No audio on my calls

franxez

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#1
hi :D, i want to know if some one could help me, im running elastix on mi server, and i have fallowing conf
sip._general_custom.conf

register = user:pass@voip.wind.net.do

[FromWind]
type=peer
host= voip.wind.net.do
fromuser= user
secret= passw
context=incoming
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
deny=0.0.0.0/0
permit=200.26.169.19
insecure=invite


[1000]
type=friend
host=dynamic
username=1000
secret=1000
nat=no
canreinvite=no
context=callcenter
mailbox=1000@local
callgroup=1
pickupgroup=1

and so goes on.

on my extensions_override_freepbx.conf

[incoming] ;
exten => s,1,Answer ;
exten => s,n,Dial(SIP/1001&SIP/1002&SIP/1003&SIP/1004&SIP/1005&SIP/1006,30,r)
exten => s,n,Hangup

ok, so when i make a call from a cell phone or a public phone, the call comes in, but there is no voice, and that's weird is the same conf i had 4 days ago and it had voice, but now im having this issue i tested the exten on 4 different pc and its all the same. so its not an soft phone issue, hope so.



In advance:cheer:

Thanks :D
 

dicko

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#2
From your two posts, I would suggest checking your Firewall/router forwarding rules. It is well explained in "Elastix Without Tears". To get very verbose output as to what is failing try

"sip debug ip 200.26.169.19" at the Asterisk CLI.
 

franxez

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#3
:lol: thats true, but i think that if calls are coming in and are receive by the extension, then there must be an other problem, but ill debug the codec's on asterisk cli. ill see what i can do, im not so experience on asterisk
but any ways ill keep u inform if i succeed or fail on this.:lol:

i apologise for my terrible english but my native lang is spanish, but english forums answer faster..

Thanks.:cheer:
 

dicko

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#4
That's why I suggested "Elastix Without Tears" it will explain the difference between Sip signalling ports (5060 UDP) and the ports used for RTP (audio - 10-20000 UDP)and also how to set up sip_nat.conf if you are behind a PNAT translating box, to suit your network configuration.
 

franxez

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#5
MoxTel*CLI>
<--- SIP read from 200.26.169.19:5060 --->
SIP/2.0 100 Trying
From: <sip:uxxxxxx@voip.wind.net.do>;tag=as096547f0
To: <sip:uxxxxxx@voip.wind.net.do>
Call-ID: 0e39d8582d9fa6ec4b279490015cd09e@192.168.50.98
CSeq: 109 REGISTER
Via: SIP/2.0/UDP 192.168.50.98:5060;received=192.168.50.98;rport=5060;branch=z9hG4bK49c294d7
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
MoxTel*CLI>
<--- SIP read from 200.26.169.19:5060 --->
SIP/2.0 200 Registration Successful
From: "SIPLineUser SIPLineUser"<sip:uxxxxxx@voip.wind.net.do>;tag=as096547f0
To: <sip:uxxxxxx@voip.wind.net.do>;tag=1039779532
Call-ID: 0e39d8582d9fa6ec4b279490015cd09e@192.168.50.98
CSeq: 109 REGISTER
Via: SIP/2.0/UDP 192.168.50.98:5060;received=192.168.50.98;rport=5060;branch=z9hG4bK49c294d7
Contact: <sip:s@192.168.50.98>;expires=115
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,com.nortelnetworks.im.encryption
Content-Length: 0
 

dicko

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#6
Then apparently the 5060 signaling is working

"rtp debug" from Asterisk CLI, will show you the audio stream (should be both ways)
 

