Call auto disconnected as agent logged in

[mudasar321]

Hi there,

I recently fresh installed ViciDial. When I try to login as agent for outbound campaign, I receive call at my softphone, as soon I answer it got disconnected automatically. Please advise. Below are the logs I captured.

[May 3 17:47:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 3 17:47:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 3 17:47:09] == Manager 'sendcron' logged on from 127.0.0.1
[May 3 17:47:09] ERROR[13269]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("vicibox10", "(null)", ...): Name or service not known
[May 3 17:47:09] WARNING[13269]: acl.c:892 resolve_first: Unable to lookup 'vicibox10'
[May 3 17:47:09] == Using SIP RTP CoS mark 5
[May 3 17:47:09] Audio is at 18278
[May 3 17:47:09] Adding codec ulaw to SDP
[May 3 17:47:09] Adding codec gsm to SDP
[May 3 17:47:09] Adding non-codec 0x1 (telephone-event) to SDP
[May 3 17:47:09] Reliably Transmitting (NAT) to 64.49.65.10:53553:
[May 3 17:47:09] INVITE sip:1002@192.168.5.226:53553;ob SIP/2.0
[May 3 17:47:09] Via: SIP/2.0/UDP 208.76.252.234:5060;branch=z9hG4bK0e3816a5;rport
[May 3 17:47:09] Max-Forwards: 70
[May 3 17:47:09] From: "ACagcW16516144011002100210021002" <sip:7185772813@208.76.252.234>;tag=as3db616b5
[May 3 17:47:09] To: <sip:1002@192.168.5.226:53553;ob>
[May 3 17:47:09] Contact: <sip:7185772813@208.76.252.234:5060>
[May 3 17:47:09] Call-ID: 2aa0efea0251eee0454d69e521faef34@208.76.252.234:5060
[May 3 17:47:09] CSeq: 102 INVITE
[May 3 17:47:09] User-Agent: Asterisk PBX 13.38.2-vici
[May 3 17:47:09] Date: Tue, 03 May 2022 21:47:09 GMT
[May 3 17:47:09] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[May 3 17:47:09] Supported: replaces, timer
[May 3 17:47:09] Remote-Party-ID: "ACagcW16516144011002100210021002" <sip:7185772813@208.76.252.234>;party=calling;privacy=off;screen=no
[May 3 17:47:09] Content-Type: application/sdp
[May 3 17:47:09] Content-Length: 282
[May 3 17:47:09]
[May 3 17:47:09] v=0
[May 3 17:47:09] o=root 366715024 366715024 IN IP4 208.76.252.234
[May 3 17:47:09] s=Asterisk PBX 13.38.2-vici
[May 3 17:47:09] c=IN IP4 208.76.252.234
[May 3 17:47:09] t=0 0
[May 3 17:47:09] m=audio 18278 RTP/AVP 0 3 101
[May 3 17:47:09] a=rtpmap:0 PCMU/8000
[May 3 17:47:09] a=rtpmap:3 GSM/8000
[May 3 17:47:09] a=rtpmap:101 telephone-event/8000
[May 3 17:47:09] a=fmtp:101 0-16
[May 3 17:47:09] a=ptime:20
[May 3 17:47:09] a=maxptime:150
[May 3 17:47:09] a=sendrecv
[May 3 17:47:09]
[May 3 17:47:09] ---
[May 3 17:47:09] -- Called 1002
[May 3 17:47:10] Retransmitting #1 (NAT) to 64.49.65.10:53553:
[May 3 17:47:10] INVITE sip:1002@192.168.5.226:53553;ob SIP/2.0
[May 3 17:47:10] Via: SIP/2.0/UDP 208.76.252.234:5060;branch=z9hG4bK0e3816a5;rport
[May 3 17:47:10] Max-Forwards: 70
[May 3 17:47:10] From: "ACagcW16516144011002100210021002" <sip:7185772813@208.76.252.234>;tag=as3db616b5
[May 3 17:47:10] To: <sip:1002@192.168.5.226:53553;ob>
[May 3 17:47:10] Contact: <sip:7185772813@208.76.252.234:5060>
