no audio on external phone

adcphones

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#1
I am getting no audio both way with external phone, using dynamic ip

sip_nat.cong

externhost=junqueirasip.selfip.net
localnet=192.168.1.0/255.255.255.0­
externrefresh=10
nat=yes
qualify=yes
canreinvite=no

can somebody help please
 

Kriss

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#2
Do you have a firewall that could drop some RTP trafic ?
Have you tried :
externip=67.205.200.91
in your /etc/asterisk/sip_nat.conf

No error on the codecs side ?
 

alben

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#3
I have a similar problem.
I have elastix 1.6..12, i have 2 trunks setup with voip providers working fine.
Local extension to extension calls work fine, but external extension registers, calls, rings, pick up but no voice.
my sip_nat.conf looks

nat=yes
externip=190.43.78.40
localnet=192.168.1.0/255.255.255.0
externrefresh=10

my elastix server is 192.168.1.46
in my zyxel router just for testing i set to default all to 192.168.1.46

i apreciate your help
 

dicko

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#4
The audio payload is not SIP, it is an rtp stream on an arbitrary port (negotiated by SIP) between 10000 and 20000 (by default) your router must allow and forward such connections to your Asterisk Server without attempting any port translation and vice versa at the router on the far-end to the extension, suspect the far-end router.

dicko
 

alben

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#5
hi,
in my router i defaulted all to my elastix, just for testing (called DMZ?) so is not my router, i the remote router i am able to make voip calls (means rtp is flowing?)
Also in a elastix IRC channel i heard that custom context might messed up elastix, i have custom context installed.
thanks again
 

dicko

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#6
If the extension is registered and can place and ring calls (all SIP) then it's not custom-contexts.

That only leaves the routers. There is nothing else to be said

you can try rtp debug from asterisk and tcpdump from bash to find out what you did wrong.

dicko

p.s.

.
.
sip_nat.cong

externhost=junqueirasip.selfip.net
localnet=192.168.1.0/255.255.255.0­
externrefresh=10
nat=yes
qualify=yes
canreinvite=no
.
.
.

the nat, qualify and canreinvite do NOT belong in this part of sip.conf and it's inclusions.
 

witekprytek

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#7
alben: what is your trunk configuration?
If you forward all traffic to your astsreisk server you do not need to use sip_nat.cfg
You can forward sip port 5060 udp to your elastix server and rtp udp 10000-20000 to your elastix server too (check the range of rtp in /etc/asterisk/rtp.conf
I have :
rtpstart=10000
rtpend=20000)

Then comment everything in sip_nat.cfg.

In your SIP trunk configuration set nat=yes.

This solution works fine for me.
 

alben

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#8
Re: Re:no audio on external phone

my sip_nat.conf look like this:

nat=yes
externip=190.43.87.41
localnet=192.168.1.0/255.255.255.0
externrefresh=10

i use trunks for voip provider only, i think i dont need trunks for extension to extension calls,

my ip is dynamic changes once a day,

thanks
 

dicko

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#9
Re: Re:no audio on external phone

My apologies for commenting on the wrong sip_nat.conf.

If your ip is dynamic then you need to use

externhost=myhostname.someorg.org

and register with a dynamic dns provider, preferably in Peru if available, or edit that file every day when it changes.
 

DaveD

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#10
Re: Re:no audio on external phone

With Elastix version 1.6 doesn't sip_nat.conf get replaced with sip_general_custom.conf
 

dicko

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#11
Re: Re:no audio on external phone

It doesn't really matter, sip.conf "includes" sip_general_custom.conf immediately before sip_nat.conf, but they are both still included. it's a matter of style, I prefer to put the nat related stuff in the appropriately named file, but as ever the "last one wins"

regards

dicko
 

alben

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#12
Re: Re:no audio on external phone

Hi dicko

my rtp debug fills up my screen with this

Got RTP packet from 192.168.1.130:16386 (type 08, seq 000094, ts 076400, len 000160)
Sent RTP packet to 192.168.1.33:16384 (type 00, seq 031484, ts 076400, len 000160)
Got RTP packet from 192.168.1.130:16386 (type 08, seq 000095, ts 076560, len 000160)
Sent RTP packet to 192.168.1.33:16384 (type 00, seq 031485, ts 076560, len 000160)
Got RTP packet from 192.168.1.130:16386 (type 08, seq 000096, ts 076720, len 000160)
Sent RTP packet to 192.168.1.33:16384 (type 00, seq 031486, ts 076720, len 000160)
Got RTP packet from 192.168.1.130:16386 (type 08, seq 000097, ts 076880, len 000160)
Sent RTP packet to 192.168.1.33:16384 (type 00, seq 031487, ts 076880, len 000160)
Got RTP packet from 192.168.1.130:16386 (type 08, seq 000098, ts 077040, len 000160)
Sent RTP packet to 192.168.1.33:16384 (type 00, seq 031488, ts 077040, len 000160)
Got RTP packet from 192.168.1.130:16386 (type 08, seq 000099, ts 077200, len 000160)

my elastix server is 192.168.1.47, .....1.130 is my ATA device but ...1.33 is not in my LAN.

this is my router screeenshot


thanks a lot
 

dicko

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#13
Re: Re:no audio on external phone

Then I suggest you start with:-

http://www.voipuser.org/forum_topic_7295.html

To understand the basics of SIP/SDP and so to identify which part of your network is misreporting to SIP (the SDP part of SIP) as to how to write the VIA's, It is not your Server and your NAT descriptions are obviously correct. Something is translating the rtp port (that will be a firewall somewhere) and the Address (probably also a firewall)

you can read the INVITE's and VIA's at the Asterisk CLI with :-

sip set debug peer <ext>.

dicko
 

alben

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#14
Hi,
following the problem that external phone has no voice,
if i am able to place voip calls from my local extension (using providers as betamax) that eliminates my local router as the source of the problem?
also if i am able to place voip calls from my remote extension(not using elastix) that leaves out that router as source of the problem

thanks again
 

dicko

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#15
Although you have eliminated the routers as possibilities, Your logs show however show otherwise.

As I said before


sip set debug peer <ext>.

to see what is misconfigured.
 

alben

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#16
following my study case, with my same network i connected another elastix 1.6 server and now i do have voice in external phones. Then, something is wrong with my first elastix server.
The following is what i found is different in both cases:

----elastix with no voice--------------------

dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 2001 to extension map
-- dialparties.agi: Extension 2001 cf is disabled
-- dialparties.agi: Extension 2001 do not disturb is disabled
> dialparties.agi: extnum 2001 has: cw: 0; hascfb: 0 [] hascfu: 0 []
dialparties.agi: ExtensionState: 0
dialparties.agi: Extension 2001 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 2001
-- dialparties.agi: dbset CALLTRACE/2001 to 1000
-- dialparties.agi: Filtered ARG3: 2001
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/1000-082c7390", "SIP/2001||tr") in new stack
-- Called 2001

----elastix with voice------------------------

dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 200 to extension map
-- dialparties.agi: Extension 200 cf is disabled
-- dialparties.agi: Extension 200 do not disturb is disabled
dialparties.agi: ExtensionState: 0
-- dialparties.agi: dbset CALLTRACE/200 to 201
-- dialparties.agi: Filtered ARG3: 200
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/201-08b03f10", "SIP/200||tr") in new stack
-- Called 200
-- SIP/200-08b5f478 is ringing
-- SIP/200-08b5f478 answered SIP/201-08b03f10
--------------------------------------------------------------

the line
dialparties.agi: extnum 2001 has: cw: 0; hascfb: 0 [] hascfu: 0 []

must be causing the problem?


thanks a lot
 

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