[Linphone-users] asterisk and linphonec setup
I looking for some of the basic setup for linphonec to connect (use) a local asterisk server. My understanding is I have to setup a proxy server (proxy add) to my asterisk server? I'm unsure if I am doing it correctly as I can not call the other sip phones in my network. I had success with minisip, but prefer to use linphone as it is much smaller. I can dial directly from one linphone to another, but rather go via the asterisk server. Examples of a working configuration of asterisk and linphonec would be greatly appreciated, as well as usage (assuming I using linphonec incorrectly). Thanks, Craig ___ Linphone-users mailing list Linphone-users@nongnu.org http://lists.nongnu.org/mailman/listinfo/linphone-users
Re: [Linphone-users] asterisk and linphonec setup
I can call between devices (direct) with version 1.2.0, however with versions 1.4 and greater I do not hear any audio. I have double check the .linphonerc setup. I actually get the dialing tones but once I answer nothing. Seems the 1.2.0 version was working better than the 1.4.0 - 1.5.1 version of linphone. When using asterisk as the exchange, the calls appear to route but now sound. Does anyone ahve any ideas? I was previously using minisip and it worked flawlessly. I would like to get linphone work as well, to evaluate. Thanks, Craig On Thursday 18 January 2007 1:01 pm, Craig Matsuura wrote: > I looking for some of the basic setup for linphonec to connect (use) a > local asterisk server. > > My understanding is I have to setup a proxy server (proxy add) to my > asterisk server? I'm unsure if I am doing it correctly as I can not call > the other sip phones in my network. I had success with minisip, but prefer > to use linphone as it is much smaller. > > I can dial directly from one linphone to another, but rather go via the > asterisk server. > > Examples of a working configuration of asterisk and linphonec would be > greatly appreciated, as well as usage (assuming I using linphonec > incorrectly). > > Thanks, > Craig > > > ___ > Linphone-users mailing list > Linphone-users@nongnu.org > http://lists.nongnu.org/mailman/listinfo/linphone-users ___ Linphone-users mailing list Linphone-users@nongnu.org http://lists.nongnu.org/mailman/listinfo/linphone-users
Re: [Linphone-users] asterisk and linphonec setup
When having problems, the best thing is to send here a log like this: linphone --verbose &>log.txt Simon Le vendredi 19 janvier 2007 21:53, Craig Matsuura a écrit : > I can call between devices (direct) with version 1.2.0, however with > versions 1.4 and greater I do not hear any audio. I have double check the > .linphonerc setup. I actually get the dialing tones but once I answer > nothing. > > Seems the 1.2.0 version was working better than the 1.4.0 - 1.5.1 version > of linphone. > > When using asterisk as the exchange, the calls appear to route but now > sound. > > Does anyone ahve any ideas? I was previously using minisip and it worked > flawlessly. I would like to get linphone work as well, to evaluate. > > Thanks, > Craig > > On Thursday 18 January 2007 1:01 pm, Craig Matsuura wrote: > > I looking for some of the basic setup for linphonec to connect (use) a > > local asterisk server. > > > > My understanding is I have to setup a proxy server (proxy add) to my > > asterisk server? I'm unsure if I am doing it correctly as I can not call > > the other sip phones in my network. I had success with minisip, but > > prefer to use linphone as it is much smaller. > > > > I can dial directly from one linphone to another, but rather go via the > > asterisk server. > > > > Examples of a working configuration of asterisk and linphonec would be > > greatly appreciated, as well as usage (assuming I using linphonec > > incorrectly). > > > > Thanks, > > Craig > > > > > > ___ > > Linphone-users mailing list > > Linphone-users@nongnu.org > > http://lists.nongnu.org/mailman/listinfo/linphone-users > > ___ > Linphone-users mailing list > Linphone-users@nongnu.org > http://lists.nongnu.org/mailman/listinfo/linphone-users ___ Linphone-users mailing list Linphone-users@nongnu.org http://lists.nongnu.org/mailman/listinfo/linphone-users
Re: [Linphone-users] asterisk and linphonec setup
Thank you Simon. Here are two logs one is from linphone 1.5 and the other is from 1.2. I calling a SIP phone and with 1.2 I make a connection and everything works. With version 1.5 I make the call the to the SIP Phone, and everything appears to work but not audio. I will also attach my .linphonerc (or .linphonec) config too. Linphone 1.5 Log | INFO3 | MESSAGE REC. CALLID:1600048906 | INFO1 | cb_rcv2xx (id=8) | INFO1 | eXosip: timer sec:3 usec:52632! ortp-message-cfg= sip:192.168.0.21, cfg->rid=1, rid=1 | INFO1 | Release a non-terminated transaction | INFO2 | free transaction ressource 8 1600048906 | INFO2 | free nict ressource | INFO1 | Release a terminated transaction | INFO2 | free transaction ressource 4 147186140 | INFO2 | free ict ressource | INFO2 | free transaction ressource 6 147186140 | INFO2 | free nict ressource | INFO1 | Release a terminated transaction | INFO2 | free transaction ressource 5 147186140 | INFO2 | free ist ressource | ERROR | module sfp: _osip_kill_transaction transaction should be released by modules! | INFO2 | free transaction ressource 3 147186140 | INFO2 | free ict ressource Linephone 1.2 Log -- | INFO3 | MESSAGE REC. CALLID:939770138 | INFO1 | cb_rcv2xx (id=8) | INFO1 | eXosip: timer sec:3 usec:978010! | INFO1 | Release a non-terminated transaction | INFO2 | free transaction ressource 8 939770138 | INFO2 | free nict ressource | INFO1 | Release a terminated transaction | INFO2 | free transaction ressource 4 1891087716 | INFO2 | free ict ressource | INFO2 | free transaction ressource 6 1891087716 | INFO2 | free nict