I tried the following setups. Although there is a minor "ringback" issue that I haven't found any solution yet.
1.) ATA->CiscoNAT->Asterisk->SER+RTPProxy->Cisco2600 2.) ATA->CiscoNAT->SER+RTPProxy1->Asterisk->SER+RTPProxy2->Cisco2600 I cannot remember if * can directly connect to Cisco2600. I know I had problems initially with it, that's why I installed the SER, and since now I'm focusing to solve the ringback issue, I didn't have time to take out SER out of my equation. ----- Original Message ----- From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 16, 2004 7:33 PM Subject: Re: [Asterisk-Users] SER & Asterisk > Thanks guys, thought SER had to 'register' to be able to use > any Asterisk contexts. > But just defining a new entry in the sip.conf with just context & ip worked! > > But now i'm stumbling on another problem.. Asterisk seems to want > to send the SIP udp packets directly to the SIP clients. > In the case of a SIP user/client behind a NAT, this obviously doesn't > work. > > SER is configured to use the wonderful RTPProxy + SER nathelper module, > and this works flawlessly (using the rewritehostport function). > > But when I try to call a phone number on the PSTN network from a SIP > client behind NAT, SER sends the invites to Asterisk, and Asterisk > makes an outbound call to the phone number, the phone rings, but when > the pstn user picks up the phone, no sound, and after a while (couple of > seconds), the call is dropped. > Asterisk spews out the following warning, > chan_sip.c:471 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 29898 (Response) > Tried searching on the voip-info wiki and mailinglists, but didn't find > a way to force Asterisk to use a SIP proxy/SER. > > Any ideas ? > > > On Fri, Jan 16, 2004 at 12:12:14AM -0800, Chris Albertson wrote: > > > > Yes, you can keep non-authorized SIP callers from accessing the > > PSTN by setting up the .conf file "correctly" as below > > but you can also > > run a fire wall on the box that Asterisk runs on. Firewall off > > SIP ports except for if they come from your SER server. > > > > > > --- Fran Boon <[EMAIL PROTECTED]> wrote: > > > [ser] > > > context=sip-legal > > > host=y.y.y.y ; IP address of SER > > > > > > Se this Wiki page for more flesh of my (not yet fully working!) > > > configs: > > > http://voip-info.org/wiki-Asterisk+cisco+FXO > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
