ViciNow seem to make calls but doesnt show anything, no live

[michaelux]

Hi all..

I just installed VicidialNow. We have a vicidial server version 2.0.4 and asterisk and asterisk 1.2.24 running in debian..

I had to move the server to a Centos Server but I got some problems , so I decided install vicidialnow.

First, I ran the upgrade script for the database, vicidial 2.0.5 use 78 tables and 2.04 use 58 tables... After that I did a mysqldump, put it in the vicidialnow and I can login in the agent and manager.

I configured asterisk to connect to the other asterisk server, the connection is OK, i can see i with iax2 show registry.

I can log in with the softphone OK.

When I login to the vicidial client my softphone ring. But I dont see any live call.

When I check the asterisk CLI it seem to do the calls, and the same in the other asterisk server, the logs show: (this is the other asterisk with iax)

Code: Select all

- Accepting AUTHENTICATED call from 192.168.0.139:
       > requested format = slin,
       > requested prefs = (gsm|ulaw),
       > actual format = ulaw,
       > host prefs = (ulaw|alaw|gsm|ilbc),
       > priority = mine
    -- Executing Dial("IAX2/vicidial-8989", "SIP/13054561901@net2phone") in new stack
    -- Called 13054561901@net2phone
    -- Accepting AUTHENTICATED call from 192.168.0.139:
       > requested format = slin,
       > requested prefs = (gsm|ulaw),
       > actual format = ulaw,
       > host prefs = (ulaw|alaw|gsm|ilbc),
       > priority = mine
    -- Executing Dial("IAX2/vicidial-15554", "SIP/13054129831@net2phone") in new stack
    -- Called 13054129831@net2phone
    -- Accepting AUTHENTICATED call from 192.168.0.139:
       > requested format = slin,
       > requested prefs = (gsm|ulaw),
       > actual format = ulaw,
       > host prefs = (ulaw|alaw|gsm|ilbc),
       > priority = mine
    -- Executing Dial("IAX2/vicidial-15774", "SIP/13052354747@net2phone") in new stack
    -- Called 13052354747@net2phone
    -- Got SIP response 408 "Request Timeout" back from 66.33.157.119
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Hangup("IAX2/vicidial-8989", "") in new stack
  == Spawn extension (vicidial, 013054561901, 2) exited non-zero on 'IAX2/vicidial-8989'
    -- Hungup 'IAX2/vicidial-8989'
    -- SIP/net2phone-092923f8 is ringing
    -- SIP/net2phone-092797f0 is ringing
    -- SIP/net2phone-092923f8 answered IAX2/vicidial-15774
  == Spawn extension (vicidial, 013052354747, 1) exited non-zero on 'IAX2/vicidial-15774'
    -- Hungup 'IAX2/vicidial-15774'
    -- SIP/net2phone-092797f0 answered IAX2/vicidial-15554
  == Spawn extension (vicidial, 013054129831, 1) exited non-zero on 'IAX2/vicidial-15554'
    -- Hungup 'IAX2/vicidial-15554

'




This is the asterisk cli of vicidialnow:

