GO HERE Anveo Outgoing Help - No Audio either way - calls connect, does RTP channel connect?

[newvoiper]

I've been having a problem with Anveo Direct outgoing calls for some time. The calls connect to the destination number, but there is no audio coming through either way after the call is answered. I have no DID phone number at Anveo. (Could that in itself be the reason why I am having a problem?). When I reached out to Anveo, they said they don't proxy the voice channel so wouldn't be able to see it or help me with this problem.

Running IncrediblePBX 2021. I've been over the set up for outgoing calls in the Anveo portal many times and can't see any problem, even though that's where I think it probably is.
In Reports--> Asterisk Logfile I observe the following when the call starts to go bad:

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04943[2021-01-12 11:27:56] VERBOSE[10377][C-00000010] app_dial.c: SIP/sbc.anveo.com-0000001e answered SIP/703-0000001d
104944[2021-01-12 11:27:56] VERBOSE[10398][C-00000010] bridge_channel.c: Channel SIP/sbc.anveo.com-0000001e joined 'simple_bridge' basic-bridge <c507fa13-098f-483f-a933-8118de5e6b2e>
104945[2021-01-12 11:27:56] VERBOSE[10377][C-00000010] bridge_channel.c: Channel SIP/703-0000001d joined 'simple_bridge' basic-bridge <c507fa13-098f-483f-a933-8118de5e6b2e>
104946[2021-01-12 11:28:27] NOTICE[1901] chan_sip.c: Disconnecting call 'SIP/sbc.anveo.com-0000001e' for lack of RTP activity in 31 seconds
104947[2021-01-12 11:28:27] VERBOSE[10398][C-00000010] bridge_channel.c: Channel SIP/sbc.anveo.com-0000001e left 'simple_bridge' basic-bridge <c507fa13-098f-483f-a933-8118de5e6b2e>
104948[2021-01-12 11:28:27] VERBOSE[10377][C-00000010] bridge_channel.c: Channel SIP/703-0000001d left 'simple_bridge' basic-bridge <c507fa13-098f-483f-a933-8118de5e6b2e>
104949[2021-01-12 11:28:27] VERBOSE[10377][C-00000010] app_macro.c: Spawn extension (macro-dialout-trunk, s, 37) exited non-zero on 'SIP/703-0000001d' in macro 'dialout-trunk'

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When I look at the siptrace in the Anveo portal, it looks like my server offers RTP port 10490, and the carrier offers RTP port 30406 (does it matter if these are the same?)
Can anyone tell what might be wrong?

I have UDP ports 5060-61 and 10,000-20,000 forwarded to IncrediblePBX 2020 by my router. Here is the entire siptrace, with sensative numbers replaced with phrases:

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/*<<<|myserverip:5060 @ 2021-01-12 16:27:52 */
INVITE sip:0XXXXXDestPhone#@sbc.anveo.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK62fc8aaa;rport
Max-Forwards: 70
From: <sip:CIDPHONE#@192.168.1.3>;tag=as169a92f8
To: <sip:mypinDestPhone#@sbc.anveo.com>
Contact: <sip:CIDPHONE#@192.168.1.3:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: IncrediblePBX-15.0.17.9.1(18.1.1)
Date: Tue, 12 Jan 2021 16:27:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "CIDPHONE#" <sip:CIDPHONE#@192.168.1.3>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 554

v=0
o=root 2010611583 2010611583 IN IP4 192.168.1.3
s=Asterisk PBX 18.1.1
c=IN IP4 192.168.1.3
t=0 0
m=audio 10490 RTP/AVP 0 9 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=ice-ufrag:61e52591392e42766f398dc50c1ffdb0
a=ice-pwd:082d4573739975e854f6630d542f278d
a=candidate:Hc0a80103 1 UDP 2130706431 192.168.1.3 10490 typ host
a=candidate:Hc0a80103 2 UDP 2130706430 192.168.1.3 10491 typ host
a=sendrecv


/*>>>|myserverIPaddress:5060 @ 2021-01-12 16:27:52 */
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:5060;received=myserverIPaddress;branch=z9hG4bK62fc8aaa;rport=5060
From: <sip:CIDPHONE#@192.168.1.3>;tag=as169a92f8
To: <sip:mypinDestPhone#@sbc.anveo.com>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: Anveo Callcontrol
Content-Length: 0


/*>>>|myserverIPaddress:5060 @ 2021-01-12 16:27:53 */
SIP/2.0 180 Ringing
Server: Anveo Callcontrol
Via: SIP/2.0/UDP 192.168.1.3:5060;received=myserverIPaddress;rport=5060;branch=z9hG4bK62fc8aaa
From: <sip:CIDPHONE#@192.168.1.3>;tag=as169a92f8
To: <sip:mypinDestPhone#@sbc.anveo.com>;tag=c27181dfb2bf8fc943389145f808e126
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: Anveo Callcontrol
Content-Length: 0


/*>>>|myserverIPaddress:5060 @ 2021-01-12 16:27:56 */
SIP/2.0 200 OK
Server: Anveo Callcontrol
Via: SIP/2.0/UDP 192.168.1.3:5060;received=myserverIPaddress;rport=5060;branch=z9hG4bK62fc8aaa
From: <sip:CIDPHONE#@192.168.1.3>;tag=as169a92f8
To: <sip:mypinDestPhone#@sbc.anveo.com>;tag=c27181dfb2bf8fc943389145f808e126
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: Anveo Callcontrol
Contact: Anonymous <sip:169.48.232.158:5060>
Content-Type: application/sdp
Content-Length: 234

v=0
o=Sonus_UAC 253599 468908 IN IP4 67.231.5.112
s=SIP Media Capabilities
c=IN IP4 67.231.5.80
t=0 0
m=audio 30406 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

/*<<<|myserverIPaddress:5060 @ 2021-01-12 16:27:56 */
ACK sip:169.48.232.158:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK1655266a;rport
Max-Forwards: 70
From: <sip:CIDPHONE#@192.168.1.3>;tag=as169a92f8
To: <sip:mypinDestPhone#@sbc.anveo.com>;tag=c27181dfb2bf8fc943389145f808e126
Contact: <sip:CIDPHONE#@192.168.1.3:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: IncrediblePBX-15.0.17.9.1(18.1.1)
Content-Length: 0

/*<<<|myserverIPaddress:5060 @ 2021-01-12 16:28:27 */
BYE sip:169.48.232.158:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK3bb69466;rport
Max-Forwards: 70
From: <sip:CIDPHONE#@192.168.1.3>;tag=as169a92f8
To: <sip:mypinDestPhone#@sbc.anveo.com>;tag=c27181dfb2bf8fc943389145f808e126
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: IncrediblePBX-15.0.17.9.1(18.1.1)
X-Asterisk-HangupCause: Requested channel not available
X-Asterisk-HangupCauseCode: 44
Content-Length: 0

/*>>>|myserverIPaddress:5060 @ 2021-01-12 16:28:27 */
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5060;received=myserverIPaddress;branch=z9hG4bK3bb69466;rport=5060

 

[wardmundy]

Have you reviewed this thread?

 

[kenn10]

You appear to be sending your internal IP address to Anveo. You need to be sending your external IP address or it won't work. Do you have NAT=Yes on your extensions?

Also, in the PBX GUI, look at Settings > Advanced SIP Settings and look at the General SIP Settings. Is your external IP address in the External Address field? If not, click on the Detect Network Settings button and save it and apply settings.

 

[newvoiper]

@wardmundy: I found the thread, but didn't think it applied to outbound routes. But since it's so easy to try, I added the [from-anveo] to /etc/asterisk/extensions_custom.conf AT THE BOTTOM OF THE FILE. Even after an IncrediblePBX 2020 reboot, I got the same results.

@kenn10: I couldn't find anywhere to add a NAT=Yes on the sip extensions in IncrediblePBX 2020. Under "Advanced" tab for the sip extension I used for the outbound call, there is a "NAT Mode" item set to "Yes - (force_rport, comedia)"

Under Settings-->Asterisk SIP Settings-->General SIP Settings, I have left NAT Settings blank because I have tried it before and didn't get any different results. I guess I can try it again and look at the SIP trace again....

 

[atsak]

You will need to have that SIP settings setup with the external IP or a dns name, or you'll get no audio. Also make sure your router doesn't have SIP ALG enabled.

 

[newvoiper]

I tried clicking "Detect Network Settings" from Settings-->Asterisk SIP Settings-->General SIP Settings-->NAT Settings, and it correctly populated my server's external IP address, and the CIDR for my local network, but after a reload I got the same results. My old router had a SIP ALG setting but the FreshTomato firmware I used today, doesn't have that setting.

 

[kenn10]

I had so much trouble with Tomato firmware, I bought a new router.

