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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: compsos on March 02, 2014, 11:55:33 AM
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Hi S
We applied 4.0.0-115 and found we could not make international calls. Whilst trying to trace that issue found
sip show peers had a strange line
sipsak 192.168.100.1 5060 Unmonitored
We found the sark_sip_main.conf had the IP as host=192.168.100.1 as the first line. Removing that still left the sipsak line but with no ip address.
Logs did not show any entries of sipsak or 192.168.100.1 or did the internet. Any ideas what sipsak is?
Removing 4.0.0-115 and going back to 4.0.0-103 and the issues including the international calls issue disappeared.
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Hi there
Without a SIP trace, I can't say what the international call issue you mention might be but I think it is unlikely to have anything to do with the sipsak trunk entry. The sipsak entry was added in 109/110 for Pacemaker/crm. The Asterisk resource agent in Pacemaker uses sipsak to "ping" asterisk with a SIP OPTIONS request every few seconds to see if it is up and responding to sip requests. The trunk is not reported by the SAIL front-end, it just acts as an anchor point for the sipsak requests. What you should see in sark_sip_main is something like this
[sipsak]
type=peer
host={local host ip}
The host ip would normally be the local ip of the asterisk box (unless you are running Pacemaker, in which case it would be the Virtual IP, which you will find in globals=>haclusterip). If it isn't showing up as the local ip, it may be that the box is not using eth0. In either case it should be benign. The actual entry is generated at line 88 in the sip generator (/opt/sark/generator/sipSQL.php) so you can comment it out if you like. On reflection, perhaps we should suppress it for non Pacemaker installs.
If you'd like me to look at the International call issue then I'll need a console log of a failing call and a SIP trace.
You can always see what changes we've made in any new release by looking at the changelog
rpm -q sail --changelog | less
Kind Regards
S
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Hi S
In the sark_sip_main we only got
host={local host ip}
So that code only partially worked. I did a grep on all /etc/asterisk files looking for sipsak and return was empty only the IP address returned anything.
As for the call problem, even sip debug showed no activity when attempting to call. Setting the debug to the peer confirmed no activity. But resetting back to 103 and no other changes.
worked.
My first thought on the unmonitored peer entry with a foreign IP address was a hack attempt and was trying to get it out as quickly as possible so have no more details on what was blocking calls.
It may have something to do with 0011 for international lines in the route.