One way / garbled audio


I am new on this forum and also quite new to FreePBX, so please bear with me if I am asking a FAQ.

I have a FreePBX box running with a PJSIP trunk to may SIP provider 1und1 (Germany).
I also created two PJSIP extensions, 100 and 200, which i can access with my two phones running Acrobits Groundwire. Woks fine so far, extensions can communicate with each other and can also be called from outside via Phone1 -> Sipgate -> 1und1 ->- FreePBX -> Phone2
Problems arise when I call from outside via Mobile Network. Phone1 uses German Telekom, Phone2 uses 1und1, which in turn uses Vodaphone (I believe).
In order to test things, I setup the inbound route to Voicemail / Ext. 100 with messages emailed to me. Phone2 works fine, I can hear recorded message from my mail program.
When dialing in with phone1, I do net hear anything, no busy message, no recording prompt. If I blindly speak a message anyhow, it is recorded and emailed to be, but it is totally garbled.
I this a codec problem?
I tried to upload the message file, but as a new user I am not allowed to do so. You can access it here:

Edit: I just learned that new Users cannot put links either. I will try as text:


Thanks for any help,


I forgot to post my codec configuration:

ulaw, alaw, gsm, g726, g722

PBX Version:
PBX Distro: 12.7.8-2008-1.sng7
Asterisk Version: 16.13.0


There is a log of Smartphone calling into conference, entering PIN and recording name, then hangup. During this whole process, I did not hear anything on the phone. I watched log in order to know when to enter PIN and record name. HTH

In an earlier experiment using the conference, I could speak into the phone, and other participants could hear me clearly, not garbled, so this does not look like a codec issue.
Note that whatever phone is use, pbx always communicates with the 1und1 sip server.


This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.