What about firewall on the machine? It has to be something like that since there is no response from testing on your public server also. On 17/03/2013, at 6:06 PM, Ted Gerold <tedgerold at icloud.com> wrote: > Nope, no response messages. results remain the same. > > > On Mar 17, 2013, at 12:05 AM, Omar Hussein <omarh2812 at gmail.com> wrote: > >> Is there still no response messages in the log with both reg_uri not set and public_addr set to your nat router public address when testing on private network. >> >> >> >> On 17/03/2013, at 4:47 PM, Ted Gerold <tedgerold at icloud.com> wrote: >> >>> >>> Same results using public_addr. tried it on both test systems. this is rough :) >>> >>> On Mar 16, 2013, at 10:30 PM, Ted Gerold <tedgerold at icloud.com> wrote: >>> >>>> 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 >>>> >>>> >>>> _______________________________________________ >>>> 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 >
