vicidial.org

Hello all,

Recently we obtained services from a new provider that uses IP authentication. We setup the carrier so that the outbound calls work, but we can't get any inbound calls and the provider has informed us that the calls are getting through from their side. I am somewhat new to this and various guides to configuration of inbound routing and all of that have confused me, so I seek help from you.

So what I'm looking for basically is step to step guide how to receive inbound calls from the outside on our server, and I'll give all the information about the server and what I did so far in hopes of solving this.

We're using Vicidial VERSION: 2.4-309a BUILD: 110430-1642, Asterisk 1.4.39.1-vici RPM. I know it's outdated but can't go through updating it currently.

Carrier info:

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[carrierName]
type=peer
host=xxx.xxx.xxx.xxx
fromdomain=xxx.xxx.xxx.xxx
outboundproxy=xxx.xxx.xxx.xxx
port=5060
insecure=port,invite
canreinvite=no
nat=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g722
dtmfmode=rfc2833
registertimeout=600
context=trunkinbound


And the dialplan is next, and it is working

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exten => _385.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _385.,2,Dial(${carrierName}/${EXTEN:0},55,tTo)
exten => _385.,3,Hangup


Now after creating the carrier I added a new In-Group and allowed it in the campaign that is using this carrier, then I added a new DID with the number the providers assigned us, and set the DID route to IN_GROUP and set the In-Group ID to the ID of the newly created In-group.

When I call this number from an outside phone I get one ring and then that busy signal (beep beep beep...).

Do I need to add anything to the dialplan extensions, or to the extensions.conf or maybe I have missed some steps in configuration of this service?

Thank you in advance, all help is appreciated.

Can you post what you see on CLI when you dial your DID?

Thank's for the answer, and there is nothing in the CLI, with or without sip debug. It seems like that calls are not coming from the side of the provider but they're saying that the calls are passing through. So I don't really understand weather it's a problem on our server or with their service, because outbound calls are working fine. Any other ideas?

does sip show peers show you the provider as REGISTERED? If NO, then check your settings with provider and make sure firewall is not blocking provider's IP and/or sip port.

Try allowguest=yes in you sip.conf

Thank you for the replies and sorry for the late answer, it's been one hell of a week.

I had a conversation with the providers and they were sending the invite request on another port instead of 5060 so that's why nothing came through. As soon as they started sending the invite on 5060 port we can see the incoming calls but there is a new problem now, the calls seem to go to queue even though an agent is in the group and has selected the inbound group like seen in the image below.



I will continue searching for the answers on google and so on, but if you can offer a quick solution I would be very thankful for that. And thanks again for the replies

I still haven't found the solution to the problem. Tried deleting the in-group and the DID and then creating them again but the same problem persists.

When I call the DID number it rings for a millisecond and then the call automatically ends. On the agent side it doesn't ring, the only change is that Calls in Queue increments. And on the reports page you can see calls waiting like in the picture below



And in the CLI I get this

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    -- Executing [##DIDNUMBER##@trunkinbound:1] AGI("SIP/Softnet_Inbound-00001263", "agi-DID_route.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
    -- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20181108093731_##DIDNUMBER##_##CALLERNUMBER##)
[Nov  8 09:37:31] ERROR[24802]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
    -- AGI Script agi-DID_route.agi completed, returning 0
    -- Executing [99909*10***DID@default:1] Answer("SIP/Softnet_Inbound-00001263", "") in new stack
    -- Executing [99909*10***DID@default:2] AGI("SIP/Softnet_Inbound-00001263", "agi-VDAD_ALL_inbound.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
    -- Executing [99909*10***DID@default:3] Hangup("SIP/Softnet_Inbound-00001263", "") in new stack
  == Spawn extension (default, 99909*10***DID, 3) exited non-zero on 'SIP/Softnet_Inbound-00001263'
    -- Executing [h@default:1] DeadAGI("SIP/Softnet_Inbound-00001263", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0

I think the problem is in routing the call which is a little difficult to understand. Maybe it's something simple and I just don't quite see it yet.

Again any help would be deeply appreciated. Thank you in advance.

Please share your configurations and is it only me who can't view your images?

Hello again, I can view the pictures, here are the links if the ones posted up don't work:

http://oi66.tinypic.com/2lu8llj.jpg
http://oi66.tinypic.com/10e1owj.jpg

Configuration stayed the same for the carrier, I deleted the DID and the in_group then created new ones again following the manual, DID is pointed to the in_group and in the traffic for the DID and the in_group you can see the calls that came but we're unanswered. I captured a debug of the call made this morning, it seems that the routing is poorly done, and that the calls are automatically answered amd hung up.

