That was it!! There was a firewall issue with both machines which I was just now able to resolve: # ./sip sip:6028675309 at outbound.vitelity.net 0 00:26:23.684 os_core_unix.c !pjlib 2.1-svn for POSIX initialized 00:26:23.693 sip_endpoint.c .Creating endpoint instance... 00:26:23.697 pjlib .select() I/O Queue created (0xc5fc40) 00:26:23.699 sip_endpoint.c .Module "mod-msg-print" registered 00:26:23.702 sip_transport. .Transport manager created. 00:26:23.705 pjsua_core.c .PJSUA state changed: NULL --> CREATED ############## Call State on slot '-1': CALLING [code = 1] Press 'h' to hangup all calls, 'q' to quit ############## Call State on slot '-1': EARLY [code = 3] ############## Call State on slot '1': CONNECTING [code = 4] ############## Call State on slot '1': CONFIRMED [code = 5] ############## Call State on slot '1': DISCONNCTD [code = 6] wh00p! Thanks SO MUCH for the help. Sorry it was something so stupid. On Mar 17, 2013, at 12:17 AM, Omar Hussein <omarh2812 at gmail.com> wrote: > 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 >>