First, try this, on your router, put the IP address of your PBX in the DMZ. This will allow full access to your PBX from the outside but the firewall on the PBX will protect it. If your calls still don't have audio, something is wrong in the PBX setup. If it does fix it, you have something wrong in the router. If it does not fix it, provide info to question below.

Secondly, did you set up the Anveo trunk as chan_sip or PJSIP? Is your extension chan_sip or PJSIP?

 

[kenn10]

Also, see this thread about disabling SIP ALG on your FreshTomato router if it has the settings described below:

Disable SIP ALG on Tomato Shibby | MCB Systems

A PBX vendor like 3CX may tell you to disable SIP ALG on your router. I’m running Tomato on my router, specifically Tomato Firmware 1.28.0000 MIPSR2-132 K26 Max. With the help of this thread, I checked and disabled SIP ALG.Under Advanced > Conntrack/Netfilter, in the Tracking/NAT Helpers...

www.mcbsys.com

 

[newvoiper]

@kenn10: OK, putting the IP address of my PBX in the DMZ, my calls still don't have audio (I waited a few minutes and tried it twice)

I use the Anveo-Out trunk for outgoing calls. It's set up as 'custom' and the only configuration set is the following in "Custom Dial String": SIP/[email protected] and I populated "Oubound Caller ID" with the value <1NNNNNNN>.

In addition to Anveo-Out, there are 4 other Anveo-x trunks set up and enabled to do conventional registration. Tell me if you want the configuration details for those. I have confirmed each of these have one of the 4 ip addresses in the Anveo FAQ for setup. (There was an extra Anveo-x trunk in the setup with a different IP which I didn't appear to need and I deleted it.)

Thank you for the link showing how to disable SIP ALG--I didn't realize the router had the capability. But I confirmed, neither setting was checked. To be doubly sure I ran lsmod and confirmed the modules shown in the link DON'T show up.

 

[kenn10]

Check these advanced SIP settings:

 

[kenn10]

On your extension, check these settings:

 

[newvoiper]

Thanks for the ideas: now I know where the NAT setting is. All the settings illustrated, are the same as what is in your pictures. I also tried a router reboot just in case, to no avail.

Is there a tool you use to grab the picture of the Asterisk Settings?

 

[kenn10]

Built into Windows 10: https://support.microsoft.com/en-us...eenshots-00246869-1843-655f-f220-97299b865f6b

 

[kenn10]

I'm still suspicious of your router firmware. Try creating a new PJSIP trunk. Name it AnveoTest or something. This thread give you screenshots of my trunk:

https://www.voip-info.org/forum/threads/ip-call-issues-when-using-anveo.25229/#post-153742

 

[newvoiper]

@kenn10 - I tried it. Every time I dial out, I get Allison's voice saying "The number you have dialed is not in service. Please check the number and try again."

Here are my pjsip set up screens--please tell me if I overlooked something there.

 

[kenn10]

Make sure you put the dial strings on the new trunk. Anveo expects 1+area code+number:

 

[kenn10]

Be sure you authorize your IP address on the Anveo portal for Outbound trunks. Leave the dialing prefix blank on this page.

 

[kenn10]

Assuming Anveo is your only trunk, delete your other outbound route and create one with the new Anveo trunk in it and set the route dialplan something like this to allow you to dial a number with 1+area code+number or just area code+number.

 

[kenn10]

Finally, log into the Asterisk CLI (asterisk -rvvvvv) and type this command:

pjsip show endpoints

The result should show the trunk being available if all is well:

 Endpoint:  Anveo-Direct                                         Not in use    0 of inf
        Aor:  Anveo-Direct                                       0
      Contact:  Anveo-Direct/sip:sbc.anveo.com:5060        c4a7c40a48 Avail        31.198
  Transport:  0.0.0.0-udp               udp      3     96  0.0.0.0:5062
   Identify:  Anveo-Direct/Anveo-Direct
        Match: 67.212.84.21/32
        Match: 176.9.39.206/32
        Match: 72.9.149.25/32
        Match: 169.48.232.158/32

 

[newvoiper]

@kenn That worked--I got the number to go through after changing the dial plan in Anveo-Test trunk (the dial plan was already as you showed in Default Outbound Route) AND changing the configuration in Anveo Portal as you showed.

Unfortunately, there was no sound either way after the call connected

Here is the snip of pjsip show endpoints from Reports-->Asterisk Info: