
variables and store the ones you want.
thx, then i am gonna go this route. I got the CDR anyway displayed via
Perl already.
This usually works pretty well, except for CDRs, which are generally a
mess no ma
Am 02.12.2013 15:02, schrieb Raghav Goud:
Hi all,
I want peer-peer communication between two clients (soft phones)
both are behind NAT.I have an accounts with SIP proxy for two clients.
To communicate these two clients do i need to use Asterisk ?
As per my understanding Asterisk acts l
I am not sure if its just me, but i am able to get only local channel
variables containing RTCP QOS values.
The Version is 1.8.14.
I want to store values of bridged channel in CDR.
Phone is Cisco 7941 SIP and with sip show channelstats i see all the
relevant information (jitter,packet loss) i w
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