Tag Archives: snom

Snom VoIP phones: OpenVPN and multicast audio

Snom voice-over-IP phones have a built-in OpenVPN client, and they can also have audio transmitted to them via multicast. However, in my case a phone logged into OpenVPN did not play back audio received from the local ethernet network

I had multicast reception configured correctly on the phone:

multicast_listen=on
mc_address=239.255.255.245:5555

(Remember that you need to push the Apply button at the bottom of the page and then the Save button at the top of the page for the setting to actually take effect.)

To diagnose the problem, you can use multicast pings. Running

ping -I eth0 -t 2 239.255.255.245

on the local network showed no reactions from the phone. However, running

ping -I tun2 -t 2 239.255.255.245

on the VPN server showed responses from the phone. So clearly the phone wanted to do multicast exclusively via VPN. The cause of this was the following line in my OpenVPN config on the server:

push "redirect-gateway def1"

The obvious solution was to replace this line with routes that excluded the multicast IP addresses:

push "route 0.0.0.0 128.0.0.0"
push "route 128.0.0.0 192.0.0.0"
push "route 192.0.0.0 224.0.0.0"

While this worked, the phone wouldn’t complete booting up the next time I power-cycled it. So I switched back to the old OpenVPN config and instead turned on multicast routing on the VPN server:

apt-get install smcroute
echo 'mgroup from eth0 group 239.255.255.245' > /etc/smcroute.conf
echo 'mroute from eth0 group 239.255.255.245 to tun0 tun1 tun2' > /etc/smcroute.conf
service smcroute restart

Multicast audio now works and it even goes across the VPN.

Below you can find some commands that use FFMPEG to send the multicast audio:

ffmpeg -re -i song.mp3 -filter_complex 'aresample=16000,asetnsamples=n=160' -acodec g722 -ac 1 -vn -f rtp udp://239.255.255.245:5555
ffmpeg -re -i song.mp3 -filter_complex 'aresample=8000,asetnsamples=n=160' -acodec pcm_mulaw -ac 1 -vn -f rtp udp://239.255.255.245:5555
ffmpeg -re -i song.mp3 -filter_complex 'aresample=8000,asetnsamples=n=160' -acodec pcm_alaw -ac 1 -vn -f rtp udp://239.255.255.245:5555

Interestingly, you don’t even need to set the rtp_codec_type setting — the phone automatically determines the codec from the stream. I wasn’t able to get the Opus codec working though, the phone just makes crackling noises when I tried.

Asterisk: Change number in To header

A long time ago, I wrote about changing the callee ID as seen by the caller using CONNECTEDLINE.
Changing the caller ID as seen by the callee is also pretty obvious using CALLERID.
That leaves two more constellations: changing the caller ID as seen by the caller (which doesn’t make sense because a phone typically doesn’t display its own number on outgoing calls)., and changing the callee ID as seen by the callee, which I’ll talk about here now.

The reason you might want to do this is because you have multiple PSTN phone numbers that ring the same SIP phone. The obvious way to solve this would be to use

exten => _X.,n,SipAddHeader(To: "123456" <sip:123456@server>)
exten => _X.,n,Dial(SIP/${EXTEN})

, but that doesn’t work because SipAddHeader doesn’t overwrite existing headers, it only adds new ones. The Snom forum mentions a hack using the Diversion header, assuming your phone does indeed display that. A much nicer way is the following:

exten => _X.,n,Dial(SIP/${EXTEN}!123456)

. The number after the exclamation mark is simply what Asterisk uses as the local part when it composes the To URI. This features is not well-documented, but from the code I guess it was introduced in Asterisk 1.6. Asterisk 1.8’s (and higher) chan_sip.c gives a short explanation:

 *  SIP Dial string syntax:
 *       SIP/devicename
 *  or   SIP/username@domain (SIP uri)
 * or   SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
 * or   SIP/devicename/extension
 *  or   SIP/devicename/extension/IPorHost
 * or   SIP/username@domain//IPorHost
 * and there is an optional [!dnid] argument you can append to alter the
 *  To: header.