vicidial.org

Ok, maybe I am missing the simple things....distracted or both. Installed new server, migrated database and all is working.....outbound calls auto and manual no issues. Never has a DID issue before.

Requested 2 new DIDs from my provider, configured the DIDs in webpage and pointed them to PHONE and created inbound carrier. The SIP phone connects, can call other extensions no issues. However when try to call the DID number, call fails, plays ss-noservice and hangs up. Have never had an issue setting up DIDs, so I must be missing something thing simple. I did notice in the sip debug a BAD EVENT 489 but cant trace it and 401 unauthorized. The BAD EVENT I think is the Zoiper softphone sending RTP, atleast that is what i found on google.

Even tried to turn off IPtables but still same result.

CARRIER
[CommPeakInBound]
disallow=all
allow=ulaw
allow=gsm
type=peer
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=port,invite
nat=force_rport,comedia
host=uswest.sip.commpeak.com

Asterisk CLI

[Dec 28 23:56:57] -- Executing [13525778992@trunkinbound:1] AGI("SIP/CommPeakInBound-00000005", "agi-DID_route.agi") in new stack
[Dec 28 23:56:57] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Dec 28 23:56:57] -- <SIP/CommPeakInBound-00000005>AGI Script agi-DID_route.agi completed, returning 0
[Dec 28 23:56:57] -- Executing [9998811112@default:1] Wait("SIP/CommPeakInBound-00000005", "2") in new stack
[Dec 28 23:56:59] -- Executing [9998811112@default:2] Answer("SIP/CommPeakInBound-00000005", "") in new stack
[Dec 28 23:57:00] > 0x7efff8024180 -- Probation passed - setting RTP source address to 45.79.75.243:31522
[Dec 28 23:57:00] -- Executing [9998811112@default:3] Playback("SIP/CommPeakInBound-00000005", "ss-noservice") in new stack
[Dec 28 23:57:00] -- <SIP/CommPeakInBound-00000005> Playing 'ss-noservice.gsm' (language 'en')
[Dec 28 23:57:02] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 28 23:57:02] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 28 23:57:02] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 28 23:57:03] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 28 23:57:05] -- Executing [9998811112@default:4] Playback("SIP/CommPeakInBound-00000005", "vm-goodbye") in new stack
[Dec 28 23:57:05] -- <SIP/CommPeakInBound-00000005> Playing 'vm-goodbye.gsm' (language 'en')
[Dec 28 23:57:05] -- Executing [9998811112@default:5] Hangup("SIP/CommPeakInBound-00000005", "") in new stack
[Dec 28 23:57:05] == Spawn extension (default, 9998811112, 5) exited non-zero on 'SIP/CommPeakInBound-00000005'
[Dec 28 23:57:05] -- Executing [h@default:1] AGI("SIP/CommPeakInBound-00000005", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Dec 28 23:57:05] -- <SIP/CommPeakInBound-00000005>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

