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 >>
