Calls forwarded outbound via SIP trunks connect but no audio 4

[SenorWatsonSan]

Howdy, everyone!

Here's what's happening:
Forwarding a call coming into the SIP trunks and then back out the SIP trunks results in no audio. The call rings on the far-end, and can be answered, but there's no audio in either direction.
A call inbound via SIP trunks to a SIP extension works just fine. (Using a softphone on my laptop registered via VPN during testing)
A call inbound via SIP trunks then forwarded outbound with a Misc Destination to my cell via the one PoTS trunk on the system works fine.

Here are relevant specs:
UCx20 at 4.5 with latest updates (just applied them while working on this).
VOIP.ms SIP trunks.

I ran Wireshark captures for working/not-working calls, thinking that would give me enough information to reach out to the IT guy to show RTP was being sent by the UCx and must be getting blocked by their Sonicwall. However, there's no RTP whatsoever during the no-audio call. I can see the call setup, 200 OKs, 100 Trying, 183 Session progress, and finally the BYE, but no RTP.

The Wireshark capture for the working call has RTP all over the place, and I can playback my audio with the Wireshark player.

Basically, the UCx seems to be producing no audio at all when I'm trying to forward calls off-site via the SIP trunks (after having come into the system via the same SIP trunks).

I did a bit of looking around for REFER settings, but I can't find anything relevant on the E-Metrotel Documentation page, nor in the VOIP.ms portal.

P.S. Ringback is also absent during the no-audio call.

Any insights would be appreciated.

 

[ucxguy]

You must have a port forwarding rule on your router for the UCx RTP port range (by default 10000-13999) - make sure your router is configured to forward RTP traffic to the UCx IP address.

 

[emtsupport]

Alan,

As the Ucxguy alluded to, make sure you have the ports forwarded. I duplicated this in the lab by removing the port forward rules for 10000-13999

 

[SenorWatsonSan]

Much obliged, guys.

I placed the request in with the IT guy, and I'll verify the results when I have more info.

Thanks again!

 

[theislandtech]

What is the forwarding configuration for the call with no audio?
I use ring groups and never had to do any port forwarding.

 

[ucxguy]

If I understand the description, the inbound route sends an incoming SIP trunk call (maybe based on a time condition) to an external number via a miscellaneous destination. The call comes in over a SIP trunk and is immediately routed out over another SIP trunk. There is only one destination and that is the external number.

If my memory serves me, calls like that were called Tandem Calls in Nortel documentation...

 

[SenorWatsonSan]

This issue came up when I created the following callflow:
VOIP.ms DID inbound route -> Time Conditions for Holiday/Business Hours -> Ring Group -> NoAnswer to Misc Dest Cell phone oudialing via VOIP.ms. (no audio)
For the meantime, this callflow is outdialing via the single PoTS failover as a band-aid.

For testing, I used a spare DID as follows:
VOIP.ms DID inbound route -> Misc Dest for my cell outdialing via VOIP.ms (this produced the same result as above; no audio. I updated the destination to my test softphone and that worked).

No word back from IT guy yet.

 

[theislandtech]

try: VOIP.ms DID inbound route -> Time Conditions for Holiday/Business Hours -> Ring Group -> no answer use the new ring group outdial Voip.ms

create the ring group
use defaults for all options
for the extension add only your cell number followed by # (use xxxxxxxxxx# if your dial plan is 10 digit dialing or add an access code to reach the voip.ms trunk if necessary)
for destination if no answer use whatever you want to handle the call

audio always good both ways

 

[SenorWatsonSan]

theislandtech,

The outdial worked just fine, but the no audio issue persists (although this time I'm actually getting ringback on the originating phone, but no voice).

I definitely am going to use that RingGroup-forward for other things, now, though.

Thanks for the suggestion

 

[theislandtech]

open the web gui
PBX->TOOLS->Configuration File Editor
click show filter
enter sip_general click search
select sip_general_custom.conf
enter progressinband = yes
click Save then click Reload Configuration

do you have your external ip and local network defined in Settings->Sip Settings?

Edit: what type of router and can you disable sip alg (also known as sip helpers)
I've found SIP ALG causes more problems than it is supposed to help!

 

[SenorWatsonSan]

Interesting....it's a mixed bag, this setup.

I already had the public IP and local network definitions, but I didn't have "progressinband = yes"

I added that to the config file as you specified and reloaded.

Call forwarding w/audio WORKS OK using this:
VOIP.ms DID -> Inbound DID route -> Misc Dest for cell.

