Charles Wang wrote:

Dear Alexey:

Can you send me your config about ser.cfg & sip.conf, extensions.conf for me to reference?

My UA1(under NAT) can make a call to Asterisk via SER, then Asterisk forward the call to PSTN(behind a CISCO 5300).

But only UA1 can talk to PSTN side, PSTN side can't talk to UA1.

Here are my sip.conf & extensions.conf.

I just want to make a call to PSTN using a UA behind NAT.

Is any config necessary to modify in my goal?

First of all, you need to add some record for incoming calls from CISCO to sip.conf...

[from-cisco] type=peer context=cisco host=61.220.190.243

Next, you need to add [cisco] section to your extensions.conf...

[cisco] exten => _., 1, Answer exten => _., 2, Dial(sip/ser/USER_NAME_HERE) exten => _., 3, Hangup

Note: It's for all calls be forwarded to same user... If you need (as i think) some more efficient scheme, use more extensions in you extensions.conf... Or do semething like that:

[cisco] exten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r)