vicidial.org

by **newx22** » Wed Apr 15, 2009 10:22 am

astguiclient 2.0.5
Asterisk 1.2.30.2

when I do screen -r <asterisk screen id>
I see these line going incredibly fast on the screen ... whats going on ?

-- Executing Dial("SIP/213.166.93.15-afd02938", "SIP/911430500@voip_provider1|40|To") in new stack
-- Called 911430500@voip_provider1
Apr 15 15:18:13 WARNING[23569]: channel.c:2781 ast_channel_make_compatible: No path to translate from SIP/voip_provider1-0a1a8180(256) to SIP/213.166.93.15-afd02938(1024)
== Spawn extension (default, 911430500, 2) exited non-zero on 'SIP/213.166.93.15-b7142d48'
-- Executing DeadAGI("SIP/213.166.93.15-b7142d48", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing AGI("SIP/213.166.93.15-b71587f8", "agi://127.0.0.1:4577/call_log") in new stack
Apr 15 15:18:13 WARNING[2547]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0xb40eea10', 9 retries!
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/213.166.93.15-b71587f8", "SIP/911430500@voip_provider1|40|To") in new stack
-- Called 911430500@voip_provider1
Apr 15 15:18:13 WARNING[23574]: channel.c:2781 ast_channel_make_compatible: No path to translate from SIP/voip_provider1-0a850ba8(256) to SIP/213.166.93.15-b71587f8(1024)
== Spawn extension (default, 911430500, 2) exited non-zero on 'SIP/213.166.93.15-b02fac40'
-- Executing DeadAGI("SIP/213.166.93.15-b02fac40", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing AGI("SIP/213.166.93.15-b714dfc0", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/213.166.93.15-b714dfc0", "SIP/911430500@voip_provider1|40|To") in new stack
-- Called 911430500@voip_provider1
Apr 15 15:18:13 WARNING[23578]: channel.c:2781 ast_channel_make_compatible: No path to translate from SIP/voip_provider1-09e58820(256) to SIP/213.166.93.15-b714dfc0(1024)
== Spawn extension (default, 911430500, 2) exited non-zero on 'SIP/213.166.93.15-b408aea8'
-- Executing DeadAGI("SIP/213.166.93.15-b408aea8", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing AGI("SIP/213.166.93.15-b02fac40", "agi://127.0.0.1:4577/call_log") in new stack
Apr 15 15:18:13 WARNING[2547]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0xafd0c748', 9 retries!
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/213.166.93.15-b02fac40", "SIP/911430500@voip_provider1|40|To") in new stack
-- Called 911430500@voip_provider1
Apr 15 15:18:13 WARNING[23582]: channel.c:2781 ast_channel_make_compatible: No path to translate from SIP/voip_provider1-0a4c66e8(256) to SIP/213.166.93.15-b02fac40(1024)
== Spawn extension (default, 911430500, 2) exited non-zero on 'SIP/213.166.93.15-b406c180'
-- Executing DeadAGI("SIP/213.166.93.15-b406c180", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

by **mflorell** » Wed Apr 15, 2009 1:01 pm

What codecs are you using?

by **newx22** » Wed Apr 15, 2009 1:15 pm

iLBC on trunk1 and g729 on another

by **mflorell** » Wed Apr 15, 2009 1:18 pm

Do you have the license installed for G729 from Digium?

by **newx22** » Wed Apr 15, 2009 1:25 pm

only two agents are using g729, the rest 25 are using iLBC or maybe ulaw .
the g729 is the free version (codec_g729-ast12-gcc4-glibc-core2.so) that I downloaded from some website.

by **mflorell** » Wed Apr 15, 2009 1:31 pm

Did you compile that library that you downloaded into Asterisk?

