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Cannot connect to SIP account

June 8, 2014, 6:47 pm
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Code: [root@vps2708-cloud ~]# telnet localhost 5060
Trying 127.0.0.1...
telnet: connect to address 127.0.0.1: Connection refused

it seem that sip port is not reachable even from the server itself.
regards.

Statistics : Posted by toure32 • on Sun Jun 08, 2014 6:47 pm • Replies 8 • Views 225
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Cannot connect to SIP account

June 8, 2014, 6:47 pm
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In that case, provide the SIP debug output.

Statistics : Posted by toure32 • on Sun Jun 08, 2014 6:47 pm • Replies 8 • Views 225
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Cannot connect to SIP account

June 8, 2014, 6:47 pm
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My serveur is on internet and My provider has assure me there is no firewall blocking SIP traffic.
regards.

Statistics : Posted by toure32 • on Sun Jun 08, 2014 6:47 pm • Replies 8 • Views 225
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Cannot connect to SIP account

June 8, 2014, 6:47 pm
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Check your firewall and angle the sip debug.

Statistics : Posted by toure32 • on Sun Jun 08, 2014 6:47 pm • Replies 8 • Views 225
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Cannot connect to SIP account

June 8, 2014, 6:47 pm
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Hi,
I've installed asterisk 10.7.0.1 and configure my sip.conf as :
Code: [general]
transport=udp

[friends_internal](!)
type=friend
host=dynamic
context=from-internal
disallow=all
allow=ulaw

[demo-alice](friends_internal)
secret=verysecretpassword ;

[demo-bob](friends_internal)
secret=othersecretpassword ;

and extension.conf as :
Code: [from-internal]
exten=>6001,1,Dial(SIP/demo-alice,20)
exten=>6002,1,Dial(SIP/demo-bob,20)


but I cannot connect to any of these SIP account... There is no out put in the asterisk consol nor in the asterisk log message.
Your help will be appreciate.

Statistics : Posted by toure32 • on Sun Jun 08, 2014 6:47 pm • Replies 8 • Views 225
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Cannot connect to SIP account

June 8, 2014, 6:47 pm
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In that case nothing is getting to port 5060 on the Asterisk machine. You need to check firewalls (including on the machine itself, routers, and the addresses used).

Statistics : Posted by toure32 • on Sun Jun 08, 2014 6:47 pm • Replies 8 • Views 225
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Cannot connect to SIP account

June 8, 2014, 6:47 pm
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Code: asterisk -rvvvvvvvvv
sip set debug on

I try to reconnect the softphone nothing appear in the console and in log message file.
regards.

Statistics : Posted by toure32 • on Sun Jun 08, 2014 6:47 pm • Replies 8 • Views 225
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Cannot connect to SIP account

June 8, 2014, 6:47 pm
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david55 wrote:In that case nothing is getting to port 5060 on the Asterisk machine. You need to check firewalls (including on the machine itself, routers, and the addresses used).


Statistics : Posted by toure32 • on Sun Jun 08, 2014 6:47 pm • Replies 13 • Views 365
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spinach.v0.11, Dynamic Sip Nat without Dyndns.org or similar

June 11, 2014, 9:19 am
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Really true!
Thank you for your reply!
We got spinach.sh v0.11 from Olaf Winkler:

Code: #!/bin/bash
doreload=0
wanip=`curl -q tnx.nl/ip`
if [ -f /tmp/oldip.txt ]; then
   oldip=`cat /tmp/oldip.txt`
   [ "$wanip" != "$oldip" ] && doreload=1
else
   doreload=1
fi
echo $wanip > /tmp/oldip.txt
if [ $doreload -eq 1 ]; then
      sed -i 's/externip=.*/externip='"$wanip"'/' /etc/asterisk/sip.conf
      asterisk -rx 'sip reload'
fi


PS. if TNX.NL is down you still got a few backup sites:
whatismyip.org
icanhazip.com
myip.dnsomatic.com
ip.appspot.com
checkip.dyndns.org:8245
whatismyip.com
jsonip.com

(you may easily run your own "backup site"!!!)

Thank you!

Statistics : Posted by qzu • on Wed Jun 11, 2014 9:19 am • Replies 2 • Views 91
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spinach.v0.11, Dynamic Sip Nat without Dyndns.org or similar

June 11, 2014, 9:19 am
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Nice idea, but at least the script should check whether the IP has changed or not - probably by saving the actual IP to a file. This will avoid non necessary reloads.

Code: #!/bin/bash
doreload=0
wanip=`curl -q tnx.nl/ip`
if [ -f /tmp/oldip.txt ]; then
   oldip=`cat /tmp/oldip.txt`
   [ "$wanip" != "$oldip" ] && doreload=1
else
   doreload=1
fi
echo $wanip > /tmp/oldip.txt
if [ $doreload -eq 1 ]; then
      sed -i 's/externip=.*/externip='"$wanip"'/' /etc/asterisk/sip.conf
      asterisk -rx 'sip reload'
fi


Statistics : Posted by qzu • on Wed Jun 11, 2014 9:19 am • Replies 2 • Views 91
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spinach.v0.11, Dynamic Sip Nat without Dyndns.org or similar

June 11, 2014, 9:19 am
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Hi to all Asterisk enthusiasts

About a month ago, Dyndns.org shut down his free dynamic dns service, disrupting service to hundreds of thousands of home server installations, shame on them!

Many Asterisk setups relied on this to overcome nat problems with dynamically assigned WAN ips (using the externhost=xyz.dyndns.org directive)

Here's a tiny shell script to automatically update your externip directive from sip.conf reflecting your actual WAN ip address:

# nano /root/spinach.sh

Code: wanip=`curl -q tnx.nl/ip`
sed -i 's/externip=.*/externip='"$wanip"'/' /etc/asterisk/sip.conf
asterisk -rx 'sip reload'

# crontab -e

Code: * * * * * /root/http://forums.asterisk.org/spinach.sh

From now on, your sip.conf externip=a.b.c.d directive will be updated every minute.
Any comments would be highly appreciated!

Statistics : Posted by qzu • on Wed Jun 11, 2014 9:19 am • Replies 2 • Views 91
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Asterisk scaling and load balance

June 11, 2014, 9:20 am
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Yes your analysis is correct. The upstream is an wholesale ITSP who does support direct media. In fact I think I read that it was their preferred method.

I agree there is a NAT issue to consider.

Another ways of achieving the end game?

Statistics : Posted by londonnet • on Wed Jun 11, 2014 9:20 am • Replies 6 • Views 132
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Asterisk scaling and load balance

June 11, 2014, 9:20 am
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Assuming you control all the servers up and downstream:

Enabling direct media in sip.conf, and not using any feature that requires media to go through Asterisk, will result in the media bypassing Asterisk if up- and down-stream peers support that.

Using Transfer, rather than Dial (note it takes actual SIP addresses) will also release the signalling path, but Transfer is relatively fragile and not much work has gone into.

If the upstream is an ITSP, it is very unlikely they will support Transfer, and if they do, they will probably charge it the same as passing the call through in tandem. Most ITSP's will not support direct media, and making it work will involve NAT issues, even if they do.

Statistics : Posted by londonnet • on Wed Jun 11, 2014 9:20 am • Replies 6 • Views 132
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Asterisk scaling and load balance

June 11, 2014, 9:20 am
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I need to put people through an IVR first.

So I could do something like this

Server0
exten => s,1,Answer()
same => n,Playback(silence/1)
same => n,Read(route,welcome,1,,1,1)
same => n,ExecIf($["${route}"="1"]?Dial(SIP/server1)
same => n,ExecIf($["${route}"="2"]?Dial(SIP/server2)
same => n,ExecIf($["${route}"="3"]?Dial(SIP/server3)
same => n,Hangup()

I assume the original server would remain in the call path rather than transfer sip and media direct to the next server. Is this correct?

How do we route the call on and drop the original server out of the call?

Statistics : Posted by londonnet • on Wed Jun 11, 2014 9:20 am • Replies 6 • Views 132
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Asterisk scaling and load balance

June 11, 2014, 9:20 am
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You don't seem to be doing load balancing, you seem to be doing call routing. That is a very basic application for Asterisk.

On the other hand, if you have very simple routing requirements, a proxy may be more efficient.

Statistics : Posted by londonnet • on Wed Jun 11, 2014 9:20 am • Replies 6 • Views 132
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Asterisk scaling and load balance

June 11, 2014, 9:20 am
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What I am trying to do is breakup tasks in the asterisk dial plan. A sip proxy wont give me the opportunity to perform any ivr functions before off loading to a que or service on another Asterisk server.

I'm trying to rout on skill not availability

Statistics : Posted by londonnet • on Wed Jun 11, 2014 9:20 am • Replies 6 • Views 132
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Asterisk scaling and load balance

June 11, 2014, 9:20 am
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Use a SIP Proxy

Statistics : Posted by londonnet • on Wed Jun 11, 2014 9:20 am • Replies 6 • Views 132
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Asterisk scaling and load balance

June 11, 2014, 9:20 am
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What I think I would like to do is recive calls into a single asterisk instance.
Answer the call, collect some digits
And then pass on the call and media to 1 of 10 other Asterisk servers depending on the digits recived.

Is this possible?

It is important the call is always sent to the same server based on the digits collected.

An enhancment would be to forward calls to a seconday in the event of a server not being available.

Thanks

Statistics : Posted by londonnet • on Wed Jun 11, 2014 9:20 am • Replies 6 • Views 132
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SIP and SCCP concurrently

June 8, 2014, 4:29 am
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See the first reply. It is normal use of Asterisk.

Statistics : Posted by clear • on Sun Jun 08, 2014 4:29 am • Replies 7 • Views 308
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SIP and SCCP concurrently

June 8, 2014, 4:29 am
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Quote:I want to connect my sccp phones on an asterisk server and i want to create a sip trunk with 3CX PBX. In this case the sccp phones from Asterisk and sip phones from 3CX will communicate each other?


So.. Can i do this or it's not possible? Image

Statistics : Posted by clear • on Sun Jun 08, 2014 4:29 am • Replies 7 • Views 308
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