Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1o
Here’s a strange one. When dialing multiple SIP extensions at the same time, only the first three listed are called:
exten => 102,1,Dial(SIP/250&SIP/241&SIP/240&SIP/210&SIP/230&SIP/231&SIP/232&SIP/233)
results in:
-- Called 250
-- Called 241
-- Called 240
-- SIP/250-da81 is ringing
-- SIP/241-925d is ringing
-- SIP/240-d15c is ringing
Nothing else. What about the other five extensions?
Full CLI:
localhost*CLI> set verbose 99
Verbosity was 7 and is now 99
localhost*CLI> dial 102
-- Executing Dial("OSS/dsp", "SIP/250&SIP/241&SIP/240&SIP/210&SIP/230&SIP/231&SIP/232&SIP/233") in new stack
-- Called 250
-- Called 241
-- Called 240
-- SIP/250-da81 is ringing
-- SIP/241-925d is ringing
-- SIP/240-d15c is ringing
localhost*CLI> hangup
== Spawn extension (from-internal, 102, 1) exited non-zero on 'OSS/dsp'
-- Executing Macro("OSS/dsp", "hangupcall") in new stack
-- Executing ResetCDR("OSS/dsp", "w") in new stack
-- Executing NoCDR("OSS/dsp", "") in new stack
-- Executing Wait("OSS/dsp", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'OSS/dsp' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'OSS/dsp'
<< Hangup on console >>
No matter how I change the order of the extensions in the Dial command, Asterisk only dials the first three. I enabled SIP debugging to the IP address associated with SIP/210 (next in Dial command) and get no output.
I’m stumped. Why does Asterisk only dial a maximum of three estensions at the same time. Any advise?
There is a new version of BRIstuff but the Changelog only says:
0.3.0-PRE-1p
- fixed answer confirmation for chan_zap (Dial(ZAP/g1c/…)
(DTMF detection was disabled.)
- added ztpty to zaptel
I’ve tried working around this issue by creating a queue and making all the extensions permament members of the queue. Same result (well, worse), only the first TWO members are dialed.
Have you tried dialing without the OSS/dsp in there? Just to eliminate that as a possible issue?
Yes:
[quote]localhost*CLI>
– Executing Dial(“SIP/501-8bbd”, “SIP/250&SIP/241&SIP/240&SIP/210&SIP/230&SIP/231&SIP/232&SIP/233”) in new stack
– Called 250
– Called 241
– SIP/250-c93a is ringing
– SIP/241-3bd8 is ringing
== Spawn extension (from-internal, 103, 1) exited non-zero on ‘SIP/501-8bbd’
[/quote]
I tried this on my Asterisk 1.2.7:
gizmo_test => {
Dial(SIP/3000&SIP/3001&SIP/3002&SIP/3100&SIP/3200);
Hangup();
};
And then tried dialing with Gizmo and got:
-- Executing Dial("SIP/proxy01.sipphone.com-b72031a8", "SIP/3000&SIP/3001&SIP/3002&SIP/3100&SIP/3200") in new stack
May 20 11:35:17 NOTICE[22621]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
-- SIP Seeding peer from astdb: '3001' at 3001@myip:5064 for 180
-- Called 3001
-- SIP Seeding peer from astdb: '3002' at 3002@myip:5063 for 180
-- Called 3002
-- SIP Seeding peer from astdb: '3100' at 3100@myip:5060 for 180
-- Called 3100
-- SIP Seeding peer from astdb: '3001' at 3001@myip:5064 for 180
-- SIP Seeding peer from astdb: '3200' at 3200@myip:5060 for 180
-- Called 3200
-- SIP/3001-4b4e is ringing
-- SIP/3002-bbae is ringing
-- SIP Seeding peer from astdb: '3002' at 3002@myip:5063 for 180
-- SIP/3200-5ca4 is ringing
-- SIP Seeding peer from astdb: '3200' at 3200@myip:5060 for 180
-- SIP/3100-598e is ringing
-- SIP Seeding peer from astdb: '3100' at 3100@myip:5060 for 180
== Spawn extension (default, gizmo_test, 1) exited non-zero on 'SIP/proxy01.sipphone.com-b72031a8'
It worked fine, the fact that it did not connect to 3000 is fine as it is not actually connected to the Asterisk server.
Strange problem you are having…
Well, after upgrading and also reverting to older versions with no luck, I decided to examine /var/log/asterisk/full . Apparently I did not have enough RTP ports available to create more SIP channels. Editing rtp.conf fixed that right-quick.
Morale of the story, read the logs first!