vicidial.org

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

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

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.

michaelux wrote: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

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

Hi,

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

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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?

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

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

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

Any idea? thanks[/code]