No or one-way audio during SIP phone – SIP phone calls
This problem usually means that one or both phones are behind a NAT firewall. Unfortunately, unless the RTP Proxy is turned on or certain “smart” SIP phones/NAT routers are used, there is no way to guarantee proper performance in such cases (see NAT Traversal section for details).
One-way audio during SIP phone – Cisco gateway call
This problem can occur if the Cisco GW is not configured properly. Please check that the GW contains the following in its IOS configuration:
sip-ua nat symmetric check-media-src
I have problems when trying to use SIP phone X made by vendor Y with PortaSIP
Unfortunately, not all of the many SIP phones available on the market today fully comply with the SIP standard, especially low-end products. We use Sipura/Linksys 941 as a reference phone, and the Sipura/ Linksys – PortaSIP combination has been thoroughly tested.
If you are unable to get your third-party vendor SIP phone working properly, follow the instructions below:
- Make sure the phone has been configured properly, with such parameters as account ID, password, SIP server address, etc. Consult the product documentation regarding other configuration settings.
- Check the PortaSIP and PortaBilling logs to ensure that there is not a problem with the account you are trying to use (for example, an expired or blocked account).
- Connect the Sipura/Linksys phone or ATA to the same network as your SIP phone. If possible, disconnect the SIP phone and use the same IP address for the Sipura/Linksys as was previously used by the third-party SIP phone. Configure the Sipura/Linksys with the same account as was used on your third-party SIP phone.
- Try to make test calls from the Sipura/Linksys.
- If you have followed the preceding steps and the problem disappears, then this means your third-party vendor SIP phone is not working according to the standard. Contact the vendor of the SIP phone, and describe the problem.
- If this problem with the Sipura/Linksys persists, contact firstname.lastname@example.org. Provide a full description of the problem, the ID of the account being used for testing, and the relevant parts of the sip.log and porta-billing.log