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Inbound SIP URI / FastAGI failed.

Added by Mo Zoubi about 5 years ago

Hello,

I'm trying to switch from GAD3 to GAD4 for Inbound operations. I have successfully installed and configured GAD4 from scratch.
I am able to make outgoing calls using the web-dialer, but I'm having issues routing the incoming calls.

With GAD3, I created the inbound campaign and ingroup; I choose DID 1000 for the ingroup and I am able to receive calls via the sip uri ":5060" (Great, right!)
However, I can't seem to replicate this with GAD4 for some reason.

I created the campaign, ingroup and DID and tried to call the URI ":5060" without any success. Then, I realized GAD4 actually registers to port 5070, so I tried to call :5070 and it actually routes to my GAD4 server BUT not to the agent queue. Here' what asterisk says:

 ----------)': Connection refused
[Mar  8 18:09:44] WARNING[8428][C-00000238]: res_agi.c:2110 launch_netscript: Couldn't connect to a                          ny host.  FastAGI failed.
[Mar  8 18:09:44] WARNING[8427][C-00000239]: func_hangupcause.c:140 hangupcause_read: Unable to fin                          d information for channel
[Mar  8 18:09:44] WARNING[8427][C-00000239]: res_agi.c:2038 handle_connection: Connecting to '127.0                          .0.1:4577' failed for url 'agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0----------                          ----------)': Connection refused
[Mar  8 18:09:44] WARNING[8427][C-00000239]: res_agi.c:2110 launch_netscript: Couldn't connect to a                          ny host.  FastAGI failed.
[Mar  8 18:22:57] WARNING[10182][C-0000023a]: func_hangupcause.c:140 hangupcause_read: Unable to find information            for channel
[Mar  8 18:22:57] WARNING[10182][C-0000023a]: res_agi.c:2038 handle_connection: Connecting to '127.0.0.1:4577' fai           led for url 'agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)': Connection ref           used
[Mar  8 18:22:57] WARNING[10182][C-0000023a]: res_agi.c:2110 launch_netscript: Couldn't connect to any host.  Fast           AGI failed.
[Mar  8 18:27:51] WARNING[10702][C-0000023b]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Mar  8 18:27:51] WARNING[10702][C-0000023b]: res_agi.c:2038 handle_connection: Connecting to '127.0.0.1:4577' failed for url 'agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)': Connection refused
[Mar  8 18:27:51] WARNING[10702][C-0000023b]: res_agi.c:2110 launch_netscript: Couldn't connect to any host.  FastAGI failed.
x32*CLI> exit

Can anyone help me figure out why the incoming calls are hitting the server but not being routed to the INGROUP. Why does FastAGI fail?


Replies (23)

RE: Inbound SIP URI / FastAGI failed. - Added by Jorge Cornejo about 5 years ago

Don't double post..I think will be harder to find a solution. You are posting the same error I did just a few hours ago. Try to search first.

RE: Inbound SIP URI / FastAGI failed. - Added by Wittie Manansala about 5 years ago

Hi,

Please post the output below:

1. service kamailio status
2. screen -ls

May we know what is the purpose of and provide us the call flow for us to review.

Thank you

RE: Inbound SIP URI / FastAGI failed. - Added by Mo Zoubi about 5 years ago

I'm having issues posting my reply.

RE: Inbound SIP URI / FastAGI failed. - Added by Mo Zoubi about 5 years ago

Hi,

Thank you for your reply.

