[RESOLVED] - Limited to dialing 3 extensions max


#1

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


#2

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.


#3

Have you tried dialing without the OSS/dsp in there? Just to eliminate that as a possible issue?


#4

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]


#5

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…


#6

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!