Hi
ne1 know if there has been recent (over easter) cvs changes to what happens
when nat =yes especially in relation to sip
some things seem to work for me that didn't before ;)
thanks
walt.
_
Express yourself instantly with MSN
Hi
I am using cvs and updating it every couple of days Unfortunately I am still
getting a 20 second timeout on sip calls placed to various providers, can
anyone see anything wrong from sip debugs? Or have any ideas what the
problem might be?
Cheers
Walt
sip debug peer of a provider:
I'm having the same problem over here, but with both, inbound/outbound
calls, I use a SER server auth my users, and when I need to use a VoIP
line that is not at my server, I use Asterisk to auth line outside my
server at my Foreign Voip server then when I get the line I can dial, but
none of
Hi
I am having alot of difficulty connecting to SIP providers (I am trying 3)
and can't seem to find anything similar in the wiki or on the lists.I
can receive inbound calls fine however on placing an outbound call the
calling phone never gets 'connected' but 2 way audio is passed for about
Sounds like you are having a codec issue with 2 of your providers. Make
sure you find out what codecs are supported and that your config is set up
accordingly.
Thanks :)
I don't think that is it though as I have tried with other codecs initially
and inbound calls work fine regardless.
My
Hi
Hope someone can help :)
I am testing 4 PSTN termination providers. 3 SIP and 1 IAX
IAX and 1 of the SIP providers work fine.
Now the wierdness:
2 SIP providers I can only get oubound calls to ring at the destination and
then nothing more. 1 gets as far as SIP code 183 (and ringing on the src
'auto-answer' script for the 79XX phones. It basically telnets into the
phone and presses the answer key
Thanks Chris
I suppose I could make a dial out command via telnet as well for the cisco.
Other option I want to try is using agents - this to allow a degree of
roaming users - course with
I too am having the same problem with =VS from last night. From my
debugging, * never attempts to start MOH. Anyone else =ound this?
Me too
Music on hold - with SIP handsets at least - stopped working for me with
asterisk 1.0.6 and cvs.
If I downgraded to 1.0.5 works fine, upgrade and it stops
I am thinking about a making a web based directory that dials a number with
one click.
From an overview picture does the below look like the correct way to go
about it:
web app sends something like the below call file to asterisk
Action: Originate
Channel: SIP/1010
Context: demo
Exten: 1234
Wondering if anyone has any tweaks to get optimum headset sound quality with
polycoms?
atm i'm trying changing the gains, aes aec, in imipmid.cfg
headsets are unamplified, UNEX GNetcom using G729 codec with SIP 1.4.1
firmware
Thanks
Walter
I had similar distortion issues and shutdown problems, I had better results
using madplay instead of mpg123 as per the wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musiconhold.conf
Cheers
Walt
_
FREE pop-up
Hi
I have changed my settings to make the whole thing consistent username =
extension = first bit of sip address
found the problem with registration is simply username/password used.
for some reason the setting on phone.cfg under reg.1.auth.userId is NOT
being passed to the phone and is just
Hi
I am just getting started with asterisk and trying out using the sample
files with a Polycom IP500 (latest sip.ld etc)
My question:
phone status says not registered:
/system status/server status Line 1 poly is not registered
console message says:
handle_request: Registration from 'sip:[EMAIL
Hi
I am struggling with hardware choices to get started with. My options are
narrowed down to SIP phones - Polycom IP500, IP600 and Cisco 7940G.
of importance is:
- functionality / integration with asterisk
- headset functionality and use
- voice quality
- build quality
Is there much of a
14 matches
Mail list logo
