Choppy sound for few first secounds

Hi there,

I’ve been working on it for days, searching web to no avail

We’ve got 2 PBX asterisk servers in Chicago and Warsaw connected via iax2 trunk, the voice quality is good while calling either extension, the problem arises when someone from Warsaw (either hardphone or softphone extension) wants to call Chicago’s PSTN sip trunk, it’s routed via iax2 to chicago pbx and via sip trunk reallinx to external numbers

Warsaw - iax2 -> Chicago - sip trunk -> external US numbers

For the first 5-15 seconds the sound quality on Warsaw phone is horrible, I guess it might be related to comfort noise frames as the warsaw asterisk CLI shows a lot of those warnings (usually the more the warning the longer the choppy sound is) US caller has perfect voice quality.

[2016-12-06 17:26:45] WARNING[27889][C-0000005c]: chan_sip.c:7299 sip_write: Can’t send 10 type frames with SIP write

I’ve tried both alaw and ulaw codecs as well as gsm. I know asterisk does not support comfort noise frames, but is it something that can be either filtered or changed? I cannot disable CNF since I have no control over external sip trunk provider like reallinx (maybe it can be done on the chicago side pbx?)

Since I’m not VoIP expert please let me know what details you need to know.