Audio Problem!

FreePBX Tips and Tricks
1

Hello,

I am getting below error messages from last few days from my asterisk server(1.8 version).

ERROR[17287]: utils.c:1215 ast_careful_fwrite: fwrite() returned error: Broken pipe

ERROR[18634]: utils.c:1215 ast_careful_fwrite: fwrite() returned error: Connection reset by peer

WARNING[2247]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response

ERROR[18652]: utils.c:1215 ast_careful_fwrite: fwrite() returned error: Connection reset by peer

And also some times outgoing call is talking longer time to get phone ring. When the conversation starts often I hear very badly the person speaking and after a while the audio come back normal.Seems that the person is still hearing me well though.

Please help in resolving audio issues.

2

First, please supply complete information, including a summary of your network topology and FreePBX version.

The errors you are receiving are due to networking issues.

3

I am using Asterisk 1.8 and FreePbx 2.9… My topology is… Asterisk is installed on Ubuntu 12.04, available on my local network, router configured with my static IP. I have forwarded all ports required for asterisk server.

1.My main problem is some time it takes long time to ring.
2. Often when the call starts I hear badly the voice of the other (distortion) after a while (15-20 sec.) the audio is nice.