ressource | INFO1 | Release a terminated transaction | INFO2 | free transaction ressource 5 1891087716 | INFO2 | free ist ressource | ERROR | module sfp: _osip_kill_transaction transaction should be released by modules! | INFO2 | free transaction ressource 3 1891087716 | INFO2 | free ict ressource Linphone 1.2 config file (.linphonec) - [net] download_bw=0 upload_bw=0 use_stun=0 use_nat=0 con_type=3 [sip] sip_port=5060 guess_hostname=1 contact=sip:[EMAIL PROTECTED] inc_timeout=15 use_info=0 use_ipv6=0 default_proxy=0 [rtp] audio_rtp_port=7078 video_rtp_port=9078 audio_jitt_comp=60 video_jitt_comp=60 [sound] playback_dev_id=1 ringer_dev_id=1 capture_dev_id=1 local_ring=/control4/share/sounds/linphone/rings/oldphone.wav remote_ring=/usr/share/sounds/linphone/ringback.wav echocancelation=0 rec_lev=80 play_lev=80 ring_lev=80 source=m [video] enabled=0 show_local=0 [audio_codec_0] mime=speex rate=16000 enabled=1 [audio_codec_1] mime=speex rate=8000 enabled=1 [audio_codec_2] mime=PCMU rate=8000 enabled=1 [audio_codec_3] mime=GSM rate=8000 enabled=1 [audio_codec_4] mime=PCMA rate=8000 enabled=1 [proxy_0] reg_proxy=sip:192.168.0.21 reg_identity=sip:[EMAIL PROTECTED] reg_expires=600 reg_sendregister=1 publish=0 [audio_codec_5] mime=1015 rate=8000 enabled=1 [auth_info_0] username=mini-touch-000FFF002DF0 passwd=secret realm="asterisk" Linephone 1.5 config file (.linphonerc) --- [net] download_bw=0 upload_bw=0 firewall_policy=0 stun_server=192.168.0.21 [sip] sip_port=5060 guess_hostname=1 contact=sip:[EMAIL PROTECTED] inc_timeout=15 use_info=0 use_ipv6=0 default_proxy=0 [rtp] audio_rtp_port=7078 video_rtp_port=9078 audio_jitt_comp=60 video_jitt_comp=60 [sound] playback_dev_id=OSS: /dev/dsp1 ringer_dev_id=OSS: /dev/dsp1 capture_dev_id=OSS: /dev/dsp1 local_ring=/control4/share/sounds/linphone/rings/oldphone.wav remote_ring=/control4/share/sounds/linphone/ringback.wav echocancelation=0 rec_lev=80 play_lev=80 ring_lev=80 [video] enabled=0 show_local=0 [audio_codec_0] mime=speex rate=16000 enabled=1 [audio_codec_1] mime=speex rate=8000 enabled=1 [audio_codec_2] mime=PCMU rate=8000 enabled=1 [audio_codec_3] mime=GSM rate=8000 enabled=1 [audio_codec_4] mime=PCMA rate=8000 enabled=1 [proxy_0] reg_proxy=sip:192.168.0.21 reg_identity=sip:[EMAIL PROTECTED] reg_expires=600 reg_sendregister=1 publish=0 [auth_info_0] username=mini-touch-000FFF00165F passwd=secret realm="asterisk" On Monday 22 January 2007 7:22 am, Simon Morlat wrote: Thanks, Craig > --verbose &>log.txt ___ Linphone-users mailing list Linphone-users@nongnu.org http://lists.nongnu.org/mailman/listinfo/linphone-users
Re: [Linphone-users] asterisk and linphonec setup
The call is being made to the asterisk server (via ext), in the linphone 1.2 case it works. linphone 1.5 appears to connect, but not audio (I do hear a brief click). Craig On Monday 22 January 2007 12:45 pm, Craig Matsuura wrote: > Thank you Simon. Here are two logs one is from linphone 1.5 and the other > is from 1.2. I calling a SIP phone and with 1.2 I make a connection and > everything works. With version 1.5 I make the call the to the SIP Phone, > and everything appears to work but not audio. > > I will also attach my .linphonerc (or .linphonec) config too. > > Linphone 1.5 Log > > > | INFO3 | MESSAGE REC. CALLID:1600048906 > | INFO1 | cb_rcv2xx (id=8) > | INFO1 | eXosip: timer sec:3 usec:52632! > > ortp-message-cfg= sip:192.168.0.21, cfg->rid=1, rid=1 > > | INFO1 | Release a non-terminated transaction > | INFO2 | free transaction ressource 8 1600048906 > | INFO2 | free nict ressource > | INFO1 | Release a terminated transaction > | INFO2 | free transaction ressource 4 147186140 > | INFO2 | free ict ressource > | INFO2 | free transaction ressource 6 147186140 > | INFO2 | free nict ressource > | INFO1 | Release a terminated transaction > | INFO2 | free transaction ressource 5 147186140 > | INFO2 | free ist ressource > | ERROR | module sfp: _osip_kill_transaction transaction > > should be released by modules! > > | INFO2 | free transaction ressource 3 147186140 > | INFO2 | free ict ressource > > Linephone 1.2 Log > -- > > | INFO3 | MESSAGE REC. CALLID:939770138 > | INFO1 | cb_rcv2xx (id=8) > | INFO1 | eXosip: timer sec:3 usec:978010! > | INFO1 | Release a non-terminated transaction > | INFO2 | free transaction ressource 8 939770138 > | INFO2 | free nict ressource > | INFO1 | Release a terminated transaction > | INFO2 | free transaction ressource 4 1891087716 > | INFO2 | free ict ressource > | INFO2 | free transaction ressource 6 1891087716 > | INFO2 | free nict ressource > | INFO1 | Release a terminated transaction > | INFO2 | free transaction ressource 5 1891087716 > | INFO2 | free ist ressource > | ERROR | module sfp: _osip_kill_transaction transaction > > should be released by modules! > > | INFO2 | free transaction ressource 3 1891087716 > | INFO2 | free ict ressource > > Linphone 1.2 config file (.linphonec) > - > > [net] > download_bw=0 > upload_bw=0 > use_stun=0 > use_nat=0 > con_type=3 > > [sip] > sip_port=5060 > guess_hostname=1 > contact=sip:[EMAIL PROTECTED] > inc_timeout=15 > use_info=0 > use_ipv6=0 > default_proxy=0 > > [rtp] > audio_rtp_port=7078 > video_rtp_port=9078 > audio_jitt_comp=60 > video_jitt_comp=60 > > [sound] > playback_dev_id=1 > ringer_dev_id=1 > capture_dev_id=1 > local_ring=/control4/share/sounds/linphone/rings/oldphone.wav > remote_ring=/usr/share/sounds/linphone/ringback.wav > echocancelation=0 > rec_lev=80 > play_lev=80 > ring_lev=80 > source=m > > [video] > enabled=0 > show_local=0 > > [audio_codec_0] > mime=speex > rate=16000 > enabled=1 > > [audio_codec_1] > mime=speex > rate=8000 > enabled=1 > > [audio_codec_2] > mime=PCMU > rate=8000 > enabled=1 > > [audio_codec_3] > mime=GSM > rate=8000 > enabled=1 > > [audio_codec_4] > mime=PCMA > rate=8000 > enabled=1 > > [proxy_0] > reg_proxy=sip:192.168.0.21 > reg_identity=sip:[EMAIL PROTECTED] > reg_expires=600 > reg_sendregister=1 > publish=0 > > [audio_codec_5] > mime=1015 > rate=8000 > enabled=1 > > [auth_info_0] > username=mini-touch-000FFF002DF0 > passwd=secret > realm="asterisk" > > Linephone 1.5 config file (.linphonerc) > --- > > [net] > download_bw=0 > upload_bw=0 > firewall_policy=0 > stun_server=192.168.0.21 > > [sip] > sip_port=5060 > guess_hostname=1 > contact=sip:[EMAIL PROTECTED] > inc_timeout=15 > use_info=0 > use_ipv6=0 > default_proxy=0 > > [rtp] > audio_rtp_port=7078 > video_rtp_port=9078 > audio_jitt_comp=60 > video_jitt_comp=60 > > [sound] > playback_dev_id=OSS: /dev/dsp1 > ringer_dev_id=OSS: /dev/dsp1 > capture_dev_id=OSS: /dev/dsp1 > local_ring=/control4/share/sounds/linphone/rings/oldphone.wav > remote_ring=/control4/share/sounds/linphone/ringback.wav > echocancelation=0 > rec_lev=80 > play_lev=80 > ring_lev=80 > > [video] > enabled=0 > show_local=0 > > [audio_codec_0] > mime=speex > rate=16000 > enabled=1 > > [audio_codec_1] > mime=speex > rate=8000 > enabled=1 > > [audio_codec_2] > mime=PCMU > rate=8000 > enabled=1 > > [audio_codec_3] > mime=GSM > rate=8000 > enabled=1 > > [audio_codec_4] > mime=PCMA > rate=8000 > enabled=1 > > [proxy_0] > reg_proxy=sip:192.168.0.21 > reg_identity=sip:[EMAIL PROTECTED] > reg_expires=600 > reg_sendregister=1 > publish=0 > > [auth_info_0] > username=mini-touch-000FFF00165F > passwd=secret > realm="asterisk" > > On Monday 22 January 2007 7:22 am, Simon Morlat wrote: > > Thanks, > Craig > > > --verbose &>log.txt > > ___ > Linphone-users
Re: [Linphone-users] asterisk and linphonec setup
I have also noticed a lack of audio levels in the configuration file for 1.5. 1.2 had *_lev= settings. Craig On Monday 22 January 2007 3:14 pm, Craig Matsuura wrote: > The call is being made to the asterisk server (via ext), in the linphone > 1.2 case it works. linphone 1.5 appears to connect, but not audio (I do > hear a brief click). > > Craig > > On Monday 22 January 2007 12:45 pm, Craig Matsuura wrote: > > Thank you Simon. Here are two logs one is from linphone 1.5 and the > > other is from 1.2. I calling a SIP phone and with 1.2 I make a > > connection and everything works. With version 1.5 I make the call the to > > the SIP Phone, and everything appears to work but not audio. > > > > I will also attach my .linphonerc (or .linphonec) config too. > > > > Linphone 1.5 Log > > > > > > | INFO3 | MESSAGE REC. CALLID:1600048906 > > | INFO1 | cb_rcv2xx (id=8) > > | INFO1 | eXosip: timer sec:3 usec:52632! > > > > ortp-message-cfg= sip:192.168.0.21, cfg->rid=1, rid=1 > > > > | INFO1 | Release a non-terminated transaction > > | INFO2 | free transaction ressource 8 > > | 1600048906 INFO2 | free nict ressource > > | INFO1 | Release a terminated transaction > > | INFO2 | free transaction ressource 4 > > | 147186140 INFO2 | free ict ressource > > | INFO2 | free transaction ressource 6 > > | 147186140 INFO2 | free nict ressource > > | INFO1 | Release a terminated transaction > > | INFO2 | free transaction ressource 5 > > | 147186140 INFO2 | free ist ressource > > | ERROR | module sfp: _osip_kill_transaction transaction > > > > should be released by modules! > > > > | INFO2 | free transaction ressource 3 > > | 147186140 INFO2 | free ict ressource > > > > Linephone 1.2 Log > > -- > > > > | INFO3 | MESSAGE REC. CALLID:939770138 > > | INFO1 | cb_rcv2xx (id=8) > > | INFO1 | eXosip: timer sec:3 usec:978010! > > | INFO1 | Release a non-terminated transaction > > | INFO2 | free transaction ressource 8 > > | 939770138 INFO2 | free nict ressource > > | INFO1 | Release a terminated transaction > > | INFO2 | free transaction ressource 4 > > | 1891087716 INFO2 | free ict ressource > > | INFO2 | free transaction ressource 6 > > | 1891087716 INFO2 | free nict ressource > > | INFO1 | Release a terminated transaction > > | INFO2 | free transaction ressource 5 > > | 1891087716 INFO2 | free ist ressource > > | ERROR | module sfp: _osip_kill_transaction transaction > > > > should be released by modules! > > > > | INFO2 | free transaction ressource 3 > > | 1891087716 INFO2 | free ict ressource > > > > Linphone 1.2 config file (.linphonec) > > - > > > > [net] > > download_bw=0 > > upload_bw=0 > > use_stun=0 > > use_nat=0 > > con_type=3 > > > > [sip] > > sip_port=5060 > > guess_hostname=1 > > contact=sip:[EMAIL PROTECTED] > > inc_timeout=15 > > use_info=0 > > use_ipv6=0 > > default_proxy=0 > > > > [rtp] > > audio_rtp_port=7078 > > video_rtp_port=9078 > > audio_jitt_comp=60 > > video_jitt_comp=60 > > > > [sound] > > playback_dev_id=1 > > ringer_dev_id=1 > > capture_dev_id=1 > > local_ring=/control4/share/sounds/linphone/rings/oldphone.wav > > remote_ring=/usr/share/sounds/linphone/ringback.wav > > echocancelation=0 > > rec_lev=80 > > play_lev=80 > > ring_lev=80 > > source=m > > > > [video] > > enabled=0 > > show_local=0 > > > > [audio_codec_0] > > mime=speex > > rate=16000 > > enabled=1 > > > > [audio_codec_1] > > mime=speex > > rate=8000 > > enabled=1 > > > > [audio_codec_2] > > mime=PCMU > > rate=8000 > > enabled=1 > > > > [audio_codec_3] > > mime=GSM > > rate=8000 > > enabled=1 > > > > [audio_codec_4] > > mime=PCMA > > rate=8000 > > enabled=1 > > > > [proxy_0] > > reg_proxy=sip:192.168.0.21 > > reg_identity=sip:[EMAIL PROTECTED] > > reg_expires=600 > > reg_sendregister=1 > > publish=0 > > > > [audio_codec_5] > > mime=1015 > > rate=8000 > > enabled=1 > > > > [auth_info_0] > > username=mini-touch-000FFF002DF0 > > passwd=secret > > realm="asterisk" > > > > Linephone 1.5 config file (.linphonerc) > > --- > > > > [net] > > download_bw=0 > > upload_bw=0 > > firewall_policy=0 > > stun_server=192.168.0.21 > > > > [sip] > > sip_port=5060 > > guess_hostname=1 > > contact=sip:[EMAIL PROTECTED] > > inc_timeout=15 > > use_info=0 > > use_ipv6=0 > > default_proxy=0 > > > > [rtp] > > audio_rtp_port=7078 > > video_rtp_port=9078 > > audio_jitt_comp=60 > > video_jitt_comp=60 > > > > [sound] > > playback_dev_id=OSS: /dev/dsp1 > > ringer_dev_id=OSS: /dev/dsp1 > > capture_dev_id=OSS: /dev/dsp1 > > local_ring=/control4/share/sounds/linphone/rings/oldphone.wav > > remote_ring=/control4/share/sounds/linphone/ringback.wav > > echocancelation=0 > > rec_lev=80 > > play_lev=80 > > ring_lev=80 > > > > [video] > > enabled=0 > > show_local=0 > > > > [audio_codec_0] > > mime=speex > > rate=16000 > > enabled=1 > > > > [audio_codec_1] > > mime=speex > > rate=8000 > > enabled=1 >
