[mmartinez]

Hi Guys,

I have an asterisk@home server, I have been able to configure it with FWD thorugh IAX2, stanaphone through sip, I have Ipkall working fine as well as outgoing calls to fwd, sipphone and 1800 numbers.
WHat I don't seem to be able to get working is incoming calls to sipphone and inphonex.

Has anyone out there been able to do get inphonex and sipphone incoming calls.

The message that I get when I dial the sipphone number is that the person you are trying to reach has not setup voicemail (although the asterisk box tells me the number is registered).

In the case of inphonex I get a similar message.
Thanks in advance for any help .

Marcias

[mberlant]

I can't help with the inphonex, but here is the register line for sipphone:

register=17475551234:999999:17475551234@proxy01.si pphone.com/17475551234

I did find a disappointing peculiarity necessary to make it work. The routing "extension number", /17475551234 must match your SIPphone number. I was disappointed that I needed to make an exception to my numbering plan for this one provider.

[chandave]

Also, in sip.conf, your section for [proxy01.sipphone.com] should include:
insecure=very

They don't send authentication information in their INVITEs.

[osalmon]

I used the register string mentioned and I not can receive calls from sipphone, I still cannot make outgoing sipphone calls.

Any help would be apreciated!

Owen

[chandave]

Originally Posted by osalmon

I used the register string mentioned and I not can receive calls from sipphone, I still cannot make outgoing sipphone calls.

Any help would be apreciated!

Owen

It would really help if you would supply the following information:

• A copy of your sip.conf that contains the registration entry you are using for sipphone (with an obscured username/password). Also, the sip.conf should include the peer section related to your sipphone connection (ie [proxy01.sipphone.com] section). The peer section should be sanitized of real account names and passwords.
• A copy of your extensions.conf that includes the dial context specified by the peer section listed in the sip.conf.
• A "sip debug peer proxy01.sipphone.com" log from your Asterisk console. We would need the list for attempts to register with proxy01.sipphone.com (you'll see a "REGISTER siproxy01.sipphone.com" and some replies). For incoming calls we would be looking for incoming invites ("INVITE 1747XXXXXXX@Your_IP_ADDR"). For outgoing calls, we would be looking for outgoing invites ("INVITE 1747XXXXXXX@proxy01.sipphone.com").

With that type of information, we can see if:

1. Your extension specified in your Asterisk register properly corresponds with an extension found in the context specified by the [proxy01.sipphone.com] peer definition.
2. You've properly turned off inbound authentication of INVITES from proxy01.sipphone.com.
3. You've properly declared authentication information for outbound INVITES being sent to proxy01.sipphone.com.