Hi Ted, If the provider does not require sending REGISTER before accepting call requests then don't set reg_uri in pjsua_acc_config struct. Regards, Omar -----Original Message----- From: Ted Gerold [mailto:tedgerold@xxxxxxxxxx] Sent: Saturday, 16 March 2013 3:44 PM To: pjsip list; Omar Hussein Subject: Re: Problems with Outbound Calls Hello Omar, Thanks so much for the reply. Regarding wireshark: Ive not heard of that I will have to look into it to see if I can use it to collect some more data. My provider does not require registration it works based on the IP address of my machine. The domain name they use is: outbound.vitelity.net however they list this on their site: Proxy: sip29.vitelity.net (66.241.99.28) Outbound Proxy: outbound.vitelity.net Since I am using the test sip.c app there is nothing in it that would allow me to test with any proxy. Is there code I could add to the account section that would accomplish the same thing? Thanks for the heads up on the password. It was just a test user so no big deal, easily deleted but should probably not be in there regardless. New sip.c example if needed: http://dl.dropbox.com/u/61083309/www/content/sip.c -Ted On Mar 15, 2013, at 7:40 PM, Omar Hussein <omarh2812 at gmail.com> wrote: > Hi Ted, > > From sip.log it seems that there are no responses to REGISTER / INVITE > messages being sent to your provider. > > Since you have it working with a softphone perhaps a wireshark trace > of the working call compared to the non working call will help. > > Do you know if the provider requires Registration? What is the domain > of your account? Sometimes the domain (sent in To: header etc. ) is > separate to the server IP address you actually need to send packets > to. If that is the case you can configure the --proxy setting in > pjsua.exe sample application. > > P.S. I hope the username/password specified in your sip.c file are not > the real ones. :) > > Regards, > Omar > > > -----Original Message----- > From: pjsip [mailto:pjsip-bounces@xxxxxxxxxxxxxxx] On Behalf Of Ted > Gerold > Sent: Saturday, 16 March 2013 12:19 PM > To: PJSip > Subject: Problems with Outbound Calls > > Hello, > > I have spent a few days now trying to make an output call with PJSAU > but I can not seem to figure out why its not working properly. > > Here is the program I'm using (I got this example from some pj site): > > https://www.dropbox.com/s/ltic4hw5hm8p3xz/sip.c > > Here is my full log file: > > https://www.dropbox.com/s/61qtwoh25jox8d9/sip.log > > At first I was getting the 'cant find sound device' which I expected > as I'm not trying to connect to an external sound device. I simply > want to make a call, play a wav and hangup. So I used: > pjsua_set_null_snd_dev() to fix that. Now I can make the call and the > other end rings just fine but nothing happens after that. No media > state changes, no call_id given. It waits about 5 seconds and > disconnects (pjsua is not disconnecting as again there is no media state change). > Then PJSUA just tries calling again. > > So to summarize: > > o call initiates > o i pick up phone and hear nothing > o no media state change in pjsua > o pjsua calls me 2 more times before quitting. > > > At this point Ive probably spent at least 10 hours on this problem and > am desperate. Any help is much appreciated. I tried to provide all > possible information. > > Also I did test my DID provider Vitelity with a soft phone app and it > worked great. > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
