TIPS One way audio

shetu

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Hello
I install chan dongle (Asterisk 13.23.1, Huweai e1550, firmware 11.609.20.03.356).
Codec uLaw, g722, gsm
I run traffic capture at Wireshark and it shows RTP send receive both way. But other caller dose not hear anything.

upload_2019-1-28_23-42-6.png
 

AndyInNYC

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One way audio is almost always a fault of NAT.
1. Make sure you have NAT as yes for the extension.
2. Make sure you have SIP ALG disabled on any routers

Good luck.
 

shetu

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Hi
This was from lan. Pbx server and extention are same netwook. I use pfsense router and i think there are no sip alg.
 

chris_c_

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Hi
This was from lan. Pbx server and extention are same netwook. I use pfsense router and i think there are no sip alg.
Try these, one at a time, and report back your results:
1. Enable only uLaw and aLaw codecs (on the Extension and Asterisk SIP Settings). https://wiki.freepbx.org/display/PHON/Codecs
2. Enable all the codecs (on the Extension and Asterisk SIP Settings).
3. Enable STUN (on Extension and Asterisk SIP Settings) and set STUN server to stun.counterpath.net and the default port is 3478.
4. Check your Asterisk SIP Settings, make sure they're sensible:
https://wiki.freepbx.org/display/FPG/Asterisk+SIP+Settings+User+Guide
 

shetu

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1. Enable only uLaw and aLaw codecs (on the Extension and Asterisk SIP Settings).
Same Caller not hear my voice. Here is my wirshark file
2. Enable all the codecs (on the Extension and Asterisk SIP Settings). (Except slin - Huge noise)
Same Caller not hear my voice. Wireshark file
3. I use yate pc client. I did not find stun option for this.

Note : there are no dahdi module.

NAT Settings
External Address : Public IP
Local Networks : LAN - 192.244.11.0/24
Host PC Windos xp (192.244.11.1), VMware 7 - (pbx server 192.244.11.11)​

RTP Port : 10000-20000
RTP Checksums : Yes
Strict RTP : Yes
Chain Sip Settings : NAT - Yes / IP Configaration - Public IP
Reinvite Behavior : no
RTP Timeout : 30
RTP Hold Timeout : 300
RTP Keep Alive : 0
Other SIP Settings : match_auth_username=yes / accept_outofcall_message=yes / outofcall_message_context=sip-message / auth_message_requests= no

Entention 706
DTMF Signaling RFC 2833
Qualify - Yes
Can Reinvite - NO
 
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