Displaying 11 results from an estimated 11 matches for "sipout".
2006 Jun 06
1
Asterisk Realtime and SIP Registration
Hi!
I use the following configuration to register my asterisk server to my SIP
provider:
register => 12345:passwd@sip.provider.com/12345
sip.conf:
[sipout-test]
type=peer
username=12345
fromuser=12345
fromdomain=provider.com
secret=passwd
insecure=very
host=sip.provider.com
qualify=yes
context=test-incoming
extensions.conf:
exten => 12345,1,Dial(SIP/10)
exten => _0NXZXXXXXX,1,Dial(SIP/${EXTEN}@sipout-test)
This works fine when I put it into t...
2004 Jan 23
3
SIP Absolute Timeout
...in for SIP? I've search all the messages in the
news letters and tried what was suggested and still have not gotten it to
work. Below is a portion of my extensions.conf. I've also been running these
test on ver 0.5.0
exten => _X.,1,Absolutetimeout(20)
exten => _X.,2,dial(SIP/${EXTEN}@SIPOUT#1)
exten => T,1,BackGround(tt-weasels)
exten => T,2,Hangup()
Thanks ahead of time for any help / suggestions.
Wes Marderness
2008 May 01
1
ast_indicate_data: Unable to handle indication 3
Hi guys,
When I try to get ring tones when dialing out with the command
Dial(SIP/sipout/${PHONE},15,r), I get the error message indicated in the
subject. I've checked my indications.conf file using the sample file
provided with asterisk 1.4.10 (the version I'm using) and it's not better.
Any idea ?
Regards.
--
Cyril SCETBON
2004 Jan 19
2
RE: current version
To be clear I meant using Chan)_h323 with Call Manager where CM is
configured
with * as a H.323 gateway, not client.
CM supports H.323 to direct calls through gateways, and in fact until
recently
that is all they supported. They now also have MGCP, but only to their
IOS
platforms, and SIP is coming soon. There are NO sccp-based gateways,
from Cisco
anyways.
Dan
-----Original Message-----
2010 Aug 27
0
Duplicate channel variables after transfer
...els and in the h extension I want to
access the SIPCALLID variable of the A channel. Every access to it gives me
the wrong value namely that of channel B1. How do i access the _second_
variable named SIPCALLID in the channel?
Extract from DumpChan() as an example:
Dumping Info For Channel: SIP/sipout-00000055<ZOMBIE>:
================================================================================
Info:
Name= SIP/sipout-00000055<ZOMBIE>
Type= SIP
UniqueID= 1282913436.108
....
Variables:
...
SIPCALLID=eae94252-ebf238ff at 172.28.4.112
...
SIPCALL...
2004 Dec 14
3
Realtime problem
I'm having trouble with the Realtime setup. I've followed the instructions on
voip-info using odbc but I get this message during asterisk boot:
Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory)
Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load
config sip.conf, SIP disabled
== Registered channel type 'SIP' (Session
2005 Aug 07
4
Configuring Asterisk@home for Sipgate.
Hi all,
I'm new to the forum. Oh no....newbie question coming, I hear you all cry!
I'm playing around with Asterisk@home and have installed software and fiddled around with sip and extensions files.
I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel.
I've
2006 Jan 11
1
[suse-isdn] Major Problems UTStarcom F1000 registering -- pls help
...nonymous@192.168.1.200>' failed for
'192.168.1.217' - Username/auth name mismatch
extract of [sip.conf]:
...................................
[UTStarcomF1000]
type=friend
bindport=5060
username=anonymous
;fromuser=anonymous
secret=welcome
mailbox=1000
canreinvite=yes
context=sipout
insecure=very
defaultip=192.168.1.217
host=dynamic
qualify=yes
nat=no
;auth=anonymous:welcome@192.168.1.217
dtmfmode=rcfa2833
....................................................
*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
UTStarcomF1000/anon...
2009 Oct 03
0
ERROR[1499]: rtp.c:2482 ast_rtcp_write_sr: RTCP SR transmission error
....22.XX.XX:
> requested format = alaw,
> requested prefs = disabled,
> actual format = alaw,
> host prefs = disabled,
> priority = reqonly
[Oct 3 17:40:55] -- Executing [0473775006 at from-BOX-YOCAN:1]
Set("IAX2/BOX-YOCAN-10022", "SIPOUT=YOCAN-3STARSNET") in new stack
[Oct 3 17:40:55] -- Executing [0473775006 at from-BOX-YOCAN:2]
NoOp("IAX2/BOX-YOCAN-10022", "call from BOX-YOCAN") in new stack
[Oct 3 17:40:55] -- Executing [0473775006 at from-BOX-YOCAN:3]
Dial("IAX2/BOX-YOCAN-10022", &qu...
2008 Jan 07
0
chan_mobile and W300i
...ult tries to redirect
context=fromsoftphone
port=5061 ; Uncomment this line if Ekiga and Asterisk are on the same host
more extensions.conf
[globals]
HOMEPHONE=XXXXXXXXX
MOBILEA=Mobile/W300i ;the mobile connected to asterisk
MOBILEB=Mobile/W300i/XXXXXXXXX ;the "remote" mobile
SIPOUT=SIP/sip-out
SOFTPHONE=SIP/1000
TIMEOUT=2
[general]
autofallthrough=yes
[fromsoftphone]
exten => 0,1,Answer()
exten => 0,n,Authenticate(XXXX)
exten => 0,n,Dial(${MOBILEB},30)
[frommobile]
exten => s,1,Answer()
exten => s,n,Authenticate(XXXX)
exten => s,n,Dial(${SOFTPHONE},45)
[...
2010 Aug 27
1
asterisk-users Digest, Vol 73, Issue 58
...gt; to
> access the SIPCALLID variable of the A channel. Every access to it gives me
> the wrong value namely that of channel B1. How do i access the _second_
> variable named SIPCALLID in the channel?
>
> Extract from DumpChan() as an example:
>
> Dumping Info For Channel: SIP/sipout-00000055<ZOMBIE>:
> ================================================================================
> Info:
> Name= SIP/sipout-00000055<ZOMBIE>
> Type= SIP
> UniqueID= 1282913436.108
> ....
> Variables:
> ...
> SIPCALLID=...