No probs. glad it's now working for you. On 17/03/2013, at 6:47 PM, Ted Gerold <tedgerold at icloud.com> wrote: > 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 >
