Opened 3 years ago
Last modified 3 years ago
#185 new defect
QuteCom-2.2-RC3 Skips words and phrases when using Text to Speech
| Reported by: | amuthiga | Owned by: | vadim |
|---|---|---|---|
| Priority: | major | Milestone: | |
| Component: | 3rd party libs | Version: | |
| Keywords: | Cc: |
Description
Hello,
I have a short IVR system building and testing on an Asterisk-1.6 box with swift text to speech engine. When using the QuteCom I hear the welcome message it then skips the next message and I can hear part of the third message. On the asterisk Verbose screen asterisk say that the text was read out. I tried using there sip clients and I can hear the text that is read out.
Attached are part of the Asterisk outputs when using QuteCom and X-lite soft phones respectively. The only difference I can see is that when using QuteCom there is the Notice
[Jan 20 11:32:57] NOTICE[13774]: channel.c:2940 ast_read: Dropping incompatible voice frame on SIP/2004-00000027 of format ulaw since our native format has changed to 0xc0002 (gsm|h261|h263)
The default Audio Codec on The QuteCom is gsm/8000. Tried the other codecs with the same results.
Hope someone can help me fix this. Than you in advance.
Attachments (1)
Change History (4)
Changed 3 years ago by amuthiga
comment:1 follow-up: ↓ 2 Changed 3 years ago by vadim
comment:2 in reply to: ↑ 1 Changed 3 years ago by amuthiga
Replying to vadim:
Can You please wireshark capture of SIP and RTP data on machine running QuteCom
Hello.
Got QuteCom to work but had to change on the user sip information on the asterisk sip.conf. Changed allow=all to allow=gsm. It seems like QuteCom fails to negotiate for the right codec like the other sip phones.
Regards,
Albert.
comment:3 Changed 3 years ago by chris-mac
- Milestone QuteCom 2.2-RC2 deleted
- Version 2.2-RC3 deleted

Can You please wireshark capture of SIP and RTP data on machine running QuteCom