vicidial.org

Tried to setup inbound using the Manual but error encountered.

[Jul 30 23:48:08] ERROR[30924]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Jul 30 23:48:08] -- AGI Script agi-DID_route.agi completed, returning 0
[Jul 30 23:48:08] -- Executing [9998811112@default:1] Wait("SIP/VOIP-00000bdc", "2") in new stack
[Jul 30 23:48:10] -- Executing [9998811112@default:2] Answer("SIP/VOIP-00000bdc", "") in new stack
[Jul 30 23:48:10] -- Executing [9998811112@default:3] Playback("SIP/VOIP-00000bdc", "ss-noservice") in new stack
[Jul 30 23:48:10] WARNING[30924]: channel.c:3409 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[Jul 30 23:48:10] WARNING[30924]: file.c:995 ast_streamfile: Unable to open ss-noservice (format 0x100 (g729)): No such file or directory
[Jul 30 23:48:10] WARNING[30924]: app_playback.c:440 playback_exec: ast_streamfile failed on SIP/VOIP-00000bdc for ss-noservice
[Jul 30 23:48:10] -- Executing [9998811112@default:4] Playback("SIP/VOIP-00000bdc", "vm-goodbye") in new stack
[Jul 30 23:48:10] WARNING[30924]: channel.c:3409 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[Jul 30 23:48:10] WARNING[30924]: file.c:995 ast_streamfile: Unable to open vm-goodbye (format 0x100 (g729)): No such file or directory
[Jul 30 23:48:10] WARNING[30924]: app_playback.c:440 playback_exec: ast_streamfile failed on SIP/VOIP-00000bdc for vm-goodbye
[Jul 30 23:48:10] -- Executing [9998811112@default:5] Hangup("SIP/VOIP-00000bdc", "") in new stack
[Jul 30 23:48:10] == Spawn extension (default, 9998811112, 5) exited non-zero on 'SIP/VOIP-00000bdc'
[Jul 30 23:48:10] -- Executing [h@default:1] DeadAGI("SIP/VOIP-00000bdc", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack



Goautodial
VERSION: 2.4-309a
BUILD: 110430-1642
© 2011 ViciDial Group
Asterisk 1.4.39.1-vici

new error:NOTICE[9975]: chan_sip.c:15566 handle_request_invite: Call from 'VOIP' to extension '3XXXXXXXXX' rejected because extension not found.

[VOIP]
type=friend
host=X.X.X.X
allow=all
qualify=yes
nat=no
dtmfmode=rfc2833
canreinvite=no
insecure=very
context=trunkibound

exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(${SIPIVOIP}/${EXTEN},,tTo)
exten => _X.,3,Hangup
*******************************************************************
[trunkinbound]
; DID call routing process
exten => _X.,1,AGI(agi-DID_route.agi)

1.check your did route setting , as per the log it is showing 9998811112 , so change the DID route to ingroup or exten and select the respective ingroup from the drop down.

2. add this under default context and check the call
exten => _3XXXXXXXXX,1,AGI(agi-DID_route.agi)

3. install g729 codec

**********inbound group
Group ID: inbounddid
Group Name: test inbound
Group Color: BLACK
Active: Y
Next Agent Call: oldest_call_finish
Drop Transfer Group: None
Call Time: 24hours
Default Transfer Group: inbounddid-test inbound
the rest default

***************DID
DID Extension: 3123456789
DID Description: 3123456789
DID Route: IN_GROUP
Record Call: Y
Extension: 9998811112
Extension Context: default
Server IP: x.x.x.x
User Unavailable Action: voicemail
User Route Settings In-Group: inbounddid-testinbound
In-Group ID: inbounddid-testinbound
Filter User Unavailable Action: voicemail
**************extensions
[default]
exten => _3123456789,1,AGI(agi-DID_route.agi)
**************
done install codec g729
---reload asterisk
---make testcall >> error still the same..please help

can u restart your system and make a test call and post the cli output here

Use sip debug to find the context the system is searching in for 3XXXXXXXX. If the context is not "trunkinbound", then the call is authenticating to the wrong account somehow and you should try to find the sip account it is actually authenticating to. ALL sip carriers should have "context=trunkinbound" in them anyway, since that setting is only used for Inbound calls (not used for outbound!).

