imcomming calls to remote extention not working

mcyclegar

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#1
I have successfully setup asterisk (elastix) and have set up 3 extentions and 1 SIP Trunk.
2 (1001, 1003) are in the same network(local) as asterisk server and 1(1002) is remotely connected.

The 2(1001,1003) extentions are able able to make outgoing calls to each other as well as outside and reveive calls as well.

Calls to remote extention from both the 1001 and 1003 are not going through as well as calls from outside are not terminating on the remote extention.

where as calls from remote extention 1002 is successfully terminating on both 1000 and 1001 extentions as well as outside using the trunk.

In simple plain works all outgoing calls from remote extentions are working and incomming to the remote extention is not.

I tried setting up on different networks with same result.

Server ip is in DMZ hence dont think the issue is with NAT.

Please help me resolve this.

Thanks in advance

Logs
[Jan 16 23:24:54] WARNING[2716] rtp.c: Unable to set TOS to 184
[Jan 16 23:24:54] VERBOSE[19367] logger.c: -- Executing [1002@from-internal:1] Macro("SIP/1003-08b96858", "exten-vm|1002|1002") in new stack
[Jan 16 23:24:54] VERBOSE[19367] logger.c: -- Executing [s@macro-exten-vm:1] Macro("SIP/1003-08b96858", "user-callerid") in new stack
[Jan 16 23:24:54] VERBOSE[19367] logger.c: -- Executing [s@macro-user-callerid:1] NoOp("SIP/1003-08b96858", "user-callerid: device 1003") in new stack

configurations

Posted: Mon Dec 29, 2008 9:21 am Post subject:

--------------------------------------------------------------------------------

Thanks for the update

This is the extensions.conf
http://pastebin.com/m3cdc9958

This is sip.conf
http://pastebin.com/m7ddeb37

sip_nat.conf
nat=yes
externhost=xxxx.gotdns.com
localnet=192.168.1.0/255.255.255.0
externrefresh=120

this is sip_additional.conf
http://pastebin.com/m754a26eb


sip_general_additional.conf
; do not edit this file, this is an auto-generated file by freepbx
; all modifications must be done from the web gui


vmexten=*97
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
limitonpeers=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
 

mcyclegar

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#2
Part of the log which i forgot to paste earlier


[Jan 16 23:24:54] WARNING[19367] rtp.c: Unable to set TOS to 184
[Jan 16 23:24:54] WARNING[19367] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Jan 16 23:24:54] VERBOSE[19367] logger.c: == Everyone is busy/congested at this time (1:0/0/1)
 

dicko

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#3
Your external extension has
qualify=yes
but is it qualified (registered)?

You really shouldn't need the serer in the DMZ as your router/firewall/NAT device will normally handle the connection between the User Agent (phone) and the server after the UA properly registers with the server (that way the client makes the connection not the server, which usually is not a problem with an off the shelf firewall/router box), and I strongly suggest you do just that, but if you feel you need to have it there then . . .

does sip show peers show the external ip of 1002 correctly, if not you might want to replace dynamic with the static ip (this will eventually break if the remote is on a dhcp served client, use dyndns or similar if so)


does sip show peer 1002 show the expected status of that extension (lots of info about the agent but definitely not UNKNOWN) ?

if so then it should work, if not you have a far end network/nat problem, because 1002 can call in but your connection to 1002, is not getting to the ip/port of the device.

if it's still not working, are there any other voip devices in the remote network that might pre-empt connections to 5060? if so move the port to where it doesn't conflict (on both ends)
 

danardf

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#4
Hi.

For information, like me for exemple, I have an box for ADSL connexion, an Linksys router/Access point.
I can, from remote, make some incoming and outgoing calls.

Exemple:
My public IP address: 62.100.110.120
My router: 192.168.1.1
My Elastix server: 192.168.1.20

I enable the map (nat) from router:
5060 (UDP) -> 192.168.1.20
10000-20000 (UDP) -> 192.168.1.20
(No DMZ, as said dicko)

you must configure sip_nat.conf file for exemple like that:
  • externip = 62.100.110.120 (Because I have an public IP address) else
    externhost=xxxx.gotdns.com (If you have an DynDNS)
    localnet = 192.168.1.0/255.255.255.0
    externrefresh=120 (also if you have an DynDNS)

Also, I don't know if you don't need the srvlookup=yes
! :unsure:

And into your extension config:

  • [102]
    type=friend
    secret=*******
    record_out=Adhoc
    record_in=Adhoc
    qualify=yes
    port=5060
    pickupgroup=
    nat=yes
    mailbox=102@default
    host=dynamic
    dtmfmode=rfc2833
    disallow=all
    dial=SIP/102
    context=from-internal
    canreinvite=no
    callgroup=
    callerid=device <102>
    allow=gsm
    allow=alaw
    allow=h261
    allow=h263
    allow=h263p
    accountcode=
    call-limit=50

And with this config, it's work fine. ;)
 

mcyclegar

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#5
Thanks all for the help.

The issue is resolved.

Thanks again for the help.

Mohammed:)
 

dicko

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#6
It's always helpful for others with the same problem to post your final resolution here.
 

Chilling_Silence

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#7
It looks like a NAT issue at the "remote" end...
 

mcyclegar

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#8
Sure i will ..if it helps someone..

My situation was very strange.

I had a actiontek router from Verizon, and even after adding the elastix IP into DMZ it was not working. i had to do port forwarding also.

If i remove one(either port forwarding or dmz) it does not work i had to keep the system in DMZ and also do port forwarding.

To me it looks like this router does not do a TRUE DMZ or i am not sure why this behavior.

Thanks
Mohammed
 

charlesrg

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#9
We had this problem here, and a brief look at the Intrusion Detection system showed that certain packets to our provider were being blocked.
Unblocking it was the solution.
 

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