So are you testing the soft phone from the same pc as pjsua? How many NICs are in the PCs and what is their ip configuration? On 17/03/2013, at 7:35 AM, Ted Gerold <tedgerold at icloud.com> wrote: > So its easier to get the wireshark data via my local network (which I'm also using to test > and getting the exact same results). Using my soft phone (which works great) I get the following > from wireshark: > > 12 7.977004000 10.0.1.4 64.2.142.214 SIP/SDP 1081 Request: INVITE sip:6028675309 at outbound.vitelity.net;transport=UDP | , with session description > 13 8.032728000 64.2.142.214 10.0.1.4 SIP 552 Status: 100 Trying | > 14 8.849383000 64.2.142.214 10.0.1.4 SIP/SDP 895 Status: 183 Session Progress | , with session description > 15 8.851479000 64.2.142.214 10.0.1.4 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x58AA0F24, Seq=12346, Time=2880 > 16 8.851546000 10.0.1.4 64.2.142.214 ICMP 190 Destination unreachable (Port unreachable) > 17 8.872092000 64.2.142.214 10.0.1.4 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x58AA0F24, Seq=12347, Time=3040 > > That looks good to me except #16. Every call I make that happens once and then it continues as normal. Maybe the issue with PJSUA > is that it hangs on that part and doesn't know what to do. When I test using PJSUA there is a hang for a few moments after it starts > the call before it proceeds to display logging output again. Course in PJSUA we aren't seeing the 100 and 183 codes so thats probably > not it. > > This is very frustrating. :( > > > > On Mar 16, 2013, at 4:20 AM, Omar Hussein <omarh2812 at gmail.com> wrote: > >> Ok. I think it would be good to check the signalling of the working >> softphone with wireshark. Then you will be able to see if the softphone >> registers to the provider, what IP address it is sending to etc. When you >> test the softphone is it from the same PC? >> >> It is just strange how the log does not show any response for the INVITE. >> Normally you should see 100 Trying, 180 Ringing responses for INVITE. The >> fact that the other phone rings means that you are sending to the correct >> server. >> >> Your PC is listening for SIP messages on 207.181.98.7:5060. The via header >> in the INVITE has this address so the provider should be sending the >> response to this address. The address is a public IP and so there is no >> NAT. >> >> Do you have a firewall blocking the response packets? Iptables etc. >> >> Regards, >> Omar >> >> >> >> -----Original Message----- >> From: Ted Gerold [mailto:tedgerold@xxxxxxxxxx] >> Sent: Saturday, 16 March 2013 6:29 PM >> To: Omar Hussein >> Cc: 'pjsip list' >> Subject: Re: Problems with Outbound Calls >> >> Same results :( >> >> >> On Mar 16, 2013, at 12:12 AM, Omar Hussein <omarh2812 at gmail.com> wrote: >> >>> 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 >
