Problems with Outbound Calls

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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
>> 
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>> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> 



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