So, here's my quandary:
1) Asterisk running CVS HEAD as of a couple days ago
2) Cisco 7960 SIP phones in a different subnet than the Asterisk server
3) NAT/Firewall device between 7960's and *
I can initiate a call from the 7960's just fine. They can call out
using our Broadvoice account and access any of the vmail stuff on *.
When calling in from the outside world and dialing one of their
extensions, however, I always get a "this user is on the phone"
message.
The console spits out this nugget:
== CDR updated on SIP/4252780761-933d
-- Executing Macro("SIP/4252780761-933d",
"stdsip|tycisco|101") in
new stack
-- Executing Dial("SIP/4252780761-933d", "SIP/tycisco")
in new stack
Apr 20 08:14:59 NOTICE[32728]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
== Everyone is busy/congested at this time (1:0/1/0)
A showing of the sip peers:
sip show peers
Name/username Host Dyn Nat ACL Mask
Port Status
rickcisco/cisco2 (Unspecified) D N 255.255.255.255
0 UNKNOWN
tycisco/cisco1 (Unspecified) D N 255.255.255.255
0 UNKNOWN
sip.broadvoice.com/425278 147.135.4.128 255.255.255.255
5060 OK (127 ms)
3 sip peers [1 online , 2 offline]
I'm sure the reason I can't call to an extension is that they are
appearing offline. How can I remedy this, however?
I'm an * newbie, so go easy on me. :^)
Thanks,
Ty Christensen
MCP, MCSP, MCSB
Master Mind Productions Inc.
www.mastermindpro.com <http://www.mastermindpro.com/>
(425) 378-7724
Looks like you have sip.conf set up to expect registrations for tycisco
since it has a D for dynamic.
You can either set up the 7960 to register with asterisk and use something
like this in sip.conf:
[tycisco]
type=friend
username= someusername
secret= somesecret
insecure=no
mailbox=757
host=dynamic
callerid=""
or just not have the 7960 register and specify its IP address using the
"host=" line instead.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of List Receiver
Sent: Wednesday, April 20, 2005 11:19 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco 7960 SIP registration???
So, here's my quandary:
1) Asterisk running CVS HEAD as of a couple days ago
2) Cisco 7960 SIP phones in a different subnet than the Asterisk server
3) NAT/Firewall device between 7960's and *
I can initiate a call from the 7960's just fine. They can call out
using our Broadvoice account and access any of the vmail stuff on *.
When calling in from the outside world and dialing one of their
extensions, however, I always get a "this user is on the phone"
message.
The console spits out this nugget:
== CDR updated on SIP/4252780761-933d
-- Executing Macro("SIP/4252780761-933d",
"stdsip|tycisco|101") in
new stack
-- Executing Dial("SIP/4252780761-933d", "SIP/tycisco")
in new stack
Apr 20 08:14:59 NOTICE[32728]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
== Everyone is busy/congested at this time (1:0/1/0)
A showing of the sip peers:
sip show peers
Name/username Host Dyn Nat ACL Mask
Port Status
rickcisco/cisco2 (Unspecified) D N 255.255.255.255
0 UNKNOWN
tycisco/cisco1 (Unspecified) D N 255.255.255.255
0 UNKNOWN
sip.broadvoice.com/425278 147.135.4.128 255.255.255.255
5060 OK (127 ms)
3 sip peers [1 online , 2 offline]
I'm sure the reason I can't call to an extension is that they are
appearing offline. How can I remedy this, however?
I'm an * newbie, so go easy on me. :^)
Thanks,
Ty Christensen
MCP, MCSP, MCSB
Master Mind Productions Inc.
www.mastermindpro.com <http://www.mastermindpro.com/>
(425) 378-7724
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I've done that...I think. :^) Here's the excerpt from sip.conf: [tycisco] type=friend username=cisco1 secret=******* qualify=200 ; Qualify peer is no more than 200ms away nat=yes ;insecure=no host=dynamic ; This device registers with us ;defaultip=192.168.0.30 canreinvite=no context=fullaccess dtmfmode=inband mailbox=101 disallow=all allow=ulaw allow=alaw allow=g729 I still get no registration when I do a sip show peers. Am I missing something simple? Thanks, Ty> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of end1r > Sent: Wednesday, April 20, 2005 8:58 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] Cisco 7960 SIP registration??? > > Looks like you have sip.conf set up to expect registrations > for tycisco since it has a D for dynamic. > > You can either set up the 7960 to register with asterisk and > use something like this in sip.conf: > > > [tycisco] > type=friend > username= someusername > secret= somesecret > insecure=no > mailbox=757 > host=dynamic > callerid="" > > or just not have the 7960 register and specify its IP address > using the "host=" line instead. > > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > List Receiver > Sent: Wednesday, April 20, 2005 11:19 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Cisco 7960 SIP registration??? > > So, here's my quandary: > > 1) Asterisk running CVS HEAD as of a couple days ago > 2) Cisco 7960 SIP phones in a different subnet than the > Asterisk server > 3) NAT/Firewall device between 7960's and * > > I can initiate a call from the 7960's just fine. They can > call out using our Broadvoice account and access any of the > vmail stuff on *. > When calling in from the outside world and dialing one of > their extensions, however, I always get a "this user is on > the phone" message. > > The console spits out this nugget: > == CDR updated on SIP/4252780761-933d > -- Executing Macro("SIP/4252780761-933d", > "stdsip|tycisco|101") in new stack > -- Executing Dial("SIP/4252780761-933d", "SIP/tycisco") > in new stack Apr 20 08:14:59 NOTICE[32728]: app_dial.c:973 > dial_exec_full: Unable to create channel of type 'SIP' (cause 3) > == Everyone is busy/congested at this time (1:0/1/0) > > A showing of the sip peers: > sip show peers > Name/username Host Dyn Nat ACL Mask > Port Status > rickcisco/cisco2 (Unspecified) D N 255.255.255.255 > 0 UNKNOWN > tycisco/cisco1 (Unspecified) D N 255.255.255.255 > 0 UNKNOWN > sip.broadvoice.com/425278 147.135.4.128 255.255.255.255 > 5060 OK (127 ms) > 3 sip peers [1 online , 2 offline] > > I'm sure the reason I can't call to an extension is that they > are appearing offline. How can I remedy this, however? > > I'm an * newbie, so go easy on me. :^) > > Thanks, > > Ty Christensen > MCP, MCSP, MCSB > Master Mind Productions Inc. > www.mastermindpro.com <http://www.mastermindpro.com/> > (425) 378-7724 > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3032 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050420/72a6b76c/smime.bin