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<--- SIP read from Carrier_IP:5060 --->
INVITE sip:##OUR_NMBR##@Our_IP:5060 SIP/2.0
Record-Route: <sip:Carrier_IP;lr;ftag=KK46vZU436yNj;did=44d.4a333e54>
Via: SIP/2.0/UDP Carrier_IP:5060;branch=z9hG4bK9337.e6c964c1.0
Via: SIP/2.0/UDP Carrier_IP2:5080;received=Carrier_IP2;rport=5080;branch=z9hG4bK4t9cDQXmrtXDB
Max-Forwards: 11
From: "anonymous" <sip:##CALLER_NUMBER##@Carrier_IP2>;tag=KK46vZU436yNj
To: <sip:##OUR_NMBR##@Carrier_IP>
Call-ID: b058e125-6277-1237-ac90-e41f137b4e08
Seq: 130752035 INVITE
Contact: <sip:mod_sofia@Carrier_IP2:5080>
User-Agent: FreeSWITCH-mod_sofia/1.6.17-34-0fc0946~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 224
P-Charging-Vector: icid-value="IBCF-PRIV-4-154217959527184712914695";orig-ioi=3GPP-UTRAN

v=0
o=FreeSWITCH 1542152409 1542152410 IN IP4 Carrier_IP2
s=FreeSWITCH
c=IN IP4 Carrier_IP2
t=0 0
m=audio 23790 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (18 headers 10 lines) ---
Sending to Carrier_IP : 5060 (NAT)
Using INVITE request as basis request - b058e125-6277-1237-ac90-e41f137b4e08
Found peer 'Netsoft1'
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100e (gsm|ulaw|alaw|g722), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port Carrier_IP2:23790
Looking for ##OUR_NMBR## in trunkinbound (domain Our_IP)
list_route: hop: <sip:Carrier_IP;lr;ftag=KK46vZU436yNj;did=44d.4a333e54>

<--- Transmitting (NAT) to Carrier_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP Carrier_IP:5060;branch=z9hG4bK9337.e6c964c1.0;received=Carrier_IP
Via: SIP/2.0/UDP Carrier_IP2:5080;received=Carrier_IP2;rport=5080;branch=z9hG4bK4t9cDQXmrtXDB
Record-Route: <sip:Carrier_IP;lr;ftag=KK46vZU436yNj;did=44d.4a333e54>
From: "anonymous" <sip:##CALLER_NUMBER##@Carrier_IP2>;tag=KK46vZU436yNj
To: <sip:##OUR_NMBR##@Carrier_IP>
Call-ID: b058e125-6277-1237-ac90-e41f137b4e08
CSeq: 130752035 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:##OUR_NMBR##@Our_IP>
Content-Length: 0


<------------>
    -- Executing [##OUR_NMBR##@trunkinbound:1] AGI("SIP/Netsoft1-000009c5", "agi-DID_route.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
    -- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20181114081315_##OUR_NMBR##_##CALLER_NUMBER##)
[Nov 14 08:13:15] ERROR[19681]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
    -- AGI Script agi-DID_route.agi completed, returning 0
    -- Executing [99909*10***DID@default:1] Answer("SIP/Netsoft1-000009c5", "") in new stack
Audio is at Our_IP port 11696
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to Carrier_IP:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP Carrier_IP:5060;branch=z9hG4bK9337.e6c964c1.0;received=Carrier_IP
Via: SIP/2.0/UDP Carrier_IP2:5080;received=Carrier_IP2;rport=5080;branch=z9hG4bK4t9cDQXmrtXDB
Record-Route: <sip:Carrier_IP;lr;ftag=KK46vZU436yNj;did=44d.4a333e54>
From: "anonymous" <sip:##CALLER_NUMBER##@Carrier_IP2>;tag=KK46vZU436yNj
To: <sip:##OUR_NMBR##@Carrier_IP>;tag=as365fc120
Call-ID: b058e125-6277-1237-ac90-e41f137b4e08
CSeq: 130752035 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:##OUR_NMBR##@Our_IP>
Content-Type: application/sdp
Content-Length: 209

v=0
o=root 2675 2675 IN IP4 Our_IP
s=session
c=IN IP4 Our_IP
t=0 0
m=audio 11696 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
    -- Executing [99909*10***DID@default:2] AGI("SIP/Netsoft1-000009c5", "agi-VDAD_ALL_inbound.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi

<--- SIP read from Carrier_IP:5060 --->
ACK sip:##OUR_NMBR##@Our_IP SIP/2.0
Via: SIP/2.0/UDP Carrier_IP:5060;branch=z9hG4bK9337.e6c964c1.2
Via: SIP/2.0/UDP Carrier_IP2:5080;received=Carrier_IP2;rport=5080;branch=z9hG4bK5325ejerN3K0p
Max-Forwards: 69
From: "anonymous" <sip:##CALLER_NUMBER##@Carrier_IP2>;tag=KK46vZU436yNj
To: <sip:##OUR_NMBR##@Carrier_IP>;tag=as365fc120
Call-ID: b058e125-6277-1237-ac90-e41f137b4e08
CSeq: 130752035 ACK
Contact: <sip:mod_sofia@Carrier_IP2:5080>
ontent-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
    -- Executing [99909*10***DID@default:3] Hangup("SIP/Netsoft1-000009c5", "") in new stack
  == Spawn extension (default, 99909*10***DID, 3) exited non-zero on 'SIP/Netsoft1-000009c5'
    -- Executing [h@default:1] DeadAGI("SIP/Netsoft1-000009c5", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
Scheduling destruction of SIP dialog 'b058e125-6277-1237-ac90-e41f137b4e08' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:Carrier_IP;lr;ftag=KK46vZU436yNj;did=44d.4a333e54> for address/port to send to
set_destination: set destination to Carrier_IP, port 5060
Reliably Transmitting (NAT) to Carrier_IP:5060:
BYE sip:mod_sofia@Carrier_IP2:5080 SIP/2.0
Via: SIP/2.0/UDP Our_IP:5060;branch=z9hG4bK61bfe397;rport
Route: <sip:Carrier_IP;lr;ftag=KK46vZU436yNj;did=44d.4a333e54>
From: <sip:##OUR_NMBR##@Carrier_IP>;tag=as365fc120
To: "anonymous" <sip:##CALLER_NUMBER##@Carrier_IP2>;tag=KK46vZU436yNj
Call-ID: b058e125-6277-1237-ac90-e41f137b4e08
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Nov 14 08:13:15] ERROR[19681]: cdr_custom.c:138 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : No such file or directory

<--- SIP read from Carrier_IP:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP Our_IP:5060;received=Our_IP;branch=z9hG4bK61bfe397;rport=5060
From: <sip:##OUR_NMBR##@Carrier_IP>;tag=as365fc120
To: "anonymous" <sip:##CALLER_NUMBER##@Carrier_IP2>;tag=KK46vZU436yNj
Call-ID: b058e125-6277-1237-ac90-e41f137b4e08
CSeq: 102 BYE
User-Agent: FreeSWITCH-mod_sofia/1.6.17-34-0fc0946~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Content-Length: 0


I just noticed that there is a second IP from the carrier present in the invites and all, and don't understand why.

Here's the AGI log for this call, don't know if it's useful but why not.