After you succeed in finding the correct sip account and routing the call to "trunkinbound", the [trunkinbound] context in extensions.conf should catch ALL calls (_X.) and send them to the vicidial scripts and you're on.

However, if that is not something you can accomplish right now ... try taking the "_" off the front of 3123456789. The _ is used to indicate "pattern", and this is not a pattern, it's an extension. Placing that line in the context the system is actually searching should result in the call being passed to the vicidial scripts (a common "workaround" if you can't figure out how to get it to trunkinbound ... it'll work that way just fine while you "figure that out").

Happy hunting.

i tried to remove the '_' << still same error encountered.

CLI "sip debug"

<--- SIP read from Y.Y.Y.Y:5060 --->
INVITE sip:3123456789@X.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bK1sansay1748347844rdb2820
Record-Route: <sip:sansay1748347844rdb2820@Y.Y.Y.Y:5060;lr;transport=udp>
To: <sip:3123456789@X.X.X.X>
From: <sip:+16615512873@Y.Y.Y.Y;isup-oli=00>;tag=sansay1748347844rdb2820
Call-ID: 679820913-0-4237353744@X.X.X.X
CSeq: 1 INVITE
Contact: <sip:+16615512873@Y.Y.Y.Y:5060>
Supported: timer,100rel
Session-Expires: 1800;refresher=uac
Min-SE: 90
Remote-Party-ID: <sip:+16615512873@4.55.13.227:5060>;privacy=off
Max-Forwards: 61
Content-Type: application/sdp
Content-Length: 301

v=0
o=Sansay-VSXi 188 1 IN IP4 Y.Y.Y.Y
s=Session Controller
c=IN IP4 X.X.X.X
t=0 0
m=audio 22734 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20

<------------->
[Aug 2 17:13:05] --- (15 headers 14 lines) ---
[Aug 2 17:13:05] Sending to Y.Y.Y.Y : 5060 (NAT)
[Aug 2 17:13:05] Using INVITE request as basis request - 679820913-0-4237353744@X.X.X.X
[Aug 2 17:13:05] Found peer 'VOIP'
[Aug 2 17:13:05] Found RTP audio format 0
[Aug 2 17:13:05] Found RTP audio format 8
[Aug 2 17:13:05] Found RTP audio format 18
[Aug 2 17:13:05] Found RTP audio format 101
[Aug 2 17:13:05] Found audio description format PCMU for ID 0
[Aug 2 17:13:05] Found audio description format PCMA for ID 8
[Aug 2 17:13:05] Found audio description format G729 for ID 18
[Aug 2 17:13:05] Found audio description format telephone-event for ID 101
[Aug 2 17:13:05] Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
[Aug 2 17:13:05] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Aug 2 17:13:05] Peer audio RTP is at port X.X.X.X:22734
[Aug 2 17:13:05] Looking for 3123456789 in trunkibound (domain X.X.X.X)
[Aug 2 17:13:05]
<--- Reliably Transmitting (no NAT) to Y.Y.Y.Y:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bK1sansay1748347844rdb2820;received=Y.Y.Y.Y
From: <sip:+16615512873@Y.Y.Y.Y;isup-oli=00>;tag=sansay1748347844rdb2820
To: <sip:3123456789@X.X.X.X>;tag=as591f9506
Call-ID: 679820913-0-4237353744@X.X.X.X
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Aug 2 17:13:05] NOTICE[3575]: chan_sip.c:15566 handle_request_invite: Call from 'VOIP' to extension '3123456789' rejected because extension not found.
[Aug 2 17:13:05] Scheduling destruction of SIP dialog '679820913-0-4237353744@X.X.X.X' in 12928 ms (Method: INVITE)
[Aug 2 17:13:05]
<--- SIP read from Y.Y.Y.Y:5060 --->
ACK sip:3123456789@X.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bK1sansay1748347844rdb2820
To: <sip:3123456789@X.X.X.X>;tag=as591f9506
From: <sip:+16615512873@Y.Y.Y.Y;isup-oli=00>;tag=sansay1748347844rdb2820
Call-ID: 679820913-0-4237353744@X.X.X.X
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