Code: Select all

 Executing AGI("Local/013052354747@default-771c,1", "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 AGI("Local/013052354747@default-771c,1", "agi-VDADtransfer.agi|8365") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
    -- AGI Script agi-VDADtransfer.agi completed, returning 0
  == Spawn extension (default, 013052354747, 3) exited non-zero on 'Local/013052354747@default-771c,2'
    -- Executing AGI("IAX2/172.22.0.5:4569-6393", "agi-VDADtransfer.agi|8365") in new stack
    -- Executing DeadAGI("Local/013052354747@default-771c,2", "agi://127.0.0.1:4577/call_log") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing DeadAGI("Local/013052354747@default-771c,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----8-----0)") in new stack
    -- AGI Script agi-VDADtransfer.agi completed, returning 0
    -- Executing AGI("IAX2/172.22.0.5:4569-6393", "agi-VDADtransfer.agi|8365") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
    -- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----8-----0) completed, returning 0
    -- AGI Script agi-VDADtransfer.agi completed, returning 0
    -- Executing Hangup("IAX2/172.22.0.5:4569-6393", "") in new stack
  == Spawn extension (default, 8365, 5) exited non-zero on 'IAX2/172.22.0.5:4569-6393'
    -- Executing DeadAGI("IAX2/172.22.0.5:4569-6393", "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 DeadAGI("IAX2/172.22.0.5:4569-6393", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16---------------)") in new stack
    -- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16---------------) completed, returning 0
    -- Hungup 'IAX2/172.22.0.5:4569-6393'
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
    -- IAX2/172.22.0.5:4569-10447 stopped sounds
    -- IAX2/172.22.0.5:4569-10447 answered Local/013054129831@default-bf67,2
       > Channel Local/013054129831@default-bf67,1 was answered.
    -- Executing AGI("Local/013054129831@default-bf67,1", "agi://127.0.0.1:4577/call_log") in new stack
  == Manager 'sendcron' logged off from 127.0.0.1
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing AGI("Local/013054129831@default-bf67,1", "agi-VDADtransfer.agi|8365") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
  == Spawn extension (default, 013054129831, 3) exited non-zero on 'Local/013054129831@default-bf67,2'
    -- Executing DeadAGI("Local/013054129831@default-bf67,2", "agi://127.0.0.1:4577/call_log") in new stack
    -- AGI Script agi-VDADtransfer.agi completed, returning 0
    -- Executing AGI("IAX2/172.22.0.5:4569-10447", "agi-VDADtransfer.agi|8365") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
    -- AGI Script agi-VDADtransfer.agi completed, returning 0
    -- Executing DeadAGI("Local/013054129831@default-bf67,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----29-----0)") in new stack
    -- Executing AGI("IAX2/172.22.0.5:4569-10447", "agi-VDADtransfer.agi|8365") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
    -- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----29-----0) completed, returning 0
    -- AGI Script agi-VDADtransfer.agi completed, returning 0
    -- Executing Hangup("IAX2/172.22.0.5:4569-10447", "") in new stack
  == Spawn extension (default, 8365, 5) exited non-zero on 'IAX2/172.22.0.5:4569-10447'
    -- Executing DeadAGI("IAX2/172.22.0.5:4569-10447", "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 DeadAGI("IAX2/172.22.0.5:4569-10447", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16---------------)") in new stack
    -- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16---------------) completed, returning 0
    -- Hungup 'IAX2/172.22.0.5:4569-10447'
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing MeetMeAdmin("Local/55558600051@default-60d2,2", "8600051|K") in new stack
    -- Hungup 'Zap/pseudo-591276188'
  == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/6674-08429110'
    -- Executing DeadAGI("SIP/6674-08429110", "agi://127.0.0.1:4577/call_log") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
Sep  8 14:03:48 NOTICE[16836]: app_meetme.c:2210 admin_exec: Conference Number not found
    -- Executing Hangup("Local/55558600051@default-60d2,2", "") in new stack
  == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-60d2,2'
    -- Executing DeadAGI("Local/55558600051@default-60d2,2", "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 DeadAGI("Local/55558600051@default-60d2,2", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16---------------)") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing DeadAGI("SIP/6674-08429110", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16---------------)") in new stack
    -- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16---------------) completed, returning 0
    -- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16---------------) completed, returning 0

I move the asterisks configuration files from the old server to vicidial, meetme, sip, extension.

If I configure iax in the asterisk configuration file, should I configure the Carriers in the manager?