[May 3 17:47:10] Call-ID: 2aa0efea0251eee0454d69e521faef34@208.76.252.234:5060
[May 3 17:47:10] CSeq: 102 INVITE
[May 3 17:47:10] User-Agent: Asterisk PBX 13.38.2-vici
[May 3 17:47:10] Date: Tue, 03 May 2022 21:47:09 GMT
[May 3 17:47:10] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[May 3 17:47:10] Supported: replaces, timer
[May 3 17:47:10] Remote-Party-ID: "ACagcW16516144011002100210021002" <sip:7185772813@208.76.252.234>;party=calling;privacy=off;screen=no
[May 3 17:47:10] Content-Type: application/sdp
[May 3 17:47:10] Content-Length: 282
[May 3 17:47:10]
[May 3 17:47:10] v=0
[May 3 17:47:10] o=root 366715024 366715024 IN IP4 208.76.252.234
[May 3 17:47:10] s=Asterisk PBX 13.38.2-vici
[May 3 17:47:10] c=IN IP4 208.76.252.234
[May 3 17:47:10] t=0 0
[May 3 17:47:10] m=audio 18278 RTP/AVP 0 3 101
[May 3 17:47:10] a=rtpmap:0 PCMU/8000
[May 3 17:47:10] a=rtpmap:3 GSM/8000
[May 3 17:47:10] a=rtpmap:101 telephone-event/8000
[May 3 17:47:10] a=fmtp:101 0-16
[May 3 17:47:10] a=ptime:20
[May 3 17:47:10] a=maxptime:150
[May 3 17:47:10] a=sendrecv
[May 3 17:47:10]
[May 3 17:47:10] ---
[May 3 17:47:10]
[May 3 17:47:10] <--- SIP read from UDP:64.49.65.10:53553 --->
[May 3 17:47:10] SIP/2.0 100 Trying
[May 3 17:47:10] Via: SIP/2.0/UDP 208.76.252.234:5060;rport=5060;received=208.76.252.234;branch=z9hG4bK0e3816a5
[May 3 17:47:10] Call-ID: 2aa0efea0251eee0454d69e521faef34@208.76.252.234:5060
[May 3 17:47:10] From: "ACagcW16516144011002100210021002" <sip:7185772813@208.76.252.234>;tag=as3db616b5
[May 3 17:47:10] To: <sip:1002@192.168.5.226;ob>
[May 3 17:47:10] CSeq: 102 INVITE
[May 3 17:47:10] Content-Length: 0
[May 3 17:47:10]
[May 3 17:47:10] <------------->
[May 3 17:47:10] --- (7 headers 0 lines) ---
[May 3 17:47:10]
[May 3 17:47:10] <--- SIP read from UDP:64.49.65.10:53553 --->
[May 3 17:47:10] SIP/2.0 180 Ringing
[May 3 17:47:10] Via: SIP/2.0/UDP 208.76.252.234:5060;rport=5060;received=208.76.252.234;branch=z9hG4bK0e3816a5
[May 3 17:47:10] Call-ID: 2aa0efea0251eee0454d69e521faef34@208.76.252.234:5060
[May 3 17:47:10] From: "ACagcW16516144011002100210021002" <sip:7185772813@208.76.252.234>;tag=as3db616b5
[May 3 17:47:10] To: <sip:1002@192.168.5.226;ob>;tag=31d0b62d43c443b4a09e1074cbc8919d
[May 3 17:47:10] CSeq: 102 INVITE
[May 3 17:47:10] Contact: "1002 ViCiDial" <sip:1002@192.168.5.226:53553;ob>
[May 3 17:47:10] Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
[May 3 17:47:10] Content-Length: 0
[May 3 17:47:10]
[May 3 17:47:10] <------------->
[May 3 17:47:10] --- (9 headers 0 lines) ---
[May 3 17:47:10] sip_route_dump: route/path hop: <sip:1002@192.168.5.226:53553;ob>
[May 3 17:47:10] -- SIP/1002-00000004 is ringing
[May 3 17:47:10]
[May 3 17:47:10] <--- SIP read from UDP:64.49.65.10:53553 --->
[May 3 17:47:10] SIP/2.0 180 Ringing
[May 3 17:47:10] Via: SIP/2.0/UDP 208.76.252.234:5060;rport=5060;received=208.76.252.234;branch=z9hG4bK0e3816a5
[May 3 17:47:10] Call-ID: 2aa0efea0251eee0454d69e521faef34@208.76.252.234:5060
[May 3 17:47:10] From: "ACagcW16516144011002100210021002" <sip:7185772813@208.76.252.234>;tag=as3db616b5
