alpercoskun
commented
1 year ago Just note that to help some others which has same/similar problems:
- Upgrading from 1.10.7 to 1.10.9 didn't solve the issue.
- Adding a 500 ms delay just before "play_and_get_digits" function solved the issue!
Any clue what happens here?
Describe the bug For an originated call from Freeswitch:
{sip_cid_type=none,sip_h_P-Asserted-Identity=sip:+380635**@10...145;user=phone,origination_uuid=68b59e96-a707-4619-xxxx-99b94169df08,ignore_early_media=true,sip_ignore_183nosdp=true,sip_h_P-Early-Media=supported,origination_caller_id_number=sip:+380635**@10...145;user=phone}sofia/gateway/${distributor(MSC ${sofia(profile external gwlist down)})}/+380638** &park
After originate is successful (answered), we try to "play_and_get_digits":
2023-03-16 23:42:47.682294 98.50% [DEBUG] switch_ivr.c:632 sofia/external/+380638** Command Execute [depth=1] play_and_get_digits(1 1 1 10000 +# /space/voice/ua0001.wav silence_stream://100 collectedDigits .+ 1000)
It immediately finishes playing the file (file is not very short):
2023-03-16 23:42:47.682294 98.50% [DEBUG] mod_commands.c:5169 (sofia/external/+380638000894) State Change CS_CONSUME_MEDIA -> CS_EXECUTE 2023-03-16 23:42:47.682294 98.50% [DEBUG] switch_ivr_play_say.c:1556 Codec Activated L16@8000hz 1 channels 20ms 2023-03-16 23:42:47.682294 98.50% [DEBUG] switch_ivr_play_say.c:2001 done playing file /space/voice/ua0001.wav
On pcap file, we don't see any RTP packets from our side. On the handset, we don't hear anything. What can be the problem?
It randomly happens, and occurs in 1 of 5 calls.
To Reproduce Steps to reproduce the behavior:
Expected behavior RTP traffic should start when we try to call "play_and_get_digits" function, and the handset should hear the voice file.
Package version or git hash
Trace logs
test_1603_2.pcap.zip fs.log.zip