by **newx22** » Wed Apr 15, 2009 1:45 pm

nop. just put it in asterisk modules directory

by **newx22** » Wed Apr 15, 2009 1:53 pm

also the output of show channels :
SIP/213.178.16.28-b3 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-08f61ab8 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-b3 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-09131430 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-ac 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-08ba7260 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-b3 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-09105788 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-b2 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-08bd6eb0 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-ac 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-08a67c58 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-b7 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-08f7b690 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-b3 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-0931c038 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-b7 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-097d9ae8 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-b2 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-08f89608 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-b3 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-08be0a08 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-b7 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-08eb28e0 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-ae 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-093f08b8 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-ae 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-08ce22a0 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-ae 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-08f54080 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-b7 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-09355db8 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-ae 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-08f22ab8 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-b2 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-0934cfa8 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-af 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-08a7fe68 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-aa 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-092a4b48 (None) Down AppDial((Outgoing Line))
SIP/213.178.16.28-b3 911430500@default:2 Ring Dial(SIP/911430500@voip_provider1|40|
SIP/voip_provider1-09695150 (None) Up Bridged Call(SIP/cc131-afe4527
SIP/cc131-afe45270 971347768@default:2 Up Dial(SIP/971347768@voip_provider1|40|
SIP/voip_provider1-09020cd8 (None) Down AppDial((Outgoing Line))
SIP/cc118-af544738 933460266@default:2 Ring Dial(SIP/933460266@voip_provider1|40|
SIP/voip_provider1-08a77578 (None) Up Bridged Call(SIP/cc101-b703e9d
SIP/cc101-b703e9d8 662441288@default:2 Up Dial(SIP/662441288@voip_provider1|40|
SIP/voip_provider1-09f068d8 (None) Up Bridged Call(SIP/cc112-b25393c
SIP/cc112-b25393c0 934141714@default:2 Up Dial(SIP/934141714@voip_provider1|40|
SIP/voip_provider1-08ce8ae0 (None) Down AppDial((Outgoing Line))
SIP/cc107-b7079ef8 986565063@default:2 Ring Dial(SIP/986565063@voip_provider1|40|
SIP/voip_provider1-08a41a68 (None) Up Bridged Call(SIP/cc133-af2b154
SIP/cc133-af2b1548 916781607@default:2 Up Dial(SIP/916781607@voip_provider1|40|
SIP/voip_provider1-093b6408 (None) Up Bridged Call(SIP/cc117-b35976e
SIP/cc117-b35976e0 934427399@default:2 Up Dial(SIP/934427399@voip_provider1|40|
SIP/voip_provider1-09316af8 8600057@default:1 Up MeetMe(8600057|F)
SIP/voip_provider1-092fe108 (None) Up Bridged Call(SIP/cc133-ab6c271
SIP/cc133-ab6c2710 917518319@default:2 Up Dial(SIP/917518319@voip_provider1|40|
Zap/pseudo-921653270 s@default:1 Rsrvd (None)
SIP/cc122-09ee6b58 8600057@default:1 Up MeetMe(8600057|F)
SIP/voip_provider1-094da5d8 (None) Up Bridged Call(SIP/cc100-b35d03d
SIP/cc100-b35d03d8 914892413@default:2 Up Dial(SIP/914892413@voip_provider1|40|
SIP/voip_provider1-08b89ec8 (None) Up Bridged Call(SIP/cc101-b70d94b
SIP/cc101-b70d94b0 987844504@default:2 Up Dial(SIP/987844504@voip_provider1|40|
Zap/pseudo-151635680 s@default:1 Rsrvd (None)
SIP/cc126-08d1c260 8600052@default:1 Up MeetMe(8600052|F)
SIP/voip_provider1-09116830 (None) Up Bridged Call(SIP/cc100-b70c478
SIP/cc100-b70c4788 934202070@default:2 Up Dial(SIP/934202070@voip_provider1|40|
SIP/voip_provider1-08b35490 (None) Up Bridged Call(SIP/cc111-afe8d96
SIP/cc111-afe8d968 952927167@default:2 Up Dial(SIP/952927167@voip_provider1|40|
SIP/voip_provider1-08ed7040 (None) Up Bridged Call(SIP/cc123-08e928e
SIP/cc123-08e928e8 666818430@default:2 Up Dial(SIP/666818430@voip_provider1|40|
Zap/pseudo-174244078 s@default:1 Rsrvd (None)
SIP/cc113-0928fc48 8600059@default:1 Up MeetMe(8600059|F)
Zap/pseudo-173764505 s@default:1 Rsrvd (None)
SIP/cc115-09809798 8600054@default:1 Up MeetMe(8600054|F)
Zap/pseudo-483933697 s@default:1 Rsrvd (None)
SIP/cc110-08e15f48 8600060@default:1 Up MeetMe(8600060|F)
SIP/voip_provider1-092dc438 (None) Up Bridged Call(SIP/voip_provider1-09167
SIP/voip_provider1-09167cd0 933981126@default:2 Up Dial(SIP/933981126@voip_provider1|40|
SIP/cc119-08eaa780 8600056@default:1 Up MeetMe(8600056|F)
Zap/pseudo-429688367 s@default:1 Rsrvd (None)
717 active channels
359 active calls
Verbosity is at least 21

I can't make 359 simultaneouse calls with 25 agents !!!!

by **newx22** » Wed Apr 15, 2009 1:59 pm

now I am at 625 simultaneouse calls...

