Kinda confused by this... I have a Cisco 7960 configured with a
couple SIP extensions configured on the phone. Just trying to dial
one extension from the other on the same phone, but when I do, I get:
-- Remote UNIX connection
-- Executing Dial("SIP/2001-ffd4", "SIP/2002") in new
stack
-- Called 2002
-- Got SIP response 486 "Busy here" back from xxx.xx.xx.xxx
-- SIP/2002-f29b is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Got SIP response 481 "Call Leg/Transaction Does Not Exist"
back from xxx.xx.xx.xxx
Any ideas?
# sip.conf
[2001]
type=friend
username=2001
secret=hjksdfg23ASDF
context=ario-extensions
host=dynamic
nat=yes
register=yes
qualify=yes
disallow=all
allow=ulaw
mailbox=2000@ario
[2002]
type=friend
username=2002
secret=hjksdfg23ASDF
context=ario-extensions
host=dynamic
nat=yes
register=yes
qualify=yes
disallow=all
allow=ulaw
mailbox=2000@ario
# extensions.conf
[ario-extensions]
exten => 2000,1,GoTo(2001,1)
exten => 2001,1,Dial(SIP/2001)
exten => 2002,1,Dial(SIP/2002)
# asterisk -rx "sip show peers"
Name/username Host Dyn Nat ACL Port Status
2002/2002 xxx.xx.xx.xxx D N 5060 OK
(198 ms)
2001/2001 xxx.xx.xx.xxx D N 5060 OK
(167 ms)
2 sip peers [2 online , 0 offline]
-- Remote UNIX connection
how baout codecs ? try enabling all for testing ..then limit.. On 6/9/06, Jason Lixfeld <jason+lists.asterisk@lixfeld.ca> wrote:> > Kinda confused by this... I have a Cisco 7960 configured with a > couple SIP extensions configured on the phone. Just trying to dial > one extension from the other on the same phone, but when I do, I get: > > -- Remote UNIX connection > -- Executing Dial("SIP/2001-ffd4", "SIP/2002") in new stack > -- Called 2002 > -- Got SIP response 486 "Busy here" back from xxx.xx.xx.xxx > -- SIP/2002-f29b is busy > == Everyone is busy/congested at this time (1:1/0/0) > -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" > back from xxx.xx.xx.xxx > > Any ideas? > > # sip.conf > [2001] > type=friend > username=2001 > secret=hjksdfg23ASDF > context=ario-extensions > host=dynamic > nat=yes > register=yes > qualify=yes > disallow=all > allow=ulaw > mailbox=2000@ario > > [2002] > type=friend > username=2002 > secret=hjksdfg23ASDF > context=ario-extensions > host=dynamic > nat=yes > register=yes > qualify=yes > disallow=all > allow=ulaw > mailbox=2000@ario > > # extensions.conf > [ario-extensions] > exten => 2000,1,GoTo(2001,1) > exten => 2001,1,Dial(SIP/2001) > exten => 2002,1,Dial(SIP/2002) > > # asterisk -rx "sip show peers" > Name/username Host Dyn Nat ACL Port Status > 2002/2002 xxx.xx.xx.xxx D N 5060 OK > (198 ms) > 2001/2001 xxx.xx.xx.xxx D N 5060 OK > (167 ms) > 2 sip peers [2 online , 0 offline] > -- Remote UNIX connection > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Mike Sales Manager http://www.theclubvoip.com Making it happen 1.888.470.7253 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060609/13a52f1f/attachment.htm
On 6/9/06, Jason Lixfeld <jason+lists.asterisk@lixfeld.ca> wrote:> Kinda confused by this... I have a Cisco 7960 configured with a > couple SIP extensions configured on the phone. Just trying to dial > one extension from the other on the same phone, but when I do, I get: >Could the phone be returning 'busy' because you are on a call in dial-state (as opposed to an established call)?
g711ulaw was the default coded on the 7960. ulaw was explicitly allowed in the sip.conf while disallowing all others, so technically that should have worked. I did, for fun allow=all in sip.conf, but still the same. On 9-Jun-06, at 2:39 PM, Mike Lynchfield wrote:> how baout codecs ? > > try enabling all for testing ..then limit.. > > > > On 6/9/06, Jason Lixfeld < jason+lists.asterisk@lixfeld.ca> > wrote:Kinda confused by this... I have a Cisco 7960 configured with a > couple SIP extensions configured on the phone. Just trying to dial > one extension from the other on the same phone, but when I do, I get: > > -- Remote UNIX connection > -- Executing Dial("SIP/2001-ffd4", "SIP/2002") in new stack > -- Called 2002 > -- Got SIP response 486 "Busy here" back from xxx.xx.xx.xxx > -- SIP/2002-f29b is busy > == Everyone is busy/congested at this time (1:1/0/0) > -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" > back from xxx.xx.xx.xxx > > Any ideas? > > # sip.conf > [2001] > type=friend > username=2001 > secret=hjksdfg23ASDF > context=ario-extensions > host=dynamic > nat=yes > register=yes > qualify=yes > disallow=all > allow=ulaw > mailbox=2000@ario > > [2002] > type=friend > username=2002 > secret=hjksdfg23ASDF > context=ario-extensions > host=dynamic > nat=yes > register=yes > qualify=yes > disallow=all > allow=ulaw > mailbox=2000@ario > > # extensions.conf > [ario-extensions] > exten => 2000,1,GoTo(2001,1) > exten => 2001,1,Dial(SIP/2001) > exten => 2002,1,Dial(SIP/2002) > > # asterisk -rx "sip show peers" > Name/username Host Dyn Nat ACL Port Status > 2002/2002 xxx.xx.xx.xxx D N 5060 OK > (198 ms) > 2001/2001 xxx.xx.xx.xxx D N 5060 OK > (167 ms) > 2 sip peers [2 online , 0 offline] > -- Remote UNIX connection > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > Mike > Sales Manager > http://www.theclubvoip.com > Making it happen > 1.888.470.7253 > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users