I can't seem to get my outgoing connections to work with IConnecthere. At one time it did with v1.0 I can register and receive calls just fine. But can't make them. Ultimately, the trace ends with a "400 Bad Request" error when you do a SIP debug. Has anyone got it to work with v1.2? Don't know if it is related to the version or not since I haven't worked with it in a while and didn't test outgoing before upgrading. -- ___________________________________________________ Play 100s of games for FREE! http://games.mail.com/
On Wednesday 07 December 2005 11:10, John Voss wrote:> I can't seem to get my outgoing connections to work with IConnecthere. At > one time it did with v1.0 > > I can register and receive calls just fine. But can't make them. > > Ultimately, the trace ends with a "400 Bad Request" error when you do a SIP > debug. > > Has anyone got it to work with v1.2? Don't know if it is related to the > version or not since I haven't worked with it in a while and didn't test > outgoing before upgrading.Im having no issues with my outgoing calls but my incoming call registration keeps locking up. it seems after awhile it stops sending reregistration packets. my extentions.conf has ; ****************** Dial Out iconnecthere *************** exten => _1NXXNXXXXXX,1,SetCallerID(ph number) exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@iconnect,60,r) exten => _1NXXNXXXXXX,3,Congestion exten => _NXXXXXX,1,SetCallerID("Dennis Gilmore" <ph number>) exten => _NXXXXXX,2,Dial(SIP/1309${EXTEN}@iconnect,60,r) exten => _NXXXXXX,3,Congestion exten => _61.,1,SetCallerID(ph number) exten => _61.,2,Dial(SIP/${EXTEN}@iconnect,60,r) exten => _61.,3,Congestion in sip.conf [iconnect] ; for routing calls outbound to the PSTN numbers via iconnecthere ; (aka deltathree) type=friend secret=<pass> username=<userid> CallerID="Dennis Gilmore <ph number>" authname=<ph number> host=natrelay.deltathree.com nat=yes dtmfmode=rfc2833 canreinvite=no disallow=all allow=ulaw allow=gsm allow=alaw #allow=G726 Dennis
On Wed, 7 Dec 2005 13:32:09 -0600 Dennis Gilmore wrote:>Im having no issues with my outgoing calls? but my incoming call registration >keeps locking up.? it seems after awhile it stops sending reregistration >packets. >my extentions.conf has >; ****************** Dial Out iconnecthere *************** >exten => _1NXXNXXXXXX,1,SetCallerID(ph number) >exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@iconnect,60,r) >exten => _1NXXNXXXXXX,3,Congestion > >exten => _NXXXXXX,1,SetCallerID("Dennis Gilmore" <ph number>) >exten => _NXXXXXX,2,Dial(SIP/1309${EXTEN}@iconnect,60,r) >exten => _NXXXXXX,3,Congestion > >exten => _61.,1,SetCallerID(ph number) >exten => _61.,2,Dial(SIP/${EXTEN}@iconnect,60,r) >exten => _61.,3,Congestion > >in sip.conf >[iconnect] >; for routing calls outbound to the PSTN numbers via iconnecthere >; (aka deltathree) >type=friend >secret=<pass> >username=<userid> >CallerID="Dennis Gilmore <ph number>" >authname=<ph number> >host=natrelay.deltathree.com >nat=yes >dtmfmode=rfc2833 >canreinvite=no >disallow=all >allow=ulaw >allow=gsm >allow=alaw >#allow=G726 > >DennisYour SIP.conf file looks much different than mine. I'll give it a try. [iconnect] type=friend secret=<pass> username=<userid> host=213.137.73.140 ;sipauth.deltathree.com permit=213.137.73.140/255.255.255.0 permit=208.170.168.0/255.255.255.0 disallow=all context=incoming allow=gsm allow=ulaw allow=alaw allow=G726 insecure=very nat=Yes canreinvite=no I don't know what your register line looks like in your SIP.conf. This is mine. register => <ph number>:<pass>:<userid>@213.137.73.140:5060 I was unable to receive calls until I added the insecure=very line. Hope it helps JV Still working on my issue -- ___________________________________________________ Play 100s of games for FREE! http://games.mail.com/
I finally got my issue resolved. It actually had nothing to do with my SIP.conf file. The problem was how I was trying to set the callerid in my extensions.conf file. Anyway, do you have other voip providers that are working? Do incoming calls work at all prior to timing out? Are you NAT'ed, or are you behind a broadband router? If you are NAT'ed and you haven't already configured * for it you may have issues like this. (from http://www.voip-info.org/wiki/view/tips) When sip is behind a NAT do not forget to specify: in sip.conf [general] nat=yes externip = X.X.X.X fromdomain = yourdomain.com localnet = 192.168.X.0/255.255.255.0 I choose to use externhost = yourdomain.com instead of externip since most broadband providers use DHCP and you address can change. I registered a domain with no-ip.com (it's free) and use that in place of yourdomain.com. They also have a client that you can load on a windows box that keeps track of your external ip and updates your domain if your ip ever changes. That way I don't have to worry about it. You may need to add srvlookup to your sip.conf to allow name resolution if you use externhost instead of externip: srvlookup=yes You will also need to setup port forwarding on your broadband router/firewall: (from http://www.voip-info.org/wiki/view/NAT+and+VOIP) SIP signaling: Ports 5060 to 5070 RTP audio: Ports 8766 to 35000 I only forward the following listening ports (read comments in the wiki for this) SIP signaling: Port 5060 RTP audio: Ports 1000 - 2000 (you can restrict this in RTP.conf) Try these wiki pages for more info: http://www.voip-info.org/wiki-Asterisk+config+rtp.conf http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions http://www.voip-info.org/wiki/view/NAT+and+VOIP ----- Original Message ----- From: "Dennis Gilmore" To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] RE:IConnecthere dial out problems Date: Wed, 7 Dec 2005 21:49:01 -0600 Once upon a time Wednesday 07 December 2005 8:42 pm, John Voss wrote:> Your SIP.conf file looks much different than mine. I'll give it a try.Hope mine helped> [iconnect] > type=friend > secret= > username= > host=213.137.73.140 ;sipauth.deltathree.com > permit=213.137.73.140/255.255.255.0 > permit=208.170.168.0/255.255.255.0 > disallow=all > context=incoming > allow=gsm > allow=ulaw > allow=alaw > allow=G726 > insecure=very > nat=Yes > canreinvite=no > > I don't know what your register line looks like in your SIP.conf. This is > mine. > > register => ::@213.137.73.140:5060 > > I was unable to receive calls until I added the insecure=very line.mine is register => ::@natrelay.deltathree.com i can receive incomming calls for a little while after a reload but after some timeouts incomming calls stop -- Dennis Gilmore, RHCE http://www.ausil.us << 2.dat >> -- ___________________________________________________ Play 100s of games for FREE! http://games.mail.com/