PJSIP call, play music and hangup in python

KT
Kaushik Turlapaty
Tue, May 31, 2016 8:33 AM

https://github.com/jrocharodrigues/sipamos/blob/master/sipamos.py

I tried this code but when I lift the call, I cannot hear any audio and I
am stuck up with this from last 4 days. Can someone pls help me with this ?
Is there something wrong with the code. If so, what it is ? Your help will
be very highly appreciated

Thanks,
Kaushik

https://github.com/jrocharodrigues/sipamos/blob/master/sipamos.py I tried this code but when I lift the call, I cannot hear any audio and I am stuck up with this from last 4 days. Can someone pls help me with this ? Is there something wrong with the code. If so, what it is ? Your help will be very highly appreciated Thanks, Kaushik
KT
Kaushik Turlapaty
Tue, May 31, 2016 8:56 AM
  This is the logic of the goal I am trying to achieve. It should

call, when confirmed, should play an audio wav file and hangup after
that

  I am able to call, but when the call is lift, I cannot hear the audio.

    if self.call.info().state == pjsua.CallState.DISCONNECTED:
        current_call = None
        print 'Current call is', current_call
        in_call = False
    elif self.call.info().state == pjsua.CallState.CONFIRMED:
  #Call is Answred
        print "Call Answred"
        wfile = wave.open("message.wav")
        time = (1.0 * wfile.getnframes ()) / wfile.getframerate ()
        print str(time) + "ms"
        wfile.close()
        call_slot = self.call.info().conf_slot
        self.wav_player_id=pjsua.Lib.instance().create_player('message.wav',loop=False)
        self.wav_slot=pjsua.Lib.instance().player_get_slot(self.wav_player_id)
        pjsua.Lib.instance().conf_connect(self.wav_slot, call_slot)
        sleep(time)
        pjsua.Lib.instance().player_destroy(self.wav_player_id)
        self.call.hangup()
        in_call = False

This is some part of the log when I run the program

17:03:26.375    pjsua_aud.c  .....Creating file player: test.wav..
17:03:26.375  wav_player.c  ......File player 'test.wav' created:
samp.rate=16000, ch=1, bufsize=4KB, filesize=166KB
17:03:26.375    pjsua_aud.c  ......Player created, id=0, slot=2
17:03:26.375    pjsua_aud.c  .....Conf connect: 2 --> 1
17:03:26.375    pjsua_aud.c  ......Set sound device: capture=-99, playback=-99
17:03:26.376    pjsua_aud.c  .......No changes in capture and playback devices
17:03:26.376  conference.c  ......Port 2 (test.wav) transmitting to
port 1 (sip:*@..com)
17:03:28.141    pjsua_aud.c  .....Destroying player 0..
17:03:28.141  pjsua_call.c  .....Call 0 hanging up: code=603..

This is the logic of the goal I am trying to achieve. It should call, when confirmed, should play an audio wav file and hangup after that I am able to call, but when the call is lift, I cannot hear the audio. if self.call.info().state == pjsua.CallState.DISCONNECTED: current_call = None print 'Current call is', current_call in_call = False elif self.call.info().state == pjsua.CallState.CONFIRMED: #Call is Answred print "Call Answred" wfile = wave.open("message.wav") time = (1.0 * wfile.getnframes ()) / wfile.getframerate () print str(time) + "ms" wfile.close() call_slot = self.call.info().conf_slot self.wav_player_id=pjsua.Lib.instance().create_player('message.wav',loop=False) self.wav_slot=pjsua.Lib.instance().player_get_slot(self.wav_player_id) pjsua.Lib.instance().conf_connect(self.wav_slot, call_slot) sleep(time) pjsua.Lib.instance().player_destroy(self.wav_player_id) self.call.hangup() in_call = False This is some part of the log when I run the program 17:03:26.375 pjsua_aud.c .....Creating file player: test.wav.. 17:03:26.375 wav_player.c ......File player 'test.wav' created: samp.rate=16000, ch=1, bufsize=4KB, filesize=166KB 17:03:26.375 pjsua_aud.c ......Player created, id=0, slot=2 17:03:26.375 pjsua_aud.c .....Conf connect: 2 --> 1 17:03:26.375 pjsua_aud.c ......Set sound device: capture=-99, playback=-99 17:03:26.376 pjsua_aud.c .......No changes in capture and playback devices 17:03:26.376 conference.c ......Port 2 (test.wav) transmitting to port 1 (sip:***********@*****.*****.com) 17:03:28.141 pjsua_aud.c .....Destroying player 0.. 17:03:28.141 pjsua_call.c .....Call 0 hanging up: code=603..
JL
JOHAN LANTZ
Tue, May 31, 2016 9:14 AM

Have not tried python but something like this works perfectly fine in C.

static pjsua_player_id player_id;
int call_playback_file(pjsua_call_id call_id, char* path, int no_loop)
{
pj_status_t status =PJ_SUCCESS;
if (call_id != PJSUA_INVALID_ID) {
pj_str_t path_as_pjstr = pj_str(path);
pjsua_player_id player_id;
status = pjsua_player_create(&path_as_pjstr, no_loop, &player_id);
if (status == PJ_SUCCESS) {
status = pjsua_conf_connect( pjsua_player_get_conf_port(player_id),
pjsua_call_get_conf_port(call_id));
if (status != PJ_SUCCESS) {
PJ_LOG(1,(THIS_FILE,"Error: Failed to connect file player to bridge with error %d", status));
}
} else {
PJ_LOG(1,(THIS_FILE,"Failed to create WAV player));
}
} else {
PU_LOG(1,(THIS_FILE,"No active call found"));
}
return status;
}


Este mensaje y sus adjuntos se dirigen exclusivamente a su destinatario, puede contener información privilegiada o confidencial y es para uso exclusivo de la persona o entidad de destino. Si no es usted. el destinatario indicado, queda notificado de que la lectura, utilización, divulgación y/o copia sin autorización puede estar prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción.

The information contained in this transmission is privileged and confidential information intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this transmission in error, do not read it. Please immediately reply to the sender that you have received this communication in error and then delete it.

Esta mensagem e seus anexos se dirigem exclusivamente ao seu destinatário, pode conter informação privilegiada ou confidencial e é para uso exclusivo da pessoa ou entidade de destino. Se não é vossa senhoria o destinatário indicado, fica notificado de que a leitura, utilização, divulgação e/ou cópia sem autorização pode estar proibida em virtude da legislação vigente. Se recebeu esta mensagem por erro, rogamos-lhe que nos o comunique imediatamente por esta mesma via e proceda a sua destruição

Have not tried python but something like this works perfectly fine in C. static pjsua_player_id player_id; int call_playback_file(pjsua_call_id call_id, char* path, int no_loop) { pj_status_t status =PJ_SUCCESS; if (call_id != PJSUA_INVALID_ID) { pj_str_t path_as_pjstr = pj_str(path); pjsua_player_id player_id; status = pjsua_player_create(&path_as_pjstr, no_loop, &player_id); if (status == PJ_SUCCESS) { status = pjsua_conf_connect( pjsua_player_get_conf_port(player_id), pjsua_call_get_conf_port(call_id)); if (status != PJ_SUCCESS) { PJ_LOG(1,(THIS_FILE,"Error: Failed to connect file player to bridge with error %d", status)); } } else { PJ_LOG(1,(THIS_FILE,"Failed to create WAV player)); } } else { PU_LOG(1,(THIS_FILE,"No active call found")); } return status; } ________________________________ Este mensaje y sus adjuntos se dirigen exclusivamente a su destinatario, puede contener información privilegiada o confidencial y es para uso exclusivo de la persona o entidad de destino. Si no es usted. el destinatario indicado, queda notificado de que la lectura, utilización, divulgación y/o copia sin autorización puede estar prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción. The information contained in this transmission is privileged and confidential information intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this transmission in error, do not read it. Please immediately reply to the sender that you have received this communication in error and then delete it. Esta mensagem e seus anexos se dirigem exclusivamente ao seu destinatário, pode conter informação privilegiada ou confidencial e é para uso exclusivo da pessoa ou entidade de destino. Se não é vossa senhoria o destinatário indicado, fica notificado de que a leitura, utilização, divulgação e/ou cópia sem autorização pode estar proibida em virtude da legislação vigente. Se recebeu esta mensagem por erro, rogamos-lhe que nos o comunique imediatamente por esta mesma via e proceda a sua destruição
GM
Gianni Massi
Tue, May 31, 2016 9:16 AM

You might need to put the conf_connect calls in the onMediaState() callback. See if that works.

Gianni

Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org] Per conto di Kaushik Turlapaty
Inviato: martedì 31 maggio 2016 10:57
A: pjsip@lists.pjsip.org
Oggetto: Re: [pjsip] PJSIP call, play music and hangup in python

  This is the logic of the goal I am trying to achieve. It should call, when confirmed, should play an audio wav file and hangup after that
  I am able to call, but when the call is lift, I cannot hear the audio. 
    if self.call.info().state == pjsua.CallState.DISCONNECTED:
        current_call = None
        print 'Current call is', current_call
        in_call = False
    elif self.call.info().state == pjsua.CallState.CONFIRMED:
        #Call is Answred
        print "Call Answred"
        wfile = wave.open("message.wav")
        time = (1.0 * wfile.getnframes ()) / wfile.getframerate ()
        print str(time) + "ms"
        wfile.close()
        call_slot = self.call.info().conf_slot
        self.wav_player_id=pjsua.Lib.instance().create_player('message.wav',loop=False)
        self.wav_slot=pjsua.Lib.instance().player_get_slot(self.wav_player_id)
        pjsua.Lib.instance().conf_connect(self.wav_slot, call_slot)
        sleep(time)
        pjsua.Lib.instance().player_destroy(self.wav_player_id)
        self.call.hangup()
        in_call = False

This is some part of the log when I run the program

17:03:26.375    pjsua_aud.c  .....Creating file player: test.wav..
17:03:26.375  wav_player.c  ......File player 'test.wav' created: samp.rate=16000, ch=1, bufsize=4KB, filesize=166KB
17:03:26.375    pjsua_aud.c  ......Player created, id=0, slot=2
17:03:26.375    pjsua_aud.c  .....Conf connect: 2 --> 1
17:03:26.375    pjsua_aud.c  ......Set sound device: capture=-99, playback=-99
17:03:26.376    pjsua_aud.c  .......No changes in capture and playback devices
17:03:26.376  conference.c  ......Port 2 (test.wav) transmitting to port 1 ( sip:***********@*****.*****.com sip:*@..com)
17:03:28.141    pjsua_aud.c  .....Destroying player 0..
17:03:28.141  pjsua_call.c  .....Call 0 hanging up: code=603..

You might need to put the conf_connect calls in the onMediaState() callback. See if that works. Gianni Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org] Per conto di Kaushik Turlapaty Inviato: martedì 31 maggio 2016 10:57 A: pjsip@lists.pjsip.org Oggetto: Re: [pjsip] PJSIP call, play music and hangup in python This is the logic of the goal I am trying to achieve. It should call, when confirmed, should play an audio wav file and hangup after that I am able to call, but when the call is lift, I cannot hear the audio. if self.call.info().state == pjsua.CallState.DISCONNECTED: current_call = None print 'Current call is', current_call in_call = False elif self.call.info().state == pjsua.CallState.CONFIRMED: #Call is Answred print "Call Answred" wfile = wave.open("message.wav") time = (1.0 * wfile.getnframes ()) / wfile.getframerate () print str(time) + "ms" wfile.close() call_slot = self.call.info().conf_slot self.wav_player_id=pjsua.Lib.instance().create_player('message.wav',loop=False) self.wav_slot=pjsua.Lib.instance().player_get_slot(self.wav_player_id) pjsua.Lib.instance().conf_connect(self.wav_slot, call_slot) sleep(time) pjsua.Lib.instance().player_destroy(self.wav_player_id) self.call.hangup() in_call = False This is some part of the log when I run the program 17:03:26.375 pjsua_aud.c .....Creating file player: test.wav.. 17:03:26.375 wav_player.c ......File player 'test.wav' created: samp.rate=16000, ch=1, bufsize=4KB, filesize=166KB 17:03:26.375 pjsua_aud.c ......Player created, id=0, slot=2 17:03:26.375 pjsua_aud.c .....Conf connect: 2 --> 1 17:03:26.375 pjsua_aud.c ......Set sound device: capture=-99, playback=-99 17:03:26.376 pjsua_aud.c .......No changes in capture and playback devices 17:03:26.376 conference.c ......Port 2 (test.wav) transmitting to port 1 ( <sip:***********@*****.*****.com> sip:***********@*****.*****.com) 17:03:28.141 pjsua_aud.c .....Destroying player 0.. 17:03:28.141 pjsua_call.c .....Call 0 hanging up: code=603..
KT
Kaushik Turlapaty
Tue, May 31, 2016 11:48 AM

Hey,

I just figured out the problem is because of ALSA ( I am using Ubuntu 14.04
). It is because of ALSA that I am not able to get the audio. Can someone
help me how to go about the whole ALSA thing to get the audio

Thanks,
Kaushik

On Tue, May 31, 2016 at 2:46 PM, Gianni Massi g.massi@infosolution.it
wrote:

You might need to put the conf_connect calls in the onMediaState()
callback. See if that works.

Gianni

Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org] *Per conto di *Kaushik
Turlapaty
Inviato: martedì 31 maggio 2016 10:57
A: pjsip@lists.pjsip.org
Oggetto: Re: [pjsip] PJSIP call, play music and hangup in python

  •  This is the logic of the goal I am trying to achieve. It should call, when confirmed, should play an audio wav file and hangup after that*
    
  •  I am able to call, but when the call is lift, I cannot hear the audio. *
    
  •    if* self.call.info().state == pjsua.CallState.DISCONNECTED:
    
          current_call = None
    
          print *'Current call is'*, current_call
    
          in_call = False
    
      elif self.call.info().state == pjsua.CallState.CONFIRMED:
    
          #Call is Answred
    
          print "Call Answred"
    
          wfile = wave.open("message.wav")
    
          time = (1.0 * wfile.getnframes ()) / wfile.getframerate ()
    
          print str(time) + "ms"
    
          wfile.close()
    
          call_slot = self.call.info().conf_slot
    
          self.wav_player_id=pjsua.Lib.instance().create_player(*'message.wav'*,loop=False)
    
          self.wav_slot=pjsua.Lib.instance().player_get_slot(self.wav_player_id)
    
          pjsua.Lib.instance().conf_connect(self.wav_slot, call_slot)
    
          sleep(time)
    
          pjsua.Lib.instance().player_destroy(self.wav_player_id)
    
          self.call.hangup()
    
          in_call = False
    

This is some part of the log when I run the program

17:03:26.375    pjsua_aud.c  .....Creating file player: test.wav..
17:03:26.375  wav_player.c  ......File player 'test.wav' created: samp.rate=16000, ch=1, bufsize=4KB, filesize=166KB
17:03:26.375    pjsua_aud.c  ......Player created, id=0, slot=2
17:03:26.375    pjsua_aud.c  .....Conf connect: 2 --> 1
17:03:26.375    pjsua_aud.c  ......Set sound device: capture=-99, playback=-99
17:03:26.376    pjsua_aud.c  .......No changes in capture and playback devices
17:03:26.376  conference.c  ......Port 2 (test.wav) transmitting to port 1 (sip:*@..com)
17:03:28.141    pjsua_aud.c  .....Destroying player 0..
17:03:28.141  pjsua_call.c  .....Call 0 hanging up: code=603..


Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Hey, I just figured out the problem is because of ALSA ( I am using Ubuntu 14.04 ). It is because of ALSA that I am not able to get the audio. Can someone help me how to go about the whole ALSA thing to get the audio Thanks, Kaushik On Tue, May 31, 2016 at 2:46 PM, Gianni Massi <g.massi@infosolution.it> wrote: > You might need to put the conf_connect calls in the onMediaState() > callback. See if that works. > > > > Gianni > > > > *Da:* pjsip [mailto:pjsip-bounces@lists.pjsip.org] *Per conto di *Kaushik > Turlapaty > *Inviato:* martedì 31 maggio 2016 10:57 > *A:* pjsip@lists.pjsip.org > *Oggetto:* Re: [pjsip] PJSIP call, play music and hangup in python > > > > * This is the logic of the goal I am trying to achieve. It should call, when confirmed, should play an audio wav file and hangup after that* > > * I am able to call, but when the call is lift, I cannot hear the audio. * > > * if* self.call.info().state == pjsua.CallState.DISCONNECTED: > > current_call = None > > print *'Current call is'*, current_call > > in_call = False > > elif self.call.info().state == pjsua.CallState.CONFIRMED: > > #Call is Answred > > print "Call Answred" > > wfile = wave.open("message.wav") > > time = (1.0 * wfile.getnframes ()) / wfile.getframerate () > > print str(time) + "ms" > > wfile.close() > > call_slot = self.call.info().conf_slot > > self.wav_player_id=pjsua.Lib.instance().create_player(*'message.wav'*,loop=False) > > self.wav_slot=pjsua.Lib.instance().player_get_slot(self.wav_player_id) > > pjsua.Lib.instance().conf_connect(self.wav_slot, call_slot) > > sleep(time) > > pjsua.Lib.instance().player_destroy(self.wav_player_id) > > self.call.hangup() > > in_call = False > > This is some part of the log when I run the program > > > 17:03:26.375 pjsua_aud.c .....Creating file player: test.wav.. > 17:03:26.375 wav_player.c ......File player 'test.wav' created: samp.rate=16000, ch=1, bufsize=4KB, filesize=166KB > 17:03:26.375 pjsua_aud.c ......Player created, id=0, slot=2 > 17:03:26.375 pjsua_aud.c .....Conf connect: 2 --> 1 > 17:03:26.375 pjsua_aud.c ......Set sound device: capture=-99, playback=-99 > 17:03:26.376 pjsua_aud.c .......No changes in capture and playback devices > 17:03:26.376 conference.c ......Port 2 (test.wav) transmitting to port 1 (sip:***********@*****.*****.com) > 17:03:28.141 pjsua_aud.c .....Destroying player 0.. > 17:03:28.141 pjsua_call.c .....Call 0 hanging up: code=603.. > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > >
GM
Gianni Massi
Tue, May 31, 2016 12:43 PM

Hi Kaushik,

If I understand the problem you have correctly, (please next time post logs/console ouputs leading to your diagnosis of the problem, they can be useful even if you’re 100% sure about the nature of the problem) you should be able to follow the instructions from step 3 in this link to solve your problem.

http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2013-December/016918.html

Hope this is useful,

Gianni

Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org] Per conto di Kaushik Turlapaty
Inviato: martedì 31 maggio 2016 13:48
A: pjsip list pjsip@lists.pjsip.org
Oggetto: Re: [pjsip] R: PJSIP call, play music and hangup in python

Hey,

I just figured out the problem is because of ALSA ( I am using Ubuntu 14.04 ). It is because of ALSA that I am not able to get the audio. Can someone help me how to go about the whole ALSA thing to get the audio

Thanks,

Kaushik

On Tue, May 31, 2016 at 2:46 PM, Gianni Massi <g.massi@infosolution.it mailto:g.massi@infosolution.it > wrote:

You might need to put the conf_connect calls in the onMediaState() callback. See if that works.

Gianni

Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org mailto:pjsip-bounces@lists.pjsip.org ] Per conto di Kaushik Turlapaty
Inviato: martedì 31 maggio 2016 10:57
A: pjsip@lists.pjsip.org mailto:pjsip@lists.pjsip.org
Oggetto: Re: [pjsip] PJSIP call, play music and hangup in python

  This is the logic of the goal I am trying to achieve. It should call, when confirmed, should play an audio wav file and hangup after that
  I am able to call, but when the call is lift, I cannot hear the audio. 
    if self.call.info().state == pjsua.CallState.DISCONNECTED:
        current_call = None
        print 'Current call is', current_call
        in_call = False
    elif self.call.info().state == pjsua.CallState.CONFIRMED:
        #Call is Answred
        print "Call Answred"
        wfile = wave.open("message.wav")
        time = (1.0 * wfile.getnframes ()) / wfile.getframerate ()
        print str(time) + "ms"
        wfile.close()
        call_slot = self.call.info().conf_slot
        self.wav_player_id=pjsua.Lib.instance().create_player('message.wav',loop=False)
        self.wav_slot=pjsua.Lib.instance().player_get_slot(self.wav_player_id)
        pjsua.Lib.instance().conf_connect(self.wav_slot, call_slot)
        sleep(time)
        pjsua.Lib.instance().player_destroy(self.wav_player_id)
        self.call.hangup()
        in_call = False

This is some part of the log when I run the program

17:03:26.375    pjsua_aud.c  .....Creating file player: test.wav..
17:03:26.375  wav_player.c  ......File player 'test.wav' created: samp.rate=16000, ch=1, bufsize=4KB, filesize=166KB
17:03:26.375    pjsua_aud.c  ......Player created, id=0, slot=2
17:03:26.375    pjsua_aud.c  .....Conf connect: 2 --> 1
17:03:26.375    pjsua_aud.c  ......Set sound device: capture=-99, playback=-99
17:03:26.376    pjsua_aud.c  .......No changes in capture and playback devices
17:03:26.376  conference.c  ......Port 2 (test.wav) transmitting to port 1 (sip:*@..com)
17:03:28.141    pjsua_aud.c  .....Destroying player 0..
17:03:28.141  pjsua_call.c  .....Call 0 hanging up: code=603..


Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org mailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Hi Kaushik, If I understand the problem you have correctly, (please next time post logs/console ouputs leading to your diagnosis of the problem, they can be useful even if you’re 100% sure about the nature of the problem) you should be able to follow the instructions from step 3 in this link to solve your problem. http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2013-December/016918.html Hope this is useful, Gianni Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org] Per conto di Kaushik Turlapaty Inviato: martedì 31 maggio 2016 13:48 A: pjsip list <pjsip@lists.pjsip.org> Oggetto: Re: [pjsip] R: PJSIP call, play music and hangup in python Hey, I just figured out the problem is because of ALSA ( I am using Ubuntu 14.04 ). It is because of ALSA that I am not able to get the audio. Can someone help me how to go about the whole ALSA thing to get the audio Thanks, Kaushik On Tue, May 31, 2016 at 2:46 PM, Gianni Massi <g.massi@infosolution.it <mailto:g.massi@infosolution.it> > wrote: You might need to put the conf_connect calls in the onMediaState() callback. See if that works. Gianni Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org <mailto:pjsip-bounces@lists.pjsip.org> ] Per conto di Kaushik Turlapaty Inviato: martedì 31 maggio 2016 10:57 A: pjsip@lists.pjsip.org <mailto:pjsip@lists.pjsip.org> Oggetto: Re: [pjsip] PJSIP call, play music and hangup in python This is the logic of the goal I am trying to achieve. It should call, when confirmed, should play an audio wav file and hangup after that I am able to call, but when the call is lift, I cannot hear the audio. if self.call.info().state == pjsua.CallState.DISCONNECTED: current_call = None print 'Current call is', current_call in_call = False elif self.call.info().state == pjsua.CallState.CONFIRMED: #Call is Answred print "Call Answred" wfile = wave.open("message.wav") time = (1.0 * wfile.getnframes ()) / wfile.getframerate () print str(time) + "ms" wfile.close() call_slot = self.call.info().conf_slot self.wav_player_id=pjsua.Lib.instance().create_player('message.wav',loop=False) self.wav_slot=pjsua.Lib.instance().player_get_slot(self.wav_player_id) pjsua.Lib.instance().conf_connect(self.wav_slot, call_slot) sleep(time) pjsua.Lib.instance().player_destroy(self.wav_player_id) self.call.hangup() in_call = False This is some part of the log when I run the program 17:03:26.375 pjsua_aud.c .....Creating file player: test.wav.. 17:03:26.375 wav_player.c ......File player 'test.wav' created: samp.rate=16000, ch=1, bufsize=4KB, filesize=166KB 17:03:26.375 pjsua_aud.c ......Player created, id=0, slot=2 17:03:26.375 pjsua_aud.c .....Conf connect: 2 --> 1 17:03:26.375 pjsua_aud.c ......Set sound device: capture=-99, playback=-99 17:03:26.376 pjsua_aud.c .......No changes in capture and playback devices 17:03:26.376 conference.c ......Port 2 (test.wav) transmitting to port 1 (sip:***********@*****.*****.com) 17:03:28.141 pjsua_aud.c .....Destroying player 0.. 17:03:28.141 pjsua_call.c .....Call 0 hanging up: code=603.. _______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list pjsip@lists.pjsip.org <mailto:pjsip@lists.pjsip.org> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
KT
Kaushik Turlapaty
Tue, May 31, 2016 1:10 PM

Hey,

This is the log

14:54:30.981 os_core_unix.c !pjlib 2.2.1-svn for POSIX initialized
14:54:30.983 sip_endpoint.c  .Creating endpoint instance...
14:54:30.984          pjlib  .select() I/O Queue created (0xca1850)
14:54:30.984 sip_endpoint.c  .Module "mod-msg-print" registered
14:54:30.984 sip_transport.  .Transport manager created.
14:54:30.984  pjsua_core.c  .PJSUA state changed: NULL --> CREATED
14:54:30.985 sip_endpoint.c  .Module "mod-pjsua-log" registered
14:54:30.985 sip_endpoint.c  .Module "mod-tsx-layer" registered
14:54:30.985 sip_endpoint.c  .Module "mod-stateful-util" registered
14:54:30.985 sip_endpoint.c  .Module "mod-ua" registered
14:54:30.985 sip_endpoint.c  .Module "mod-100rel" registered
14:54:30.985 sip_endpoint.c  .Module "mod-pjsua" registered
14:54:30.985 sip_endpoint.c  .Module "mod-invite" registered
14:54:31.030    alsa_dev.c  ..ALSA driver found 17 devices
14:54:31.030    alsa_dev.c  ..ALSA initialized
14:54:31.030          pjlib  ..select() I/O Queue created (0xcb5968)
14:54:31.035    pjsua_vid.c  ..Initializing video subsystem..
14:54:31.155    v4l2_dev.c  ...Video4Linux2 has 1 devices
14:54:31.155 colorbar_dev.c  ...Colorbar video src initialized with 1
device(s):
14:54:31.155 colorbar_dev.c  ... 0: Colorbar generator

On Tue, May 31, 2016 at 6:13 PM, Gianni Massi g.massi@infosolution.it
wrote:

Hi Kaushik,

If I understand the problem you have correctly, (please next time post
logs/console ouputs leading to your diagnosis of the problem, they can be
useful even if you’re 100% sure about the nature of the problem) you should
be able to follow the instructions from step 3 in this link to solve your
problem.

http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2013-December/016918.html

Hope this is useful,

Gianni

Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org] *Per conto di *Kaushik
Turlapaty
Inviato: martedì 31 maggio 2016 13:48
A: pjsip list pjsip@lists.pjsip.org
Oggetto: Re: [pjsip] R: PJSIP call, play music and hangup in python

Hey,

I just figured out the problem is because of ALSA ( I am using Ubuntu
14.04 ). It is because of ALSA that I am not able to get the audio. Can
someone help me how to go about the whole ALSA thing to get the audio

Thanks,

Kaushik

On Tue, May 31, 2016 at 2:46 PM, Gianni Massi g.massi@infosolution.it
wrote:

You might need to put the conf_connect calls in the onMediaState()
callback. See if that works.

Gianni

Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org] *Per conto di *Kaushik
Turlapaty
Inviato: martedì 31 maggio 2016 10:57
A: pjsip@lists.pjsip.org
Oggetto: Re: [pjsip] PJSIP call, play music and hangup in python

  •  This is the logic of the goal I am trying to achieve. It should call, when confirmed, should play an audio wav file and hangup after that*
    
  •  I am able to call, but when the call is lift, I cannot hear the audio. *
    
  •    if* self.call.info().state == pjsua.CallState.DISCONNECTED:
    
          current_call = None
    
          print *'Current call is'*, current_call
    
          in_call = False
    
      elif self.call.info().state == pjsua.CallState.CONFIRMED:
    
          #Call is Answred
    
          print "Call Answred"
    
          wfile = wave.open("message.wav")
    
          time = (1.0 * wfile.getnframes ()) / wfile.getframerate ()
    
          print str(time) + "ms"
    
          wfile.close()
    
          call_slot = self.call.info().conf_slot
    
          self.wav_player_id=pjsua.Lib.instance().create_player(*'message.wav'*,loop=False)
    
          self.wav_slot=pjsua.Lib.instance().player_get_slot(self.wav_player_id)
    
          pjsua.Lib.instance().conf_connect(self.wav_slot, call_slot)
    
          sleep(time)
    
          pjsua.Lib.instance().player_destroy(self.wav_player_id)
    
          self.call.hangup()
    
          in_call = False
    

This is some part of the log when I run the program

17:03:26.375    pjsua_aud.c  .....Creating file player: test.wav..
17:03:26.375  wav_player.c  ......File player 'test.wav' created: samp.rate=16000, ch=1, bufsize=4KB, filesize=166KB
17:03:26.375    pjsua_aud.c  ......Player created, id=0, slot=2
17:03:26.375    pjsua_aud.c  .....Conf connect: 2 --> 1
17:03:26.375    pjsua_aud.c  ......Set sound device: capture=-99, playback=-99
17:03:26.376    pjsua_aud.c  .......No changes in capture and playback devices
17:03:26.376  conference.c  ......Port 2 (test.wav) transmitting to port 1 (sip:*@..com)
17:03:28.141    pjsua_aud.c  .....Destroying player 0..
17:03:28.141  pjsua_call.c  .....Call 0 hanging up: code=603..


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

Hey, This is the log 14:54:30.981 os_core_unix.c !pjlib 2.2.1-svn for POSIX initialized 14:54:30.983 sip_endpoint.c .Creating endpoint instance... 14:54:30.984 pjlib .select() I/O Queue created (0xca1850) 14:54:30.984 sip_endpoint.c .Module "mod-msg-print" registered 14:54:30.984 sip_transport. .Transport manager created. 14:54:30.984 pjsua_core.c .PJSUA state changed: NULL --> CREATED 14:54:30.985 sip_endpoint.c .Module "mod-pjsua-log" registered 14:54:30.985 sip_endpoint.c .Module "mod-tsx-layer" registered 14:54:30.985 sip_endpoint.c .Module "mod-stateful-util" registered 14:54:30.985 sip_endpoint.c .Module "mod-ua" registered 14:54:30.985 sip_endpoint.c .Module "mod-100rel" registered 14:54:30.985 sip_endpoint.c .Module "mod-pjsua" registered 14:54:30.985 sip_endpoint.c .Module "mod-invite" registered 14:54:31.030 alsa_dev.c ..ALSA driver found 17 devices 14:54:31.030 alsa_dev.c ..ALSA initialized 14:54:31.030 pjlib ..select() I/O Queue created (0xcb5968) 14:54:31.035 pjsua_vid.c ..Initializing video subsystem.. 14:54:31.155 v4l2_dev.c ...Video4Linux2 has 1 devices 14:54:31.155 colorbar_dev.c ...Colorbar video src initialized with 1 device(s): 14:54:31.155 colorbar_dev.c ... 0: Colorbar generator On Tue, May 31, 2016 at 6:13 PM, Gianni Massi <g.massi@infosolution.it> wrote: > Hi Kaushik, > > > > If I understand the problem you have correctly, (please next time post > logs/console ouputs leading to your diagnosis of the problem, they can be > useful even if you’re 100% sure about the nature of the problem) you should > be able to follow the instructions from step 3 in this link to solve your > problem. > > > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2013-December/016918.html > > > > Hope this is useful, > > Gianni > > > > *Da:* pjsip [mailto:pjsip-bounces@lists.pjsip.org] *Per conto di *Kaushik > Turlapaty > *Inviato:* martedì 31 maggio 2016 13:48 > *A:* pjsip list <pjsip@lists.pjsip.org> > *Oggetto:* Re: [pjsip] R: PJSIP call, play music and hangup in python > > > > Hey, > > I just figured out the problem is because of ALSA ( I am using Ubuntu > 14.04 ). It is because of ALSA that I am not able to get the audio. Can > someone help me how to go about the whole ALSA thing to get the audio > > Thanks, > > Kaushik > > > > On Tue, May 31, 2016 at 2:46 PM, Gianni Massi <g.massi@infosolution.it> > wrote: > > You might need to put the conf_connect calls in the onMediaState() > callback. See if that works. > > > > Gianni > > > > *Da:* pjsip [mailto:pjsip-bounces@lists.pjsip.org] *Per conto di *Kaushik > Turlapaty > *Inviato:* martedì 31 maggio 2016 10:57 > *A:* pjsip@lists.pjsip.org > *Oggetto:* Re: [pjsip] PJSIP call, play music and hangup in python > > > > * This is the logic of the goal I am trying to achieve. It should call, when confirmed, should play an audio wav file and hangup after that* > > * I am able to call, but when the call is lift, I cannot hear the audio. * > > * if* self.call.info().state == pjsua.CallState.DISCONNECTED: > > current_call = None > > print *'Current call is'*, current_call > > in_call = False > > elif self.call.info().state == pjsua.CallState.CONFIRMED: > > #Call is Answred > > print "Call Answred" > > wfile = wave.open("message.wav") > > time = (1.0 * wfile.getnframes ()) / wfile.getframerate () > > print str(time) + "ms" > > wfile.close() > > call_slot = self.call.info().conf_slot > > self.wav_player_id=pjsua.Lib.instance().create_player(*'message.wav'*,loop=False) > > self.wav_slot=pjsua.Lib.instance().player_get_slot(self.wav_player_id) > > pjsua.Lib.instance().conf_connect(self.wav_slot, call_slot) > > sleep(time) > > pjsua.Lib.instance().player_destroy(self.wav_player_id) > > self.call.hangup() > > in_call = False > > This is some part of the log when I run the program > > > 17:03:26.375 pjsua_aud.c .....Creating file player: test.wav.. > 17:03:26.375 wav_player.c ......File player 'test.wav' created: samp.rate=16000, ch=1, bufsize=4KB, filesize=166KB > 17:03:26.375 pjsua_aud.c ......Player created, id=0, slot=2 > 17:03:26.375 pjsua_aud.c .....Conf connect: 2 --> 1 > 17:03:26.375 pjsua_aud.c ......Set sound device: capture=-99, playback=-99 > 17:03:26.376 pjsua_aud.c .......No changes in capture and playback devices > 17:03:26.376 conference.c ......Port 2 (test.wav) transmitting to port 1 (sip:***********@*****.*****.com) > 17:03:28.141 pjsua_aud.c .....Destroying player 0.. > 17:03:28.141 pjsua_call.c .....Call 0 hanging up: code=603.. > > > _______________________________________________ > 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 > >
GM
Gianni Massi
Tue, May 31, 2016 2:05 PM

Everything looks in order for me, you don’t have the common problem I expected, with portaudio being used instead of alsa, so no need to follow the link I gave you. Have you tried my previous suggestion about conf_connect?

Gianni

Da: pjsip [ mailto:pjsip-bounces@lists.pjsip.org mailto:pjsip-bounces@lists.pjsip.org] Per conto di Kaushik Turlapaty
Inviato: martedì 31 maggio 2016 15:10
A: pjsip list pjsip@lists.pjsip.org
Oggetto: Re: [pjsip] R: R: PJSIP call, play music and hangup in python

Hey,

This is the log

14:54:30.981 os_core_unix.c !pjlib 2.2.1-svn for POSIX initialized
14:54:30.983 sip_endpoint.c  .Creating endpoint instance...
14:54:30.984          pjlib  .select() I/O Queue created (0xca1850)
14:54:30.984 sip_endpoint.c  .Module "mod-msg-print" registered
14:54:30.984 sip_transport.  .Transport manager created.
14:54:30.984  pjsua_core.c  .PJSUA state changed: NULL --> CREATED
14:54:30.985 sip_endpoint.c  .Module "mod-pjsua-log" registered
14:54:30.985 sip_endpoint.c  .Module "mod-tsx-layer" registered
14:54:30.985 sip_endpoint.c  .Module "mod-stateful-util" registered
14:54:30.985 sip_endpoint.c  .Module "mod-ua" registered
14:54:30.985 sip_endpoint.c  .Module "mod-100rel" registered
14:54:30.985 sip_endpoint.c  .Module "mod-pjsua" registered
14:54:30.985 sip_endpoint.c  .Module "mod-invite" registered
14:54:31.030    alsa_dev.c  ..ALSA driver found 17 devices
14:54:31.030    alsa_dev.c  ..ALSA initialized
14:54:31.030          pjlib  ..select() I/O Queue created (0xcb5968)
14:54:31.035    pjsua_vid.c  ..Initializing video subsystem..
14:54:31.155    v4l2_dev.c  ...Video4Linux2 has 1 devices
14:54:31.155 colorbar_dev.c  ...Colorbar video src initialized with 1 device(s):
14:54:31.155 colorbar_dev.c  ... 0: Colorbar generator

On Tue, May 31, 2016 at 6:13 PM, Gianni Massi <g.massi@infosolution.it mailto:g.massi@infosolution.it > wrote:

Hi Kaushik,

If I understand the problem you have correctly, (please next time post logs/console ouputs leading to your diagnosis of the problem, they can be useful even if you’re 100% sure about the nature of the problem) you should be able to follow the instructions from step 3 in this link to solve your problem.

http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2013-December/016918.html

Hope this is useful,

Gianni

Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org mailto:pjsip-bounces@lists.pjsip.org ] Per conto di Kaushik Turlapaty
Inviato: martedì 31 maggio 2016 13:48
A: pjsip list <pjsip@lists.pjsip.org mailto:pjsip@lists.pjsip.org >
Oggetto: Re: [pjsip] R: PJSIP call, play music and hangup in python

Hey,

I just figured out the problem is because of ALSA ( I am using Ubuntu 14.04 ). It is because of ALSA that I am not able to get the audio. Can someone help me how to go about the whole ALSA thing to get the audio

Thanks,

Kaushik

On Tue, May 31, 2016 at 2:46 PM, Gianni Massi <g.massi@infosolution.it mailto:g.massi@infosolution.it > wrote:

You might need to put the conf_connect calls in the onMediaState() callback. See if that works.

Gianni

Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org mailto:pjsip-bounces@lists.pjsip.org ] Per conto di Kaushik Turlapaty
Inviato: martedì 31 maggio 2016 10:57
A: pjsip@lists.pjsip.org mailto:pjsip@lists.pjsip.org
Oggetto: Re: [pjsip] PJSIP call, play music and hangup in python

  This is the logic of the goal I am trying to achieve. It should call, when confirmed, should play an audio wav file and hangup after that
  I am able to call, but when the call is lift, I cannot hear the audio. 
    if self.call.info().state == pjsua.CallState.DISCONNECTED:
        current_call = None
        print 'Current call is', current_call
        in_call = False
    elif self.call.info().state == pjsua.CallState.CONFIRMED:
        #Call is Answred
        print "Call Answred"
        wfile = wave.open("message.wav")
        time = (1.0 * wfile.getnframes ()) / wfile.getframerate ()
        print str(time) + "ms"
        wfile.close()
        call_slot = self.call.info().conf_slot
        self.wav_player_id=pjsua.Lib.instance().create_player('message.wav',loop=False)
        self.wav_slot=pjsua.Lib.instance().player_get_slot(self.wav_player_id)
        pjsua.Lib.instance().conf_connect(self.wav_slot, call_slot)
        sleep(time)
        pjsua.Lib.instance().player_destroy(self.wav_player_id)
        self.call.hangup()
        in_call = False

This is some part of the log when I run the program

17:03:26.375    pjsua_aud.c  .....Creating file player: test.wav..
17:03:26.375  wav_player.c  ......File player 'test.wav' created: samp.rate=16000, ch=1, bufsize=4KB, filesize=166KB
17:03:26.375    pjsua_aud.c  ......Player created, id=0, slot=2
17:03:26.375    pjsua_aud.c  .....Conf connect: 2 --> 1
17:03:26.375    pjsua_aud.c  ......Set sound device: capture=-99, playback=-99
17:03:26.376    pjsua_aud.c  .......No changes in capture and playback devices
17:03:26.376  conference.c  ......Port 2 (test.wav) transmitting to port 1 (sip:*@..com)
17:03:28.141    pjsua_aud.c  .....Destroying player 0..
17:03:28.141  pjsua_call.c  .....Call 0 hanging up: code=603..


Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org mailto: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 mailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Everything looks in order for me, you don’t have the common problem I expected, with portaudio being used instead of alsa, so no need to follow the link I gave you. Have you tried my previous suggestion about conf_connect? Gianni Da: pjsip [ <mailto:pjsip-bounces@lists.pjsip.org> mailto:pjsip-bounces@lists.pjsip.org] Per conto di Kaushik Turlapaty Inviato: martedì 31 maggio 2016 15:10 A: pjsip list <pjsip@lists.pjsip.org> Oggetto: Re: [pjsip] R: R: PJSIP call, play music and hangup in python Hey, This is the log 14:54:30.981 os_core_unix.c !pjlib 2.2.1-svn for POSIX initialized 14:54:30.983 sip_endpoint.c .Creating endpoint instance... 14:54:30.984 pjlib .select() I/O Queue created (0xca1850) 14:54:30.984 sip_endpoint.c .Module "mod-msg-print" registered 14:54:30.984 sip_transport. .Transport manager created. 14:54:30.984 pjsua_core.c .PJSUA state changed: NULL --> CREATED 14:54:30.985 sip_endpoint.c .Module "mod-pjsua-log" registered 14:54:30.985 sip_endpoint.c .Module "mod-tsx-layer" registered 14:54:30.985 sip_endpoint.c .Module "mod-stateful-util" registered 14:54:30.985 sip_endpoint.c .Module "mod-ua" registered 14:54:30.985 sip_endpoint.c .Module "mod-100rel" registered 14:54:30.985 sip_endpoint.c .Module "mod-pjsua" registered 14:54:30.985 sip_endpoint.c .Module "mod-invite" registered 14:54:31.030 alsa_dev.c ..ALSA driver found 17 devices 14:54:31.030 alsa_dev.c ..ALSA initialized 14:54:31.030 pjlib ..select() I/O Queue created (0xcb5968) 14:54:31.035 pjsua_vid.c ..Initializing video subsystem.. 14:54:31.155 v4l2_dev.c ...Video4Linux2 has 1 devices 14:54:31.155 colorbar_dev.c ...Colorbar video src initialized with 1 device(s): 14:54:31.155 colorbar_dev.c ... 0: Colorbar generator On Tue, May 31, 2016 at 6:13 PM, Gianni Massi <g.massi@infosolution.it <mailto:g.massi@infosolution.it> > wrote: Hi Kaushik, If I understand the problem you have correctly, (please next time post logs/console ouputs leading to your diagnosis of the problem, they can be useful even if you’re 100% sure about the nature of the problem) you should be able to follow the instructions from step 3 in this link to solve your problem. http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2013-December/016918.html Hope this is useful, Gianni Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org <mailto:pjsip-bounces@lists.pjsip.org> ] Per conto di Kaushik Turlapaty Inviato: martedì 31 maggio 2016 13:48 A: pjsip list <pjsip@lists.pjsip.org <mailto:pjsip@lists.pjsip.org> > Oggetto: Re: [pjsip] R: PJSIP call, play music and hangup in python Hey, I just figured out the problem is because of ALSA ( I am using Ubuntu 14.04 ). It is because of ALSA that I am not able to get the audio. Can someone help me how to go about the whole ALSA thing to get the audio Thanks, Kaushik On Tue, May 31, 2016 at 2:46 PM, Gianni Massi <g.massi@infosolution.it <mailto:g.massi@infosolution.it> > wrote: You might need to put the conf_connect calls in the onMediaState() callback. See if that works. Gianni Da: pjsip [mailto:pjsip-bounces@lists.pjsip.org <mailto:pjsip-bounces@lists.pjsip.org> ] Per conto di Kaushik Turlapaty Inviato: martedì 31 maggio 2016 10:57 A: pjsip@lists.pjsip.org <mailto:pjsip@lists.pjsip.org> Oggetto: Re: [pjsip] PJSIP call, play music and hangup in python This is the logic of the goal I am trying to achieve. It should call, when confirmed, should play an audio wav file and hangup after that I am able to call, but when the call is lift, I cannot hear the audio. if self.call.info().state == pjsua.CallState.DISCONNECTED: current_call = None print 'Current call is', current_call in_call = False elif self.call.info().state == pjsua.CallState.CONFIRMED: #Call is Answred print "Call Answred" wfile = wave.open("message.wav") time = (1.0 * wfile.getnframes ()) / wfile.getframerate () print str(time) + "ms" wfile.close() call_slot = self.call.info().conf_slot self.wav_player_id=pjsua.Lib.instance().create_player('message.wav',loop=False) self.wav_slot=pjsua.Lib.instance().player_get_slot(self.wav_player_id) pjsua.Lib.instance().conf_connect(self.wav_slot, call_slot) sleep(time) pjsua.Lib.instance().player_destroy(self.wav_player_id) self.call.hangup() in_call = False This is some part of the log when I run the program 17:03:26.375 pjsua_aud.c .....Creating file player: test.wav.. 17:03:26.375 wav_player.c ......File player 'test.wav' created: samp.rate=16000, ch=1, bufsize=4KB, filesize=166KB 17:03:26.375 pjsua_aud.c ......Player created, id=0, slot=2 17:03:26.375 pjsua_aud.c .....Conf connect: 2 --> 1 17:03:26.375 pjsua_aud.c ......Set sound device: capture=-99, playback=-99 17:03:26.376 pjsua_aud.c .......No changes in capture and playback devices 17:03:26.376 conference.c ......Port 2 (test.wav) transmitting to port 1 (sip:***********@*****.*****.com) 17:03:28.141 pjsua_aud.c .....Destroying player 0.. 17:03:28.141 pjsua_call.c .....Call 0 hanging up: code=603.. _______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list pjsip@lists.pjsip.org <mailto: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 <mailto:pjsip@lists.pjsip.org> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org