[root@x32 ~]# service kamailio status -l
Redirecting to /bin/systemctl status  -l kamailio.service
? kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server
   Loaded: loaded (/usr/lib/systemd/system/kamailio.service; enabled; vendor preset: disabled)
   Active: active (running) since Mon 2019-03-11 21:24:28 UTC; 36s ago
 Main PID: 18396 (kamailio)
   CGroup: /system.slice/kamailio.service
           +-18396 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18397 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18398 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18399 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18400 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18401 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18402 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18403 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18404 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18405 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18406 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18407 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18408 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18409 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18410 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18411 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18412 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18413 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18414 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18415 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18416 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18417 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18418 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18419 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18420 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18421 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4
           +-18422 /usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 64 -M 4

Mar 11 21:24:29 x32.iyap.ca /usr/sbin/kamailio[18418]: INFO: rtpengine [rtpengine.c:2209]: rtpp_test(): rtp proxy <udp:127.0.0.1:5066> found, support for it enabled
Mar 11 21:24:29 x32.iyap.ca /usr/sbin/kamailio[18416]: INFO: rtpengine [rtpengine.c:2209]: rtpp_test(): rtp proxy <udp:127.0.0.1:5066> found, support for it enabled
Mar 11 21:24:29 x32.iyap.ca /usr/sbin/kamailio[18415]: INFO: rtpengine [rtpengine.c:2209]: rtpp_test(): rtp proxy <udp:127.0.0.1:5066> found, support for it enabled
Mar 11 21:24:29 x32.iyap.ca /usr/sbin/kamailio[18414]: INFO: rtpengine [rtpengine.c:2209]: rtpp_test(): rtp proxy <udp:127.0.0.1:5066> found, support for it enabled
Mar 11 21:24:29 x32.iyap.ca /usr/sbin/kamailio[18414]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop:  using epoll_lt io watch method (config)
Mar 11 21:24:29 x32.iyap.ca /usr/sbin/kamailio[18420]: INFO: rtpengine [rtpengine.c:2209]: rtpp_test(): rtp proxy <udp:127.0.0.1:5066> found, support for it enabled
Mar 11 21:24:29 x32.iyap.ca /usr/sbin/kamailio[18412]: INFO: rtpengine [rtpengine.c:2209]: rtpp_test(): rtp proxy <udp:127.0.0.1:5066> found, support for it enabled
Mar 11 21:24:29 x32.iyap.ca /usr/sbin/kamailio[18417]: INFO: rtpengine [rtpengine.c:2209]: rtpp_test(): rtp proxy <udp:127.0.0.1:5066> found, support for it enabled
Mar 11 21:24:29 x32.iyap.ca /usr/sbin/kamailio[18421]: INFO: rtpengine [rtpengine.c:2209]: rtpp_test(): rtp proxy <udp:127.0.0.1:5066> found, support for it enabled
Mar 11 21:24:29 x32.iyap.ca /usr/sbin/kamailio[18419]: INFO: rtpengine [rtpengine.c:2209]: rtpp_test(): rtp proxy <udp:127.0.0.1:5066> found, support for it enabled

And


[root@x32 ~]# screen -ls
There are screens on:
        11075.ASTVDadapt        (Detached)
        12561.asterisk  (Detached)
2 Sockets in /var/run/screen/S-root.
sip::5060 is the SIP URI which I am currently used to receiving calls with on GoAutoDial v3.
It consists of [IN_GROUP DID] @ [SERVER DOMAIN] : [PORT USED BY ASTERISK]

Caller dials 1-800-XXX-XXXX >> Call routed to sip::5060 >> GoAutoDial inbound call accepted. >> Placed in queue for next available agent.

It does not seem to work with GoAutoDial 4

Also, on another note, upon editing /etc/kamailio/kamailio.cfg to replace all instances of "vaglxc01.goautodial.com" with "mydomain.com" / I can no longer hear "You are the only person in this conference" nor can I make any outbound calls anymore. I have followed the information found here https://goautodial.org/boards/3/topics/14605 and updated mysql>kamailio>subscribers & still no luck :(

RE: Inbound SIP URI / FastAGI failed. - Added by Mo Zoubi about 5 years ago

UPDATE:
I got outbound calling working, and fixed my initial issue; but still having a problem with INBOUND CALLS. Can anyone help?

