Hi, thanks for the hint, but I don't have issue with Asterisk. I use tcpbindaddr=192.168.10.10:7365 and Asterisk seem to work fine. Also, PJSIP seems to work fine (using the proper TCP port) until it is time for the BYE message. Then it creates a new TCP transport on port 5060 and tries to send the message to 5060. I do not know why PJSIP creates new TCP transport at that moment with the default port. Thanks, Filip On Jul 3, 2014, at 6:24 PM, Dennis Guse <dennis.guse at alumni.tu-berlin.de> wrote: > Check the documentation: http://svn.digium.com/svn/asterisk/trunk/configs/sip.conf.sample > tcpenable=no ; Enable server for incoming TCP connections (default is no) > tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) > ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) > > > --- > Dennis Guse > > > On Thu, Jul 3, 2014 at 6:08 PM, Yosi Taguri <yosi at taguri.com> wrote: > Aterisk is hardcoded into 5060. You may forward other ports to it but basically it will create a contact header with 5060. We had to change asterisk code base in order to really work with another port. This sucks! > ? > Sent from Mailbox > > > On Thu, Jul 3, 2014 at 7:05 PM, pjsip-request at lists.pjsip.org <pjsip-request at lists.pjsip.org> wrote: > Send pjsip mailing list submissions to > pjsip at lists.pjsip.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > or, via email, send a message with subject or body 'help' to > pjsip-request at lists.pjsip.org > > You can reach the person managing the list at > pjsip-owner at lists.pjsip.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of pjsip digest..." > > > Today's Topics: > > 1. Re: Changing PJSIP to work over TCP on port different than > 5060 (Filip Planinsky) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 2 Jul 2014 20:49:38 +0200 > From: Filip Planinsky <filip@xxxxxxxxxxxx> > To: pjsip at lists.pjsip.org > Subject: Re: Changing PJSIP to work over TCP on port different > than 5060 > Message-ID: <B6AC1124-9129-4C6F-9DF7-38B2B66E166E at fonetize.com> > Content-Type: text/plain; charset="us-ascii" > > 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 > >> > >> > >> > >> > >> > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20140702/e0dce27f/attachment-0001.html> > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > ------------------------------ > > End of pjsip Digest, Vol 83, Issue 2 > ************************************ > > > > _______________________________________________ > 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 -------------- next part -------------- An HTML attachment was scrubbed... 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