vicidial.org

I want to configure to connect to my Provider

But they use SIP Authentication: don't use username/password

This is my configure:

String: register => username:password@66.228.113.152:5060

[VoipInvite]
disallow=all
allow=g729
allow=gsm
allow=ulaw
type=friend
host=66.228.113.152
dtmfmode=rfc2833
qualify=yes
insecure=very
canreinvite=no

Golobal string: SIPgoautodial = SIP/goautodial

Configure above is correct ?

change this
Golobal string: SIPgoautodial = SIP/goautodial

to
Golobal string: SIPgoautodial = SIP/VoipInvite

My New Confgiure:

[VoipInvite]
disallow=all
allow=g729
allow=gsm
allow=ulaw
type=friend
host=66.228.113.152
dtmfmode=rfc2833
qualify=yes
insecure=very
canreinvite=no


SIPVoipInvite= SIP/VoipInvite

exten => _1XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1XXXXXXXXXX,2,Dial(${SIPVoipInvite}/${EXTEN:1},,tTo)
exten => _1XXXXXXXXXX,3,Hangup

above is correct ? And I need Registration String ? because We use IP Authentication (don't use username and Password) and how to test ?

I had the same situation where registration string (username/password) was not used. So don't put anything there because it can mess things up when connecting to the provider. That happened in my case.
If sip show peers in asterisk console shows that the sip connection to your provider is "Ok", then it should work.

And when I am calling out my dialplan is like

exten => _yourcallpattern.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _yourcallpattern.,n,Dial(SIP/${EXTEN}@hostip:5060)
exten => _yourcallpattern.,n,Hangup

Hope this helps.

rsaaris wrote:I had the same situation where registration string (username/password) was not used. So don't put anything there because it can mess things up when connecting to the provider. That happened in my case.
If sip show peers in asterisk console shows that the sip connection to your provider is "Ok", then it should work.

And when I am calling out my dialplan is like

exten => _yourcallpattern.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _yourcallpattern.,n,Dial(SIP/${EXTEN}@hostip:5060)
exten => _yourcallpattern.,n,Hangup

Hope this helps.

Thank for help

means that I don't need to setting: Registration String, Globals String, Account Entry ?
I only fill out dialplan the same above ?

I try to type sip show peers but I don't see the Sip connection that I regerdit on Vicidial

dinhthuc wrote:
Thank for help

means that I don't need to setting: Registration String, Globals String, Account Entry ?
I only fill out dialplan the same above ?

I try to type sip show peers but I don't see the Sip connection that I regerdit on Vicidial

Account entry and dialplan entry should be all you need If carrier uses ip based authentication with sip trunk. One good tool is to use asterisk console command sip set debug ip hostip:port. And to contact your carrier and ask if they see any activity in their end.

Registration string can be blank for IP peering.
You still need account entry, globals string and dialplan entry.

Make sure you have active = Y and wait a few minutes for the configuration in mysql to be pushed to asterisk. You should see in your sip-vicidial.conf, extensions-vicidial.conf the new entries after approx a minute.

I have a main PBX that is connected to the carrier and from there the Vicidial is connected with SIP using IP authentication just as you describe. Each end knows the IP of the other end and there is no registration or username. I just finished upgrading from ver 2.0.9 to ver 2.2.1 and all I did was copy over the mysql database and then run the upgrade process. The SIP trunk settings are carried over with no hassle. Actually the upgrade process is very easy, much less pain than was initially anticipiated.

I connected directly from my office(vicidial server) to my provider via optical cable. On my side I have a optical converter and connect my lan cable directly from optical converter to vicidial server in NIC eth0. (this NIC is only for telephony i.e sip trunk, another nic eth1 is for my LAN i.e for my agents)
Configuration of eth0 is

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IP: 217.75.206.97
SUB: 255.255.255.192
G: 217.75.206.65

This is my sip carrier configuration:
CARRIER ID

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logosoft

ACCOUNT ENTRY

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[logosoft]
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=g729
type=peer
host=217.75.205.49
dtmfmode=rfc2833
qualify=yes
insecure=very
canreinvite=no


GLOBAL STRING

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SIPlogosoft = SIP/logosoft


DIALPLAN ENTRY

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exten => _9062723254,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9062723254,2,Dial(SIP/${EXTEN:1}@logosoft,,tTo)
exten => _9062723254,3,Hangup


"sip show peers" show me:

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go*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status

8002/8002                  192.168.1.6      D   N      19243    OK (110 ms)
8001/8001                  192.168.1.7      D   N      13266    OK (104 ms)
logosoft                   217.75.205.49        N      5060     UNREACHABLE
21 sip peers [Monitored: 2 online, 19 offline Unmonitored: 0 online, 0 offline]


sip debug show me:

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[Feb 15 16:54:14] WARNING[4697]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0x844ce54 (len 498) to 217.75.205.49:5060 returned -2: Network is unreachable
[Feb 15 16:54:18] Really destroying SIP dialog '4067fd5168c653d7120c84dc7356f14a@127.0.0.1' Method: OPTIONS
[Feb 15 16:54:28] WARNING[4697]: acl.c:541 ast_ouraddrfor: Cannot connect
[Feb 15 16:54:28] Reliably Transmitting (NAT) to 217.75.205.49:5060:
OPTIONS sip:217.75.205.49;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK3ba094c0;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as660bb335
To: <sip:217.75.205.49;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 21df3e4e5a359b72289fd2c7230f5dde@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 15 Feb 2013 21:54:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Feb 15 16:54:28] WARNING[4697]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0x844ce54 (len 498) to 217.75.205.49:5060 returned -2: Network is unreachable
[Feb 15 16:54:32] Really destroying SIP dialog '21df3e4e5a359b72289fd2c7230f5dde@127.0.0.1' Method: OPTIONS
go*CLI>


I don't know what could be the problem, please could you give me some suggestions, advice ??

One interesting information, when I change in account entry, qualify=yes to qualify=no I get from sip show peers

go*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status

8002/8002 192.168.1.6 D N 19243 OK (110 ms)
8001/8001 192.168.1.7 D N 13266 OK (104 ms)
logosoft 217.75.205.49 N 5060 Unmonitored
21 sip peers [Monitored: 2 online, 19 offline Unmonitored: 0 online, 0 offline]

and sip debug show me nothing !?

Uhhh. I SOLVED my problem. Problem was about nic A and nic B. Nic A have a two ports , eth0 and eth1 and nic B also have a two ports eth2 and eth3.
First time I connected both lan cables in eth0 and eth1 (i.e I connected all in one nic, nic A) and there was a problem!

Another time (correct way)
I connected one lan cable from router to eth0 - NIC A (it will be uses for my agents), and second cable from optical converter to eth3-NIC B (it will be uses only for sip trunk) and everything is OK now ! Great!