The URI routes to the server >> server to the ingroup >> hold audio files play fine >> as soon as agent is available for call, it's dropped and says DEAD CALL on agent screen.
It can't be a codec issue because outgoing calls work perfectly fine. It's the inbound calls I'm still having an issue with.

    -- Executing [99909*78***DID@default:2] AGI("SIP/voipms-00000005", "agi-VDAD_ALL_inbound.agi") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
    -- <SIP/voipms-00000005> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000005> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- Music class default requested but no musiconhold loaded.
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/58600051@default-00000008;2
[Mar 12 20:53:47] WARNING[4511][C-00000012]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("Local/58600051@default-00000008;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack
  == Manager 'sendcron' logged off from 127.0.0.1
    -- <Local/58600051@default-00000008;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0
  == Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600051@default-00000008;1'
[Mar 12 20:53:47] WARNING[4510][C-00000013]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("Local/58600051@default-00000008;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack
    -- <Local/58600051@default-00000008;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0
    -- <SIP/voipms-00000005> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000005> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000005> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000005> Playing 'generic_hold.gsm' (escape_digits=) (sample_offset 0) (language 'en')
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Called 104*207*151*246*78600051@default
    -- Executing [104*207*151*246*78600051@default:1] Goto("Local/104*207*151*246*78600051@default-00000009;2", "default,78600051,1") in new stack
    -- Goto (default,78600051,1)
    -- Executing [78600051@default:1] Konference("Local/104*207*151*246*78600051@default-00000009;2", "8600051,qR") in new stack
    -- Local/104*207*151*246*78600051@default-00000009;1 answered
    -- Executing [83047777777777@vicidial-auto:1] Answer("Local/104*207*151*246*78600051@default-00000009;1", "") in new stack
    -- Executing [83047777777777@vicidial-auto:2] Playback("Local/104*207*151*246*78600051@default-00000009;1", "ding") in new stack
    -- <Local/104*207*151*246*78600051@default-00000009;1> Playing 'ding.gsm' (language 'en')
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Executing [83047777777777@vicidial-auto:3] Hangup("Local/104*207*151*246*78600051@default-00000009;1", "") in new stack
  == Spawn extension (vicidial-auto, 83047777777777, 3) exited non-zero on 'Local/104*207*151*246*78600051@default-00000009;1'
[Mar 12 20:53:53] WARNING[5527][C-00000016]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@vicidial-auto:1] AGI("Local/104*207*151*246*78600051@default-00000009;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
    -- <Local/104*207*151*246*78600051@default-00000009;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Mar 12 20:53:53] WARNING[5528][C-00000015]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("Local/104*207*151*246*78600051@default-00000009;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
    -- <Local/104*207*151*246*78600051@default-00000009;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
    -- <SIP/voipms-00000005> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000005> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Called 58600051@default
    -- Executing [58600051@default:1] Konference("Local/58600051@default-0000000a;2", "8600051,qLR") in new stack
    -- Local/58600051@default-0000000a;1 answered
    -- Executing [8309@default:1] Answer("Local/58600051@default-0000000a;1", "") in new stack
    -- Executing [8309@default:2] Monitor("Local/58600051@default-0000000a;1", "wav,20190313-045353_6138794444_12904133_agent002") in new stack
    -- Executing [8309@default:3] Wait("Local/58600051@default-0000000a;1", "3600") in new stack
  == Manager 'sendcron' logged off from 127.0.0.1
    -- <SIP/voipms-00000005> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000005> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000005> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000005> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000005>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
    -- Executing [104*207*151*246*8600051@default:1] Goto("SIP/voipms-00000005", "default,8600051,1") in new stack
    -- Goto (default,8600051,1)
    -- Executing [8600051@default:1] Konference("SIP/voipms-00000005", "8600051,R") in new stack
[Mar 12 20:53:55] ERROR[5501][C-00000014]: member.c:404 member_exec: unable to create member
  == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/voipms-00000005'
[Mar 12 20:53:55] WARNING[5501][C-00000014]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("SIP/voipms-00000005", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack
    -- <SIP/voipms-00000005>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1

RE: Inbound SIP URI / FastAGI failed. - Added by Wittie Manansala about 5 years ago

Hi,

Please post the below information for us to review the issue:

1. Complete Carrier Settings
2. nano /etc/asterisk/musiconhold.conf

Thanks

RE: Inbound SIP URI / FastAGI failed. - Added by Mo Zoubi about 5 years ago

Thanks for your reply, as requested here are the Carrier Settings:

Registration String:

register => USERNAME:PASSWORD@newyork8.voip.ms:5060/USERNAME

Account Entry:

[voipms]
disallow=all
allow=gsm
allow=ulaw
type=peer
canreinvite=no
context=trunkinbound
qualify=yes
insecure=invite
nat=yes
host=newyork8.voip.ms
username=*******
fromuser=*******
secret=******
trustrpid=yes
sendrpid=yes

Dial Plan:

exten => _3003149926.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _3003149926.,2,Dial(SIP/${EXTEN:10}@voipms,,tTo)
exten => _3003149926.,3,Hangup

I should note, that I have the same exact settings (different username) working perfectly fine with GOAutoDial v3, but it fails with V4. Im not sure, but this may be a common bug with AppKonference INBOUND calls according to:
https://sourceforge.net/p/appkonference/discussion/924974/thread/fcf9ba72/

Your generous help is kindly appreciated!

RE: Inbound SIP URI / FastAGI failed. - Added by Mo Zoubi about 5 years ago

Sorry, almost forgot:

[root@x32 ~]# cat /etc/asterisk/musiconhold.conf
;
; Music on Hold -- Sample Configuration
;

[old-default]
mode=files
directory=/var/lib/asterisk/mohmp3

[old-quiet]
mode=files
directory=/var/lib/asterisk/quiet-mp3

#include musiconhold-vicidial.conf

; valid mode options:
; quietmp3      -- default
; mp3           -- loud
; mp3nb         -- unbuffered
; quietmp3nb    -- quiet unbuffered
; custom        -- run a custom application
; files         -- read files from a directory in any Asterisk supported format

I don't think it's an issue with hold, because the inbound call is actually SUCCESSFULLY accepted by the server, placed on hold without issue and the caller will hear "Your call is important to us..." but the moment that the agent accepts the call (resumes calls / ends pause to accept call) the call will DROP DEAD. No issues with outgoing calls though..

RE: Inbound SIP URI / FastAGI failed. - Added by Levy Ryan Nolasco about 5 years ago

Edit the entry below

FROM

[old-default]
mode=files
directory=/var/lib/asterisk/mohmp3

TO

[default]
mode=files
directory=/var/lib/asterisk/mohmp3

RE: Inbound SIP URI / FastAGI failed. - Added by Mo Zoubi about 5 years ago

Hello,

As advised, I have updated my musiconhold.conf file.

[root@x32 ~]# cat /etc/asterisk/musiconhold.conf
;
; Music on Hold -- Sample Configuration
;

[default]
mode=files
directory=/var/lib/asterisk/mohmp3

[old-quiet]
mode=files
directory=/var/lib/asterisk/quiet-mp3

#include musiconhold-vicidial.conf

; valid mode options:
; quietmp3      -- default
; mp3           -- loud
; mp3nb         -- unbuffered
; quietmp3nb    -- quiet unbuffered
; custom        -- run a custom application
; files         -- read files from a directory in any Asterisk supported format

Unfortunately, it did not help with my issue receiving incoming calls. Here's the asterisk log:

Connected to Asterisk 13.17.2-vici currently running on x32 (pid = 3873)
  == Using SIP RTP CoS mark 5
       > 0x7fc578017690 -- Strict RTP learning after remote address set to: 23.29.136.40:12310
    -- Executing [10000@trunkinbound:1] AGI("SIP/voipms-00000001", "agi-DID_route.agi") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
    -- <SIP/voipms-00000001>AGI Script agi-DID_route.agi completed, returning 0
    -- Executing [99909*78***DID@default:1] Answer("SIP/voipms-00000001", "") in new stack
       > 0x7fc578017690 -- Strict RTP switching to RTP target address 23.29.136.40:12310 as source
    -- Executing [99909*78***DID@default:2] AGI("SIP/voipms-00000001", "agi-VDAD_ALL_inbound.agi") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
    -- <SIP/voipms-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Called 127*000*000*001*78600051@default
    -- Executing [127*000*000*001*78600051@default:1] Goto("Local/127*000*000*001*78600051@default-00000001;2", "default,78600051,1") in new stack
    -- Goto (default,78600051,1)
    -- Executing [78600051@default:1] Konference("Local/127*000*000*001*78600051@default-00000001;2", "8600051,qR") in new stack
    -- Local/127*000*000*001*78600051@default-00000001;1 answered
    -- Executing [83047777777777@vicidial-auto:1] Answer("Local/127*000*000*001*78600051@default-00000001;1", "") in new stack
    -- Executing [83047777777777@vicidial-auto:2] Playback("Local/127*000*000*001*78600051@default-00000001;1", "ding") in new stack
    -- <Local/127*000*000*001*78600051@default-00000001;1> Playing 'ding.gsm' (language 'en')
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Executing [83047777777777@vicidial-auto:3] Hangup("Local/127*000*000*001*78600051@default-00000001;1", "") in new stack
  == Spawn extension (vicidial-auto, 83047777777777, 3) exited non-zero on 'Local/127*000*000*001*78600051@default-00000001;1'
[Mar 15 22:34:42] WARNING[4321][C-00000004]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@vicidial-auto:1] AGI("Local/127*000*000*001*78600051@default-00000001;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
    -- <Local/127*000*000*001*78600051@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Mar 15 22:34:42] WARNING[4322][C-00000003]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("Local/127*000*000*001*78600051@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
    -- <Local/127*000*000*001*78600051@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
    -- <SIP/voipms-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
       > 0x7fc578017690 -- Strict RTP learning complete - Locking on source address 23.29.136.40:12310
    -- <SIP/voipms-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/voipms-00000001>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
    -- Executing [127*000*000*001*8600051@default:1] Goto("SIP/voipms-00000001", "default,8600051,1") in new stack
    -- Goto (default,8600051,1)
    -- Executing [8600051@default:1] Konference("SIP/voipms-00000001", "8600051,R") in new stack
[Mar 15 22:34:44] ERROR[4314][C-00000002]: member.c:404 member_exec: unable to create member
  == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/voipms-00000001'
[Mar 15 22:34:44] WARNING[4314][C-00000002]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("SIP/voipms-00000001", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack
    -- <SIP/voipms-00000001>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Called 58600051@default
    -- Executing [58600051@default:1] Konference("Local/58600051@default-00000002;2", "8600051,qLR") in new stack
    -- Local/58600051@default-00000002;1 answered
    -- Executing [8309@default:1] Answer("Local/58600051@default-00000002;1", "") in new stack
    -- Executing [8309@default:2] Monitor("Local/58600051@default-00000002;1", "wav,20190316-063442_6138794444_81629854_agent002") in new stack
    -- Executing [8309@default:3] Wait("Local/58600051@default-00000002;1", "3600") in new stack

RE: Inbound SIP URI / FastAGI failed. - Added by Mo Zoubi about 5 years ago

Any suggestions? Has anyone actually been able to successfully receive calls with GAD v4?

RE: Inbound SIP URI / FastAGI failed. - Added by Jorge Cornejo about 5 years ago

I am stuck at similar step, when I do a tcpdump all I can see is SIP incoming call to port 5060 and then it answers a 404. I assume it is Kamailio answering as Asterisk shows nothing
What do you get?

