No audio on Asterisk SIP call

Question

I almost managed to init a 2 sided call (click to call): 1st to my office and the 2nd to my cell using Michal Niklas answer (thanks Michal) on Asterisk click to call.

The major ISSUE is that the 2 call participants can not hear one each other, I used from-internal context for both of them. The system status web UI shows me 4 active channels and 2 external calls when the call is connected with both sides.

I configured the Channel: local/MY OFFICE PHONE@from-internal Extension: MY CELL PHONE

when I set the Channel to Sip/1 and check it with soft phone it works great!

Thanks for helping...

score 12 · Answer 1 · answered Jul 25 '13 at 01:50

12

seems to be NAT issue. here are some suggestions for sip.conf

put nat=yes in user definition and in [general] tag, put externip and localnet

e.g.

[general]
externip=<your global IP>
localnet=192.168.2.0/255.255.0.0

answered Jul 25 '13 at 01:50

Suresh Kumar

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Hello, I have done as you have mention in your answer, then also I am not getting voice.. I am getting Beep sound ... but no voice any help? – Nik Jun 06 '18 at 11:50

score 1 · Answer 2 · answered May 24 '16 at 21:26

1

configure in /etc/asterisk/rtp.conf

[general]
rtpstart=10000
rtpend=20000

go to asterisk cli

rasterisk -vvvvvvvvvvvvvvvv
core reload

answered May 24 '16 at 21:26

jose miguel rivera rodríguez

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score 1 · Answer 3 · answered Apr 05 '11 at 16:04

1

Sounds like a NAT issue; did you correctly forward the RTP ports? Did you configure the STUN server in the softphone (if any)?

answered Apr 05 '11 at 16:04

Andrea Spadaccini

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well, I haven't done that. what would be the best approach to figure this out? – Sam Apr 07 '11 at 07:56
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@Sam here is a brief intro to SIP/NAT/Asterisk: http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions – Andrea Spadaccini Apr 07 '11 at 08:03

score 1 · Answer 4 · answered Apr 14 '19 at 01:11

1

In my case, I had to go change the chan sip settings (freepbx.tld/admin/config.php?display=sipsettings) from nat to route

answered Apr 14 '19 at 01:11

Pedro Lobito

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score 1 · Answer 5 · answered Mar 06 '20 at 12:00

Besides NAT problems I've also faced this issues on 3 cases:

1) Missconfigured parameter localnet: on /etc/asterisk/sip.conf make sure you set the network address for the phones. You can alos add multiple networks, for example: localnet=172.16.1.0/24 localnet=192.168.1.0/24

2) Not answered calls: in your dialplan (/etc/asterisk/extensions.conf) you need to answer calls before they can send any audio (RTP), make sure every call runs through an Answer().

3) Multiple answers in a single call: a call can be answered only a single time, in some asterisk versions you won't receive audio if a call is answered twice or more times so make sure you don't.

Anyways, why is Asterisk placing 2 calls? The general workaround for a click to call is: to call Alice and when Alice answers (and only when it answers) place a call to Bob, which will be connected directly to Alice, so you should have only 1 call with 2 channels anytime.

Maybe this can help: https://community.asterisk.org/t/how-to-make-a-click-to-call/74080/2

score -1 · Answer 6 · answered Oct 28 '15 at 16:48

-1

Try adding/changing you're IPAddress in sip_nat.conf

vi /etc/asterisk/sip_nat.conf

Add/change

nat=yes
externip=XXX.XXX.XXX.XXX

Save, quit and restart

answered Oct 28 '15 at 16:48

Slipstream

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No audio on Asterisk SIP call

6 Answers6