I also get a warning about codec:

Code: Select all

Sep  8 14:02:37 WARNING[2155]: channel.c:2403 set_format: Unable to find a codec translation path from ulaw to ilbc
Sep  8 14:02:37 WARNING[2155]: channel.c:2403 set_format: Unable to find a codec translation path from ulaw to ilbc
       > Channel SIP/6674-08429110 was answered.
    -- Executing MeetMe("SIP/6674-08429110", "8600051") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '8600051'
    -- Playing 'conf-onlyperson' (language 'en')
Sep  8 14:02:37 NOTICE[16536]: rtp.c:587 ast_rtp_read: Unknown RTP codec 126 received
Sep  8 14:02:37 NOTICE[16536]: rtp.c:587 ast_rtp_read: Unknown RTP codec 126 received
Sep  8 14:02:37 NOTICE[16536]: rtp.c:587 ast_rtp_read: Unknown RTP codec 126 received
  == Manager 'sendcron' logged off from 127.0.0.1
Sep  8 14:02:40 WARNING[16536]: channel.c:2403 set_format: Unable to find a codec translation path from ulaw to ilbc
Sep  8 14:02:40 WARNING[16536]: file.c:195 ast_stopstream: Unable to restore format back to 1024


What else should I configure and check?

Thanks

[suneyo21]

But the calls push thru? or running behind the background the problem is that there is no "LIVE CALL" popping out the screen??

[gardo]

There's your problem. You're other Asterisk box is using ilbc as codec which your VicidialNOW box doesn't have. Try setting the codecs in both your Asterisk boxes to ulaw.

I also get a warning about codec:

Code: Select all

Sep  8 14:02:37 WARNING[2155]: channel.c:2403 set_format: Unable to find a codec translation path from ulaw to ilbc
Sep  8 14:02:37 WARNING[2155]: channel.c:2403 set_format: Unable to find a codec translation path from ulaw to ilbc
       > Channel SIP/6674-08429110 was answered.
    -- Executing MeetMe("SIP/6674-08429110", "8600051") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '8600051'
    -- Playing 'conf-onlyperson' (language 'en')
Sep  8 14:02:37 NOTICE[16536]: rtp.c:587 ast_rtp_read: Unknown RTP codec 126 received
Sep  8 14:02:37 NOTICE[16536]: rtp.c:587 ast_rtp_read: Unknown RTP codec 126 received
Sep  8 14:02:37 NOTICE[16536]: rtp.c:587 ast_rtp_read: Unknown RTP codec 126 received
  == Manager 'sendcron' logged off from 127.0.0.1
Sep  8 14:02:40 WARNING[16536]: channel.c:2403 set_format: Unable to find a codec translation path from ulaw to ilbc
Sep  8 14:02:40 WARNING[16536]: file.c:195 ast_stopstream: Unable to restore format back to 1024


What else should I configure and check?

Thanks

[michaelux]

Ohh thank you, I saw that but as it is a warning, but why vicidialnow doesnt use ilbc? if vicidialnow doesnt has ilbc why doesnt try to use ulaw as second choice? :S

Can I install ilbc in vicidialnow? . Change the option in the other server could be dangerous because there are running another applications...

thanks

[michaelux]

Hi,

I checked the other server, the iax.conf and the user we have to do the iax tunnel has this:

Code: Select all

username=vicidial
secret=xxxx
auth=plaintext
type=friend
host=dynamic
context=vicidial
peercontext=default
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc

The first option is ULAW and the last is ilbc :s so the asterisk should try to use ulaw firts, second alaw , then gsm and the last ilbc, isnt it?

[gardo]

Check your iax.conf entries in your VicidialNOW server. The order of codecs might not be the same.

[michaelux]

Hi, here is the iax.conf in the vicidialnow...

But, this same configuration file was in the old server and it connected to the other asterisk server and work fine :s

ulaw is allow .. so what can be wrong? ..

Code: Select all

[general]
bindport=4569
iaxcompat=yes
bandwidth=high
allow=all
allow=gsm                      ; Always allow GSM, it's cool :)
allow=ulaw
jitterbuffer=no
tos=lowdelay
context=cong

register  => vicidial:aaaaaa@172.22.X.X

[newbot] ;decia newbot
username=vicidial
secret=aaaaaa
auth=plaintext
type=friend
host=dynamic
context=default
peercontext=vicidial
disallow=all
allow=ulaw
allow=alaw

[michaelux]

I still have the problem, I can not listen anything in the softphone :s

Any idea? thanks[/code]