SIP debug
[Dec 29 00:26:18] -- Executing [13525778992@trunkinbound:1] AGI("SIP/CommPeakInBound-00000006", "agi-DID_route.agi") in new stack
[Dec 29 00:26:18] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Dec 29 00:26:19] -- <SIP/CommPeakInBound-00000006>AGI Script agi-DID_route.agi completed, returning 0
[Dec 29 00:26:19] -- Executing [9998811112@default:1] Wait("SIP/CommPeakInBound-00000006", "2") in new stack
[Dec 29 00:26:21] -- Executing [9998811112@default:2] Answer("SIP/CommPeakInBound-00000006", "") in new stack
[Dec 29 00:26:21] Audio is at 12814
[Dec 29 00:26:21] Adding codec 100003 (ulaw) to SDP
[Dec 29 00:26:21] Adding non-codec 0x1 (telephone-event) to SDP
[Dec 29 00:26:21]
[Dec 29 00:26:21] <--- Reliably Transmitting (NAT) to 45.79.73.196:5060 --->
[Dec 29 00:26:21] SIP/2.0 200 OK
[Dec 29 00:26:21] Via: SIP/2.0/UDP 45.79.73.196:5060;branch=z9hG4bK284a.21cf0c86.0;received=45.79.73.196;rport=5060
[Dec 29 00:26:21] From: "anonymous" <sip:anonymous@did.commpeak.net>;tag=Z2HUF242jU7rp
[Dec 29 00:26:21] To: <sip:13525778992@sip.commpeak.net>;tag=as564432c3
[Dec 29 00:26:21] Call-ID: 1c44aa1d-85cd-1237-cc9d-da94cfe4b10e
[Dec 29 00:26:21] CSeq: 132694525 INVITE
[Dec 29 00:26:21] Server: Asterisk PBX 11.22.0-vici
[Dec 29 00:26:21] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Dec 29 00:26:21] Supported: replaces, timer
[Dec 29 00:26:21] Contact: <sip:13525778992@192.99.55.242:5060>
[Dec 29 00:26:21] Content-Type: application/sdp
[Dec 29 00:26:21] Content-Length: 241
[Dec 29 00:26:21]
[Dec 29 00:26:21] v=0
[Dec 29 00:26:21] o=root 951440689 951440689 IN IP4 192.99.55.242
[Dec 29 00:26:21] s=Asterisk PBX 11.22.0-vici
[Dec 29 00:26:21] c=IN IP4 192.99.55.242
[Dec 29 00:26:21] t=0 0
[Dec 29 00:26:21] m=audio 12814 RTP/AVP 0 101
[Dec 29 00:26:21] a=rtpmap:0 PCMU/8000
[Dec 29 00:26:21] a=rtpmap:101 telephone-event/8000
[Dec 29 00:26:21] a=fmtp:101 0-16
[Dec 29 00:26:21] a=ptime:20
[Dec 29 00:26:21] a=sendrecv
[Dec 29 00:26:21]
[Dec 29 00:26:21] <------------>
[Dec 29 00:26:21]
[Dec 29 00:26:21] <--- SIP read from UDP:45.79.73.196:5060 --->
[Dec 29 00:26:21] ACK sip:13525778992@192.99.55.242:5060 SIP/2.0
[Dec 29 00:26:21] Via: SIP/2.0/UDP 45.79.73.196:5060;branch=z9hG4bK284a.21cf0c86.2
[Dec 29 00:26:21] Max-Forwards: 69
[Dec 29 00:26:21] From: "anonymous" <sip:anonymous@did.commpeak.net>;tag=Z2HUF242jU7rp
[Dec 29 00:26:21] To: <sip:13525778992@sip.commpeak.net>;tag=as564432c3
[Dec 29 00:26:21] Call-ID: 1c44aa1d-85cd-1237-cc9d-da94cfe4b10e
[Dec 29 00:26:21] CSeq: 132694525 ACK
[Dec 29 00:26:21] Contact: <sip:commpeak@45.79.73.196;did=02d.9a56a7a2>
[Dec 29 00:26:21] Content-Length: 0
[Dec 29 00:26:21]
[Dec 29 00:26:21] <------------->
[Dec 29 00:26:21] --- (9 headers 0 lines) ---
[Dec 29 00:26:21] > 0x7efff801cb20 -- Probation passed - setting RTP source address to 45.79.89.12:29800
[Dec 29 00:26:21] -- Executing [9998811112@default:3] Playback("SIP/CommPeakInBound-00000006", "ss-noservice") in new stack
[Dec 29 00:26:21] -- <SIP/CommPeakInBound-00000006> Playing 'ss-noservice.gsm' (language 'en')
[Dec 29 00:26:22]
[Dec 29 00:26:22] <--- SIP read from UDP:120.29.100.71:56075 --->
[Dec 29 00:26:22] PUBLISH sip:9176@symphotel.ngtechnologies.net;transport=UDP SIP/2.0
[Dec 29 00:26:22] Via: SIP/2.0/UDP 120.29.100.71:56075;branch=z9hG4bK-524287-1---959b9da6f278ab5c
[Dec 29 00:26:22] Max-Forwards: 70
[Dec 29 00:26:22] Contact: <sip:9176@120.29.100.71:56075;transport=UDP>
[Dec 29 00:26:22] To: <sip:9176@symphotel.ngtechnologies.net;transport=UDP>
[Dec 29 00:26:22] From: <sip:9176@symphotel.ngtechnologies.net;transport=UDP>;tag=f0687717
[Dec 29 00:26:22] Call-ID: dWMNy-iiK__D8dLdHFsO_Q..
[Dec 29 00:26:22] CSeq: 1 PUBLISH
[Dec 29 00:26:22] Expires: 600
[Dec 29 00:26:22] Content-Type: application/pidf+xml
[Dec 29 00:26:22] User-Agent: Z 3.15.40006 rv2.8.20
[Dec 29 00:26:22] Event: presence
[Dec 29 00:26:22] Allow-Events: presence, kpml, talk
[Dec 29 00:26:22] Content-Length: 282
[Dec 29 00:26:22]
[Dec 29 00:26:22] <?xml version="1.0" encoding="UTF-8"?>
[Dec 29 00:26:22] <presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:9176@symphotel.ngtechnologies.net;transport=UDP"> <tuple id="9176" > <status><basic>open</basic></status> <note>On the phone</note> </tuple>
[Dec 29 00:26:22] </presence>
[Dec 29 00:26:22] <------------->
[Dec 29 00:26:22] --- (14 headers 3 lines) ---
[Dec 29 00:26:22] Sending to 120.29.100.71:56075 (NAT)
[Dec 29 00:26:22]
[Dec 29 00:26:22] <--- Transmitting (NAT) to 120.29.100.71:56075 --->
[Dec 29 00:26:22] SIP/2.0 489 Bad Event
[Dec 29 00:26:22] Via: SIP/2.0/UDP 120.29.100.71:56075;branch=z9hG4bK-524287-1---959b9da6f278ab5c;received=120.29.100.71;rport=56075
[Dec 29 00:26:22] From: <sip:9176@symphotel.ngtechnologies.net;transport=UDP>;tag=f0687717
[Dec 29 00:26:22] To: <sip:9176@symphotel.ngtechnologies.net;transport=UDP>;tag=as4cb29256
[Dec 29 00:26:22] Call-ID: dWMNy-iiK__D8dLdHFsO_Q..
[Dec 29 00:26:22] CSeq: 1 PUBLISH
[Dec 29 00:26:22] Server: Asterisk PBX 11.22.0-vici
[Dec 29 00:26:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Dec 29 00:26:22] Supported: replaces, timer
[Dec 29 00:26:22] Content-Length: 0
[Dec 29 00:26:22]
[Dec 29 00:26:22]
[Dec 29 00:26:22] <------------>
[Dec 29 00:26:22] Really destroying SIP dialog 'dWMNy-iiK__D8dLdHFsO_Q..' Method: PUBLISH
[Dec 29 00:26:22]
[Dec 29 00:26:22] <--- SIP read from UDP:120.29.100.71:56075 --->
[Dec 29 00:26:22] SUBSCRIBE sip:9176@symphotel.ngtechnologies.net;transport=UDP SIP/2.0
[Dec 29 00:26:22] Via: SIP/2.0/UDP 120.29.100.71:56075;branch=z9hG4bK-524287-1---a03293d1b04c14d8
[Dec 29 00:26:22] Max-Forwards: 70
[Dec 29 00:26:22] Contact: <sip:9176@120.29.100.71:56075;transport=UDP>
[Dec 29 00:26:22] To: <sip:9176@symphotel.ngtechnologies.net;transport=UDP>
[Dec 29 00:26:22] From: <sip:9176@symphotel.ngtechnologies.net;transport=UDP>;tag=fa5eea78
[Dec 29 00:26:22] Call-ID: y9O_Tt_JIUkN9y0DjPKVZw..
[Dec 29 00:26:22] CSeq: 1 SUBSCRIBE
[Dec 29 00:26:22] Expires: 600
[Dec 29 00:26:22] Accept: application/watcherinfo+xml
[Dec 29 00:26:22] User-Agent: Z 3.15.40006 rv2.8.20
[Dec 29 00:26:22] Event: presence.winfo
[Dec 29 00:26:22] Allow-Events: presence, kpml, talk
[Dec 29 00:26:22] Content-Length: 0
[Dec 29 00:26:22]
[Dec 29 00:26:22] <------------->
[Dec 29 00:26:22] --- (14 headers 0 lines) ---
[Dec 29 00:26:22] Sending to 120.29.100.71:56075 (NAT)
[Dec 29 00:26:22] Creating new subscription
[Dec 29 00:26:22] Sending to 120.29.100.71:56075 (NAT)
[Dec 29 00:26:22] list_route: hop: <sip:9176@120.29.100.71:56075;transport=UDP>
[Dec 29 00:26:22] Found peer '9176' for '9176' from 120.29.100.71:56075
[Dec 29 00:26:22]
[Dec 29 00:26:22] <--- Transmitting (NAT) to 120.29.100.71:56075 --->
[Dec 29 00:26:22] SIP/2.0 401 Unauthorized
[Dec 29 00:26:22] Via: SIP/2.0/UDP 120.29.100.71:56075;branch=z9hG4bK-524287-1---a03293d1b04c14d8;received=120.29.100.71;rport=56075
[Dec 29 00:26:22] From: <sip:9176@symphotel.ngtechnologies.net;transport=UDP>;tag=fa5eea78
[Dec 29 00:26:22] To: <sip:9176@symphotel.ngtechnologies.net;transport=UDP>;tag=as6b3282fa
[Dec 29 00:26:22] Call-ID: y9O_Tt_JIUkN9y0DjPKVZw..
[Dec 29 00:26:22] CSeq: 1 SUBSCRIBE
[Dec 29 00:26:22] Server: Asterisk PBX 11.22.0-vici
[Dec 29 00:26:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Dec 29 00:26:22] Supported: replaces, timer
[Dec 29 00:26:22] WWW-Authenticate: Digest algorithm=MD5, realm="symphotel.ngtechnologies.net", nonce="4121b65b"
[Dec 29 00:26:22] Content-Length: 0
[Dec 29 00:26:22]
[Dec 29 00:26:22]
[Dec 29 00:26:22] <------------>
[Dec 29 00:26:22] Scheduling destruction of SIP dialog 'y9O_Tt_JIUkN9y0DjPKVZw..' in 15360 ms (Method: SUBSCRIBE)
[Dec 29 00:26:22]
[Dec 29 00:26:22] <--- SIP read from UDP:120.29.100.71:56075 --->
[Dec 29 00:26:22] SUBSCRIBE sip:9176@symphotel.ngtechnologies.net;transport=UDP SIP/2.0
[Dec 29 00:26:22] Via: SIP/2.0/UDP 120.29.100.71:56075;branch=z9hG4bK-524287-1---d74cb4f64409f73b
[Dec 29 00:26:22] Max-Forwards: 70
[Dec 29 00:26:22] Contact: <sip:9176@120.29.100.71:56075;transport=UDP>
[Dec 29 00:26:22] To: <sip:9176@symphotel.ngtechnologies.net;transport=UDP>
[Dec 29 00:26:22] From: <sip:9176@symphotel.ngtechnologies.net;transport=UDP>;tag=fa5eea78
[Dec 29 00:26:22] Call-ID: y9O_Tt_JIUkN9y0DjPKVZw..
[Dec 29 00:26:22] CSeq: 2 SUBSCRIBE
[Dec 29 00:26:22] Expires: 600
[Dec 29 00:26:22] Accept: application/watcherinfo+xml
[Dec 29 00:26:22] User-Agent: Z 3.15.40006 rv2.8.20
[Dec 29 00:26:22] Authorization: Digest username="9176",realm="symphotel.ngtechnologies.net",nonce="4121b65b",uri="sip:9176@symphotel.ngtechnologies.net;transport=UDP",response="02084f11866778aac20a93201b4bafa8",algorithm=MD5
[Dec 29 00:26:22] Event: presence.winfo
[Dec 29 00:26:22] Allow-Events: presence, kpml, talk
[Dec 29 00:26:22] Content-Length: 0
[Dec 29 00:26:22]
[Dec 29 00:26:22] <------------->
[Dec 29 00:26:22] --- (15 headers 0 lines) ---
[Dec 29 00:26:22] Creating new subscription
[Dec 29 00:26:22] Sending to 120.29.100.71:56075 (NAT)
[Dec 29 00:26:22] Found peer '9176' for '9176' from 120.29.100.71:56075
[Dec 29 00:26:22]
[Dec 29 00:26:22] <--- Transmitting (NAT) to 120.29.100.71:56075 --->
[Dec 29 00:26:22] SIP/2.0 489 Bad Event
[Dec 29 00:26:22] Via: SIP/2.0/UDP 120.29.100.71:56075;branch=z9hG4bK-524287-1---d74cb4f64409f73b;received=120.29.100.71;rport=56075
[Dec 29 00:26:22] From: <sip:9176@symphotel.ngtechnologies.net;transport=UDP>;tag=fa5eea78
[Dec 29 00:26:22] To: <sip:9176@symphotel.ngtechnologies.net;transport=UDP>;tag=as6b3282fa
[Dec 29 00:26:22] Call-ID: y9O_Tt_JIUkN9y0DjPKVZw..
[Dec 29 00:26:22] CSeq: 2 SUBSCRIBE
[Dec 29 00:26:22] Server: Asterisk PBX 11.22.0-vici
[Dec 29 00:26:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Dec 29 00:26:22] Supported: replaces, timer
[Dec 29 00:26:22] Content-Length: 0
[Dec 29 00:26:22]
[Dec 29 00:26:22]
[Dec 29 00:26:22] <------------>
[Dec 29 00:26:22] Really destroying SIP dialog 'y9O_Tt_JIUkN9y0DjPKVZw..' Method: SUBSCRIBE
[Dec 29 00:26:26] -- Executing [9998811112@default:4] Playback("SIP/CommPeakInBound-00000006", "vm-goodbye") in new stack
[Dec 29 00:26:26] -- <SIP/CommPeakInBound-00000006> Playing 'vm-goodbye.gsm' (language 'en')
[Dec 29 00:26:27] -- Executing [9998811112@default:5] Hangup("SIP/CommPeakInBound-00000006", "") in new stack
[Dec 29 00:26:27] == Spawn extension (default, 9998811112, 5) exited non-zero on 'SIP/CommPeakInBound-00000006'
[Dec 29 00:26:27] -- Executing [h@default:1] AGI("SIP/CommPeakInBound-00000006", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Dec 29 00:26:27] -- <SIP/CommPeakInBound-00000006>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Dec 29 00:26:27] Scheduling destruction of SIP dialog '1c44aa1d-85cd-1237-cc9d-da94cfe4b10e' in 6400 ms (Method: ACK)
[Dec 29 00:26:27] set_destination: Parsing <sip:commpeak@45.79.73.196;did=02d.9a56a7a2> for address/port to send to
[Dec 29 00:26:27] set_destination: set destination to 45.79.73.196:5060
[Dec 29 00:26:27] Reliably Transmitting (NAT) to 45.79.73.196:5060:
[Dec 29 00:26:27] BYE sip:commpeak@45.79.73.196;did=02d.9a56a7a2 SIP/2.0
[Dec 29 00:26:27] Via: SIP/2.0/UDP 192.99.55.242:5060;branch=z9hG4bK0adbe943;rport
[Dec 29 00:26:27] Max-Forwards: 70
[Dec 29 00:26:27] From: <sip:13525778992@sip.commpeak.net>;tag=as564432c3
[Dec 29 00:26:27] To: "anonymous" <sip:anonymous@did.commpeak.net>;tag=Z2HUF242jU7rp
[Dec 29 00:26:27] Call-ID: 1c44aa1d-85cd-1237-cc9d-da94cfe4b10e
[Dec 29 00:26:27] CSeq: 102 BYE
[Dec 29 00:26:27] User-Agent: Asterisk PBX 11.22.0-vici
[Dec 29 00:26:27] X-Asterisk-HangupCause: Normal Clearing
[Dec 29 00:26:27] X-Asterisk-HangupCauseCode: 16
[Dec 29 00:26:27] Content-Length: 0


Scratch install Centos6
VERSION: 2.14-695a
BUILD: 181116-1133
SVN Version: 3057

why did you not share the link/instructions for your installation (since you didn't install from an .iso installer, it's quite relevant ... unless you did install from an .iso installer, in which case you left that information out, lol).

13525778992 ... is this the DID you configured? Or did you configure 3525778992?

I wont analyze your Asterisk sip trace as the CLI shows clearly that the to the DID is reaching Asterisk

[Dec 28 23:56:57] -- Executing [13525778992@trunkinbound:1] AGI("SIP/CommPeakInBound-00000005", "agi-DID_route.agi") in new stack

You have the DID 1352577899 set to EXTEN and by Default is 9998811112, no-service. that's why is not working.


Change that on your vicidial DID MENU

williamconley wrote:why did you not share the link/instructions for your installation (since you didn't install from an .iso installer, it's quite relevant ... unless you did install from an .iso installer, in which case you left that information out, lol).

13525778992 ... is this the DID you configured? Or did you configure 3525778992?

The install was based in the old Ray Soloman guide, which i have built 20 servers with no issues before. And 3525778992 was configured in the webpage.


https://ibb.co/6P5PKXD

ambiorixg12 wrote:I wont analyze your Asterisk sip trace as the CLI shows clearly that the to the DID is reaching Asterisk

[Dec 28 23:56:57] -- Executing [13525778992@trunkinbound:1] AGI("SIP/CommPeakInBound-00000005", "agi-DID_route.agi") in new stack

You have the DID 1352577899 set to EXTEN and by Default is 9998811112, no-service. that's why is not working.


Change that on your vicidial DID MENU

Is set to phone, not extension

https://ibb.co/6P5PKXD

your DID reach Asterisk as 1352577899 not like 352577899, so please change it

Watch your logs
[Dec 28 23:56:57] -- Executing [13525778992@trunkinbound:1] AGI("SIP/CommPeakInBound-00000005", "agi-DID_route.agi") in new stack

UPDATE

Ok, saw a post from William earlier and created a new DID using the 13525778992 (included the 1), then changed the DID Route to EXTENSION and replaced 9998811112 with my actual phone 9176....and it works. If I have the DID Route as PHONE is always say non service.

UPDATE UPDATED

Changed DID route back to PHONE and rebooted....now routing works correctly. Now rings however not going to voicemail even after 10 rings

Anyone know how to configure the voicemail to pickup the call after say 30sec ringing? is a setting or config in asterisk missing?

Phones --> select DID route phone --> Phone Ring Timeout --> change it to "30"

Phone Ring Timeout -This is the amount of time, in seconds, that the phone will ring in the dialplan before sending the call to voicemail. Default is 60 seconds.

Don't forget to rebuild your conf files in server settings. Wait a couple of minutes, before your changes take effect.

Thanks so much. Knew i was just missing something simple