[May 3 17:47:10] To: <sip:1002@192.168.5.226;ob>;tag=31d0b62d43c443b4a09e1074cbc8919d
[May 3 17:47:10] CSeq: 102 INVITE
[May 3 17:47:10] Contact: "1002 ViCiDial" <sip:1002@192.168.5.226:53553;ob>
[May 3 17:47:10] Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
[May 3 17:47:10] Content-Length: 0
[May 3 17:47:10]
[May 3 17:47:10] <------------->
[May 3 17:47:10] --- (9 headers 0 lines) ---
[May 3 17:47:10] sip_route_dump: route/path hop: <sip:1002@192.168.5.226:53553;ob>
[May 3 17:47:10] -- SIP/1002-00000004 is ringing
[May 3 17:47:11]
[May 3 17:47:11] <--- SIP read from UDP:64.49.65.10:53553 --->
[May 3 17:47:11] SIP/2.0 200 OK
[May 3 17:47:11] Via: SIP/2.0/UDP 208.76.252.234:5060;rport=5060;received=208.76.252.234;branch=z9hG4bK0e3816a5
[May 3 17:47:11] Call-ID: 2aa0efea0251eee0454d69e521faef34@208.76.252.234:5060
[May 3 17:47:11] From: "ACagcW16516144011002100210021002" <sip:7185772813@208.76.252.234>;tag=as3db616b5
[May 3 17:47:11] To: <sip:1002@192.168.5.226;ob>;tag=31d0b62d43c443b4a09e1074cbc8919d
[May 3 17:47:11] CSeq: 102 INVITE
[May 3 17:47:11] Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
[May 3 17:47:11] Contact: "1002 ViCiDial" <sip:1002@192.168.5.226:53553;ob>
[May 3 17:47:11] Supported: replaces, 100rel, timer, norefersub
[May 3 17:47:11] Content-Type: application/sdp
[May 3 17:47:11] Content-Length: 317
[May 3 17:47:11]
[May 3 17:47:11] v=0
[May 3 17:47:11] o=- 3860621231 3860621232 IN IP4 192.168.5.226
[May 3 17:47:11] s=pjmedia
[May 3 17:47:11] b=AS:84
[May 3 17:47:11] t=0 0
[May 3 17:47:11] a=X-nat:0
[May 3 17:47:11] m=audio 4026 RTP/AVP 0 101
[May 3 17:47:11] c=IN IP4 192.168.5.226
[May 3 17:47:11] b=TIAS:64000
[May 3 17:47:11] a=rtcp:4027 IN IP4 192.168.5.226
[May 3 17:47:11] a=sendrecv
[May 3 17:47:11] a=rtpmap:0 PCMU/8000
[May 3 17:47:11] a=rtpmap:101 telephone-event/8000
[May 3 17:47:11] a=fmtp:101 0-16
[May 3 17:47:11] a=ssrc:59319544 cname:640877684b511bad
[May 3 17:47:11] <------------->
[May 3 17:47:11] --- (11 headers 15 lines) ---
[May 3 17:47:11] Got SDP version 3860621232 and unique parts [- 3860621231 IN IP4 192.168.5.226]
[May 3 17:47:11] Found RTP audio format 0
[May 3 17:47:11] Found RTP audio format 101
[May 3 17:47:11] Found audio description format PCMU for ID 0
[May 3 17:47:11] Found audio description format telephone-event for ID 101
[May 3 17:47:11] Capabilities: us - (ulaw|gsm), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
[May 3 17:47:11] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[May 3 17:47:11] > 0x7fe15806abb0 -- Strict RTP learning after remote address set to: 192.168.5.226:4026
[May 3 17:47:11] Peer audio RTP is at port 192.168.5.226:4026
[May 3 17:47:11] sip_route_dump: route/path hop: <sip:1002@192.168.5.226:53553;ob>
[May 3 17:47:11] Transmitting (NAT) to 64.49.65.10:53553:
[May 3 17:47:11] ACK sip:1002@192.168.5.226:53553;ob SIP/2.0
[May 3 17:47:11] Via: SIP/2.0/UDP 208.76.252.234:5060;branch=z9hG4bK73550bc4;rport
[May 3 17:47:11] Max-Forwards: 70
