I am using the auto answer function of the pjsua on win32. It answers like it should but I always get 1 audible ring. I would like to not get an audible ring at all. Is there a way to do this with the configuration file or what is the source file to change this behavior.

Danny Brown wrote: > I am using the auto answer function of the pjsua on win32. It answers > like it should but I always get 1 audible ring. I would like to not > get an audible ring at all. Is there a way to do this with the > configuration file or what is the source file to change this behavior. With "--auto-answer 200", pjsua will send 100 and followed immediately by 200. Caller should not play any ring as no 180 is sent. -benny

here is my config file passed to pjsua_vc6d.exe --id=sip:MAST100@192.168.13.100 --registrar=sip:192.168.13.100 --username=MAST100 --realm=asterisk --password=1234 --quality=3 --auto-answer=200 --ec-tail=600 it registers, will accept calls but always rings once before answering. What am I missing? The way I am using this client is; I have an asterisk server running that is scripted to initiate a call from your sip client to another sip device. So, asterisk dials pjsua, once it answers it dials the other end and makes the connection. I do not want the caller (in this case pjsua) to actually ring ( at least not audibly). Thanks for your help. Dan On 10/24/07, Benny Prijono <bennylp@pjsip.org> wrote: > Danny Brown wrote: > > I am using the auto answer function of the pjsua on win32. It answers > > like it should but I always get 1 audible ring. I would like to not > > get an audible ring at all. Is there a way to do this with the > > configuration file or what is the source file to change this behavior. > > With "--auto-answer 200", pjsua will send 100 and followed > immediately by 200. Caller should not play any ring as no 180 is sent. > > -benny > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >

Danny Brown wrote: > here is my config file passed to pjsua_vc6d.exe > --id=sip:MAST100@192.168.13.100 > --registrar=sip:192.168.13.100 > --username=MAST100 > --realm=asterisk > --password=1234 > --quality=3 > --auto-answer=200 > --ec-tail=600 > > it registers, will accept calls but always rings once before > answering. What am I missing? The way I am using this client is; > I have an asterisk server running that is scripted to initiate a call > from your sip client to another sip device. So, asterisk dials pjsua, > once it answers it dials the other end and makes the connection. I do > not want the caller (in this case pjsua) to actually ring ( at least > not audibly). Thanks for your help. Dan Sorry I'm totally confused here. There is no such thing as audible ringing in pjsua! The poor sample application just doesn't have this feature! -benny > On 10/24/07, Benny Prijono <bennylp@pjsip.org> wrote: >> Danny Brown wrote: >>> I am using the auto answer function of the pjsua on win32. It answers >>> like it should but I always get 1 audible ring. I would like to not >>> get an audible ring at all. Is there a way to do this with the >>> configuration file or what is the source file to change this behavior. >> With "--auto-answer 200", pjsua will send 100 and followed >> immediately by 200. Caller should not play any ring as no 180 is sent. >> >> -benny >> >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip@lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -- Benny Prijono http://www.pjsip.org

In this scenario once PJSUA answers, Asterisk would be streaming the audio to PJSUA, so if you hear a ring on PJSUA, it would be coming from Asterisk. Not so? ----- Original Message ----- From: "Benny Prijono" <bennylp@pjsip.org> To: "pjsip embedded/DSP SIP discussion" <pjsip@lists.pjsip.org> Sent: Thursday, October 25, 2007 5:04 AM Subject: Re: [pjsip] prevent audible ring on client when called > Danny Brown wrote: > > here is my config file passed to pjsua_vc6d.exe > > --id=sip:MAST100@192.168.13.100 > > --registrar=sip:192.168.13.100 > > --username=MAST100 > > --realm=asterisk > > --password=1234 > > --quality=3 > > --auto-answer=200 > > --ec-tail=600 > > > > it registers, will accept calls but always rings once before > > answering. What am I missing? The way I am using this client is; > > I have an asterisk server running that is scripted to initiate a call > > from your sip client to another sip device. So, asterisk dials pjsua, > > once it answers it dials the other end and makes the connection. I do > > not want the caller (in this case pjsua) to actually ring ( at least > > not audibly). Thanks for your help. Dan > > Sorry I'm totally confused here. There is no such thing as audible > ringing in pjsua! The poor sample application just doesn't have this > feature! > > -benny > > > > On 10/24/07, Benny Prijono <bennylp@pjsip.org> wrote: > >> Danny Brown wrote: > >>> I am using the auto answer function of the pjsua on win32. It answers > >>> like it should but I always get 1 audible ring. I would like to not > >>> get an audible ring at all. Is there a way to do this with the > >>> configuration file or what is the source file to change this behavior. > >> With "--auto-answer 200", pjsua will send 100 and followed > >> immediately by 200. Caller should not play any ring as no 180 is sent. > >> > >> -benny > >> > >> > >> _______________________________________________ > >> Visit our blog: http://blog.pjsip.org > >> > >> pjsip mailing list > >> pjsip@lists.pjsip.org > >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >> > > > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org > > > > pjsip mailing list > > pjsip@lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > -- > Benny Prijono > http://www.pjsip.org > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >

Oh, maybe you are right. Do you think it is sending the ring in response to the 100 sent out from pjsua. If so, anyway to bypass sending the 100? And just send the 200. On 10/25/07, Lafras Henning <lafras@xietel.com> wrote: > In this scenario once PJSUA answers, Asterisk would be streaming the audio > to PJSUA, > so if you hear a ring on PJSUA, it would be coming from Asterisk. Not so? > > > ----- Original Message ----- > From: "Benny Prijono" <bennylp@pjsip.org> > To: "pjsip embedded/DSP SIP discussion" <pjsip@lists.pjsip.org> > Sent: Thursday, October 25, 2007 5:04 AM > Subject: Re: [pjsip] prevent audible ring on client when called > > > > Danny Brown wrote: > > > here is my config file passed to pjsua_vc6d.exe > > > --id=sip:MAST100@192.168.13.100 > > > --registrar=sip:192.168.13.100 > > > --username=MAST100 > > > --realm=asterisk > > > --password=1234 > > > --quality=3 > > > --auto-answer=200 > > > --ec-tail=600 > > > > > > it registers, will accept calls but always rings once before > > > answering. What am I missing? The way I am using this client is; > > > I have an asterisk server running that is scripted to initiate a call > > > from your sip client to another sip device. So, asterisk dials pjsua, > > > once it answers it dials the other end and makes the connection. I do > > > not want the caller (in this case pjsua) to actually ring ( at least > > > not audibly). Thanks for your help. Dan > > > > Sorry I'm totally confused here. There is no such thing as audible > > ringing in pjsua! The poor sample application just doesn't have this > > feature! > > > > -benny > > > > > > > On 10/24/07, Benny Prijono <bennylp@pjsip.org> wrote: > > >> Danny Brown wrote: > > >>> I am using the auto answer function of the pjsua on win32. It answers > > >>> like it should but I always get 1 audible ring. I would like to not > > >>> get an audible ring at all. Is there a way to do this with the > > >>> configuration file or what is the source file to change this behavior. > > >> With "--auto-answer 200", pjsua will send 100 and followed > > >> immediately by 200. Caller should not play any ring as no 180 is sent. > > >> > > >> -benny > > >> > > >> > > >> _______________________________________________ > > >> Visit our blog: http://blog.pjsip.org > > >> > > >> pjsip mailing list > > >> pjsip@lists.pjsip.org > > >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > >> > > > > > > _______________________________________________ > > > Visit our blog: http://blog.pjsip.org > > > > > > pjsip mailing list > > > pjsip@lists.pjsip.org > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > -- > > Benny Prijono > > http://www.pjsip.org > > > > > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org > > > > pjsip mailing list > > pjsip@lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >

Are you hearing the ring on PJSUA or on the other device? If you hear the ring on PJSUA it would be because the other device (not PJSUA) is sending a 180 to Asterisk, and Asterisk is producing a ring for that device. I do not know Asterisk well but I am sure you are able to set in the asterisk dial plan for it not to produce the ring back. ----- Original Message ----- From: "Danny Brown" <danbrwn@gmail.com> To: "pjsip embedded/DSP SIP discussion" <pjsip@lists.pjsip.org> Sent: Thursday, October 25, 2007 5:38 PM Subject: Re: [pjsip] prevent audible ring on client when called > Oh, maybe you are right. Do you think it is sending the ring in > response to the 100 sent out from pjsua. If so, anyway to bypass > sending the 100? And just send the 200. > > On 10/25/07, Lafras Henning <lafras@xietel.com> wrote: > > In this scenario once PJSUA answers, Asterisk would be streaming the audio > > to PJSUA, > > so if you hear a ring on PJSUA, it would be coming from Asterisk. Not so? > > > > > > ----- Original Message ----- > > From: "Benny Prijono" <bennylp@pjsip.org> > > To: "pjsip embedded/DSP SIP discussion" <pjsip@lists.pjsip.org> > > Sent: Thursday, October 25, 2007 5:04 AM > > Subject: Re: [pjsip] prevent audible ring on client when called > > > > > > > Danny Brown wrote: > > > > here is my config file passed to pjsua_vc6d.exe > > > > --id=sip:MAST100@192.168.13.100 > > > > --registrar=sip:192.168.13.100 > > > > --username=MAST100 > > > > --realm=asterisk > > > > --password=1234 > > > > --quality=3 > > > > --auto-answer=200 > > > > --ec-tail=600 > > > > > > > > it registers, will accept calls but always rings once before > > > > answering. What am I missing? The way I am using this client is; > > > > I have an asterisk server running that is scripted to initiate a call > > > > from your sip client to another sip device. So, asterisk dials pjsua, > > > > once it answers it dials the other end and makes the connection. I do > > > > not want the caller (in this case pjsua) to actually ring ( at least > > > > not audibly). Thanks for your help. Dan > > > > > > Sorry I'm totally confused here. There is no such thing as audible > > > ringing in pjsua! The poor sample application just doesn't have this > > > feature! > > > > > > -benny > > > > > > > > > > On 10/24/07, Benny Prijono <bennylp@pjsip.org> wrote: > > > >> Danny Brown wrote: > > > >>> I am using the auto answer function of the pjsua on win32. It answers > > > >>> like it should but I always get 1 audible ring. I would like to not > > > >>> get an audible ring at all. Is there a way to do this with the > > > >>> configuration file or what is the source file to change this behavior. > > > >> With "--auto-answer 200", pjsua will send 100 and followed > > > >> immediately by 200. Caller should not play any ring as no 180 is sent. > > > >> > > > >> -benny > > > >> > > > >> > > > >> _______________________________________________ > > > >> Visit our blog: http://blog.pjsip.org > > > >> > > > >> pjsip mailing list > > > >> pjsip@lists.pjsip.org > > > >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > >> > > > > > > > > _______________________________________________ > > > > Visit our blog: http://blog.pjsip.org > > > > > > > > pjsip mailing list > > > > pjsip@lists.pjsip.org > > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > > > > -- > > > Benny Prijono > > > http://www.pjsip.org > > > > > > > > > _______________________________________________ > > > Visit our blog: http://blog.pjsip.org > > > > > > pjsip mailing list > > > pjsip@lists.pjsip.org > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > > > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org > > > > pjsip mailing list > > pjsip@lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >

Yeah, I hear it on pjsua. I am investigating how to prevent this from happening on asterisk. So far no luck. But thanks for pointing me in the right direction. On 10/25/07, Lafras Henning <lafras@xietel.com> wrote: > Are you hearing the ring on PJSUA or on the other device? > > If you hear the ring on PJSUA it would be because the other device (not > PJSUA) is sending a 180 to Asterisk, and Asterisk is producing a ring for > that device. > I do not know Asterisk well but I am sure you are able to set in the > asterisk dial plan for it not to produce the ring back. > > ----- Original Message ----- > From: "Danny Brown" <danbrwn@gmail.com> > To: "pjsip embedded/DSP SIP discussion" <pjsip@lists.pjsip.org> > Sent: Thursday, October 25, 2007 5:38 PM > Subject: Re: [pjsip] prevent audible ring on client when called > > > > Oh, maybe you are right. Do you think it is sending the ring in > > response to the 100 sent out from pjsua. If so, anyway to bypass > > sending the 100? And just send the 200. > > > > On 10/25/07, Lafras Henning <lafras@xietel.com> wrote: > > > In this scenario once PJSUA answers, Asterisk would be streaming the > audio > > > to PJSUA, > > > so if you hear a ring on PJSUA, it would be coming from Asterisk. Not > so? > > > > > > > > > ----- Original Message ----- > > > From: "Benny Prijono" <bennylp@pjsip.org> > > > To: "pjsip embedded/DSP SIP discussion" <pjsip@lists.pjsip.org> > > > Sent: Thursday, October 25, 2007 5:04 AM > > > Subject: Re: [pjsip] prevent audible ring on client when called > > > > > > > > > > Danny Brown wrote: > > > > > here is my config file passed to pjsua_vc6d.exe > > > > > --id=sip:MAST100@192.168.13.100 > > > > > --registrar=sip:192.168.13.100 > > > > > --username=MAST100 > > > > > --realm=asterisk > > > > > --password=1234 > > > > > --quality=3 > > > > > --auto-answer=200 > > > > > --ec-tail=600 > > > > > > > > > > it registers, will accept calls but always rings once before > > > > > answering. What am I missing? The way I am using this client is; > > > > > I have an asterisk server running that is scripted to initiate a > call > > > > > from your sip client to another sip device. So, asterisk dials > pjsua, > > > > > once it answers it dials the other end and makes the connection. I > do > > > > > not want the caller (in this case pjsua) to actually ring ( at least > > > > > not audibly). Thanks for your help. Dan > > > > > > > > Sorry I'm totally confused here. There is no such thing as audible > > > > ringing in pjsua! The poor sample application just doesn't have this > > > > feature! > > > > > > > > -benny > > > > > > > > > > > > > On 10/24/07, Benny Prijono <bennylp@pjsip.org> wrote: > > > > >> Danny Brown wrote: > > > > >>> I am using the auto answer function of the pjsua on win32. It > answers > > > > >>> like it should but I always get 1 audible ring. I would like to > not > > > > >>> get an audible ring at all. Is there a way to do this with the > > > > >>> configuration file or what is the source file to change this > behavior. > > > > >> With "--auto-answer 200", pjsua will send 100 and followed > > > > >> immediately by 200. Caller should not play any ring as no 180 is > sent. > > > > >> > > > > >> -benny > > > > >> > > > > >> > > > > >> _______________________________________________ > > > > >> Visit our blog: http://blog.pjsip.org > > > > >> > > > > >> pjsip mailing list > > > > >> pjsip@lists.pjsip.org > > > > >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > >> > > > > > > > > > > _______________________________________________ > > > > > Visit our blog: http://blog.pjsip.org > > > > > > > > > > pjsip mailing list > > > > > pjsip@lists.pjsip.org > > > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > > > > > > > -- > > > > Benny Prijono > > > > http://www.pjsip.org > > > > > > > > > > > > _______________________________________________ > > > > Visit our blog: http://blog.pjsip.org > > > > > > > > pjsip mailing list > > > > pjsip@lists.pjsip.org > > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > > > > > > > > _______________________________________________ > > > Visit our blog: http://blog.pjsip.org > > > > > > pjsip mailing list > > > pjsip@lists.pjsip.org > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org > > > > pjsip mailing list > > pjsip@lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >