Zoiper no audio?

[killmasta93]

Hi,
I was wondering if someone could help me with Zoiper no audio.

So I was reading other forums that i need to add on sip config and sip nat config from the pictures, and then the zoiper can work the audio but the issue then is when someone calls at my Sip trunk the ivr sounds then it hangs up I tried checking the error code using set core verbose 15

 Spawn extension (ivr-3, s, 12) exited non-zero on 'SIP/UNE-OUT-00006ba8'
    -- Executing [h@ivr-3:1] Hangup("SIP/UNE-OUT-00006ba8", "") in new stack
  == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/UNE-OUT-00006ba8'

But if i delete the what i add on the sip config and sip nat config it works again. Not sure if its a NAT thing


Thank you

 

[killmasta93]

Allright so I am not sure if I should add nat=no on my trunk? I have been reading other places that if i try to use zoiper outside of a NAT on the trunk should be nat=no

 

[surp]

You didn't specify how you are connecting over the network. Is the sip phone on the same subnet as Elastix and are they both using private IP's?

Usually you should set nat=yes on everything. That is the most compatible and will usually work regardless of if they are required or not. So on the extension setting and the trunk setting in Elastix. Those sip_nat.conf settings are needed if Elastix is behind NAT and you are trying to connect to it from outside the NAT. You will also need to forward the SIP/RTP ports on the firewall if that is the case.

 

[killmasta93]

Hi Thank you for the reply,

There is no audio when i use zoiper on the iphone using my 3g but there is audio when its connected to the same network as elastix. So then i added the network settings on sip.config and sip nat config then audio works on 3g but then if i leave it like that, when someone calls to the pbx the IVR answers normally and after 5 seconds it disconnects the call. But if i leave it like before then the calls do not get disconnected but there is no audio with zoiper using 3g (back to square one). Tomorrow I will add the nat=yes on the trunk but my question is howcome some configuration people use on the trunk settings nat=no?

Thank you

 

[surp]

Hi Thank you for the reply,

There is no audio when i use zoiper on the iphone using my 3g but there is audio when its connected to the same network as elastix. So then i added the network settings on sip.config and sip nat config then audio works on 3g but then if i leave it like that, when someone calls to the pbx the IVR answers normally and after 5 seconds it disconnects the call. But if i leave it like before then the calls do not get disconnected but there is no audio with zoiper using 3g (back to square one). Tomorrow I will add the nat=yes on the trunk but my question is howcome some configuration people use on the trunk settings nat=no?

Thank you

It sounds like your whole problem is because Elastix is behind a firewall. So definitely you need the sip_nat.conf setting. Also definitely you need nat=yes on the extension setting on the PBX and the trunk setting on the PBX. Your no audio is almost definitely a NAT issue. Probably because you do not have that extension set to nat=yes on the PBX.

You would probably be able to duplicate it on a softphone besides zoiper. I doubt it is anything to do with zoiper assuming you are using all defaults. Also check that you are not using any STUN settings on zoiper. That could make it work in some situations but not others.

One more thing. You appear to be configuring the trunk for g729. Do you have a g729 codec installed? Elastix doesn't come with it and you also need to purchase a license from digium to make the digium one work. Codec issues are the other major reason for no audio.
https://www.digium.com/products/software/g729-codec

 

[killmasta93]

Hi,
Thank you for the reply, unfortunately tried all the possible outcomes and nothing


So i tried adding nat=yes to the outgoing trunk and incoming and then I went to sip nat.config and added
nat=yes
externalip=201.xxx.xx.xx
externhost=201.xxx.xx.xx
localnet=192.168.1.0/255.255.255.0

Then i tried putting nat=no on the trunk outgoing and incoming settings and the calls incoming to the pbx sounds occupied.
then tried to put nat=yes only on the sip nat.config and nothing

then i click save then reload. The zoiper now works no issue but then when i try to call the PBX line it sounds that its occupied the line not sure what i did wrong. I drew out a network diagram maybe explaining a bit more. The speedTouch is what my internet provider put but i have no access to it The worst part is that the previous person did not really leave any explication to the setup so im somewhat trying to decipher. The last part about the codec not sure why they have it configured like that and im sure that they did not buy that codec

Thank you again

 

[surp]

Hi,
Thank you for the reply, unfortunately tried all the possible outcomes and nothing


So i tried adding nat=yes to the outgoing trunk and incoming and then I went to sip nat.config and added
nat=yes
externalip=201.xxx.xx.xx
externhost=201.xxx.xx.xx
localnet=192.168.1.0/255.255.255.0

Then i tried putting nat=no on the trunk outgoing and incoming settings and the calls incoming to the pbx sounds occupied.
then tried to put nat=yes only on the sip nat.config and nothing

then i click save then reload. The zoiper now works no issue but then when i try to call the PBX line it sounds that its occupied the line not sure what i did wrong. I drew out a network diagram maybe explaining a bit more. The speedTouch is what my internet provider put but i have no access to it The worst part is that the previous person did not really leave any explication to the setup so im somewhat trying to decipher. The last part about the codec not sure why they have it configured like that and im sure that they did not buy that codec

Thank you again

Can't help you with pfsense. Make sure it is not doing anythng with SIP or blocking any outbound.

You don't have the second NIC subnet (172.22.102.0/255.255.255.0) as a localnet in your sip_nat.conf. Not sure why you are using 2 NIC's there as it shouldn't matter if the pstn gateway is behind the firewall. It shouldn't interfere with your LAN or cause any security problems either. Also not sure why your PSTN gateway is labelled with yet another subnet but whatever the reason I think that should also be added as localnet in sip_nat.conf as well. Don't use that second nic and the subnets on them if you don't have a good reason for it. KISS!

 

[killmasta93]

Hi,
Thank you for the reply,
So you were completely right I figured out that the issue is with the 172.22.102.162 static IP that is given to the Network Card.

I found out that when i changed the sip nat.config to that IP theres no issue with the IVR connection after 5 seconds but then the audio though zoiper does not work but when i put the 201.xx.xx.xx (WAN) works zoiper but the IVR connections drops after 5 seconds.

So then I realize that Elastix thinks the WAN IP is 172.22.102.162 thats why when I put the 201.xx.xx.xx My real WAN ip it drops the calls when people call the PBX. So i called my lSP to see if they could help me out on their part of the trunk.

The PSTN line is outside of my firewall thats why i need 2 NIC one that is connected to their VOIP speedtouch and the other NIC is my LAN

Thank you again I will keep you posted

 

[killmasta93]

Also just a quick question would it be possible on sip nat.config to make it local net also 172.22.102.162 I tried it but no luck

like I would to make elastix somewhat ignore the 172.22.102.162 WAN when in reality the WAN is 201.xx.xx.xx

Is there another possibility besides VPN?

nat=yes
externalip=201.xxx.xx.xx
externhost=201.xxx.xx.xx
localnet=192.168.1.0/255.255.255.0
localnet=172.22.102.162/255.255.248

Thank you

 

[killmasta93]

I have been reading few other sites and been trouble shooting i re changed my trunk settings and my extensions.
But now i got this so i think im making progress see pictures

http://postimg.org/gallery/p0xpmd9k/

granstream(LAN IP) voice to zoiper(3G) ----One way audio

but on zoiper(3G) to granstream(lan)---no audio

any idea why on the granstream phone (LAN) they cannot hear from what the person is saying on the zoiper(3g) but when talking from the granstream(lan) you can hear on the zoiper(3g)

Thank you

 

[danardf]

Hi

Why nat=no in your config trunk?
You could check what are the codecs used during call:
CLI> sip show channels

Next, if the channel is correct check the RTP using:

CLI> rtp set debug ip an_ip@

• make a call
• stop the call

CLI> rtp set debug off

You should see the ip address source and dest in both senses up and down.

In any cases, Problem with audio = problem with RTP.

Check id there's no filter for the port UDP between 10000 & 20000

 

[killmasta93]

Hi,
Thank you for the reply. I assumed that nat=no on my trunk settings because it is not behind a NAT only my trunk or did i just completely messed up?

I guess whats the most frustrating part is that it work with i add but then my PSTN line (pbx) wont work because it conflict with the eth0 IP of the PBX :cry:

nat=yes
externalip=201.xxx.xx.xx
localnet=192.168.1.0/255.255.255.0

then i did sip show channels and got this picture

then did rtp set debug 179.15.52.229

but go nothing when i call after 4 seconds it hangs up because there is no audio.

and for the audio codec

* SIP Call
  Curr. trans. direction:  Incoming
  Call-ID:                IL-dWPnSPpMRfGpD6Pv-5w..
  Owner channel ID:       <none>
  Our Codec Capability:   (gsm|ulaw|alaw)
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   (gsm|ulaw|alaw|g726|g729|speex|speex16|ilbc|g722|speex32)
  Joint Codec Capability:   (gsm|ulaw|alaw)
  Format:                 (nothing)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    179.15.52.229:46482
  Received Address:       179.15.52.229:46482
  SIP Transfer mode:      open
  Force rport:            Yes
  Audio IP:               192.168.1.20 (local)
  Our Tag:                as29c10373
  Their Tag:              59648f4a
  SIP User agent:         Zoiper r31361
  Username:               300
  Peername:               300
  Original uri:           sip:[email protected]:46482
  Caller-ID:              300
  Need Destroy:           No
  Last Message:           Rx: INVITE
  Promiscuous Redir:      No
  Route:                  <sip:[email protected]:46482;transport=UDP>
  DTMF Mode:              rfc2833
  SIP Options:            norefersub outbound path replaces replace timer
  Session-Timer:          Inactive
  Transport:              UDP
  Media:                  RTP

 

[danardf]

Do you think really that SIP and RTP protocol are on TCP ?
Maybe SIP, but here i think that's under UDP, right?
RTP is always on UDP here.
So no need to use TCP here.

If you have nothing into the row "format", then you have some routing problems sure.
You have nothing when you are launching rtp debug mode? Really a routing issue.

 

[killmasta93]

Hi,
Thank you for the reply, thats odd because on another pbx that has a trunk GSM which is on the same subnet as my elastix the format shows nothing but audio works perfectly. I do have a feeling its a routing issue on the pbx that has no audio because the trunk is outside of the sub net of the LAN IP. I thought by putting canreinvite=no on the trunk would ignore the fact of the sip nat.config when i said externip=201.xx.xx.xx. :shock:

 * SIP Call
  Curr. trans. direction:  Incoming
  Call-ID:                9bb8b8981ccd9748d179d6c69dcd2411
  Owner channel ID:       <none>
  Our Codec Capability:   (gsm|ulaw|alaw)
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   (nothing)
  Joint Codec Capability:   (nothing)
  Format:                 (nothing)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    188.138.33.xx:5076
  Received Address:       188.138.33.xx:5076
  SIP Transfer mode:      open
  Force rport:            Yes
  Audio IP:               201.233.135.xxx (local)
  Our Tag:                as1f33ca03
  Their Tag:              48cdfd5f
  SIP User agent:         sipcli/v1.8
  Caller-ID:              100
  Need Destroy:           No
  Last Message:           Rx: INVITE
  Promiscuous Redir:      No
  Route:                  N/A
  DTMF Mode:              rfc2833
  SIP Options:            (none)
  Session-Timer:          Inactive
  Transport:              UDP
  Media:                  None

 

[danardf]

Last Message: Rx: INVITE
Something is bad here.
An INVITE is sent with no reply.
It should have an ACK as Last Message.

In your logs, no informations about the codecs used the other side.
Something is wrong arround SDP exchange.

 

[killmasta93]

i just realized something when i enter though putty

it shows my VOIP provider IP for the configuration of the eth0

shouldnt it show my LAN IP?

 

[surp]

You seem to be making a mountain out of a mole hill.

First of all NAT=yes.

MAKE EVERYTHING NAT=yes. I already mentioned that before and you seem to have ignored that advice based on assumptions.

NAT=yes almost always works no matter what you configuration even if it is not necessary. NAT=no will not work for a lot of situations.

Also just used default ulaw or alaw codec. Do not enable any other codecs. Do not try do anything with any other codec when troubleshooting.

 

[killmasta93]

so true...but trust me done everything so i just gave up and ended up using iax2 and is working flawless lolz..

But prob later next month im going to format it and start from scratch because the person who installed it really screwed the routing tables.

Thank you again for everything

 

[danardf]

IAX2 doesn't need NAT.
So, it should not have any problem around the NAT, because it doesn't use it.
And more, IAX2 protocol and RTP are one the same port.