[May 3 17:47:11] From: "ACagcW16516144011002100210021002" <sip:7185772813@208.76.252.234>;tag=as3db616b5
[May 3 17:47:11] To: <sip:1002@192.168.5.226:53553;ob>;tag=31d0b62d43c443b4a09e1074cbc8919d
[May 3 17:47:11] Contact: <sip:7185772813@208.76.252.234:5060>
[May 3 17:47:11] Call-ID: 2aa0efea0251eee0454d69e521faef34@208.76.252.234:5060
[May 3 17:47:11] CSeq: 102 ACK
[May 3 17:47:11] User-Agent: Asterisk PBX 13.38.2-vici
[May 3 17:47:11] Content-Length: 0
[May 3 17:47:11]
[May 3 17:47:11]
[May 3 17:47:11] ---
[May 3 17:47:11] -- SIP/1002-00000004 answered
[May 3 17:47:11] -- Executing [8600053@default:1] MeetMe("SIP/1002-00000004", "8600053,F") in new stack
[May 3 17:47:11] WARNING[13270][C-00000007]: app_meetme.c:1663 build_conf: Unable to open DAHDI pseudo device
[May 3 17:47:11] == Spawn extension (default, 8600053, 1) exited non-zero on 'SIP/1002-00000004'
[May 3 17:47:11] -- Executing [h@default:1] AGI("SIP/1002-00000004", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK)") in new stack
[May 3 17:47:11] -- <SIP/1002-00000004>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK) completed, returning 0
[May 3 17:47:11] Scheduling destruction of SIP dialog '2aa0efea0251eee0454d69e521faef34@208.76.252.234:5060' in 23552 ms (Method: INVITE)
[May 3 17:47:11] Reliably Transmitting (NAT) to 64.49.65.10:53553:
[May 3 17:47:11] BYE sip:1002@192.168.5.226:53553;ob SIP/2.0
[May 3 17:47:11] Via: SIP/2.0/UDP 208.76.252.234:5060;branch=z9hG4bK4c6db99a;rport
[May 3 17:47:11] Max-Forwards: 70
[May 3 17:47:11] From: "ACagcW16516144011002100210021002" <sip:7185772813@208.76.252.234>;tag=as3db616b5
[May 3 17:47:11] To: <sip:1002@192.168.5.226:53553;ob>;tag=31d0b62d43c443b4a09e1074cbc8919d
[May 3 17:47:11] Call-ID: 2aa0efea0251eee0454d69e521faef34@208.76.252.234:5060
[May 3 17:47:11] CSeq: 103 BYE
[May 3 17:47:11] User-Agent: Asterisk PBX 13.38.2-vici
[May 3 17:47:11] X-Asterisk-HangupCause: Normal Clearing
[May 3 17:47:11] X-Asterisk-HangupCauseCode: 16
[May 3 17:47:11] Content-Length: 0
[May 3 17:47:11]
[May 3 17:47:11]
[May 3 17:47:11] ---
[May 3 17:47:12]
[May 3 17:47:12] <--- SIP read from UDP:64.49.65.10:53553 --->
[May 3 17:47:12] SIP/2.0 200 OK
[May 3 17:47:12] Via: SIP/2.0/UDP 208.76.252.234:5060;rport=5060;received=208.76.252.234;branch=z9hG4bK4c6db99a
[May 3 17:47:12] Call-ID: 2aa0efea0251eee0454d69e521faef34@208.76.252.234:5060
[May 3 17:47:12] From: "ACagcW16516144011002100210021002" <sip:7185772813@208.76.252.234>;tag=as3db616b5
[May 3 17:47:12] To: <sip:1002@192.168.5.226;ob>;tag=31d0b62d43c443b4a09e1074cbc8919d
[May 3 17:47:12] CSeq: 103 BYE
[May 3 17:47:12] Content-Length: 0
[May 3 17:47:12]
[May 3 17:47:12] <------------->
[May 3 17:47:12] --- (7 headers 0 lines) ---
[May 3 17:47:12] Really destroying SIP dialog '2aa0efea0251eee0454d69e521faef34@208.76.252.234:5060' Method: INVITE
[May 3 17:47:12] == Manager 'sendcron' logged off from 127.0.0.1
[May 3 17:47:16]
[May 3 17:47:16] <--- SIP read from UDP:64.49.65.10:53553 --->
[May 3 17:47:16]
[May 3 17:47:16] <------------->