franxez

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#7
sorry took so long.
here it is.
--- (9 headers 0 lines) ---
-- SIP/1001-0842d088 is ringing
Got RTP packet from 200.26.169.38:42996 (type 00, seq 040064, ts 000040, len 000160)
Sent RTP packet to 200.26.169.40:42996 (type 00, seq 020374, ts 000160, len 000160)
Got RTP packet from 200.26.169.38:42996 (type 00, seq 040065, ts 000200, len 000160)
Sent RTP packet to 200.26.169.40:42996 (type 00, seq 020375, ts 000320, len 000160)
Got RTP packet from 200.26.169.38:42996 (type 00, seq 040066, ts 000360, len 000160)
Sent RTP packet to 200.26.169.40:42996 (type 00, seq 020376, ts 000480, len 000160)
Got RTP packet from 200.26.169.38:42996 (type 00, seq 040067, ts 000520, len 000160)
Sent RTP packet to 200.26.169.40:42996 (type 00, seq 020377, ts 000640, len 000160)
Got RTP packet from 200.26.169.38:42996 (type 00, seq 040068, ts 000680, len 000160)
Sent RTP packet to 200.26.169.40:42996 (type 00, seq 020378, ts 000800, len 000160)
Got RTP packet from 200.26.169.38:42996 (type 00, seq 040069, ts 000840, len 000160)
Sent RTP packet to 200.26.169.40:42996 (type 00, seq 020379, ts 000960, len 000160)
Got RTP packet from 200.26.169.38:42996 (type 00, seq 040070, ts 001000, len 000160)
Sent RTP packet to 200.26.169.40:42996 (type 00, seq 020380, ts 001120, len 000160)
Got RTP packet from 200.26.169.38:42996 (type 00, seq 040071, ts 001160, len 000160)
Sent RTP packet to 200.26.169.40:42996 (type 00, seq 020381, ts 001280, len 000160)
Got RTP packet from 200.26.169.38:42996 (type 00, seq 040072, ts 001320, len 000160)
Sent RTP packet to 200.26.169.40:42996 (type 00, seq 020382, ts 001440, len 000160)
Got RTP packet from 200.26.169.38:42996 (type 00, seq 040073, ts 001480, len 000160)
Sent RTP packet to 200.26.169.40:42996 (type 00, seq 020383, ts 001600, len 000160)
Got RTP packet from 200.26.169.38:42996 (type 00, seq 040074, ts 001640, len 000160)
Sent RTP packet to 200.26.169.40:42996 (type 00, seq 020384, ts 001760, len 000160)
Got RTP packet from 200.26.169.38:42996 (type 00, seq 040075, ts 001800, len 000160)
Sent RTP packet to 200.26.169.40:42996 (type 00, seq 020385, ts 001920, len 000160)
Got RTP packet from 200.26.169.38:42996 (type 00, seq 040076, ts 001960, len 000160)
Sent RTP packet to 200.26.169.40:42996 (type 00, seq 020386, ts 002080, len 000160)
Got RTP packet from 200.26.169.38:42996 (type 00, seq 040077, ts 002120, len 000160)
Sent RTP packet to 200.26.169.40:42996 (type 00, seq 020387, ts 002240, len 000160)
 

dicko

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#8
That indicates that the audio is being passed between two hosts on the Wind Telecom - Softswitch, in Santa Domingo.

I this would appear to be a bridged call with you as a media relay where one host (.38) on Wind is talking to another (.40), (a call involving an internal extension would have one leg at something like 192.168.x.y or whatever your local network is,Please post your sip_nat.conf (obfuscating your public ip/hostname)).


If indeed you are calling one hosted extension to another hosted extension

canreinvite=yes

would probably work better, and alleviate you the necessity to bridge that call.

If this trace was from en external to an internal endpoint, then indeed you need to reexamine your "sip via header rewrites" (as defined in sip_nat.conf)
 

franxez

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#9
my sip_nat.conf is empty, as i told before i never had a problem like this. one more thing the server is connected to a dlink Wireless Router, and the sever is on the DMZ, before that i could not receive calls so that was my solution.
 

dicko

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#10
IMHO opinion YOUR sip_nat.conf (and I see no mention in this thread that it is) should NOT be empty, and you should NEVER have your Elastix box in the DMZ or you WILL be compromised within a week (you need to change default commonly known passwords for all your HTTP/HTTPS services etc. etc. you should never have 80 or 443 or 4445 5038 etc. . . . . exposed haphazardly as you do right now or you be reading up on "Toata dragostea mea pentru diavola" in a few days. Again read "Elastix Without Tears" and it will almost certainly work, if you follow the tried and true methodology contained within.
 

franxez

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#11
ok i understand, i took some precautions before using the server on the DMZ, but, its true what you say im aware of that. i will read the book to see if i can fix that. thanks for your advice
 

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