Calls problem (in adapt/ratio mode)

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Calls problem (in adapt/ratio mode)

Postby mumin37 » Fri Feb 28, 2014 7:33 am

Hi,
First of all I'd like to tell that is not my first installation of Vicidial, I was doing it couple times, but it is first time I've got such problem.
I've got weird problem, because I can't perform calls from Campaign in autodial mode.
I've got my provider's settings in Carriers, carrier is registered successfully, I can dial manually (from agc), I can dial manually from softphone, but when I try to call in ratio or adapt mode there is problem - phone is ringing, when I answering, I've got sipsilence and hangup. I don't know what's going on if manual dial works perfect.

I was trying to make phone call via 2 different providers, effect is the same.

910 is my prefix,
48 is phone code which is ommited in dialplan:
Code: Select all
exten => _91048XXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91048XXXXXX.,2,Dial(${extravoip}/${EXTEN:5},60,tTor)
exten => _91048XXXXXX.,3,Hangup



Code: Select all
[2014-02-28 13:14:20]     -- Executing [91048608418345@default:1] AGI("Local/91048608418345@default-38c8,2", "agi://127.0.0.1:4577                                   /call_log") in new stack
[2014-02-28 13:14:20]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[2014-02-28 13:14:20]     -- Executing [91048608418345@default:2] Dial("Local/91048608418345@default-38c8,2", "SIP/extravoip/60841                                   8345|60|tTor") in new stack
[2014-02-28 13:14:20]     -- Called extravoip/608418345
[2014-02-28 13:14:20]     -- SIP/extravoip-0000001f is ringing
[2014-02-28 13:14:23]     -- SIP/extravoip-0000001f is making progress passing it to Local/91048608418345@default-38c8,2
[2014-02-28 13:14:23]     -- SIP/extravoip-0000001f is ringing
[2014-02-28 13:14:23]     -- SIP/extravoip-0000001f is making progress passing it to Local/91048608418345@default-38c8,2
[2014-02-28 13:14:28]     -- Got SIP response 405 "Method Not Allowed" back from 192.168.1.62
[2014-02-28 13:14:30]     -- SIP/extravoip-0000001f answered Local/91048608418345@default-38c8,2
[2014-02-28 13:14:30]        > Channel Local/91048608418345@default-38c8,1 was answered.
[2014-02-28 13:14:30]     -- Executing [8368@default:1] Playback("Local/91048608418345@default-38c8,1", "sip-silence") in new stac                                   k
[2014-02-28 13:14:30]     -- <Local/91048608418345@default-38c8,1> Playing 'sip-silence' (language 'en')
[2014-02-28 13:14:30] WARNING[10023]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[2014-02-28 13:14:30]     -- Executing [h@default:1] DeadAGI("Local/91048608418345@default-38c8,2", "agi://127.0.0.1:4577/call_log                                   --HVcauses--PRI-----NODEBUG-----16-----ANSWER-----10-----0") in new stack
[2014-02-28 13:14:30]     -- Executing [8368@default:2] AGI("SIP/extravoip-0000001f", "agi://127.0.0.1:4577/call_log") in new stac                                   k
[2014-02-28 13:14:30]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0


My Configuration:
Vicibox 3.1.15 on Dell 1950 2x Core2Quad, 8GB ram
dialing via eth0 - 192.168.0.252
internal network - eth1 - 192.168.1.252
I was trying to perform ADMIN_update_server_ip.py in both ways.
In every configuration I've got the same problem also from internal network and from external

My sip show peer if needed:
Code: Select all
asterisk -rx "sip show peer 113"


  * Name       : 113
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : default
  Subscr.Cont. : <Not set>
  Language     : en
  Accountcode  : 113
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Mailbox      : 1013
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "TELEMARKETING Olga Szulc" <>
  MaxCallBR    : 384 kbps
  Expire       : 36
  Insecure     : no
  Nat          : Always
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : Yes
  Subscriptions: Yes
  Overlap dial : No
  Forward Loop : Yes
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       :
  Addr->IP     : 89.72.181.195 Port 59283
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 113
  SIP Options  : (none)
  Codecs       : 0x6 (gsm|ulaw)
  Codec Order  : (ulaw:20,gsm:20)
  Auto-Framing:  No
  Status       : OK (188 ms)
  Useragent    : 3CXPhone 6.0.26523.0
  Reg. Contact : sip:113@89.72.181.195:59283;transport=UDP;rinstance=ca0088738a8f30ba



Vicidial VERSION: 2.4-357a BUILD: 120125-2107

Do you know what can I do with this problem?
mumin37
 
Posts: 31
Joined: Wed Jan 04, 2012 3:38 am

Re: Calls problem (in adapt/ratio mode)

Postby williamconley » Fri Feb 28, 2014 5:32 pm

Are you using an old installation .iso cd you had laying around? The latest vicibox cd is 5.0.3 and has one bug left that causes the wrong asterisk version in admin->servers (compared to "asterisk -V").