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method (7.X.X?) and vicidial version with build (VERSION: 2.X-XXXx ... BUILD: #####-####).

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "manual/from scratch" you must post your operating system with version (and the .iso version from which you installed your original operating system) plus a link to the installation instructions you used. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3)

app_meetme.c:1663 build_conf: Unable to open DAHDI pseudo device

It would appear your installation failed. Or perhaps you are not done with the installation.

* Did you reboot upon completion and verify that "screen -list" resulted in more than 10 screens?
* Did you follow the Vicidial Manager's Manual from Page One until you reached your "fail" point? Without skipping anything or making up your own steps?

4) I *Highly* recommend starting over. Perhaps more than once. You'll learn from each new iteration and become more comfortable with the (FREE!!) installation process.

5) There are many possibilities for the failure presented, among those: The meetme module is complaining about the DAHDI device failure. So DAHDI may not be properly installed, meetme might not be properly installed, or the system may simply need a reboot.

6) In any event: This was not a "general discussion", this was a request for Support. Moving the thread to the Support board accordingly.

[carpenox]

run these commands

modprobe dahdi
dahdi_cfg -vvv

report back what the second one says

[mudasar321]

Hi @carpenox,

Here is the output for command "dahdi_cfg -vvv"

DAHDI Tools Version - 3.1.0

DAHDI Version: 3.1.0
Echo Canceller(s):
Configuration
======================


Channel map:


0 channels to configure.

[mudasar321]

Hi @carpenox,

I run those commands and now it's working perfect. I don't know how it made it difference. What is the function of these commands?

[williamconley]

modprobe dahdi loads the dahdi functions. should happen during boot, but if you didn't reboot after the installation ...

[mudasar321]

It was rebooted multiple times but still not loading dahdi on boot. Eventually I need to add this command "modprobe dahdi" in cron job to run on boot, otherwise not loading dahdi. Anywhere in configuration to run this on boot?

[williamconley]

I'd consider a re-install to see if you can get a valid installation.

Check the vicibox install log and see if there were any reported errors. That may not be the only flaw, it can be a headache to find problems one at a time as you go along, rather than a reinstall until you get a good, clean, installation with zero errors.

And it's free, so there's that.

[mudasar321]

Got it, thanks