When I do a forward call to a internal line it works.
I see in the CLI that there a 2 channels.
When I forward a call to a outside line, I see in the CLI 4 channels.
I think that could be the problem.
But the sollution I haven't found yet.

Hi folks,

I have the same problem here. I'm forwarding an incoming sip call to an external phone (over sip) and have no audio in both directions. If I use Follow-Me and the 'Confirm Calls'-hack, it works (but this can not be a solution). Forwarding ISDN calls makes no problems, also forwarding over ISDN is Ok.

It seems to me that there is a problem in FreePBX. Would be great if one of the developer can look at it.

Regards,
Martin

I was experiencing this because the SIP Trunk provider. I switched to a different trunk by a different provider and it worked as planned. Not sure why my original trunk provider doesn't work though.

I hit this problem last week and fortunately I can do packet captures on my router. The problem is a NAT issue.

When you initiate a call from inside to outside, you always have no way audio until the first RTP packet is sent from inside to outside. NAT routers will not allow an inbound (outside to inside) packet if it does not already have a translation created for that source, destination IP and port pair.

In the case of call forwarding from an inbound SIP call to an outbound SIP call, the call path needs to go from the calling outside phone to the called outside phone. In this case, the PBX is simply relaying RTP traffic from the inbound call to the outbound call, so the PBX will not initiate either RTP stream. Since the PBX on the inside of the NAT router does not initiate the call, it cannot receive RTP traffic from the calling or called phone.

This is why when you check the "Confirm Calls" it works. In this case, the PBX initiates one of the RTP streams to the called phone. Once you accept the call, the PBX now can initiate the second RTP stream to the calling phone.

There are two ways to fix this.

1. Use IAX - you will not see this issue since the RTP traffic uses the same source and destination ports as the control traffic.
2. Forward RTP ports to PBX - Forward the UDP port range in /etc/asterisk/rtp.conf to your PBX.

I have the same issue but when I set the "Play Music On Hold?" to "default" (it plays music instead of ringing) audio works.

--UPDATE--

I put the server on a public IP and it works.
It looks like its a NAT issue.

We started encountering this problem today - but did not have it before we upgraded to the latest update from FreePBX (2.6.0.3).

We have SIP trunks with DIDs, terminating on extensions that are also provisioned via SIP.

Prior to the upgrade, we could use Unconditional Forwarding to send all calls to an external number and everything worked just fine.

Today, I tried to setup another user exactly the same way, and the calls go through - the external number rings and they pick up the phone, but there is no audio at all on either end, just dead air. I added a 3rd line to test it out, and it happened the same way. Just on a whim, I tried using Follow Me instead of Call Forwarding and got exactly the same results.

Following some advice I saw on another thread, I tried going into the Trunk setup for my SIP trunks and I checked the box for "Block Foreign Caller IDs" just in case that might be a problem, but that had no effect on the problem. We still had no audio on any calls being forwarded.

Here is the completely weird part:

If I call into our main number and use the auto-attendant to dial the extension number rather than calling the DID number, and the call is forwarded via that method, everything works fine! We get full audio on both ends.

In all tests, these are the only calls taking place on the server at the time - this is not an issue of capacity. There appears to be some problem within the system itself, when a call comes in via direct external DID incoming call and gets directed back out to an external number that is preventing the voice channels from being bridged together properly.

I can report that after making the above mentioned changes, then reloading Asterisk, my new box works just as it should. There is a bit of latency however, so I am still trying to figure that out.

I can confirm this did not work for me on a FreePBX 2.8.11 Asterisk 1.6.2.13 installation.

Identical situation with different results. Forwarding through an AA with a 1 second timeout and no loops to the off-site number creates an automatic path, but this is unfeasible for all end users to have and administer.

The box has 2 providers on it, and is only having the issue with one (whom is diligently working with me)

Hi,

I encountered the same problem with originating calls to 2 external numbers via Callcentric SIP trunk: No audio. Asterisk is behind NAT firewall.

However, after trying all the settings, it finally worked. So here is my full configuration steps, just in case someone need to verify his settings.

--------------------------------------------------

0. Open port in Router/NAT modem, point to Asterisk internal IP

UDP 5060 (This is for remote extension - If you dont have remote extension, it is strongly recommended to close this port)(*)
UDP 10000-20000 (this is for Asterisk)
UDP 50000-65535 (this is for callcentric)(**)

(*): I tried to close this port, and the audio is still good.
(**): this is one of the key things: Use "sip set debug on" and watch the SIP package returned from Callcentric or your SIP provider in response to Asterisk INVITE. Notice the port.

