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Asterisk setup should be able to have 4 active calls, but allows only 2
Posted on 2013-01-01
How to get Asterisk to failover to maximize active calls. When I make my fist two calls they both go on one trunk, but when I make my 3rd and 4th call, instead of going to the second trunk, I get a busy signal. I know both trunks are working separately. I have an outbound route setup with the two trunks in the trunk sequence. I'm not sure what other information to give you right now.
Setup FreePBX\PBX-in a flash disro
VIOP.ms trunk and DID
Viatalk trunk and DID
Setup FreePBX\PBX-in a flash disro
VIOP.ms trunk and DID
Viatalk trunk and DID
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5 Comments
Well,
This is a little dated but you might want to look here --
http://www.freepbx.org/boo
Then start the CLI and check the status of your peer and Registry
> sip show peers
> sip show registry
Lastly, look for the dial() in /var/log/asterisk/full and see what kind of errors you are getting.
========================
NOTE: If you post any log data back here use code tags, or place it in a text file and attach it.
This is a little dated but you might want to look here --
http://www.freepbx.org/boo
Then start the CLI and check the status of your peer and Registry
> sip show peers
> sip show registry
Lastly, look for the dial() in /var/log/asterisk/full and see what kind of errors you are getting.
========================
NOTE: If you post any log data back here use code tags, or place it in a text file and attach it.
Here is the log data for the call that got the busy signal. As you can see it is just trying the first trunk. As the time this call was placed there were already two calls on Viatalk:
SIP SHOW PEERS
SIP SHOW REGISTRY
@Sandeep I'll check it out
[2013-01-02 12:43:59] VERBOSE[719] logger.c: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/108-09578178", "SIP/viatalk/8122822787|300|") in new stack
[2013-01-02 12:43:59] WARNING[719] rtp.c: Unable to set TOS to 184
[2013-01-02 12:43:59] VERBOSE[719] logger.c: -- Called viatalk/8122822787
[2013-01-02 12:43:59] VERBOSE[3494] logger.c: -- Got SIP response 603 "Declined" back from 216.246.105.146
[2013-01-02 12:43:59] VERBOSE[719] logger.c: -- SIP/viatalk-0962fa98 is busy
[2013-01-02 12:43:59] VERBOSE[719] logger.c: == Everyone is busy/congested at this time (1:1/0/0)
[2013-01-02 12:43:59] DEBUG[719] app_macro.c: Executed application: Dial
[2013-01-02 12:43:59] VERBOSE[719] logger.c: -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/108-09578178", "Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 21") in new stack
[2013-01-02 12:43:59] DEBUG[719] app_macro.c: Executed application: Noop
[2013-01-02 12:43:59] VERBOSE[719] logger.c: -- Executing [s@macro-dialout-trunk:21] Goto("SIP/108-09578178", "s-BUSY|1") in new stack
[2013-01-02 12:43:59] VERBOSE[719] logger.c: -- Goto (macro-dialout-trunk,s-BUSY,1)
[2013-01-02 12:43:59] DEBUG[719] app_macro.c: Executed application: Goto
[2013-01-02 12:43:59] VERBOSE[719] logger.c: -- Executing [s-BUSY@macro-dialout-trunk:1] NoOp("SIP/108-09578178", "Dial failed due to trunk reporting BUSY - giving up") in new stack
[2013-01-02 12:43:59] DEBUG[719] app_macro.c: Executed application: Noop
[2013-01-02 12:43:59] VERBOSE[719] logger.c: -- Executing [s-BUSY@macro-dialout-trunk:2] PlayTones("SIP/108-09578178", "busy") in new stack
[2013-01-02 12:43:59] DEBUG[719] app_macro.c: Executed application: Playtones
SIP SHOW PEERS
Name/username Host Dyn Nat ACL Port Status
Voipms/xxxxxx xxx.xxx.xxx.xxxx N 5060 OK (260 ms)
viatalk/xxxxxxxxxx xxx.xxx.xxx.xxx N 5060 OK (71 ms)
972/972 192.168.0.150 D A 5060 OK (89 ms)
540 (Unspecified) D N A 0 UNKNOWN
324/324 (Unspecified) D N A 0 UNKNOWN
216/216 192.168.0.163 D N A 5060 OK (10 ms)
1080/1080 192.168.0.128 D A 5060 OK (12 ms)
108/108 192.168.0.142 D N A 5060 OK (4 ms)
8 sip peers [Monitored: 6 online, 2 offline Unmonitored: 0 online, 0 offline]
SIP SHOW REGISTRY
Host Username Refresh State Reg.Time
atlanta.voip.ms:5060 xxxxxx 105 Registered Wed, 02 Jan 2013 13:14:15
chicago-1f.vtnoc.net:5060 xxxxxxxxxx 105 Registered Wed, 02 Jan 2013 13:14:48
@Sandeep I'll check it out
Okay,
Based on this I think you need to check on two things --
In the Out Route for this dial pattern.. you have both trunks specified under the Trunk Sequence ?
In the Trunk Setup do you have Maximum Channels set to two ? In both trunks do you have the correct Dial Rules, and outbound Dial Prefixes respectfully.
SIP response 603 is not really a Busy, if the above settings are correct you may have identified a bug in the FreePBX code generation scripts and if so opening a ticket would be the correct action to take to get a patch for it.
http://www.freepbx.org/trac/simpleticket
-----------------------
Based on this I think you need to check on two things --
-- Got SIP response 603 "Declined" back from 216.246.105.146
In the Out Route for this dial pattern.. you have both trunks specified under the Trunk Sequence ?
In the Trunk Setup do you have Maximum Channels set to two ? In both trunks do you have the correct Dial Rules, and outbound Dial Prefixes respectfully.
SIP response 603 is not really a Busy, if the above settings are correct you may have identified a bug in the FreePBX code generation scripts and if so opening a ticket would be the correct action to take to get a patch for it.
http://www.freepbx.org/trac/simpleticket
-----------------------
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