Re: [Linphone-users] asterisk and linphonec setup
The logs are incomplete ! use linphone --verbose &>log.txt and send as attachement the log.txt file. Simon Le lundi 22 janvier 2007 20:45, Craig Matsuura a écrit : > Thank you Simon. Here are two logs one is from linphone 1.5 and the other > is from 1.2. I calling a SIP phone and with 1.2 I make a connection and > everything works. With version 1.5 I make the call the to the SIP Phone, > and everything appears to work but not audio. > > I will also attach my .linphonerc (or .linphonec) config too. > > Linphone 1.5 Log > > > | INFO3 | MESSAGE REC. CALLID:1600048906 > | INFO1 | cb_rcv2xx (id=8) > | INFO1 | eXosip: timer sec:3 usec:52632! > > ortp-message-cfg= sip:192.168.0.21, cfg->rid=1, rid=1 > > | INFO1 | Release a non-terminated transaction > | INFO2 | free transaction ressource 8 1600048906 > | INFO2 | free nict ressource > | INFO1 | Release a terminated transaction > | INFO2 | free transaction ressource 4 147186140 > | INFO2 | free ict ressource > | INFO2 | free transaction ressource 6 147186140 > | INFO2 | free nict ressource > | INFO1 | Release a terminated transaction > | INFO2 | free transaction ressource 5 147186140 > | INFO2 | free ist ressource > | ERROR | module sfp: _osip_kill_transaction transaction > > should be released by modules! > > | INFO2 | free transaction ressource 3 147186140 > | INFO2 | free ict ressource > > Linephone 1.2 Log > -- > > | INFO3 | MESSAGE REC. CALLID:939770138 > | INFO1 | cb_rcv2xx (id=8) > | INFO1 | eXosip: timer sec:3 usec:978010! > | INFO1 | Release a non-terminated transaction > | INFO2 | free transaction ressource 8 939770138 > | INFO2 | free nict ressource > | INFO1 | Release a terminated transaction > | INFO2 | free transaction ressource 4 1891087716 > | INFO2 | free ict ressource > | INFO2 | free transaction ressource 6 1891087716 > | INFO2 | free nict ressource > | INFO1 | Release a terminated transaction > | INFO2 | free transaction ressource 5 1891087716 > | INFO2 | free ist ressource > | ERROR | module sfp: _osip_kill_transaction transaction > > should be released by modules! > > | INFO2 | free transaction ressource 3 1891087716 > | INFO2 | free ict ressource > > Linphone 1.2 config file (.linphonec) > - > > [net] > download_bw=0 > upload_bw=0 > use_stun=0 > use_nat=0 > con_type=3 > > [sip] > sip_port=5060 > guess_hostname=1 > contact=sip:[EMAIL PROTECTED] > inc_timeout=15 > use_info=0 > use_ipv6=0 > default_proxy=0 > > [rtp] > audio_rtp_port=7078 > video_rtp_port=9078 > audio_jitt_comp=60 > video_jitt_comp=60 > > [sound] > playback_dev_id=1 > ringer_dev_id=1 > capture_dev_id=1 > local_ring=/control4/share/sounds/linphone/rings/oldphone.wav > remote_ring=/usr/share/sounds/linphone/ringback.wav > echocancelation=0 > rec_lev=80 > play_lev=80 > ring_lev=80 > source=m > > [video] > enabled=0 > show_local=0 > > [audio_codec_0] > mime=speex > rate=16000 > enabled=1 > > [audio_codec_1] > mime=speex > rate=8000 > enabled=1 > > [audio_codec_2] > mime=PCMU > rate=8000 > enabled=1 > > [audio_codec_3] > mime=GSM > rate=8000 > enabled=1 > > [audio_codec_4] > mime=PCMA > rate=8000 > enabled=1 > > [proxy_0] > reg_proxy=sip:192.168.0.21 > reg_identity=sip:[EMAIL PROTECTED] > reg_expires=600 > reg_sendregister=1 > publish=0 > > [audio_codec_5] > mime=1015 > rate=8000 > enabled=1 > > [auth_info_0] > username=mini-touch-000FFF002DF0 > passwd=secret > realm="asterisk" > > Linephone 1.5 config file (.linphonerc) > --- > > [net] > download_bw=0 > upload_bw=0 > firewall_policy=0 > stun_server=192.168.0.21 > > [sip] > sip_port=5060 > guess_hostname=1 > contact=sip:[EMAIL PROTECTED] > inc_timeout=15 > use_info=0 > use_ipv6=0 > default_proxy=0 > > [rtp] > audio_rtp_port=7078 > video_rtp_port=9078 > audio_jitt_comp=60 > video_jitt_comp=60 > > [sound] > playback_dev_id=OSS: /dev/dsp1 > ringer_dev_id=OSS: /dev/dsp1 > capture_dev_id=OSS: /dev/dsp1 > local_ring=/control4/share/sounds/linphone/rings/oldphone.wav > remote_ring=/control4/share/sounds/linphone/ringback.wav > echocancelation=0 > rec_lev=80 > play_lev=80 > ring_lev=80 > > [video] > enabled=0 > show_local=0 > > [audio_codec_0] > mime=speex > rate=16000 > enabled=1 > > [audio_codec_1] > mime=speex > rate=8000 > enabled=1 > > [audio_codec_2] > mime=PCMU > rate=8000 > enabled=1 > > [audio_codec_3] > mime=GSM > rate=8000 > enabled=1 > > [audio_codec_4] > mime=PCMA > rate=8000 > enabled=1 > > [proxy_0] > reg_proxy=sip:192.168.0.21 > reg_identity=sip:[EMAIL PROTECTED] > reg_expires=600 > reg_sendregister=1 > publish=0 > > [auth_info_0] > username=mini-touch-000FFF00165F > passwd=secret > realm="asterisk" > > On Monday 22 January 2007 7:22 am, Simon Morlat wrote: > > Thanks, > Craig > > > --verbose &>log.txt > > ___ > Linphone-users mailing list > Linphone-users@nongnu.org > http://lists.nongnu.org/ma
Re: [Linphone-users] asterisk and linphonec setup