RE: Inbound SIP URI / FastAGI failed. - Added by Mo Zoubi about 5 years ago

Yes, I got the same thing for port 5060, but it works for port 5070 - GoAutoDial will answer and place the call on hold, play music and audio files .. however as soon as agent answers call drops dead.

RE: Inbound SIP URI / FastAGI failed. - Added by Jorge Cornejo about 5 years ago

What do you mean port 5070? You are calling directly to Asterisk?

RE: Inbound SIP URI / FastAGI failed. - Added by Mo Zoubi about 5 years ago

Yes.
I’m using Goautodial for inbound operations. I currently have another server running goautodial v3.3 and it’s running perfectly smooth on port 5060. I tried to mimic the inbound call flow with v4 but it didn’t work. So, I changed the sip URI to port 5070 and so far the calls have been able to hold in que but will drop right before connecting to the agent. Port 5060 won’t work at all, no log in asterisk. My call flow is:

1-800 >> sip::5070 >> goAutodial >> ingroup (hold) >> agent.

Call passes all the way to agent before dropping dead. Has anyone been able to use inbound at all?

RE: Inbound SIP URI / FastAGI failed. - Added by Wittie Manansala about 5 years ago

Hi Mo,

Try this:

Registration String:

register => USERNAME:/YOUR_DID_Number

Thanks

RE: Inbound SIP URI / FastAGI failed. - Added by Mo Zoubi about 5 years ago

Thanks Manasalsa for your reply, but that didn't work.
I managed to get it working by doing the following:

1. Rebuild AppKonference
2. Removed GSM support from trunk.

Inbound is working now. Thank you.

RE: Inbound SIP URI / FastAGI failed. - Added by Jorge Cornejo about 5 years ago

Well port 5060 is for Kamailio that somehow (still in the darkness) should forward traffic to Asterisk, but as you, this doesn't happens at all. Obviously there is a missing conf between Kamailio and Asterisk.
That is why your 5070 (Asterisk in v4) does work nice.
Thanks for sharing your solution. I am still stuck at even making GoAutoDial to accept a Twilio trunk.

RE: Inbound SIP URI / FastAGI failed. - Added by Jorge Cornejo about 5 years ago

Did you had to manually edit any asterisk file for Inbound trunk?

RE: Inbound SIP URI / FastAGI failed. - Added by Mo Zoubi about 5 years ago

Hi,

No I did not change any of the asterisk files for inbound trunk.
Honestly, I don't think FastAGI is working well on the machine. Still testing it.
I wonder when the official GAD4 iso will be available. Right now, the pre-release iso does not work when uploaded to vultr cloud. However, gad3 iso works fine.\\

RE: Inbound SIP URI / FastAGI failed. - Added by Rafael Piza almost 5 years ago

I was getting this error in FastAgi and I do not know how to fix it.

res_agi.c:2038 handle_connection: Connecting to '127.0.0.1:4577' failed for url 'agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)': Connection refused
res_agi.c:2110 launch_netscript: Couldn't connect to any host.  FastAGI failed.

RE: Inbound SIP URI / FastAGI failed. - Added by Wittie Manansala almost 5 years ago

Hi Rafael,

Please provide us details how you encountered the issue.

Thanks

RE: Inbound SIP URI / FastAGI failed. - Added by Anusha Chaturanga about 3 years ago

Hello

Its work for me after changing the conferences to ConfBridge but need to check other things as well

extensions-goautodial.conf

; VICIDIAL konferences
exten => _86000[5-9]X,1,ConfBridge(${EXTEN})
exten => _86000[5-9]X,n,Hangup()
exten => _8600[1-2]XX,1,ConfBridge(${EXTEN})
exten => _8600[1-2]XX,n,Hangup()

    (1-23/23)
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