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2018-11-14 08:13:15|VDfastAGI|begin|+++++++++++++++++ FastAGI Start ++++++++++++++++++++++++++++++++++++++++
2018-11-14 08:13:15|VDfastAGI|begin|Perl Environment Dump:
2018-11-14 08:13:15|VDfastAGI|begin|0|--debug
2018-11-14 08:13:15|VDfastAGI|begin|URL HVcauses: |PRI|NODEBUG|16|||0|
2018-11-14 08:13:15|VDfastAGI|begin|AGI Environment Dump:
2018-11-14 08:13:15|VDfastAGI|begin| -- accountcode =
2018-11-14 08:13:15|VDfastAGI|begin| -- callerid = CALLER_NUMBER
2018-11-14 08:13:15|VDfastAGI|begin| -- calleridname = Y1140813150000229060
2018-11-14 08:13:15|VDfastAGI|begin| -- callingani2 = 0
2018-11-14 08:13:15|VDfastAGI|begin| -- callingpres = 0
2018-11-14 08:13:15|VDfastAGI|begin| -- callingtns = 0
2018-11-14 08:13:15|VDfastAGI|begin| -- callington = 0
2018-11-14 08:13:15|VDfastAGI|begin| -- channel = SIP/Netsoft1-000009c5
2018-11-14 08:13:15|VDfastAGI|begin| -- context = default
2018-11-14 08:13:15|VDfastAGI|begin| -- dnid = OUR_NUMBER
2018-11-14 08:13:15|VDfastAGI|begin| -- enhanced = 0.0
2018-11-14 08:13:15|VDfastAGI|begin| -- extension = h
2018-11-14 08:13:15|VDfastAGI|begin| -- language = en
2018-11-14 08:13:15|VDfastAGI|begin| -- network = yes
2018-11-14 08:13:15|VDfastAGI|begin| -- network_script = call_log--HVcauses--PRI-----NODEBUG-----16---------------
2018-11-14 08:13:15|VDfastAGI|begin| -- priority = 1
2018-11-14 08:13:15|VDfastAGI|begin| -- rdnis = unknown
2018-11-14 08:13:15|VDfastAGI|begin| -- request = agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------
2018-11-14 08:13:15|VDfastAGI|begin| -- type = SIP
2018-11-14 08:13:15|VDfastAGI|begin| -- uniqueid = 1542179595.11870
2018-11-14 08:13:15|VDfastAGI|begin|AGI Variables: |1542179595.11870|SIP/Netsoft1-000009c5|h|SIP|Y1140813150000229060|
2018-11-14 08:13:15|VDfastAGI|call_log|Process to run: |agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------|call_log|END|
2018-11-14 08:13:15|VDfastAGI|call_log||CALL HUNG UP|
2018-11-14 08:13:15|VDfastAGI|call_log|URL HVcauses: |PRI|NODEBUG|16|||0|
2018-11-14 08:14:22|VDfastAGI|begin|+++++++++++++++++ FastAGI Start ++++++++++++++++++++++++++++++++++++++++

It seems that the calls from other providers do the same in their campaign, but currently we have no need for the other providers, just this one. That gives me a reason to believe that it's a problem in general with the server or with the routing system. Got any ideas?

I can give you more info about the configuration but i need some info on what to give you, kinda new in all of this. Thanks in advance

Update on the problem, it seems that the only DID that works is the default one. Tried changing the default DID to point to an ingroup and it does so, but the results are same as before. When the call is made it just answers and ends (and the caller is charged for calling).

So it's not a problem with just one DID and one ingroup, it's with all the DID's that take calls from different carriers (sipgate, sipcall, softnet).

Am really stuck and don't know how to proceed with this, seems that the default DID picks up all calls and that none are routed to a correct DID even though some were working a few months back (sipgate). All carriers have context=trunkinbound in them, all DID's are set to in_group and are pointing to a desired in_group but there is still no progress.

I don't even know what to ask anymore

Perhaps another example would be good.

Code: Select all

2018-11-14 08:13:15|VDfastAGI|call_log||CALL HUNG UP|

According to this, the caller hung up. You should be testing with a call you control on both sides.

Hello William, thanks for the response, here's another even though it also has that line "2018-11-23 08:53:27|VDfastAGI|call_log||CALL HUNG UP|"

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2018-11-23 08:53:27|VDfastAGI|begin|+++++++++++++++++ FastAGI Start ++++++++++++++++++++++++++++++++++++++++
2018-11-23 08:53:27|VDfastAGI|begin|Perl Environment Dump:
2018-11-23 08:53:27|VDfastAGI|begin|0|--debug
2018-11-23 08:53:27|VDfastAGI|begin|URL HVcauses: |PRI|NODEBUG|16|||0|
2018-11-23 08:53:27|VDfastAGI|begin|AGI Environment Dump:
2018-11-23 08:53:27|VDfastAGI|begin| -- accountcode =
2018-11-23 08:53:27|VDfastAGI|begin| -- callerid = 00387########   < my number
2018-11-23 08:53:27|VDfastAGI|begin| -- calleridname = Y1230853270000230065
2018-11-23 08:53:27|VDfastAGI|begin| -- callingani2 = 0
2018-11-23 08:53:27|VDfastAGI|begin| -- callingpres = 0
2018-11-23 08:53:27|VDfastAGI|begin| -- callingtns = 0
2018-11-23 08:53:27|VDfastAGI|begin| -- callington = 0
2018-11-23 08:53:27|VDfastAGI|begin| -- channel = SIP/Netsoft1-0000219a
2018-11-23 08:53:27|VDfastAGI|begin| -- context = default
2018-11-23 08:53:27|VDfastAGI|begin| -- dnid = 385########  < our DID
2018-11-23 08:53:27|VDfastAGI|begin| -- enhanced = 0.0
2018-11-23 08:53:27|VDfastAGI|begin| -- extension = h
2018-11-23 08:53:27|VDfastAGI|begin| -- language = en
2018-11-23 08:53:27|VDfastAGI|begin| -- network = yes
2018-11-23 08:53:27|VDfastAGI|begin| -- network_script = call_log--HVcauses--PRI-----NODEBUG-----16---------------
2018-11-23 08:53:27|VDfastAGI|begin| -- priority = 1
2018-11-23 08:53:27|VDfastAGI|begin| -- rdnis = unknown
2018-11-23 08:53:27|VDfastAGI|begin| -- request = agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------
2018-11-23 08:53:27|VDfastAGI|begin| -- type = SIP
2018-11-23 08:53:27|VDfastAGI|begin| -- uniqueid = 1542959607.41364
2018-11-23 08:53:27|VDfastAGI|begin|AGI Variables: |1542959607.41364|SIP/Netsoft1-0000219a|h|SIP|Y1230853270000230065|
2018-11-23 08:53:27|VDfastAGI|call_log|Process to run: |agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------|call_log|END|
2018-11-23 08:53:27|VDfastAGI|call_log||CALL HUNG UP|
2018-11-23 08:53:27|VDfastAGI|call_log|URL HVcauses: |PRI|NODEBUG|16|||0|


I called from my personal phone and was not charged for calling, on the phone side it looks like someone answered and immediately hung up. On the server side we have same as before

Code: Select all

    -- Executing [38516700128@trunkinbound:1] AGI("SIP/Netsoft1-0000219a", "agi-DID_route.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
    -- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20181123085327_385#########_00387#########)
[Nov 23 08:53:27] ERROR[30448]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
    -- AGI Script agi-DID_route.agi completed, returning 0
    -- Executing [99909*1***DID@default:1] Answer("SIP/Netsoft1-0000219a", "") in new stack
    -- Executing [99909*1***DID@default:2] AGI("SIP/Netsoft1-0000219a", "agi-VDAD_ALL_inbound.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
    -- Executing [99909*1***DID@default:3] Hangup("SIP/Netsoft1-0000219a", "") in new stack
  == Spawn extension (default, 99909*1***DID, 3) exited non-zero on 'SIP/Netsoft1-0000219a'
    -- Executing [h@default:1] DeadAGI("SIP/Netsoft1-0000219a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0


And on the agent's side he can only see Calls in Queue: 1 above the form fields, like seen in the pictures in my previous posts. Do you have any idea on what might be the issue in this case?

You're getting the hang of this. The command to hang up likely arrived via SIP, and the perl (agi) script merely did what it was told. Try sip debug instead of AGI debug. The sip handshake may tell you specifically what happened (like: wrong codec?) or it may just say "hang up". But at least you'll know it wasn't Vicidial.

SIP debug is not as easy to work with as AGI debug, but it's still doable. Also there are other tools available for SIP capture that can make it easier ... once you learn the tools, of course.

Ive noticed some carriers like a 1 before the 10 digit number and some just like the 10 digit number. Have you tried creating the DID’s both ways?

I posted a sip debug in one of my previous posts, posting new one made this morning. I get the overall logic of debugs, and I did couple of tcpdumps and read them with wireshark but still I couldn't find what exactly is the problem with the calls. I can see that the call was successful and that our server responded to the invite from carrier's ip, after which "agi-DID_route.agi" script was ran and returned 0 which probably means that it didn't find the correct route to the DID and was redirected to the default one, and at that time the call was automatically answered. Then server sent the OK, and the carrier returned ACK, after which the server played "sip-silence" and executed "agi-VDAD_ALL_inbound.agi" also returning 0 and hanging up the call. I cannot really see why the call was hung up or even answered when there was no ringing or answering the call in the agent's interface.

Code: Select all

<--- SIP read from xxx.carrierIP.xxx:5060 --->
INVITE sip:385XXXXXXXX@xxx.ourIP.xxx:5060 SIP/2.0
Record-Route: <sip:xxx.carrierIP.xxx;lr;ftag=g82ZSmBe7vDyH;did=b81.df76fc11>
Via: SIP/2.0/UDP xxx.carrierIP.xxx:5060;branch=z9hG4bK248c.5fc448d2.0
Via: SIP/2.0/UDP xxx.carrierIP2(RTP).xxx:5080;received=xxx.carrierIP2(RTP).xxx;rport=5080;branch=z9hG4bKp8vUjm5gpjp9H
Max-Forwards: 67
From: "00387XXXXXXX" <sip:00387XXXXXXX@xxx.carrierIP2(RTP).xxx>;tag=g82ZSmBe7vDyH
To: <sip:385XXXXXXXX@xxx.carrierIP.xxx>
Call-ID: 285e20b3-6d7a-1237-ac90-e41f137b4e08
CSeq: 131357297 INVITE
Contact: <sip:mod_sofia@xxx.carrierIP2(RTP).xxx:5080>
User-Agent: FreeSWITCH-mod_sofia/1.6.17-34-0fc0946~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 248

v=0
o=FreeSWITCH 1543361038 1543361039 IN IP4 xxx.carrierIP2(RTP).xxx
s=FreeSWITCH
c=IN IP4 xxx.carrierIP2(RTP).xxx
t=0 0
m=audio 25684 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (17 headers 11 lines) ---
Sending to xxx.carrierIP.xxx : 5060 (NAT)
Using INVITE request as basis request - 285e20b3-6d7a-1237-ac90-e41f137b4e08
Found peer 'Netsoft1'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100e (gsm|ulaw|alaw|g722), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port xxx.carrierIP2(RTP).xxx:25684
Looking for 385XXXXXXXX in trunkinbound (domain xxx.ourIP.xxx)
list_route: hop: <sip:xxx.carrierIP.xxx;lr;ftag=g82ZSmBe7vDyH;did=b81.df76fc11>

<--- Transmitting (NAT) to xxx.carrierIP.xxx:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.carrierIP.xxx:5060;branch=z9hG4bK248c.5fc448d2.0;received=xxx.carrierIP.xxx
Via: SIP/2.0/UDP xxx.carrierIP2(RTP).xxx:5080;received=xxx.carrierIP2(RTP).xxx;rport=5080;branch=z9hG4bKp8vUjm5gpjp9H
Record-Route: <sip:xxx.carrierIP.xxx;lr;ftag=g82ZSmBe7vDyH;did=b81.df76fc11>
From: "00387XXXXXXX" <sip:00387XXXXXXX@xxx.carrierIP2(RTP).xxx>;tag=g82ZSmBe7vDyH
To: <sip:385XXXXXXXX@xxx.carrierIP.xxx>
Call-ID: 285e20b3-6d7a-1237-ac90-e41f137b4e08
CSeq: 131357297 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:385XXXXXXXX@xxx.ourIP.xxx>
Content-Length: 0


<------------>
    -- Executing [385XXXXXXXX@trunkinbound:1] AGI("SIP/Netsoft1-00000127", "agi-DID_route.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
    -- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20181128082842_385XXXXXXXX_00387XXXXXXX)
[Nov 28 08:28:42] ERROR[12628]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
    -- AGI Script agi-DID_route.agi completed, returning 0
    -- Executing [99909*1***DID@default:1] Answer("SIP/Netsoft1-00000127", "") in new stack
Audio is at xxx.ourIP.xxx port 11244
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to xxx.carrierIP.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.carrierIP.xxx:5060;branch=z9hG4bK248c.5fc448d2.0;received=xxx.carrierIP.xxx
Via: SIP/2.0/UDP xxx.carrierIP2(RTP).xxx:5080;received=xxx.carrierIP2(RTP).xxx;rport=5080;branch=z9hG4bKp8vUjm5gpjp9H
Record-Route: <sip:xxx.carrierIP.xxx;lr;ftag=g82ZSmBe7vDyH;did=b81.df76fc11>
From: "00387XXXXXXX" <sip:00387XXXXXXX@xxx.carrierIP2(RTP).xxx>;tag=g82ZSmBe7vDyH
To: <sip:385XXXXXXXX@xxx.carrierIP.xxx>;tag=as4c552568
Call-ID: 285e20b3-6d7a-1237-ac90-e41f137b4e08
CSeq: 131357297 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:385XXXXXXXX@xxx.ourIP.xxx>
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 20467 20467 IN IP4 xxx.ourIP.xxx
s=session
c=IN IP4 xxx.ourIP.xxx
t=0 0
m=audio 11244 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
    -- Executing [99909*1***DID@default:2] AGI("SIP/Netsoft1-00000127", "agi-VDAD_ALL_inbound.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi

<--- SIP read from xxx.carrierIP.xxx:5060 --->
ACK sip:385XXXXXXXX@xxx.ourIP.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.carrierIP.xxx:5060;branch=z9hG4bK248c.5fc448d2.2
Via: SIP/2.0/UDP xxx.carrierIP2(RTP).xxx:5080;received=xxx.carrierIP2(RTP).xxx;rport=5080;branch=z9hG4bKQHpmmFpmKUcvD
Max-Forwards: 69
From: "00387XXXXXXX" <sip:00387XXXXXXX@xxx.carrierIP2(RTP).xxx>;tag=g82ZSmBe7vDyH
To: <sip:385XXXXXXXX@xxx.carrierIP.xxx>;tag=as4c552568
Call-ID: 285e20b3-6d7a-1237-ac90-e41f137b4e08
CSeq: 131357297 ACK
Contact: <sip:mod_sofia@xxx.carrierIP2(RTP).xxx:5080>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
    -- AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
    -- Executing [99909*1***DID@default:3] Hangup("SIP/Netsoft1-00000127", "") in new stack
  == Spawn extension (default, 99909*1***DID, 3) exited non-zero on 'SIP/Netsoft1-00000127'
    -- Executing [h@default:1] DeadAGI("SIP/Netsoft1-00000127", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
Scheduling destruction of SIP dialog '285e20b3-6d7a-1237-ac90-e41f137b4e08' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:xxx.carrierIP.xxx;lr;ftag=g82ZSmBe7vDyH;did=b81.df76fc11> for address/port to send to
set_destination: set destination to xxx.carrierIP.xxx, port 5060
Reliably Transmitting (NAT) to xxx.carrierIP.xxx:5060:
BYE sip:mod_sofia@xxx.carrierIP2(RTP).xxx:5080 SIP/2.0
Via: SIP/2.0/UDP xxx.ourIP.xxx:5060;branch=z9hG4bK5d745309;rport
Route: <sip:xxx.carrierIP.xxx;lr;ftag=g82ZSmBe7vDyH;did=b81.df76fc11>
From: <sip:385XXXXXXXX@xxx.carrierIP.xxx>;tag=as4c552568
To: "00387XXXXXXX" <sip:00387XXXXXXX@xxx.carrierIP2(RTP).xxx>;tag=g82ZSmBe7vDyH
Call-ID: 285e20b3-6d7a-1237-ac90-e41f137b4e08
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Nov 28 08:28:42] ERROR[12628]: cdr_custom.c:138 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : No such file or directory

<--- SIP read from xxx.carrierIP.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.ourIP.xxx:5060;received=xxx.ourIP.xxx;branch=z9hG4bK5d745309;rport=5060
From: <sip:385XXXXXXXX@xxx.carrierIP.xxx>;tag=as4c552568
To: "00387XXXXXXX" <sip:00387XXXXXXX@xxx.carrierIP2(RTP).xxx>;tag=g82ZSmBe7vDyH
Call-ID: 285e20b3-6d7a-1237-ac90-e41f137b4e08
CSeq: 102 BYE
User-Agent: FreeSWITCH-mod_sofia/1.6.17-34-0fc0946~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '285e20b3-6d7a-1237-ac90-e41f137b4e08' Method: ACK
Reliably Transmitting (NAT) to xxx.carrierIP.xxx:5060:
OPTIONS sip:xxx.carrierIP.xxx;cpd=on SIP/2.0
Via: SIP/2.0/UDP xxx.ourIP.xxx:5060;branch=z9hG4bK60e47378;rport
From: "asterisk" <sip:asterisk@xxx.ourIP.xxx>;tag=as7095d392
To: <sip:xxx.carrierIP.xxx;cpd=on>
Contact: <sip:asterisk@xxx.ourIP.xxx>
Call-ID: 402e5fa423a3ba8e2a3718406e77ae17@xxx.ourIP.xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 28 Nov 2018 07:28:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---

<--- SIP read from xxx.carrierIP.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.ourIP.xxx:5060;received=xxx.ourIP.xxx;branch=z9hG4bK60e47378;rport=5060
From: "asterisk" <sip:asterisk@xxx.ourIP.xxx>;tag=as7095d392
To: <sip:xxx.carrierIP.xxx;cpd=on>;tag=bf07a8cd232250793f0c37fff9a4b265.b882
Call-ID: 402e5fa423a3ba8e2a3718406e77ae17@xxx.ourIP.xxx
CSeq: 102 OPTIONS
Server: OpenSIPS (2.1.1 (x86_64/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '402e5fa423a3ba8e2a3718406e77ae17@xxx.ourIP.xxx' Method: OPTIONS
Reliably Transmitting (NAT) to 213.91.96.127:43544:
OPTIONS sip:8001@213.91.96.127:43544;rinstance=cfd093b8bc4fdd90;transport=UDP;cpd=on SIP/2.0
Via: SIP/2.0/UDP xxx.ourIP.xxx:5060;branch=z9hG4bK6af6e27b;rport
From: "asterisk" <sip:asterisk@xxx.ourIP.xxx>;tag=as595c5625
To: <sip:8001@213.91.96.127:43544;rinstance=cfd093b8bc4fdd90;transport=UDP;cpd=on>
Contact: <sip:asterisk@xxx.ourIP.xxx>
Call-ID: 4f384a263678f23656c3adbc5e9bdc8d@xxx.ourIP.xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
ate: Wed, 28 Nov 2018 07:28:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---

<--- SIP read from 213.91.96.127:43544 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.ourIP.xxx:5060;branch=z9hG4bK6af6e27b;rport=5060
Contact: <sip:213.91.96.127:43544>
To: <sip:8001@213.91.96.127:43544;rinstance=cfd093b8bc4fdd90;transport=UDP;cpd=on>;tag=38517830
From: "asterisk" <sip:asterisk@xxx.ourIP.xxx>;tag=as595c5625
Call-ID: 4f384a263678f23656c3adbc5e9bdc8d@xxx.ourIP.xxx
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.15.40006 rv2.8.20
Allow-Events: presence, kpml, talk
Content-Length: 0


385XXXXXXXX - DID
00387XXXXXXX - Calling number (my number)

Carrier told us that we should send the calls in next format 385 XXX XXX XXX, and no I didn't try creating DID's both ways, will try it out.

Seems like it's dying during the inbound agi script. Two possibilities:

* Shortcut: Verify that the Route on the DID is "ingroup" and that there is an ingroup selected for the DID (from the dropdown in modify DID) and that this ingroup actually exists (yep: deleting the ingroup will cause the call to fail with Zero CLI information).
* Long Version: Troubleshoot the ingroup script.

Yap, the route on the DID is "ingroup" and the right group is selected, and in the campaign allow inbound and closers is enabled. I guess the route is correct cause the traffic report for the DID shows that calls have been made to the DID, and also the report for the ingroup shows that the calls were routed to it.

So i guess that the inbound script is not doing what it's supposed to do. I looked it up a little and can somewhat understand most of the script, but I didn't edit it previously but it worked couple of months ago (i mean the inbound calls worked). Can I change the script without doing any more damage to the server, that is could I find the original version of the script somewhere and paste it in?

Code: Select all

ll /usr/src/astguiclient/trunk/agi/

or

Code: Select all

locate agi-VDAD_ALL_inbound.agi

and my personal favorite:

Code: Select all

cp agi-VDAD_ALL_inbound.agi agi-VDAD_ALL_inbound.agi.ORIG

Save a new version each time you change it. If all you're doing is adding new debug lines, you should be ok. But I guarantee you'll break it at least once. Don't feel bad. We all miss a semicolon from time to time or put a new command between the "if" and the stuff being executed by the If and break the If.

If you use an editor (netbeans, etc) that allows editing on your workstation with automatic upload, be sure the execute bit is on after each reupload. (chmod +x filename or "preserve settings" or similar in some editors)

Update, still haven't found the way to resolve this issue, it seems the inbound calls are only routed correctly when routed from DID to EXTENSION. Routing to IN-GROUPS and to AGENTs is not working on any DID. If anyone has experienced a problem like this or read somewhere a solution I would deeply appreciate it. Best regards

rickytrix wrote:Update, still haven't found the way to resolve this issue, it seems the inbound calls are only routed correctly when routed from DID to EXTENSION. Routing to IN-GROUPS and to AGENTs is not working on any DID. If anyone has experienced a problem like this or read somewhere a solution I would deeply appreciate it. Best regards

Update on the problem, it seems that the only DID that works is the default one. Tried changing the default DID to point to an ingroup and it does so, but the results are same as before. When the call is made it just answers and ends (and the caller is charged for calling).

You likely have two problems stemming from using a funky carrier.

1) Identifying the DID during an inbound call can be a little tricky. Some carries merely provide the 10 digit number as the extension dialed during the inbound call to identify the DID. Others will add the country code (1 in the US) in front of the DID, and others will also add a "+" so it may be "+" then "country code" then actual phone number. Strange carriers will even be so non-standard that the DID is in a SIP header instead (such as DNID). So the first thing to do is find the DID and forward the call to 'DID@trunkinbound' so the Vicidial system gets the DID and can run with it.

2) Next up is to find why the call terminates at the moment it terminates. A single CLI output with sip debug to determine WHO kills the call is the first step.