Le mardi 23 janvier 2007 00:56, Craig Matsuura a écrit : > I have also noticed a lack of audio levels in the configuration file for > 1.5. 1.2 had *_lev= settings. Actually it's a feature: many people told me that we were tired of linphone changing audio levels at startup... So I removed the storage of those levels. Simon > > Craig > > On Monday 22 January 2007 3:14 pm, Craig Matsuura wrote: > > The call is being made to the asterisk server (via ext), in the linphone > > 1.2 case it works. linphone 1.5 appears to connect, but not audio (I do > > hear a brief click). > > > > Craig > > > > On Monday 22 January 2007 12:45 pm, Craig Matsuura wrote: > > > Thank you Simon. Here are two logs one is from linphone 1.5 and the > > > other is from 1.2. I calling a SIP phone and with 1.2 I make a > > > connection and everything works. With version 1.5 I make the call the > > > to the SIP Phone, and everything appears to work but not audio. > > > > > > I will also attach my .linphonerc (or .linphonec) config too. > > > > > > Linphone 1.5 Log > > > > > > > > > | INFO3 | MESSAGE REC. CALLID:1600048906 > > > | INFO1 | cb_rcv2xx (id=8) > > > | INFO1 | eXosip: timer sec:3 usec:52632! > > > > > > ortp-message-cfg= sip:192.168.0.21, cfg->rid=1, rid=1 > > > > > > | INFO1 | Release a non-terminated transaction > > > | INFO2 | free transaction ressource 8 > > > | 1600048906 INFO2 | free nict ressource > > > | INFO1 | Release a terminated transaction > > > | INFO2 | free transaction ressource 4 > > > | 147186140 INFO2 | free ict ressource > > > | INFO2 | free transaction ressource 6 > > > | 147186140 INFO2 | free nict ressource > > > | INFO1 | Release a terminated transaction > > > | INFO2 | free transaction ressource 5 > > > | 147186140 INFO2 | free ist ressource > > > | ERROR | module sfp: _osip_kill_transaction > > > | transaction > > > > > > should be released by modules! > > > > > > | INFO2 | free transaction ressource 3 > > > | 147186140 INFO2 | free ict ressource > > > > > > Linephone 1.2 Log > > > -- > > > > > > | INFO3 | MESSAGE REC. CALLID:939770138 > > > | INFO1 | cb_rcv2xx (id=8) > > > | INFO1 | eXosip: timer sec:3 usec:978010! > > > | INFO1 | Release a non-terminated transaction > > > | INFO2 | free transaction ressource 8 > > > | 939770138 INFO2 | free nict ressource > > > | INFO1 | Release a terminated transaction > > > | INFO2 | free transaction ressource 4 > > > | 1891087716 INFO2 | free ict ressource > > > | INFO2 | free transaction ressource 6 > > > | 1891087716 INFO2 | free nict ressource > > > | INFO1 | Release a terminated transaction > > > | INFO2 | free transaction ressource 5 > > > | 1891087716 INFO2 | free ist ressource > > > | ERROR | module sfp: _osip_kill_transaction > > > | transaction > > > > > > should be released by modules! > > > > > > | INFO2 | free transaction ressource 3 > > > | 1891087716 INFO2 | free ict ressource > > > > > > Linphone 1.2 config file (.linphonec) > > > - > > > > > > [net] > > > download_bw=0 > > > upload_bw=0 > > > use_stun=0 > > > use_nat=0 > > > con_type=3 > > > > > > [sip] > > > sip_port=5060 > > > guess_hostname=1 > > > contact=sip:[EMAIL PROTECTED] > > > inc_timeout=15 > > > use_info=0 > > > use_ipv6=0 > > > default_proxy=0 > > > > > > [rtp] > > > audio_rtp_port=7078 > > > video_rtp_port=9078 > > > audio_jitt_comp=60 > > > video_jitt_comp=60 > > > > > > [sound] > > > playback_dev_id=1 > > > ringer_dev_id=1 > > > capture_dev_id=1 > > > local_ring=/control4/share/sounds/linphone/rings/oldphone.wav > > > remote_ring=/usr/share/sounds/linphone/ringback.wav > > > echocancelation=0 > > > rec_lev=80 > > > play_lev=80 > > > ring_lev=80 > > > source=m > > > > > > [video] > > > enabled=0 > > > show_local=0 > > > > > > [audio_codec_0] > > > mime=speex > > > rate=16000 > > > enabled=1 > > > > > > [audio_codec_1] > > > mime=speex > > > rate=8000 > > > enabled=1 > > > > > > [audio_codec_2] > > > mime=PCMU > > > rate=8000 > > > enabled=1 > > > > > > [audio_codec_3] > > > mime=GSM > > > rate=8000 > > > enabled=1 > > > > > > [audio_codec_4] > > > mime=PCMA > > > rate=8000 > > > enabled=1 > > > > > > [proxy_0] > > > reg_proxy=sip:192.168.0.21 > > > reg_identity=sip:[EMAIL PROTECTED] > > > reg_expires=600 > > > reg_sendregister=1 > > > publish=0 > > > > > > [audio_codec_5] > > > mime=1015 > > > rate=8000 > > > enabled=1 > > > > > > [auth_info_0] > > > username=mini-touch-000FFF002DF0 > > > passwd=secret > > > realm="asterisk" > > > > > > Linephone 1.5 config file (.linphonerc) > > > --- > > > > > > [net] > > > download_bw=0 > > > upload_bw=0 > > > firewall_policy=0 > > > stun_server=192.168.0.21 > > > > > > [sip] > > > sip_port=5060 > > > guess_hostname=1 > > > contact=sip:[EMAIL PROTECTED] > > > inc_timeout=15 > > > use_info=0 > > > use_ipv6=0 > > > default_proxy=0 > > > > > > [rtp] > > > audio_r
Re: [Linphone-users] asterisk and linphonec setup
linephonec only appears to have a -d . usage: linphonec [-c file] [-s sipaddr] [-a] [-V] [-d level ] [-l logfile] linphonec -v -c file specify path of configuration file. -d levelbe verbose. 0 is no output. 6 is all output -l logfile specify the log file for your SIP phone -s sipaddress specify the sip call to do at startup -a enable auto answering for incoming calls -V enable video (disabled by default) -v or --version display version and exits. I used - 6 and -l log.txt Anything else you can recommend to get you more info? Thanks, Craig On Wednesday 24 January 2007 8:21 am, Simon Morlat wrote: > The logs are incomplete ! > use linphone --verbose &>log.txt and send as attachement the log.txt file. > > Simon > > Le lundi 22 janvier 2007 20:45, Craig Matsuura a écrit : > > Thank you Simon. Here are two logs one is from linphone 1.5 and the > > other is from 1.2. I calling a SIP phone and with 1.2 I make a > > connection and everything works. With version 1.5 I make the call the to > > the SIP Phone, and everything appears to work but not audio. > > > > I will also attach my .linphonerc (or .linphonec) config too. > > > > Linphone 1.5 Log > > > > > > | INFO3 | MESSAGE REC. CALLID:1600048906 > > | INFO1 | cb_rcv2xx (id=8) > > | INFO1 | eXosip: timer sec:3 