I am running the current 1.4 release of Asterisk, it has a SIP config registered to my Voipfone account. I map UDP and TCP port 5060 to my asterisk server inside my NAT.
A lot of stuff is working correctly.
I am testing inbound, by calling my Voipfone 056 number from a landline.
The internal extension rings.
I get through to my voicemail.
As soon as I start recording a message, the Asterisk CLI gets flooded with this warning :
[Dec 9 16:09:34] WARNING: rtp.c:891 ast_rtcp_read: RTCP Read too short
Only the first second or two is recorded.
If I pick up the extension instead of letting it go to VM, audio seems fine.
Has anyone got an idea what is going wrong?
- Information & Feedback
- - Voipfone Status Page
- - News & Product Announcements
- - Feedback & Bug Reports
- Help and Support
- - Voipfone General Information and FAQs
- - Hardware Support
- - - Phone Setup Guides
- - - - Snom Phones
- - - - Siemens Phones
- - - - Linksys Phones
- - - Adaptor Setup Guides
- - - Network Setup Guides
- - Softphone Support
- - - Softphone Setup Guides
- - - Download Zoiper Softphone
- - Voipfone Virtual PBX Support
- - Routers, Firewalls & NAT
- - Broadband Support
- - - Getting Started, Router Config
- - - Voipfone Broadband FAQ
- - Asterisk and other hardware PBX Support
- - CRM (Customer Relationship Management) Integration
- - Voipfone Address Book Dialler (TAPI Plugin)
- - Browser Dialler Plugins
- Voipfone Community
- - Voipfone General Discussion Forum
- - Suggestion for New Voipfone Products, Services and Features
- - - Vote for the Voipfone features you want us to develop
- - Voipfone Trading Post
- - Virtual Assistants Forum
- - Transcription Services
Who is online
Users browsing this forum: No registered users and 0 guests