Otherwise, the primary difference between autodial and manual dial is that one may use a different callerid.

Would be nice if you showed a WORKING example and BROKEN example side by side (ie: manual dial vs autodial) or at least at the same time so we can compare.

Of course, your sip.conf should have an externip entry because you're on a private network (both ip addresses are private).
Vicidial Installation and Repair, plus Hosting and Colocation
Newest Product: Vicidial Agent Only Beep - Beta
http://www.PoundTeam.com # 352-269-0000 # +44(203) 769-2294
williamconley
 
Posts: 20258
Joined: Wed Oct 31, 2007 4:17 pm
Location: Davenport, FL (By Disney!)

Re: Calls problem (in adapt/ratio mode)

Postby mumin37 » Sun Mar 02, 2014 5:13 pm

Hi, I'm using vicibox 3.1.15 because there is also one old server, which works perfect, but in different gateway. I used 3.1.15 also because i wrote some php, which is selecting some info from database and I didn't have time to think if newer version of Vicidial has some changes in database. There are examples:

Autodial (not working correctly)
Code: Select all
[2014-03-02 22:59:47]   == Parsing '/etc/asterisk/manager.conf': [2014-03-02 22:59:47] Found
[2014-03-02 22:59:47]   == Manager 'sendcron' logged on from 127.0.0.1
[2014-03-02 22:59:47]     -- Executing [92448608418345@default:1] AGI("Local/92448608418345@default-ea17,2", "agi://127.0.0.1:4577/call_log") in new stack
[2014-03-02 22:59:47]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[2014-03-02 22:59:47]     -- Executing [92448608418345@default:2] Dial("Local/92448608418345@default-ea17,2", "SIP/supervoip4/608418345|60|tTor") in new stack
[2014-03-02 22:59:47]     -- Called supervoip4/608418345
[2014-03-02 22:59:48]     -- SIP/supervoip4-00000001 is making progress passing it to Local/92448608418345@default-ea17,2
[2014-03-02 22:59:56]     -- SIP/supervoip4-00000001 answered Local/92448608418345@default-ea17,2
[2014-03-02 22:59:56]        > Channel Local/92448608418345@default-ea17,1 was answered.
[2014-03-02 22:59:56]     -- Executing [8368@default:1] Playback("Local/92448608418345@default-ea17,1", "sip-silence") in new stack
[2014-03-02 22:59:56]     -- <Local/92448608418345@default-ea17,1> Playing 'sip-silence' (language 'en')
[2014-03-02 22:59:56] WARNING[3521]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[2014-03-02 22:59:56]     -- Executing [h@default:1] DeadAGI("Local/92448608418345@default-ea17,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----9-----0") in new stack
[2014-03-02 22:59:56]     -- Executing [8368@default:2] AGI("SIP/supervoip4-00000001", "agi://127.0.0.1:4577/call_log") in new stack
[2014-03-02 22:59:56]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[2014-03-02 22:59:56]     -- Executing [8368@default:3] AGI("SIP/supervoip4-00000001", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[2014-03-02 22:59:56]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[2014-03-02 22:59:56]     -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[2014-03-02 22:59:56]     -- Executing [8368@default:4] AGI("SIP/supervoip4-00000001", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[2014-03-02 22:59:56]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[2014-03-02 22:59:57]     -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[2014-03-02 22:59:57]     -- Executing [8368@default:5] Hangup("SIP/supervoip4-00000001", "") in new stack
[2014-03-02 22:59:57]   == Spawn extension (default, 8368, 5) exited non-zero on 'SIP/supervoip4-00000001'
[2014-03-02 22:59:57]     -- Executing [h@default:1] DeadAGI("SIP/supervoip4-00000001", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[2014-03-02 22:59:57]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[2014-03-02 22:59:57]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----9-----0 completed, returning 0
[2014-03-02 22:59:57]   == Spawn extension (default, 92448608418345, 2) exited non-zero on 'Local/92448608418345@default-ea17,2'
[2014-03-02 22:59:57]   == Parsing '/etc/asterisk/manager.conf': [2014-03-02 22:59:57] Found
[2014-03-02 22:59:57]   == Manager 'sendcron' logged on from 127.0.0.1
[2014-03-02 22:59:57]     -- Executing [92448608418345@default:1] AGI("Local/92448608418345@default-9f39,2", "agi://127.0.0.1:4577/call_log") in new stack
[2014-03-02 22:59:57]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[2014-03-02 22:59:57]     -- Executing [92448608418345@default:2] Dial("Local/92448608418345@default-9f39,2", "SIP/supervoip4/608418345|60|tTor") in new stack




And Manual dial (working):
Code: Select all
[2014-03-02 23:01:03]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-7f92,2", "8600051|F") in new stack
[2014-03-02 23:01:03]        > Channel Local/8600051@default-7f92,1 was answered.
[2014-03-02 23:01:03]     -- Executing [91048608418345@default:1] AGI("Local/8600051@default-7f92,1", "agi://127.0.0.1:4577/call_log") in new stack
[2014-03-02 23:01:03]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[2014-03-02 23:01:03]     -- Executing [91048608418345@default:2] Dial("Local/8600051@default-7f92,1", "SIP/extravoip/608418345|60|tTor") in new stack
[2014-03-02 23:01:03]     -- Called extravoip/608418345
[2014-03-02 23:01:03]     -- SIP/extravoip-00000003 is ringing
[2014-03-02 23:01:04]   == Parsing '/etc/asterisk/manager.conf': [2014-03-02 23:01:04] Found
[2014-03-02 23:01:04]   == Manager 'sendcron' logged on from 127.0.0.1
[2014-03-02 23:01:04]     -- Executing [58600051@default:1] MeetMe("Local/58600051@default-7355,2", "8600051|Fmq") in new stack
[2014-03-02 23:01:04]        > Channel Local/58600051@default-7355,1 was answered.
[2014-03-02 23:01:04]     -- Executing [8309@default:1] Answer("Local/58600051@default-7355,1", "") in new stack
[2014-03-02 23:01:04]     -- Executing [8309@default:2] Monitor("Local/58600051@default-7355,1", "wav|20140302-230102_608418345") in new stack
[2014-03-02 23:01:04]     -- Executing [8309@default:3] Wait("Local/58600051@default-7355,1", "3600") in new stack
[2014-03-02 23:01:05]   == Manager 'sendcron' logged off from 127.0.0.1
[2014-03-02 23:01:06]   == Manager 'sendcron' logged off from 127.0.0.1
[2014-03-02 23:01:06]   == Parsing '/etc/asterisk/manager.conf': [2014-03-02 23:01:06] Found
[2014-03-02 23:01:06]   == Manager 'sendcron' logged on from 127.0.0.1
[2014-03-02 23:01:07]   == Manager 'sendcron' logged off from 127.0.0.1
[2014-03-02 23:01:07]     -- SIP/extravoip-00000003 is making progress passing it to Local/8600051@default-7f92,1
[2014-03-02 23:01:07]     -- SIP/extravoip-00000003 is ringing
[2014-03-02 23:01:07]     -- SIP/extravoip-00000003 is making progress passing it to Local/8600051@default-7f92,1
[2014-03-02 23:01:11]     -- SIP/extravoip-00000003 answered Local/8600051@default-7f92,1
[2014-03-02 23:01:37]     -- Executing [h@default:1] DeadAGI("Local/8600051@default-7f92,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----34-----26") in new stack
[2014-03-02 23:01:37] WARNING[3691]: res_agi.c:2212 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI
[2014-03-02 23:01:37]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----34-----26 completed, returning 0
[2014-03-02 23:01:37]   == Spawn extension (default, 91048608418345, 2) exited non-zero on 'Local/8600051@default-7f92,1'
[2014-03-02 23:01:37]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-7f92,2'
[2014-03-02 23:01:37]     -- Executing [h@default:1] DeadAGI("Local/8600051@default-7f92,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[2014-03-02 23:01:37]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0



I tried to set externip, but it didn't work. There is also hard to use extern ip, because my provider gives only dynamic IP, which is very often changed :/
I tried to set it in DMZ, but it also didn't work.
Also I've copied for testing all asterisk conf files from second server. It also didn't give good effect.
I think I've tried everything.

Thanks in advance
mumin37
 
Posts: 31
Joined: Wed Jan 04, 2012 3:38 am


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