<------------->
[Aug 2 17:13:05] --- (8 headers 0 lines) ---
[Aug 2 17:13:05] Really destroying SIP dialog '679820913-0-4237353744@X.X.X.X' Method: ACK
[Aug 2 17:13:07] == Parsing '/etc/asterisk/manager.conf': [Aug 2 17:13:07] Found
[Aug 2 17:13:07] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 2 17:13:07] == Manager 'sendcron' logged off from 127.0.0.1

Looking for 3123456789 in trunkibound (domain X.X.X.X)

Probably ought to be "trunkinbound" if you want it to work. (add the n!)

This is likely a mistake in your carrier settings, account entry, "context=trunkibound" instead of "context=trunkinbound".

Hi,

Now.. It works..Thanks a million..

ANd 1 more question where can i find the location of the inbound recs? I cannot found it in monitorDONE..PLEASE. in admin page i canot click it like the outbound recs..

You will find the Vicidial Manager's Manual a marvelous tool. The recordings are managed in crontab -e (three entries: mix,move,compress; optional fourth entry to send to ftp server). After the mix/move/compress scripts are done, they will be in your monitorDONE/MP3 folder (if MP3 is the compression in use ... of course). The scripts do take a few minutes to function, though, so be patient. Searching the various folders can be enlightening as can reading the manual and options for each of the scripts.

Hi,

I'ts been 3days and still the recordings is in mix folder and when i double check the recs the customer and agent voice is in separate files. In admin page under DID, still i cant click or download the files..

### recording mixing/compressing/ftping scripts
0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl
#0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl --MIX
#0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_VDonly.pl
1,4,7,10,13,16,19,22,25,28,31,34,37,40,43,46,49,52,55,58 * * * * /usr/share/astguiclient/AST_CRON_audio_2_compress.pl --MP3
#2,5,8,11,14,17,20,23,26,29,32,35,38,41,44,47,50,53,56,59 * * * * /usr/share/astguiclient/AST_CRON_audio_3_ftp.pl --GSM

did you try with

Code: Select all

### recording mixing/compressing/ftping scripts
#0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl
0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl --MIX
0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_VDonly.pl

Also, you can run any of these files with " --debugX" and see what they do.

Hi,

I followed ur suggestion above and it works.. tnx..

ANd sirs we downloaded the mp3 recordings of all outbound recs(admin page) all we can hear is hissing sound. no voice, just noise. The only recs we can use is from monitorDONE/ORIG in gsm format.Please help

the orig file should not have gsm files in it. but i don't deal with recordings in goautodial configured systems very often. do any of the recording/mixing crontab entries have a GSM notation (not including the ftp one, which isn't active anyway)

Hi,

Below are the only entry i can found.

### recording mixing/compressing/ftping scripts
#0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl
0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl --MIX
0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_VDonly.pl
1,4,7,10,13,16,19,22,25,28,31,34,37,40,43,46,49,52,55,58 * * * * /usr/share/astguiclient/AST_CRON_audio_2_compress.pl --MP3
#2,5,8,11,14,17,20,23,26,29,32,35,38,41,44,47,50,53,56,59 * * * * /usr/share/astguiclient/AST_CRON_audio_3_ftp.pl --GSM

### keepalive script for astguiclient processes
* * * * * /usr/share/astguiclient/ADMIN_keepalive_ALL.pl

### kill Hangup script for Asterisk updaters
* * * * * /usr/share/astguiclient/AST_manager_kill_hung_congested.pl

### updater for voicemail
* * * * * /usr/share/astguiclient/AST_vm_update.pl

### updater for conference validator
* * * * * /usr/share/astguiclient/AST_conf_update.pl

### flush queue DB table every hour for entries older than 1 hour
11 * * * * /usr/share/astguiclient/AST_flush_DBqueue.pl -q

### fix the vicidial_agent_log once every hour and the full day run at night
33 * * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl
50 0 * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl --last-24hours
## uncomment below if using QueueMetrics
#*/5 * * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl --only-qm-live-call-check

