Choppy lines or call breaks on Asterisk.
Choppy line is a common problem when we use SIP signaling. The main reasons for the issue are Bandwidth, Codec, Lots of SIP Trunks registered, and Jitter.
Basic checklist for Choppy Lines
Check if Codec ULAW, ALAW or G729 is allowed on your SIP Trunks. If allowed try to use any single codec to avoid choppy lines. For example if you are using G729 then remove ULAW parameter from DIDforSale SIP Trunks. If you are using ULAW then remove g729.This can be done by adding the below parameters on DIDforSale SIP trunk.
allow=g729 (Use either one of the codec you want)
Next step is to check whether you have lots of unwanted SIP Trunks registered on PBX, as this can cause bandwidth utilization. Remove any SIP Trunk that you don’t need.
Another contributing factor could be Jitter buffer.You can enable jitter buffer to avoid this issue. For enabling jitter buffer go to
and search for jenable and jbforce and uncomment the below parameters.
If still you are facing issues take a tcpdump of rtp ports and send to us.
Command to take tcpdump on linux
tcpdump -i etho -p portrange 10000-20000 > rtpcap.pcap
and send this pcap file to firstname.lastname@example.org