When a tenant dials out or to other extensions in the same tenant all is fine and dandy. But if Tenant A calls Tenant B there is no audio. All the SIP signaling seems to go fine but no audio.
That said when a Tenant calls a Tenant it goes out to the upstream carrier and back into the server to Tenant B.
Two things, first is there a way to setup MTE so Tenant to Tenant calls never leave the server? If they do leave the server and come back in why would there be no audio?
I do have a SonicWall TZ in front of these servers.
If Server A Tenant A calls to Server B Tenant A the calls is fine. It goes out to the carrier, back in to the other server and everything is fine. Only when calls originating and terminating on the same server is there a problem
-Adam
sonicwall destroys QoS
Erik,
I have captures that show QoS functions just fine on SonicWALLs. I used them for all our customers and both of our DMZs. DSCP values are passed just fine. In addition i have load tested on a fully saturated T1 with a TZ170/180/190 and the inbound/outbound bandwidth management on the firewall and it pulls back the non specific traffic and lets the voice sip/rtp run through just fine.
Only issue i have see with the SonicWALLs is if you use their SIP proxy which is crap.
Michael
Sounds good.
I know IpTables but I have not seen performance hits yet on QOS (I am not running thousands of tenants yet)
I have other servers in the rack which I do want behind the firewall so I will have to plan on re-architecting the rack network to seperate out the sip related equipment
Thanks, I will check on the re-invite option and report back
Still Not Working
Still not getting audio
I have not had problems with the Sonicwall but are there setting I should be checking for?
I have two SIP servers behind this firewall.
MTE & FreePBX
I have no problems when a call from the FreePBX goes to someone on the MTE and the other way around, it is only tenant to tenant within the MTE
HELP please!
-Adam
you have the same problem if
you have the same problem if the freepbx machine calls a did that routes to the freepbx machine?
Still dead
I still can not get tenant to tenant calling to function properly. Can someone out there help?
What version of
What version of Asterisk?
This exact issue started occurring as soon as we upgraded to 1.4.29, we reverted back to 1.4.28 for now until we can research the issue better. Between tenants however, I refer to an OpenSIPS server (our central routing authority basically), which redirects/loops back to the from-outside context again if the DID is on the same cluster.
I wouldn't blame the Sonicwall as long as it is running Enhanced OS (nor should your tenant->tenant traffic leave your server), we deploy hundreds of these as a reseller and the only time they cause major issues are with the old Standard OS which doesn't support increasing UDP timeouts. All new ones are Enhanced OS though.
Assistance??
nocadmin, Any chance we can chat on the phone. This issue is still killing me and today I had a tenant really b*tching abou the problem. I could use a little guidance to resolve this quickly (willing to compensate if needed)
I do run Enhanced OS
Asterisk is actually at 1.4.24
3rd Lane at 6.0.1.78
please give me a call if you would. 413.854.2109
-adam
Try Asterisk 1.4.28, which is
Try Asterisk 1.4.28, which is my current build with tenant to tenant audio working. As soon as we went to 1.4.29 we had the exact same issue.
I had the same no audio issue with an earlier version, but can't recall when it was.
I can't help much as my outgoing dialing is very complex/modified from what Thirdlane provides, we run every outbound call via OpenSIPS first, authenticate with an external billing server, etc so the audio path to Asterisk may not be the same.
suggestion
What is the best way for me to upgrade to 1.4.28 without breaking anything? is there a good howto? I am getting very desperate here.!!
-Adam
if you're that desperate I
if you're that desperate I would think you'd hire some meat to bust through this issue... the back and forth on the forum would take days or weeks. Do you have weeks?
is that an offer
eeman, are you offering? I am trying to find someone who can help. Can you suggest someone?
Curious, any chance this is a ztdummy problem? I noticed when trying to setup a conference last night that the ztdummy is not running. Could that have an effect on my tenant to tenant problem and or conference problem?
Should I switch to Dahdi and if so is there a decent howto out there? How do I upgrade ASterisk to a newer 1.4 version? does thirdlane have a process or is it just a generic asterisk upgrade?
email me at eeman at
email me at eeman at bluegrass dot net
the dummy driver is required for conferences but shouldnt be required just to bridge a call between two tenants. Missing a dummy driver will cause garbled audio.
Fixed!
Turns out the RTP stream was not leaving my network because of something with the carrier. The carrier is now forcing the RTP to follow the call and that has fixed the problem.
Just curious if this makes sense to anyone. The fix makes sense to me, however, why thirdlane (or asterisk) could not deal with routing the RTP stream from A to B internally has me perplexed.
-adam
first huge mistake.. you have a sonicwall in front of your server... its been proven that sonicwalls destroy QoS. I have the packet captures to prove it :) just learn how to use IPtables rules, its more than enough firewall for your PBX
secondly make sure you have disabled reinvites everywhere as thats also been cause of noaudio transfers.