Im really starting to get the hang of Asterisk, however, I still have
one issue...
My SIP Client can dial other extensions just fine, but no extension can
ring the Sip client...
Here is the pertinent info:
SIP.CONF,
[general]
port = 5060
bindaddr = 192.168.0.5 ;ip of asterisk server
context = default
[301]
username=301
context=local
type=friend
secret=test
insecure=yes
host=dynamic
----------------------------------------
EXTENSIONS.CONF
[local]
exten => _1XX,1,Dial,ZAP/1/BYEXTENSION
exten => 301,1,Dial,SIP/sip:301@192.168.0.5 ; again, ip of * server
blah blah blah below this..
----------------------------------------
Console Debug:
When 301 is Dialed:
--Executing Dial("OSS/dsp", "SIP/sip:301@192.168.0.5") in
new stack
Called sip:301@192.168.0.5
Got SIP response 482 "Loop Detected" back from 192.168.0.5
No one is available to answer at this time
WARNING[114703]: File pbx.c, Line 1268 (ast_pbx_run): Timeout, but no
rule 't' in context 'local'
----------------------------------------
Problem is, the SIP Client never rang.....
Now...If I change the extensions.conf to read:
Exten => 301,1,Dial,SIP/sip:301@192.168.0.109
Then it works fine...problem is, 192.168.0.109 is a DHCP'd ip address to
the sip client machine...It will change occasionally...
Any Ideas? Thanks!
Ben Bawkon
Replace exten => 301,1,Dial,SIP/sip:301@192.168.0.5 with: exten => 301,1,Dial,SIP/301 --Mike On Wednesday, March 26, 2003, at 08:40 AM, Benjamin J. Bawkon wrote:> Im really starting to get the hang of Asterisk, however, I still have > one issue... > > My SIP Client can dial other extensions just fine, but no extension can > ring the Sip client... > > Here is the pertinent info: > SIP.CONF, > [general] > port = 5060 > bindaddr = 192.168.0.5 ;ip of asterisk server > context = default > > [301] > username=301 > context=local > type=friend > secret=test > insecure=yes > host=dynamic > > ---------------------------------------- > EXTENSIONS.CONF > > [local] > exten => _1XX,1,Dial,ZAP/1/BYEXTENSION > exten => 301,1,Dial,SIP/sip:301@192.168.0.5 ; again, ip of * server > > blah blah blah below this.. > > ---------------------------------------- > Console Debug: > > When 301 is Dialed: > > --Executing Dial("OSS/dsp", "SIP/sip:301@192.168.0.5") in new stack > Called sip:301@192.168.0.5 > Got SIP response 482 "Loop Detected" back from 192.168.0.5 > No one is available to answer at this time > WARNING[114703]: File pbx.c, Line 1268 (ast_pbx_run): Timeout, but no > rule 't' in context 'local' > > ---------------------------------------- > Problem is, the SIP Client never rang..... > > Now...If I change the extensions.conf to read: > Exten => 301,1,Dial,SIP/sip:301@192.168.0.109 > > Then it works fine...problem is, 192.168.0.109 is a DHCP'd ip address > to > the sip client machine...It will change occasionally... > > Any Ideas? Thanks! > Ben Bawkon > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Mike Reiling Systems & Network Administrator SoftCoin, Inc. 2000 Sierra Point Parkway Brisbane, CA 94005 650-624-3869 - P 650-624-3899 - F It might look like I'm doing nothing, but at the cellular level I'm really quite busy.
I have it working.
This is what my extensions.conf looks like....
exten => 301,1,Dial,sip/301
That 301 corresponds with the [301] section in sip.conf
-----Original Message-----
From: Benjamin J. Bawkon [mailto:bbawkon@malibutech.com]
Sent: 26 March 2003 16:41
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Dialing SIP
Im really starting to get the hang of Asterisk, however, I still have
one issue...
My SIP Client can dial other extensions just fine, but no extension can
ring the Sip client...
Here is the pertinent info:
SIP.CONF,
[general]
port = 5060
bindaddr = 192.168.0.5 ;ip of asterisk server
context = default
[301]
username=301
context=local
type=friend
secret=test
insecure=yes
host=dynamic
----------------------------------------
EXTENSIONS.CONF
[local]
exten => _1XX,1,Dial,ZAP/1/BYEXTENSION
exten => 301,1,Dial,SIP/sip:301@192.168.0.5 ; again, ip of * server
blah blah blah below this..
----------------------------------------
Console Debug:
When 301 is Dialed:
--Executing Dial("OSS/dsp", "SIP/sip:301@192.168.0.5") in
new stack
Called sip:301@192.168.0.5
Got SIP response 482 "Loop Detected" back from 192.168.0.5
No one is available to answer at this time
WARNING[114703]: File pbx.c, Line 1268 (ast_pbx_run): Timeout, but no
rule 't' in context 'local'
----------------------------------------
Problem is, the SIP Client never rang.....
Now...If I change the extensions.conf to read:
Exten => 301,1,Dial,SIP/sip:301@192.168.0.109
Then it works fine...problem is, 192.168.0.109 is a DHCP'd ip address to
the sip client machine...It will change occasionally...
Any Ideas? Thanks!
Ben Bawkon
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users