One way / garbled audio

(Heinrich) #1


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,


(Heinrich) #2

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


(Communication Technologies) #3

(Heinrich) #4

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.