No Voice between 2 Servers

[oonyxx]

Hi,
i have created a trunk between two servers in remote locations so i can transfer calls. The calls, voice, everything was working perfectly until two days ago. No changes i have made to the servers or whatsoever, but suddenly the calls can still transfer but there is no voice at all (!). I rechecked again the trunks on both sides and i checked also the NAT rules between the two IP's, but no luck. I also contacted both ISP's to make sure that they have not blocked any port. It is the weirdest problem i have faced until now. For your information i am sending you the configurations on both servers.

Server 1 - Sending Calls Trunk Configuration

[transferbr]
disallow=all
allow=ulaw,alaw
type=friend
host=<IP of the Receiving Server>
dtmfmode=auto
insecure=port,invite
canreinvite=no
nat=yes
context=trunkinbound

Global String: transfer1 = SIP/transferbr

Dialplan Entry:
exten => _778,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _778,2,Dial(SIP/transferbr/778,,tTo)
exten => _778,3,Hangup

Server 2 - Receiving Calls Trunk Configuration

[receivesip]
disallow=all
allow=ulaw,alaw
type=peer
host=<IP of the server Sending Calls>
dtmfmode=auto
canreinvite=no
insecure=port,invite
nat=yes
context=default

Dialplan Entry:
exten => _778,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _778,2,Dial(SIP/cc101,,tTo)
exten => _778,3,Hangup

I can see the trunk listed in "sip show peers" in both servers and i can also transfer the call but neither of both parts can hear any voice.

Any suggestions of how to determine what the problem is or what is causing it?

PS: on both routers i have NATed port range 5000-20000 for the link between the two servers.


Thanks in Advance!

[Noah]

Question:
Are these servers "inside" with LAN ip or publicly addressed boxes?
Are the public IP's static?
Do you have externip set in sip.conf for each box?
Have you run a sip set debub ip yet xxx.xxx.xxx.xxx on either server?

[williamconley]

Hi,
i have created a trunk between two servers in remote locations so i can transfer calls. The calls, voice, everything was working perfectly until two days ago. No changes i have made to the servers or whatsoever, but suddenly the calls can still transfer but there is no voice at all (!). I rechecked again the trunks on both sides and i checked also the NAT rules between the two IP's, but no luck. I also contacted both ISP's to make sure that they have not blocked any port. It is the weirdest problem i have faced until now. For your information i am sending you the configurations on both servers.

...

I can see the trunk listed in "sip show peers" in both servers and i can also transfer the call but neither of both parts can hear any voice.

Any suggestions of how to determine what the problem is or what is causing it?

PS: on both routers i have NATed port range 5000-20000 for the link between the two servers.

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method (7.X.X?) and vicidial version with build (VERSION: 2.X-XXXx ... BUILD: #####-####).

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "manual/from scratch" you must post your operating system with version (and the .iso version from which you installed your original operating system) plus a link to the installation instructions you used. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) Port range is 10000-25000 not 5000-20000. FYI

4) Given the odds, you have suffered a router update/configuration change. This may have nothing to do with the Vicidial servers themselves.

5) Some routers even have 'DOS Protection' algorithms that are meant to block attackers automatically ... but they can be set off by SIP service due to it's constantly changing port requirements.

6) Remember that, technically, SIP can not navigate through two routers. The protocol is old and was not designed to survive traveling through two routers. The externip=XXXX value in sip.conf is an example of the oddity of this protocol (and often contributes to "one-way sound"). If your public IP at either end changed, externip could well be your problem. But if someone changed out the physical router, but the new router approaches the SIP algorithm different ... *poof* no sound. Happens a lot. Some work better with it off, some with it on, and some just don't have it.

7) You should whitelist protect your Vicidial servers and then put each on a public IP with permission only to talk to Carriers and each other. This removes NAT concerns completely and takes those extra "could be interfering" routers out of the pathway completely.