1248 active channels
625 active calls
Verbosity is at least 21
-- Nobody picked up in 40000 ms
-- Remote UNIX connection

by **newx22** » Wed Apr 15, 2009 2:05 pm

2501.astshell (04/15/2009 03:36:41 PM) (Detached)
what is this screen related to ?
when I open it i get :

[remote detached]
sh-3.2#

by **mflorell** » Wed Apr 15, 2009 2:15 pm

That screen is normal, it is used to launch the asterisk-contained screen and detach from it so everything works properly.

by **newx22** » Wed Apr 15, 2009 2:25 pm

so is there any tests I can do to find out whats wrong with my dialing ?

by **mflorell** » Wed Apr 15, 2009 2:57 pm

Well, I am guessing that since you did not install the G729 library properly that it is not working.

We do not support the use of the free G729 library because it is illegal to use for non-academic/testing purposes without paying for it. I would recommend buying the Digium G729 licenses so you can get support from them.

by **newx22** » Thu Apr 16, 2009 1:20 am

I don't really need g729, it was suggested to me by a voip provider from belgium, who told me that I don't need to have it installed on my server if eyebeam phone paid version has it because asterisk would not need to transcoding it & only bridge it over to voip provider.

I followed your advice & made some test today & the links work with alaw.
so now I left only alaw in my allowed codecs in sip.conf
So do you think this will remove the problem I am having with the predictive dialing getting crazy & making 600 simultaneouse calls ? because in manual calling I don't have any problem.

by **mflorell** » Thu Apr 16, 2009 6:25 am

Well, try it out and let us know if it works. If you have problems please post the settings for auto dialing(the stuff in the green section of the Campaign Detail screen).

by **newx22** » Thu Apr 16, 2009 7:14 am

here is the green zone for one campain :

Dial Method: ADAPT_HARD_LIMIT
Auto Dial Level: 1.087 (0 = off) ADAPT OVERRIDE unchecked
Available Only Tally: N
Drop Percentage Limit: 40%
Maximum Adapt Dial Level: 3.5 number only
Latest Server Time: 2200 4 digits only
Adapt Intensity Modifier: 2 - More Intense
Dial Level Difference Target: 0 -- 0 Balanced
Concurrent Transfers: Auto
Queue Priority: 50-Higher

by **mflorell** » Thu Apr 16, 2009 11:00 am

Please post some Asterisk CLI output from when you are having these problems.

by **newx22** » Sat Apr 18, 2009 2:45 am

first size of log files :
d1:/var/log/astguiclient# du -hs *
2.7M action_full.2009-04-15
3.5M action_full.2009-04-16
2.9M action_full.2009-04-17
4.7M action_launch.2009-04-15
5.2M action_launch.2009-04-16
4.2M action_launch.2009-04-17
4.7M action_process.2009-04-15
5.4M action_process.2009-04-16
4.4M action_process.2009-04-17
1.7M action_send.2009-04-15
1.9M action_send.2009-04-16
1.5M action_send.2009-04-17
28M adapt.2009-04-14
32M adapt.2009-04-15
24M adapt.2009-04-16
22M adapt.2009-04-17
29M agiout.2009-04-15
6.0M agiout.2009-04-16
4.0K archive
59M core.2519
1.9G FASTagiout.2009-04-15
55M FASTagiout.2009-04-16
52M FASTagiout.2009-04-17
320K hopper.2009-04-15
292K hopper.2009-04-16
236K hopper.2009-04-17
84K hopper.2009-04-18
522M listen.2009-04-15
25M listen.2009-04-16
22M listen.2009-04-17
12K listen.2009-04-18
4.0K listen_process.2009-04-15

Day 2009-04-15 was a predictive day, while 16 a 17 were progressive.

I posted some output on my first post.

vicidial.org

error in predictive dialing ?

error in predictive dialing ?

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