Hi Bill,
thanks for the hint.
I will check it.
Filip
Sent from my iPhone
> On 04.07.2014, at 15:15, Bill Gardner <billg at wavearts.com> wrote:
>
> Pjsip will send bye to contact address sent in invite from asterisk. Check that. Bill
>
> Sent from my iPhone
>
>> On Jul 2, 2014, at 2:49 PM, Filip Planinsky <filip at fonetize.com> wrote:
>>
>> Hi There,
>>
>> I will try to bump up this question one more try.
>> Generally, if you use TCP transport on port 5060 - there is no problem.
>> However, if you try to change the port, the you get into troubles.
>> Details are below in my previous emails.
>>
>> If someone could help - will be much appreciated.
>>
>> Thanks,
>> Filip
>>
>>> On Jun 9, 2014, at 7:22 PM, Filip Planinsky <filip at fonetize.com> wrote:
>>>
>>> Hi Guys,
>>>
>>> any hints on how to change the remote SIP port for PJSIP?
>>> My Asterisk is listening on TCP port 6533 and it seems that PJSIP is having trouble to work with it in some cases.
>>> I can't use UDP - because of the iOS App, which requires TCP in order to run in background.
>>>
>>> Details are below.
>>> If you could help - it will be greatly appreciated!
>>>
>>> Thanks,
>>> Filip
>>>
>>>
>>>> On Jun 7, 2014, at 12:12 AM, Filip Planinsky <filip at fonetize.com> wrote:
>>>>
>>>> Hello,
>>>>
>>>>
>>>> I am trying to change the default SIP port 5060 to some other port.
>>>> I am using TCP, because I use PJSIP on an iOS App, which must run in background mode. And TCP is required by iOS for this to happen.
>>>>
>>>>
>>>> I tried this and it seemed to be working ok.
>>>> I have added to pjsua_config struct:
>>>>
>>>> cfg.outbound_proxy[0] = pj_str("sip:XXX.XXX.XXX.XXX:6533;transport=tcp");
>>>>
>>>> before execution the pjsua_init(&cfg, &log_cfg, NULL) function
>>>>
>>>> By setting the outbound proxy to pjsua_config struct I achieved 2 things:
>>>> 1. adding transport=tcp to all request
>>>> 2. changing the remote port from 5060 to 6533
>>>>
>>>> I also added the port to the registrar URI in pjsua_acc_config.reg_uri="sip:XXX.XXX.XXX.XXX:6533"
>>>>
>>>> This way the App is able to:
>>>> 1. Send REGISTER
>>>> 2. Receive OPTIONS from Asterisk
>>>> 3. Receive INVITE from Asterisk and establish a call, which has audio in both directions
>>>> 4. Obviously, respond to Asterisk with SIP/200 OK and ACK, when needed.
>>>>
>>>>
>>>> However, I did find that the App is not capable to send BYE requests.
>>>> I use the App only to answer calls - by design it doesn't have the ability to make calls - so I do not know if it is capable to send INVITE requests to the Asterisk.
>>>>
>>>> 23:18:16.307 pjsua_call.c !Call 0 hanging up: code=0..
>>>> 23:18:16.308 tcpc0x15ba6c14 ....TCP client transport created
>>>> 23:18:16.309 tcpc0x15ba6c14 ....TCP transport 192.168.1.11:61873 is connecting to XXX.XXX.XXX.XXX:5060...
>>>> 23:18:16.309 pjsua_core.c ....TX 476 bytes Request msg BYE/cseq=6547 (tdta0x15abf400) to TCP XXX.XXX.XXX.XXX:5060
>>>> 23:18:16.364 tsx0x15ba6064 Failed to send Request msg BYE/cseq=6547 (tdta0x15abf400)! err=120061 (Connection refused)
>>>>
>>>> It tries to send the BYE request to poet 5060, which is not possible as the Asterisk doesn't listen to this port.
>>>> I think the problem is that it says "TCP client transport created" and therefore uses the default port.
>>>>
>>>>
>>>> What is the proper way to change the TCP port from 5060 to any other port?
>>>>
>>>>
>>>> Thanks,
>>>> Filip
>>
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