No audio when using:
VOIP.ms DID -> Inbound DID route -> Ring Group with my cell and outdial digit as member + #
VOIP.ms DID -> Test ext set to unconditional forward to my cell outdial VOIP.ms.

I tried to enter the "progressinband = yes" under SIP settings, and it took the change, but the Ring Group scheme and the Misc Destination scheme both had the same old issue.

Entering the "progressinband = yes" into the config file works, but throws a scary-looking error under SIP Settings; "ERRORS, Settings in /etc/asterisk/sip_general_custom.conf may override these. Those settings should be removed."

 

[ucxguy]

The router configuration is the solution.

The progressinband won't work (most likely), because this is a NAT issue (if I am right .

A simple workaround would be to get the incoming call answered before it gets forwarded. To do so, you could pass it to an announcement (for example created from the built-in recording en/wait-moment), and the destination of the announcement would be the misc. destination. The call would get answered to play the announcement, the caller would hear "please wait a moment" and then the outbound call would be made. There should be full voice path that way.

 

[theislandtech]

Just logged into an older asterisk (1.8)and changed the progressinband to no and lost audio on a forwarded call to a cell phone.
On another asterisk (13) system i have 6 did's pointing to 6 exts that have followme activated and progressinband did not work, had to play an announcement to get audio.
a couple of other system I had to change the 'rtpkeepalive' from 0 to 1 for audio on ring group calls to external cell phones
All in all - 1 shoe doesn't fit everybody
the only thing common on all my installs is I never have to forward ports
In the case of remote extensions, I run OpenVPN on my router and the yealink phones use the builtin openvpn client to connect to the server

 

[ucxguy]

An explanation why the Asterisk NAT traversal implementation needs in some specific scenarios RTP ports forwarded would be quite technical and is IMHO beyond the scope of this forum. You can get by without the port forwarding rules, but you will encounter one way voice path or no voice path in such scenarios. The tandem call is one of them. There are workarounds that can be used to get the voice path working, but they are typically specific to each problem scenario and can have limitations.

You can certainly experiment and adopt appropriate workarounds to resolve such voice path issues as they arise. You can also save the trouble and time, enable proper port forwarding rules for the RTP port range and eliminate potential voice path issues before they happen. There are many ways to skin a cat...

 

[SenorWatsonSan]

theislandtech,
It's a Sonicwall. We've got a doc we hand to the IT folk on installs, and the SIP ALG advisement is a part of it. SIP ALG should be off.

ucxguy,
Port forwarding is our preferred method. We always try to get the IT guy to forward all the ports we would like opened (and if you can forward one to 443 of the UCx you can configure that sucker from your smartphone!!!)
Unfortunatley, no word yet from the IT guy, so I decided to try the announcement-workaround.

I went to add a recording of a ringtone, and discovered my XLite Softphone stopped working. It registers ok, shows up as Online in the Extension Report, but when I try to make a call from it, it's silent for a few seconds then disconnects.
The Log shows the following after I can see the UCx receive the digits and try to play a recording:
[2016-06-24 20:22:51] VERBOSE[5182][C-0000000a] file.c: -- <SIP/299-0000000c> Playing 'demo-echotest.slin' (language 'en')
[2016-06-24 20:22:56] WARNING[2230] chan_sip.c: Retransmission timeout reached on transmission 79961MDQ0MTlmMGFhNDAyNDcwOTRhN2Q1YmNmMmEyMzAyNjk for seqno 2 (Critical Response) -- See

https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 6399ms with no response
[2016-06-24 20:22:56] WARNING[2230] chan_sip.c: Hanging up call 79961MDQ0MTlmMGFhNDAyNDcwOTRhN2Q1YmNmMmEyMzAyNjk - no reply to our critical packet (see

https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

This Softphone was working fine, earlier, and still works fine registered to another UCx on the VPN. So that leaves something to do with the UCx, but I removed the addition to sip_general_custom.conf

I've rebooted the UCx, my laptop (I've got a couple VPNs I jump in and out of and sometimes I get weird network issues that a reboot clears), verified the edits to the config file are gone (they are, and the error from earlier about that config is gone), I've unregistered the softphone and reregistered, deleted and rebuilt the extension. No change.

Any insight would be appreciated.

I think I've had enough backtalk from machines today. I'll let y'all know how it goes.

Thanks again for your time and attention.

 

[getsomesun1]

So, what was the final resolution?????