FreePBX | Register | Issues | Wiki | Portal | Support

Remote extension audio problems


(Diego Diaz) #1

Hi people,

im havig a real nasty issue , got an extension , a sip extension in another city , where i have the router configurated, all open ports sip and rtp to allow voice trafic, it is registered, but when i call no audio is transmitted, look what i found when i did the call

<— SIP read from UDP:190.248.187.222:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.250.5:5060;branch=z9hG4bK003e9d60;rport=5060;received=116.48.147.188
From: “Grupo Over-17435737” sip:17435737@192.168.250.5;tag=as775bf58f
To: sip:1034@192.168.1.1:5060;tag=819468181
Call-ID: 6e9cda4c5629634b3c60bb7f66afde8d@192.168.250.5:5060
CSeq: 102 INVITE
Contact: sip:1034@192.168.1.1:5060
Supported: replaces, path, timer
User-Agent: Grandstream GXP1405 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 264

v=0
o=1034 8000 8000 IN IP4 192.168.1.1
s=SIP Call
c=IN IP4 192.168.1.1
t=0 0
m=audio 5004 RTP/AVP 0 8 111 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (12 headers 13 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 111
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G726-32 for ID 111
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(ulaw|alaw|g726)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.1:5004
list_route: hop: sip:1034@192.168.1.1:5060
set_destination: Parsing sip:1034@192.168.1.1:5060 for address/port to send to
set_destination: set destination to 192.168.1.1:5060
Transmitting (NAT) to 190.248.187.222:5060:
ACK sip:1034@192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.5:5060;branch=z9hG4bK0f2ee1d1;rport
Max-Forwards: 70
From: “Grupo Over-17435737” sip:17435737@192.168.250.5;tag=as775bf58f
To: sip:1034@192.168.1.1:5060;tag=819468181
Contact: sip:17435737@192.168.250.5:5060
Call-ID: 6e9cda4c5629634b3c60bb7f66afde8d@192.168.250.5:5060
CSeq: 102 ACK
User-Agent: FPBX-12.0.76.4(11.21.2)
Content-Length: 0


<— SIP read from UDP:190.249.187.222:5060 —>

<------------->
Scheduling destruction of SIP dialog ‘6e9cda4c5629634b3c60bb7f66afde8d@192.168.250.5:5060’ in 72960 ms (Method: INVITE)
set_destination: Parsing sip:1034@192.168.1.1:5060 for address/port to send to
set_destination: set destination to 192.168.1.1:5060
Reliably Transmitting (NAT) to 190.249.187.222:5060:
BYE sip:1034@192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.5:5060;branch=z9hG4bK64b13e09;rport
Max-Forwards: 70
From: “Grupo -17435737” sip:17435737@192.168.250.5;tag=as775bf58f
To: sip:1034@192.168.1.1:5060;tag=819468181
Call-ID: 6e9cda4c5629634b3c60bb7f66afde8d@192.168.250.5:5060
CSeq: 103 BYE
User-Agent: FPBX-12.0.76.4(11.21.2)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<— SIP read from UDP:190.249.187.222:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.250.5:5060;branch=z9hG4bK64b13e09;rport=5060;received=181.48.147.188
From: “Grupo Over-17435737” sip:17435737@192.168.250.5;tag=as775bf58f
To: sip:1034@192.168.1.1:5060;tag=819468181
Call-ID: 6e9cda4c5629634b3c60bb7f66afde8d@192.168.250.5:5060
CSeq: 103 BYE
Contact: sip:1034@192.168.1.1:5060
Supported: replaces, path, timer
User-Agent: Grandstream GXP1405 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


(Dave Burgess) #2

No audio is almost always a problem with the NAT and RTP configuration. This can be exacerbated by SIP-ALG settings in intervening routers, but it’s almost always a problem with NAT setting and RTP.

I’d guess that there’s at least one configuration problem in the remote network, especially within the external address and associated non-routable addresses.


(Diego Diaz) #3

so what can i do?


(Dave Burgess) #4

Fix that.


(Diego Diaz) #5

you mean , add a static route in my server , that points to the ip of the remote place where the another extension is?


(Dave Burgess) #6

That might be enough. There might also be problems with the actual addresses getting correctly in the phone that prevents it from communicating with the remote phone through two NAT routers (this can be a problem for many phones). You may need to add a STUN server for the remote phone (assuming you don’t have static addresses set up everywhere)…

To me, it looks like a ‘standard’ routing problem, although much of what you have set up may well be working.