usec:52632! > > > > ortp-message-cfg= sip:192.168.0.21, cfg->rid=1, rid=1 > > > > | INFO1 | Release a non-terminated transaction > > | INFO2 | free transaction ressource 8 > > | 1600048906 INFO2 | free nict ressource > > | INFO1 | Release a terminated transaction > > | INFO2 | free transaction ressource 4 > > | 147186140 INFO2 | free ict ressource > > | INFO2 | free transaction ressource 6 > > | 147186140 INFO2 | free nict ressource > > | INFO1 | Release a terminated transaction > > | INFO2 | free transaction ressource 5 > > | 147186140 INFO2 | free ist ressource > > | ERROR | module sfp: _osip_kill_transaction transaction > > > > should be released by modules! > > > > | INFO2 | free transaction ressource 3 > > | 147186140 INFO2 | free ict ressource > > > > Linephone 1.2 Log > > -- > > > > | INFO3 | MESSAGE REC. CALLID:939770138 > > | INFO1 | cb_rcv2xx (id=8) > > | INFO1 | eXosip: timer sec:3 usec:978010! > > | INFO1 | Release a non-terminated transaction > > | INFO2 | free transaction ressource 8 > > | 939770138 INFO2 | free nict ressource > > | INFO1 | Release a terminated transaction > > | INFO2 | free transaction ressource 4 > > | 1891087716 INFO2 | free ict ressource > > | INFO2 | free transaction ressource 6 > > | 1891087716 INFO2 | free nict ressource > > | INFO1 | Release a terminated transaction > > | INFO2 | free transaction ressource 5 > > | 1891087716 INFO2 | free ist ressource > > | ERROR | module sfp: _osip_kill_transaction transaction > > > > should be released by modules! > > > > | INFO2 | free transaction ressource 3 > > | 1891087716 INFO2 | free ict ressource > > > > Linphone 1.2 config file (.linphonec) > > - > > > > [net] > > download_bw=0 > > upload_bw=0 > > use_stun=0 > > use_nat=0 > > con_type=3 > > > > [sip] > > sip_port=5060 > > guess_hostname=1 > > contact=sip:[EMAIL PROTECTED] > > inc_timeout=15 > > use_info=0 > > use_ipv6=0 > > default_proxy=0 > > > > [rtp] > > audio_rtp_port=7078 > > video_rtp_port=9078 > > audio_jitt_comp=60 > > video_jitt_comp=60 > > > > [sound] > > playback_dev_id=1 > > ringer_dev_id=1 > > capture_dev_id=1 > > local_ring=/control4/share/sounds/linphone/rings/oldphone.wav > > remote_ring=/usr/share/sounds/linphone/ringback.wav > > echocancelation=0 > > rec_lev=80 > > play_lev=80 > > ring_lev=80 > > source=m > > > > [video] > > enabled=0 > > show_local=0 > > > > [audio_codec_0] > > mime=speex > > rate=16000 > > enabled=1 > > > > [audio_codec_1] > > mime=speex > > rate=8000 > > enabled=1 > > > > [audio_codec_2] > > mime=PCMU > > rate=8000 > > enabled=1 > > > > [audio_codec_3] > > mime=GSM > > rate=8000 > > enabled=1 > > > > [audio_codec_4] > > mime=PCMA > > rate=8000 > > enabled=1 > > > > [proxy_0] > > reg_proxy=sip:192.168.0.21 > > reg_identity=sip:[EMAIL PROTECTED] > > reg_expires=600 > > reg_sendregister=1 > > publish=0 > > > > [audio_codec_5] > > mime=1015 > > rate=8000 > > enabled=1 > > > > [auth_info_0] > > username=mini-touch-000FFF002DF0 > > passwd=secret > > realm="asterisk" > > > > Linephone 1.5 config file (.linphonerc) > > --- > > > > [net] > > download_bw=0 > > upload_bw=0 > > firewall_policy=0 > > stun_server=192.168.0.21 > > > > [sip] > > sip_port=5060 > > guess_hostname=1 > > contact=sip:[EMAIL PROTECTED] > > inc_timeout=15 > > use_info=0 > > use_ipv6=0 > > default_proxy=0 > > > > [rtp] > > audio_rtp_port=7078 > > video_rtp_port=9078 > > audio_jitt_comp=60 > > video_jitt_comp=60 > > > > [sound] > >
Re: [Linphone-users] asterisk and linphonec setup
Here is a better log, this is from version 1.5 calling a sip phone via asterisk ext. Thanks, Craig ortp-message-oRTP-0.13.0 initialized. ortp-message-Registering all filters... ortp-message-Registering all soundcard handlers ortp-message-Card OSS: /dev/dsp added ortp-message-Card OSS: /dev/dsp added ortp-message-Card OSS: /dev/dsp0 added ortp-message-Card OSS: /dev/dsp1 added ortp-message-Card OSS: /dev/dsp2 added ortp-message-Loading plugins ortp-message-Cannot open directory /control4/lib/mediastreamer/plugins: No such file or directory ortp-message-ms_init() done | INFO1 | Outgoing interface to reach 15.128.128.93 is 192.168.0.101. | INFO1 | eXosip: Reseting timer to 15s before waking up! | INFO1 | Outgoing interface to reach 192.168.0.21 is 192.168.0.101. | INFO2 | allocating transaction ressource 1 1636995824 | INFO2 | allocating NICT context | INFO2 | IPv4 address detected: 192.168.0.21 | INFO2 | DNS resolution with 192.168.0.21:5060 | INFO1 | Message sent: REGISTER sip:192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK1957117508 From: ;tag=1057953707 To: Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Contact: Max-Forwards: 5 User-Agent: Linphone-1.6.0/eXosip Expires: 600 Content-Length: 0 (len=16 sizeof(addr)=128 28) | INFO1 | cb_sndregister (id=1) | INFO1 | eXosip: timer sec:0 usec:492884! | INFO1 | Received message: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK1957117508;received=192.168.0.101 From: ;tag=1057953707 To: Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 | INFO3 | MESSAGE REC. CALLID:1636995824 | INFO1 | cb_rcv1xx (id=1) | INFO1 | eXosip: timer sec:0 usec:484075! | INFO1 | Received message: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK1957117508;received=192.168.0.101 From: ;tag=1057953707 To: ;tag=as7c4f6434 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="57d4853c" Content-Length: 0 | INFO3 | MESSAGE REC. CALLID:1636995824 | INFO1 | cb_rcv4xx (id=1) | INFO1 | eXosip: timer sec:4 usec:12! ortp-message-REGISTRATION_FAILURE ortp-message-cfg= sip:192.168.0.21, cfg->rid=1, rid=1 | INFO2 | free nict ressource | INFO2 | INFO: authinfo: "asterisk" "asterisk" | INFO1 | authinfo: mini-touch-000FFF00165F | INFO2 | allocating transaction ressource 2 1636995824 | INFO2 | allocating NICT context | INFO2 | IPv4 address detected: 192.168.0.21 | INFO2 | DNS resolution with 192.168.0.21:5060 | INFO1 | Message sent: REGISTER sip:192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK1200015116 From: ;tag=1057953707 To: Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER Contact: Authorization: Digest username="mini-touch-000FFF00165F", realm="asterisk", nonce="57d4853c", uri="sip:192.168.0.21", response="5da2bc5215e774c3ec0b46920debd29d", algorithm=MD5 Max-Forwards: 5 User-Agent: Linphone-1.6.0/eXosip Expires: 600 Content-Length: 0 (len=16 sizeof(addr)=128 28) | INFO1 | cb_sndregister (id=2) | INFO1 | eXosip: timer sec:0 usec:497286! | INFO1 | Received message: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK1200015116;received=192.168.0.101 From: ;tag=1057953707 To: Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 | INFO3 | MESSAGE REC. CALLID:1636995824 | INFO1 | cb_rcv1xx (id=2) | INFO1 | eXosip: timer sec:0 usec:491994! | INFO1 | Received message: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK1200015116;received=192.168.0.101 From: ;tag=1057953707 To: ;tag=as7c4f6434 Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 600 Contact: ;expires=600 Date: Wed, 24 Jan 2007 19:02:46 GMT Content-Length: 0 | INFO3 | MESSAGE REC. CALLID:1636995824 | INFO1 | cb_rcv2xx (id=2) | INFO1 | eXosip: timer sec:4 usec:24! ortp-message-cfg= sip:192.168.0.21, cfg->rid=1, rid=1 | INFO1 | Outgoing interface to reach 192.168.0.21 is 192.168.0.101. | INFO1 | Outgoing interface to reach 192.168.0.21 is 192.168.0.101. | INFO2 | allocating transaction ressource 3 1037700762 | INFO2 | allocating ICT context | INFO2 | IPv4 address detected: 192.168.0.21 | INFO2 | DNS resolution with 192.168.0.21:5060 | INFO1 | Message sent: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK23191995 From: ;tag=70234470 To: Call-ID: [EMAIL PROTECTED] CSeq: 20 INVITE Contact: Max-Forwards: 5 User-Agent: Linphone-1.6.0/eXosip Subject: Phone call Expires: 120 Allow: INVITE, ACK, CANCEL, BYE, OPTION
Re: [Linphone-users] asterisk and linphonec setup
I replied on another email, however I will attach the logs for version 1.5 on my device. thanks, Craig On Wednesday 24 January 2007 8:21 am, Simon Morlat wrote: > The logs are incomplete ! > use linphone --verbose &>log.txt and send as attachement the log.txt file. > > Simon > > Le lundi 22 janvier 2007 20:45, Craig Matsuura a écrit : > > Thank you Simon. Here are two logs one is from linphone 1.5 and the > > other is from 1.2. I calling a SIP phone and with 1.2 I make a > > connection and everything works. With version 1.5 I make the call the to > > the SIP Phone, and everything appears to work but not audio. > > > > I will also attach my .linphonerc (or .linphonec) config too. > > > > Linphone 1.5 Log > > > > > > | INFO3 | MESSAGE REC. CALLID:1600048906 > > | INFO1 | cb_rcv2xx (id=8) > > | INFO1 | eXosip: timer sec:3 usec:52632! > > > > ortp-message-cfg= sip:192.168.0.21, cfg->rid=1, rid=1 > > > > | INFO1 | Release a non-terminated transaction > > | INFO2 | free transaction ressource 8 > > | 1600048906 INFO2 | free nict ressource > > | INFO1 | Release a terminated transaction > > | INFO2 | free transaction ressource 4 > > | 147186140 INFO2 | free ict ressource > > | INFO2 | free transaction ressource 6 > > | 147186140 INFO2 | free nict ressource > > | INFO1 | Release a terminated transaction > > | INFO2 | free transaction ressource 5 > > | 147186140 INFO2 | free ist ressource > > | ERROR | module sfp: _osip_kill_transaction transaction > > > > should be released by modules! > > > > | INFO2 | free transaction ressource 3 > > | 147186140 INFO2 | free ict ressource > > > > Linephone 1.2 Log > > -- > > > > | INFO3 | MESSAGE REC. CALLID:939770138 > > | INFO1 | cb_rcv2xx (id=8) > > | INFO1 | eXosip: timer sec:3 usec:978010! > > | INFO1 | Release a non-terminated transaction > > | INFO2 | free transaction ressource 8 > > | 939770138 INFO2 | free nict ressource > > | INFO1 | Release a terminated transaction > > | INFO2 | free transaction ressource 4 > > | 1891087716 INFO2 | free ict ressource > > | INFO2 | free transaction ressource 6 > > | 1891087716 INFO2 | free nict ressource > > | INFO1 | Release a terminated transaction > > | INFO2 | free transaction ressource 5 > > | 1891087716 INFO2 | free ist ressource > > | ERROR | module sfp: _osip_kill_transaction transaction > > > > should be released by modules! > > > > | INFO2 | free transaction ressource 3 > > | 1891087716 INFO2 | free ict ressource > > > > Linphone 1.2 config file (.linphonec) > > - > > > > [net] > > download_bw=0 > > upload_bw=0 > > use_stun=0 > > use_nat=0 > > con_type=3 > > > > [sip] > > sip_port=5060 > > guess_hostname=1 > > contact=sip:[EMAIL PROTECTED] > > inc_timeout=15 > > use_info=0 > > use_ipv6=0 > > default_proxy=0 > > > > [rtp] > > audio_rtp_port=7078 > > video_rtp_port=9078 > > audio_jitt_comp=60 > > video_jitt_comp=60 > > > > [sound] > > playback_dev_id=1 > > ringer_dev_id=1 > > capture_dev_id=1 > > local_ring=/control4/share/sounds/linphone/rings/oldphone.wav > > remote_ring=/usr/share/sounds/linphone/ringback.wav > > echocancelation=0 > > rec_lev=80 > > play_lev=80 > > ring_lev=80 > > source=m > > > > [video] > > enabled=0 > > show_local=0 > > > > [audio_codec_0] > > mime=speex > > rate=16000 > > enabled=1 > > > > [audio_codec_1] > > mime=speex > > rate=8000 > > enabled=1 > > > > [audio_codec_2] > > mime=PCMU > > rate=8000 > > enabled=1 > > > > [audio_codec_3] > > mime=GSM > > rate=8000 > > enabled=1 > > > > [audio_codec_4] > > mime=PCMA > > rate=8000 > > enabled=1 > > > > [proxy_0] > > reg_proxy=sip:192.168.0.21 > > reg_identity=sip:[EMAIL PROTECTED] > > reg_expires=600 > > reg_sendregister=1 > > publish=0 > > > > [audio_codec_5] > > mime=1015 > > rate=8000 > > enabled=1 > > > > [auth_info_0] > > username=mini-touch-000FFF002DF0 > > passwd=secret > > realm="asterisk" > > > > Linephone 1.5 config file (.linphonerc) > > --- > > > > [net] > > download_bw=0 > > upload_bw=0 > > firewall_policy=0 > > stun_server=192.168.0.21 > > > > [sip] > > sip_port=5060 > > guess_hostname=1 > > contact=sip:[EMAIL PROTECTED] > > inc_timeout=15 > > use_info=0 > > use_ipv6=0 > > default_proxy=0 > > > > [rtp] > > audio_rtp_port=7078 > > video_rtp_port=9078 > > audio_jitt_comp=60 > > video_jitt_comp=60 > > > > [sound] > > playback_dev_id=OSS: /dev/dsp1 > > ringer_dev_id=OSS: /dev/dsp1 > > capture_dev_id=OSS: /dev/dsp1 > > local_ring=/control4/share/sounds/linphone/rings/oldphone.wav > > remote_ring=/control4/share/sounds/linphone/ringback.wav > > echocancelation=0 > > rec_lev=80 > > play_lev=80 > > ring_lev=80 > > > > [video] > > enabled=0 > > show_local=0 > > > > [audio_codec_0] > > mime=speex > > rate=16000 > > enabled=1 > > > > [audio_codec_1] > > mime=speex > > rate=8000 > > enabled=1 > > > > [audio_codec_2] > > mime=PCMU > > rate=8000 > > enabled=1 > > > > [a
Re: [Linphone-users] asterisk and linphonec setup
Hi Craig, I see two problems in your logs: - linphone fails to read sound from the oss devices (actually the driver returns no samples): if an alsa driver exists for your device, prefer to use. Maybe your oss driver isn't full-duplex ? - the asterisk server does re-INVITE and currently linphone does not support this. Check if there is way to disable the re-INVITE feature from asterisk ? Simon Le mercredi 24 janvier 2007 20:11, Craig Matsuura a écrit : > I replied on another email, however I will attach the logs for version 1.5 > on my device. > > thanks, > Craig > > On Wednesday 24 January 2007 8:21 am, Simon Morlat wrote: > > The logs are incomplete ! > > use linphone --verbose &>log.txt and send as attachement the log.txt > > file. > > > > Simon > > > > Le lundi 22 janvier 2007 20:45, Craig Matsuura a écrit : > > > Thank you Simon. Here are two logs one is from linphone 1.5 and the > > > other is from 1.2. I calling a SIP phone and with 1.2 I make a > > > connection and everything works. With version 1.5 I make the call the > > > to the SIP Phone, and everything appears to work but not audio. > > > > > > I will also attach my .linphonerc (or .linphonec) config too. > > > > > > Linphone 1.5 Log > > > > > > > > > | INFO3 | MESSAGE REC. CALLID:1600048906 > > > | INFO1 | cb_rcv2xx (id=8) > > > | INFO1 | eXosip: timer sec:3 usec:52632! > > > > > > ortp-message-cfg= sip:192.168.0.21, cfg->rid=1, rid=1 > > > > > > | INFO1 | Release a non-terminated transaction > > > | INFO2 | free transaction ressource 8 > > > | 1600048906 INFO2 | free nict ressource > > > | INFO1 | Release a terminated transaction > > > | INFO2 | free transaction ressource 4 > > > | 147186140 INFO2 | free ict ressource > > > | INFO2 | free transaction ressource 6 > > > | 147186140 INFO2 | free nict ressource > > > | INFO1 | Release a terminated transaction > > > | INFO2 | free transaction ressource 5 > > > | 147186140 INFO2 | free ist ressource > > > | ERROR | module sfp: _osip_kill_transaction > > > | transaction > > > > > > should be released by modules! > > > > > > | INFO2 | free transaction ressource 3 > > > | 147186140 INFO2 | free ict ressource > > > > > > Linephone 1.2 Log > > > -- > > > > > > | INFO3 | MESSAGE REC. CALLID:939770138 > > > | INFO1 | cb_rcv2xx (id=8) > > > | INFO1 | eXosip: timer sec:3 usec:978010! > > > | INFO1 | Release a non-terminated transaction > > > | INFO2 | free transaction ressource 8 > > > | 939770138 INFO2 | free nict ressource > > > | INFO1 | Release a terminated transaction > > > | INFO2 | free transaction ressource 4 > > > | 1891087716 INFO2 | free ict ressource > > > | INFO2 | free transaction ressource 6 > > > | 1891087716 INFO2 | free nict ressource > > > | INFO1 | Release a terminated transaction > > > | INFO2 | free transaction ressource 5 > > > | 1891087716 INFO2 | free ist ressource > > > | ERROR | module sfp: _osip_kill_transaction > > > | transaction > > > > > > should be released by modules! > > > > > > | INFO2 | free transaction ressource 3 > > > | 1891087716 INFO2 | free ict ressource > > > > > > Linphone 1.2 config file (.linphonec) > > > - > > > > > > [net] > > > download_bw=0 > > > upload_bw=0 > > > use_stun=0 > > > use_nat=0 > > > con_type=3 > > > > > > [sip] > > > sip_port=5060 > > > guess_hostname=1 > > > contact=sip:[EMAIL PROTECTED] > > > inc_timeout=15 > > > use_info=0 > > > use_ipv6=0 > > > default_proxy=0 > > > > > > [rtp] > > > audio_rtp_port=7078 > > > video_rtp_port=9078 > > > audio_jitt_comp=60 > > > video_jitt_comp=60 > > > > > > [sound] > > > playback_dev_id=1 > > > ringer_dev_id=1 > > > capture_dev_id=1 > > > local_ring=/control4/share/sounds/linphone/rings/oldphone.wav > > > remote_ring=/usr/share/sounds/linphone/ringback.wav > > > echocancelation=0 > > > rec_lev=80 > > > play_lev=80 > > > ring_lev=80 > > > source=m > > > > > > [video] > > > enabled=0 > > > show_local=0 > > > > > > [audio_codec_0] > > > mime=speex > > > rate=16000 > > > enabled=1 > > > > > > [audio_codec_1] > > > mime=speex > > > rate=8000 > > > enabled=1 > > > > > > [audio_codec_2] > > > mime=PCMU > > > rate=8000 > > > enabled=1 > > > > > > [audio_codec_3] > > > mime=GSM > > > rate=8000 > > > enabled=1 > > > > > > [audio_codec_4] > > > mime=PCMA > > > rate=8000 > > > enabled=1 > > > > > > [proxy_0] > > > reg_proxy=sip:192.168.0.21 > > > reg_identity=sip:[EMAIL PROTECTED] > > > reg_expires=600 > > > reg_sendregister=1 > > > publish=0 > > > > > > [audio_codec_5] > > > mime=1015 > > > rate=8000 > > > enabled=1 > > > > > > [auth_info_0] > > > username=mini-touch-000FFF002DF0 > > > passwd=secret > > > realm="asterisk" > > > > > > Linephone 1.5 config file (.linphonerc) > > > --- > > > > > > [net] > > > download_bw=0 > > > upload_bw=0 > > > firewall_policy=0 > > > stun_server=192.168.0.21 > > > > > > [sip] > > > sip_port=5
