Transcoding failure?

I am getting problems with one way audio on outbound calls. (I’m in the UK), and I think it is a transcoding issue.

If I make a call the internal part uses uLaw, and the SIP trunk bit uses aLaw, and I get one way audio. However, if I remove the uLaw option from the Peer details in the Trunk settings on the pbx it all works fine.

This sounds like a transcoding problem to me? Does FreePBX do transcoding? Or should I just remove all support for uLaw in the PBX?

If someone could educate me on this matter I’d appreciate it.

What codecs are active in your Asterisk SIP settings (freePBX-settings)?

Ahg, good question. ulaw, alaw, gsm, g726, g722 in that order.

Do you have any extension settings regarding the codec, like this


Yes, I’m using the Sangoma s305 phones and they all seem to be defaulted to use ulaw as the first codec. So, they are PCMU,PCMA,G722,iLBC,GSM_FR, G726-32 int hat order.

and the outgoing trunk settings? Mine look like this…you might have to add ulaw if supported in the U.K. … I always put g722 first in the Asterisk SIP codec list…


EDIT: I meant the freePBX extension settings…any codec limitations?

Sorry, realised as soon as I’d posted the last message!.. So no, nothing in Disallowed Codecs, or Allowed Codecs in the FreePBX settings.

Do these settings override everything else?

Asterisk should be able to transcode between ulaw and alaw without issue. We can troubleshoot this if desired.

However, transcoding causes a (very slight) quality reduction and increases the server load, so you should avoid it where possible. Set up your devices and Asterisk to have alaw as the first priority codec. Normal calls will then not be transcoded.

However, it’s possible that when you call a ulaw country such as the US, your trunking provider might supply (and require) only ulaw audio, if they don’t have the ability to transcode. If you make such calls, you will need to find out why transcoding in Asterisk is not working.

I am not sure…I am just a user…I always check the trunk, the extensions and the Asterisk SIP settings…dont know exactly which overrides which :wink:

How do you configure your phones? Is there a web-gui? Maybe you have to change the codec order of your phones too…

Thank you Stewart, I was coming round to that conclusion myself. I have managed to force a call to aLaw on both sides, so no transcoding, and there is a noticeable improvement in quality.

When you say set up the devices for alaw, you mean the handsets themselves, or do you mean in the settings in FreePBX in each extension?

I think you have to change the handsets. Do you use Endpoint manager or the phone web-gui?

I use the web-gui for the phones. There’s only 15 or so, so it’s not a pain to set them individually.

Test it on one phone first… :wink:
In Germany/Austria I always put g722 first, followed by alaw (Asterisk SIP, Trunk and extensions)…but I dont know about the U.K.

Thanks Mr. Darwin, much appreciated!

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