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