1. Adjust the Callcentric SIP trunk as follow (notice it is slightly different than mentioned in Callcentric support page)
TRUNK NAME: default_sip_trunk

PEER DETAILS:
context=from-trunk
fromdomain=callcentric.com
fromuser=1777XXXXXXX
authuser=1777XXXXXXX
username=1777XXXXXXX
host=callcentric.com
defaultuser=1777XXXXXXX
insecure=very
secret=******
type=friend
dtmfmode=rfc2833
qualify=yes

REGISTER: 1777XXXXXXX:******@callcentric.com/1777XXXXXXX

Make sure it is registered in Asterisk

2. Add in sip_general_custom.conf (using Trixbox Config File Editor Tool)

srvlookup=yes
session-timers=refuse ; for call-centric
session-expires=180 ; for call-centric
session-minse=90 ; for call-centric
session-refresher=uas ; for call-centric
nat=yes
externip=1.2.3.4 ; this is your public IP
localnet=192.168.0.0/255.255.255.0 ; this is Asterisk local subnet
progressinband=yes

3. Add in extension_custom.conf
[bridge-external-call]
exten => _X.,1,NoOp(Gonna make outbound call - bridge 2 external calls - nttranbao@yahoo.com - Jun 2011)
exten => _X.,n,Playback(vm-dialout) ; this is required in order to have audio. Very important
exten => _X.,n,Dial(SIP/default_sip_trunk/${EXTEN},30,)
exten => _X.,n,Hangup

4. Add in manager_custom.conf
[manager-user]
; these are for auto-bridging 2 calls
;nrtranbao - Jun 2011
secret=blablabla
deny=0.0.0.0/0.0.0.0 ; deny originate from any other hosts
permit=127.0.0.1/255.255.255.255 ; only allow originate from localhost
read=all
write=all

To originate calls, use the below command in Asterisk CLI
originate SIP/default_sip_trunk/11-digit-caller extension 11-digit-callee@bridge-external-call

Asterisk will first call the caller, and when answered, bridge to the context "bridge-external-call". This context will play a message (***), then dial 2nd number. Audio works fine.

(***)(KEY POINT - thanks to this post: http://lists.digium.com/pipermail/asterisk-users/2009-April/229699.html)

Note: if you upgrade FreePBX in the future you may need to add these lines again, depending on if they have fixed the issue or not. This worked for me, YMMV.This is better than overriding in extensions_override_freepbx.conf in my opinion.

Follow Me Fix for DID’s

Add the code "$ext->add($contextname, $grpnum, '',new ext_answer('')); " in to /var/www/html/admin/modules/findmefollow/functions.inc.php

//
// If the followme is configured for extension dialing to go to the the extension and not followme then
// go there. This is often used in VmX Locater functionality when the user does not want the followme
// to automatically be called but only if chosen by the caller as an alternative to going to voicemail
//
$ext->add($contextname, $grpnum, '',new ext_answer(''));
$ext->add($contextname, $grpnum, '', new ext_gotoif('$[ "${DB(AMPUSER/'.$grpnum.'/followme/ddial)}" = "EXTENSION" ]', 'ext-local,'.$grpnum.',1'));
$ext->add($contextname, $grpnum, 'FM'.$grpnum, new ext_macro('user-callerid'));
$ext->add($contextname, $grpnum, '', new ext_set('__EXTTOCALL','${EXTEN}'));
$ext->add($contextname, $grpnum, '', new ext_set('__PICKUPMARK','${EXTEN}'));
// block voicemail until phone is answered at which point a macro should be called on the answering // line to clear this flag so that subsequent transfers can occur, if already set by a the caller

FreePBX 2.8.1.4
elastix 2.3.0 5
asterisk 1.8.11.0 0

on a fresh build

Was getting this problem with mobile dialing into PBX and being routed out via Follow-me to an extension that was also a mobile number. No audio in either direction going through an Oztell trunk both ways. Could see no errors in the logs and calls held until terminated. If I rang from a local extension it worked fine. just inbound calls being re-routed out that got effected.

I forwarded 5060 and ports 10,000 to 20,000 on the firewall for SIP and RTP but still no luck.

then added in NAT settings on FREEPBX under the tools/Asterisk SIP Settings
Set for using NAT on a dynamic ip (no-ip.com)

and that fixed it.