vicidial.org

Hello, i am new on board, after researching and working hard for 2 weeks, i finally managed to install vicidial on Ubuntu. I am aware of vicibox or goautodial. but i need to run it on ubuntu and learn how it works. i am able to create ever thing and even able to log in to vicidal, when i hit submit on agent interface to login to vicidial my zoiper rings, but when i accept the call, it drops and says normal clearing. i have not started to set up my Voip. can that be the problem?

Vicidial 2.8-382c Build 131010-2149 | Asterisk 1.8.10.1 | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

if your zoiper is sip, turn on sip debug for your IP address. if it is set up to use iax, turn on iax debug. the debug will show the handshake and the likely missing codec that kills the call.

happy hunting!

thanks for your quick reply,

i went on with a diff scratch installation, every thing was ok but as i tried to change some configuration,now the Zoiper does not ring upon submission at all, her is my debug in CLI:


<--- SIP read from 78.157.34.182:5060 --->
SIP/2.0 423 Interval Too Brief
From: <sip:9515199@78.157.34.182:5060>;tag=as6a4b55c0
To: <sip:9515199@78.157.34.182:5060>;tag=bdc86509
CSeq: 104 REGISTER
Call-ID: 0e72f70d5f6eb3a6310f80665344771d@127.0.1.1
Via: SIP/2.0/UDP 78.157.34.182:5060;branch=z9hG4bK57360341;received=188.136.156.113;rport=5060
Min-Expires: 300
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
-- Got SIP response 423 "Interval Too Brief" back from 78.157.34.182
Really destroying SIP dialog '0e72f70d5f6eb3a6310f80665344771d@127.0.1.1' Method: REGISTER
Reliably Transmitting (no NAT) to 78.157.34.182:5060:
OPTIONS sip:78.157.34.182;cpd=on SIP/2.0
Via: SIP/2.0/UDP 78.157.34.182:5060;branch=z9hG4bK10a8df1f;rport
From: "asterisk" <sip:asterisk@78.157.34.182>;tag=as01a3bad8
To: <sip:78.157.34.182;cpd=on>
Contact: <sip:asterisk@78.157.34.182>
Call-ID: 1be53c6b2a58337809c6e71e3ad66784@78.157.34.182
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 21 Nov 2013 01:03:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
Retransmitting #1 (no NAT) to 78.157.34.182:5060:
OPTIONS sip:78.157.34.182;cpd=on SIP/2.0
Via: SIP/2.0/UDP 78.157.34.182:5060;branch=z9hG4bK10a8df1f;rport
From: "asterisk" <sip:asterisk@78.157.34.182>;tag=as01a3bad8
To: <sip:78.157.34.182;cpd=on>
Contact: <sip:asterisk@78.157.34.182>
Call-ID: 1be53c6b2a58337809c6e71e3ad66784@78.157.34.182
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 21 Nov 2013 01:03:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
Retransmitting #2 (no NAT) to 78.157.34.182:5060:
OPTIONS sip:78.157.34.182;cpd=on SIP/2.0
Via: SIP/2.0/UDP 78.157.34.182:5060;branch=z9hG4bK10a8df1f;rport
From: "asterisk" <sip:asterisk@78.157.34.182>;tag=as01a3bad8
To: <sip:78.157.34.182;cpd=on>
Contact: <sip:asterisk@78.157.34.182>
Call-ID: 1be53c6b2a58337809c6e71e3ad66784@78.157.34.182
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 21 Nov 2013 01:03:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog '1be53c6b2a58337809c6e71e3ad66784@78.157.34.182' Method: OPTIONS

<--- SIP read from 192.168.0.104:5060 --->

if your phone isn't registered, there will be no output in sip debug because no sip call will be attempted.

thank you william, i had to do my 11th installation from scratch on Ubuntu, now every thing is ok with dialer, but i am trying to create my carrier like this:
register => 9515199:XXXXXXX@78.157.34.182:5060

[trunk-out]
type=friend
secret=XXXXXX
username=9515199
host=78.157.34.182
canreinvite=no
insecure=very
qualify=no
nat=no
context=from-trunk

Global string "i left it empty"

exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(9515199:XXXXXX@78.157.34.182:5060/${EXTEN:2},,tTor)
exten => _X.,3,Hangup

and here is the VLI output:


ubuntu*CLI> sip show registry
Host Username Refresh State Reg.Time
78.157.34.182:5060 9515199 120 Request Sent

ubuntu*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
gs102/gs102 192.168.0.101 D N 5060 OK (4 ms)
trunk-out/9515199 78.157.34.182 5060 Unmonitored

sip debug

<--- SIP read from 192.168.0.101:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK6c05b4a9;rport=5060
Contact: <sip:192.168.0.101:5060>
To: <sip:gs102@192.168.0.101:5060;rinstance=a3299db79a209a3a;transport=UDP;cpd=on>;tag=96844258
From: "asterisk"<sip:asterisk@192.168.0.100>;tag=as5481f338
Call-ID: 096c51b52852bbdb38aa529e0362d4dd@192.168.0.100
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, X-cisco-serviceuri
User-Agent: Zoiper rev.14091
Allow-Events: presence, kpml
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '096c51b52852bbdb38aa529e0362d4dd@192.168.0.100' Method: OPTIONS
[Nov 25 00:56:45] NOTICE[12191]: chan_sip.c:8178 sip_reregister: -- Re-registration for 9515199@78.157.34.182
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 78.157.34.182:5060:
REGISTER sip:78.157.34.182 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK652b0b25;rport
From: <sip:9515199@78.157.34.182>;tag=as1b97a37b
To: <sip:9515199@78.157.34.182>
Call-ID: 3cffed0a3a311215005a786241d540e0@127.0.1.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:s@192.168.0.100>
Event: registration
Content-Length: 0


---

<--- SIP read from 78.157.34.182:5060 --->
SIP/2.0 423 Interval Too Brief
From: <sip:9515199@78.157.34.182:5060>;tag=as1b97a37b
To: <sip:9515199@78.157.34.182:5060>;tag=64acd4f9
CSeq: 102 REGISTER
Call-ID: 3cffed0a3a311215005a786241d540e0@127.0.1.1
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK652b0b25;rport=5060
Min-Expires: 300
Content-Length: 0

please advice how can i register my sip here...

amirmehdi wrote:thank you william, i had to do my 11th installation from scratch on Ubuntu, now every thing is ok with dialer, but i am trying to create my carrier like this:
...
please advice how can i register my sip here...

If this is not related to "Zoiper drops upon vicidial login", perhaps it would be more appropriate to create a new thread?

And before you do, search the forums for creating a carrier and you will find LOTS of entries. Try several of them. We only charge $25 to set up a standard carrier. Find and follow the rules for doing so if you want to do it yourself, don't make it up.

SIP/2.0 423 Interval Too Brief

SIP registration interval is too short. This is controlled on asterisks side in the sip.conf.

Code: Select all

;maxexpiry=3600                 ; Maximum allowed time of incoming registrations
;minexpiry=60                   ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120              ; Default length of incoming/outgoing registration

and in Zoiper it's in the SIP account settings (you need to turn on 'show advanced settings' to see it). It defaults to 3600.