No inbound audio after IVR transfer

General Help
1

Hello, I need some help please!

After making an IVR selection or interrupting with an extension number, my call is transferred to an extension but there is no inbound audio. I can hear the caller but the caller can not hear me.

I’m able to make internal extension to extension calls with no problems.

I am able to have my did ring directly an an extension (not going through the IVR) and my audio is fine both ways.

And I can call out and audio is fine both ways.

We want to setup and use the IVR, but it’s dropping the audio once the transfer connection is made.

We are using SIPSTATION trunks and from everything I’ve read, we have our nat and firewall and port forwarding and everything setup correctly.

Thank you!

2

If someone could answer this for Robinson, it would help two of us.

We built the current distribution (2.9.0.7) and are experiencing the same IRV issue as robinson. Everything is working fine except the IRV selection of groups or extensions.

The one thing that does work through the IRV are transfers to conferences. With that, we are fairly confident the firewall is port forwarding correctly.

BTW, the SIPSTATION tool does not include IRV inbound routes. We have tried creating the inbould route via SIPSTATION and then use Inbound Routes page to redirect to the IRV.

We are running static IP with “Device and User Mode” turned on behind a NAT server.

Appreciate the help, --Mark

3

Try changing “Reinvite Behavior” to “Update” on the Asterisk SIP Settings page under Tools. It worked for us.

Hope it works for you, as well. --Mark

4

Mark–after several frustrating weeks, delays with help from SIPSTATION and paying for support as well, your suggestion worked! THANK YOU so much!

5

Mark, thanks for the fix, took care of a one-way audio after external call got internally transferred issue for me!

6

For anyone who is interested, I know why this was happening. After many hours of playing with wireshark, I found the answer.

The asterisk system jumps the gun on sending out the audio stream to sipstation before the sipstation server can send out its ACK after giving the 200 OK on the codec handshake. Because of this out of place ACK, the asterisk system halts the stream from the outside.

One way audio call:
|Time |Sipstation |Asterisk |Sipstation Proxy
| | | |
|16.552 |INVITE SDP (g711U) | |
| |(5060)—>(5060) | |
| | | |
|16.586 |100 Trying | |
| |(5060)<—(5060) | |
| | | |
|16.586 |200 OK SDP (g711U) | |
| |(5060)<—(5060) | |
| | | |
|16.586 | |RTP (g711U) |
| | |(10862)---->(12144)|
| | | |
|16.636 |ACK | |
| |(5060)—>(5060) | |
| | | |
|16.648 | |RTP (g711U) |
| | |(10862)<----(12144)|

Where as a good call looks like this:

|Time |Sipstation |Asterisk |Sipstation Proxy
| | | |
|9.960 |INVITE SDP (g711U) | |
| |(5060)—>(5060) | |
| | | |
|9.961 |100 Trying | |
| |(5060)<—(5060) | |
| | | |
|9.965 |200 OK SDP (g711U) | |
| |(5060)<—(5060) | |
| | | |
|9.966 |ACK | |
| |(5060)—>(5060) | |
| | | |
|10.005 | |RTP (g711U) |
| | |(10862)---->(12144)|
| | | |
|10.086 | |RTP (g711U) |
| | |(10862)<----(12144)|

7

Hi. I’m experiencing one-way audio only when calling into my custom IVR. Direct dialing to my extension is fine.
In my case Reinvite Behavior=update did not solve the problem.
Everything ok except the IVR so I believe this is not coming from Firewall-NAT issue.

Origination Provider --> FreePBX --> IVR -->Press1 --> Ext1000 : No audio to caller
Origination Provider --> FreePBX --> Ext1000 : Working fine.
FreePBX version: 2.8.1.4

Appreciate if someone gives me a hint again.

Thank you

8

Just be sure your internal router has SIP ALG disabled.

Also, if you are using a DSL connection (with a Netopia) as the gateway, you must turn off SIP ALG on this device too.

This requires a “telnet” connection to the device, so some syntax to turn of SIP ALG can be done.

Shawn