No the soft phone is on a windows pc (10.0.1.4) and the test sip.c app is on a debian box (10.0.1.10). Both only have one NIC. I have 5060 forwarded to the debian box but no ports are forwarded to the windows box. On Mar 16, 2013, at 2:22 PM, Omar Hussein <omarh2812 at gmail.com> wrote: > 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 >>