## uncomment below if using Vtiger
1 1 * * * /usr/share/astguiclient/Vtiger_optimize_all_tables.pl --quiet

### updater for VICIDIAL hopper
* * * * * /usr/share/astguiclient/AST_VDhopper.pl -q

### adjust the GMT offset for the leads in the vicidial_list table
1 1 * * * /usr/share/astguiclient/ADMIN_adjust_GMTnow_on_leads.pl --debug

### reset several temporary-info tables in the database
2 1 * * * /usr/share/astguiclient/AST_reset_mysql_vars.pl

### optimize the database tables within the asterisk database
3 1 * * * /usr/share/astguiclient/AST_DB_optimize.pl

## adjust time on the server with ntp
30 * * * * /usr/sbin/ntpdate -u pool.ntp.org 2>/dev/null 1>&2

### VICIDIAL agent time log weekly and daily summary report generation
2 0 * * 0 /usr/share/astguiclient/AST_agent_week.pl
22 0 * * * /usr/share/astguiclient/AST_agent_day.pl

### VICIDIAL campaign export scripts (OPTIONAL)
#32 0 * * * /usr/share/astguiclient/AST_VDsales_export.pl
#42 0 * * * /usr/share/astguiclient/AST_sourceID_summary_export.pl

### remove old recordings more than 14 days old
#24 0 * * * /usr/bin/find /var/spool/asterisk/monitorDONE/ORIG/ -maxdepth 2 -type f -mtime +14 -print | xargs rm -f

### roll logs monthly on high-volume dialing systems
#30 1 1 * * /usr/share/astguiclient/ADMIN_archive_log_tables.pl

### remove old vicidial logs and asterisk logs more than 2 days old
28 0 * * * /usr/bin/find /var/log/astguiclient -maxdepth 1 -type f -mtime +2 -print | xargs rm -f
29 0 * * * /usr/bin/find /var/log/asterisk -maxdepth 3 -type f -mtime +2 -print | xargs rm -f
30 0 * * * /usr/bin/find / -maxdepth 1 -name "screenlog.0*" -mtime +4 -print | xargs rm -f

### keepalive script for GoAutoDial processes
* * * * * /usr/share/goautodial/keepalive_goautodial.pl

### logs cleanup for GoAutoDial
8 1 * * * /usr/share/goautodial/go_clean.pl

### asterisk logs access for GoAutoDial
* * * * * /usr/share/goautodial/go_astlogs.pl

### asterisk db daily backup
30 4 * * * /usr/bin/mysqldump -u root -pvicidialnow --databases asterisk > /usr/share/goautodial/godbbackup/

Campaign Rec extension - This field allows for a custom recording extension to be used with VICIDIAL. This allows you to use different extensions depending upon how long you want to allow a maximum recording and what type of codec you want to record in. The default exten is 8309 which if you follow the SCRATCH_INSTALL examples will record in the WAV format for upto one hour. Another option included in the examples is 8310 which will record in GSM format for upto one hour.

Hi,

I activated gsm format cause of the file size compare to wav.. Does it also affect the mp3 in admin page?

and sir another thing..

HOw can i setup 2 or more sip to receive this inbound call with out using inbound group? In admin page u can set 1 phone/sip to route the DID.

Just like with freepbx ring all..

currently,,this campaign use manual dial method.. and the inbound was set to 1 agent where the softphone rings everytime thers an inbound.. Now they want addtional agent receiving the inbound but in admin page we can route the DID to a single phone number only.. Please if there any way where they can do manual dial and the same time there are agents will receive inbound calls..

you could try with & between two sip phones, but you would do better to create an ingroup with on-hook remote agents and ringall. more control, plus viewing in Real Time. And the ability to easily turn each remote agent on/off and even add new ones.

Hi,

I tried the & between two sip(in adminpage"DID page").. Phone Extension:123&456.. and i submit and the result is Phone Extension:123456.. Pls advice

sip1: 123
sip2: 456
page: Modify DID

got it sir.. need to add in extensions... tnx

Now my prob is the recs.. hope any can help me

ANy one please need help with my recss..

I tried switch back to wav files and the recs in admin page work well.. Look like the gsm to mp3 prob here..