the 207 was part of the tests from the public server. the 10.0 addresses were part of the tests from my private network server which I had to use to get the wireshark data. I have not come across public_addr before. I am looking into that now. On Mar 16, 2013, at 10:25 PM, Omar Hussein <omarh2812 at gmail.com> wrote: > One thing that I don't understand is that in sip.log > > 18:10:13.837 pjsua_core.c SIP UDP socket reachable at 207.181.98.7:5060 > 18:10:13.837 udp0x1ee6ad0 SIP UDP transport started, published address is > 207.181.98.7:5060 > > This shows that UDP transport is binding to 207.181.98.7:5060 but this is > not your IP address of 10.0.1.10 (or was it at the time of testing in > sip.log). > The published address is what will be sent in the via header, contact header > etc. This can be changed to the public IP address of your NAT router by > setting public_addr in pjsua_transport_config struct. > > Regards, > Omar > > > > -----Original Message----- > From: Ted Gerold [mailto:tedgerold@xxxxxxxxxx] > Sent: Sunday, 17 March 2013 8:26 AM > To: Omar Hussein > Cc: pjsip list > Subject: Re: Problems with Outbound Calls > > 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 >>> > >
