Asterisk Server – No Sound
hello folks, this might be a simple question…
I just installed asterisk in a debian server. All seems to be running fine, but the audio sent by the server. If I have one of my registered peers call and extension (102) that plays back audio (extension.conf and sip.conf coffee-pasted below), Asterisk answers and prints no errors. Its `sip show channels` prints:
Peer User/ANR Call ID Format Hold Last Message Expiry
Peer peer.ip 1001 1…-5060 (ulaw) No Rx: ACK
1001
But I hear nothing at the peer’s end.
When one peer calls another, sound comes through just fine. So my hunch is that is something to do with the audio supplied by the server. Do I need to have alsa installed??
Any hint?
sip.conf:
[general]
context = unauthenticated bindport = 5060
bindaddr = 0.0.0.0
tcpbindaddr = 0.0.0.0
tcpenable = yes videosupport = no textsupport=yes alwaysauthreject=yes allowguest=no
[1001] ; grandstream 1
context = home type = friend callerid = One <1001>
secret = XYZ
host = dynamic mailbox = 1001
disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically nat=force_rport
[1005] ; mobile context = home type = friend callerid = Five <1005>
secret = XYZ
host = dynamic mailbox = 1005
disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically nat=force_rport
extensions.conf:
[home]
exten = 102,1,Answer()
same = n,Wait(1)
same = n,Playback(beep)
same = n,Wait(1)
same = n,Playback(im-sorry)
same = n,Wait(1)
same = n,Playback(number-not-answering)
same = n,Wait(1)
same = n,Hangup()
exten => 1001,1,Dial(SIP/1001) ; grandstream phone exten => 1005,1,Dial(SIP/1005) ; mobile
14 thoughts on - Asterisk Server – No Sound
If this is copy and paste, then your dialplan is broken (= should be =>)
But to debug, enable logging (core set verbose 5), when needed debugging
(core set debug 5) and sip logging (sip set debug on / pjsip set logger on).
Tell us about your networking arrangement – are both phones and the Asterisk machine on the same network?
Is there a router in between any of them?
Is there any NAT involved?
No.
Antony.
Le 06/06/2017 à 16:25, Daniel Tryba a écrit :
Well, no. = or => are the same.
—
Daniel
—
Thank you Daniel for pointing out the errors and debug option. Both fixed and on. It made no difference. There are no errors printed and still no sound on ppers
Now to Antony questions:
Nop. They are in 2 different networks. The phones in one and the Asterisk machine in another. Yes. In the phones network. Yes in the phones’ network. They both have different private IP address and one public IP. So I thought.
Thanks guys!!
Okay, that is why you have audio between the two phones, then – they can see each other directly, on the same network, and nothing is interfering with the traffic between them.
Okay, I suspect that this NATting router is not passing the UDP packets from the server back to the phones correctly, based on the SIP connection (when the phone makes the call).
SIP is on UDP 5060; audio is on UDP 10,000 – 20,000.
If it’s a Linux router, you need to make sure you are allowing FORWARDed traffic which matches ESTABLISHED, RELATED.
If it’s not a Linux router, you need to find out how to get it to support SIP
and RTSP.
Good luck,
Antony.
Thanks Anthony.
I did it on the server, according to https://www.voip-info.org/wiki/view/port+forwarding
However after doing it, when running Asterisk I get the following message sudo asterisk -vvvvvvr No ethernet interface found for seeding global EID. You will have to set it manually. Unable to access the running directory (No such file or directory). Changing to ‘/’ for compatibility.
How and where can it be set?
My server ifconfig:
lo Link encap:Local Loopback
inet addr:127.0.0.1 Mask:255.0.0.0
inet6 addr: ::1/128 Scope:Host
UP LOOPBACK RUNNING MTU:65536 Metric:1
RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:36041459269 (33.5 GiB) TX bytes:36041459269 (33.5 GiB)
venet0 Link encap:UNSPEC HWaddr
00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
inet addr:127.0.0.1 P-t-P:127.0.0.1 Bcast:0.0.0.0
Mask:255.255.255.255
inet6 addr: ::2/128 Scope:Compat
inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1
RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:61233254724 (57.0 GiB) TX bytes:106403959440 (99.0 GiB)
venet0:0 Link encap:UNSPEC HWaddr
00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
inet addr:server.ip.add.r P-t-P:server.ip.add.r Bcast:server.ip.add.r Mask:255.255.255.255
UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1
Try to use the echo app. If you can listen your echo, so it is something in the network.
Regards, Marcelo H. Terres
IM: mhterres@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres
Any ideas. After configuring port forwarding on the server (machine making nat) to forward connections originated from external clients to the machine running asterisk, as explained in https://www.voip-info.org/wiki/view/port+forwarding My peers were unable to register.
And When running Asterisk I am getting:
No ethernet interface found for seeding global EID. You will have to set it manually. Unable to access the running directory (No such file or directory). Changing to ‘/’ for compatibility.
Any advice what to do next?
thanks a
Which Asterisk version are you using?
Marcelo H. Terres
IM: mhterres@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres
Looks like it comes com pbx_dundi.c.
Why are you using dundi?
Regards, Marcelo H. Terres
IM: mhterres@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres
I think you need to configure the IPs in your server. You just have localhost… Marcelo H. Terres
IM: mhterres@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres
I am using version: 14.5.0
No, Im not using Dundi. Can you a bit more informative when you say I “need to configure the IPs in your server”?
thanks!
a
Well, based on the config that you sent, your server just have the localhost IP (127.0.0.1)
Marcelo H. Terres
IM: mhterres@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres
And it is worst (and that could be the reason of the error).
127.0.0.1 is configured in 2 interfaces (lo and venet0), just with different network masks.
Marcelo H. Terres
IM: mhterres@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres