Time out issue and static noise with originate command
Hi all... I have a user account on FS, sometimes when I register with Zoiper or X-Lite it works, but after some time it unregisters itself, I'm not sure why... this didn't happened on Asterisk for me. Does FS handles NAT automatically? This is how I have it setup: softphone => NAT (router) => FS In Asterisk I used to have nat=yes in sip.conf, is there a equivalent for this in FS? Why could my softphone be deregistering itself? This is my user account on FS: Maitei:/usr/local/freeswitch# cat conf/directory/default/diego.xml <include> <user id="diego" mailbox="1000"> <params> <param name="password" value="1234"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> </variables> </user> </include> Another issue... when I use the originate command, with a WAV or GSM file... all I get is static noise, is there a way that this could work?, this also works on Asterisk just fine for me: originate sofia/internal/diego &playback('/root/nin.mp3.wav') But I get static noise on FS. Thanks, Diego
So you have FreeSWITCH behind nat? on the "internal" profile? If thats the case you'll need to set the sip-ext-ip and ext-rtp-ip so that it can communicate with the outside world properly. /b On Jul 5, 2008, at 10:55 PM, Diego Viola wrote:
softphone => NAT (router) => FS
FreeSWITCH is not behind NAT, only my softphone... looks like it was only my ISP doing nasty things again. Thanks for your answer though, it works again... I tried this and happened with Asterisk too, so I think it was a network issue, but it works now... thanks. What about the originate thing, do you know why is not playing the sound file with playback? Diego On Sun, Jul 6, 2008 at 11:41 AM, Brian West <brian@freeswitch.org> wrote:
So you have FreeSWITCH behind nat? on the "internal" profile? If thats the case you'll need to set the sip-ext-ip and ext-rtp-ip so that it can communicate with the outside world properly.
/b
On Jul 5, 2008, at 10:55 PM, Diego Viola wrote:
softphone => NAT (router) => FS
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Sure, here I send you the file... all I hear is static noise when I do: originate sofia/internal/diego &playback('/home/diego/welcometorac.gsm') My softphone rings but I hear the static noise... from the XML IVR (ivr.conf.xml) with playback works well... Thanks, Diego On Sun, Jul 6, 2008 at 1:04 PM, Brian West <brian@freeswitch.org> wrote:
What kind of file is it and can you send me the file?
/b
On Jul 6, 2008, at 11:41 AM, Diego Viola wrote:
What about the originate thing, do you know why is not playing the sound file with playback?
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Uhm, I just originated a call to my landline and it works great... but when I originate to my cellphone it doesn't work. I get this message: freeswitch@Maitei> 2008-07-06 20:26:57 [NOTICE] sofia.c:2085 sofia_handle_sip_i_state() Hangup sofia/internal/011595982343565 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2008-07-06 20:26:57 [WARNING] mod_sndfile.c:212 sndfile_file_seek() File is not seekable 2008-07-06 20:26:57 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 5 (sofia/internal/011595982343565) Ended 2008-07-06 20:26:57 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/internal/011595982343565 [CS_HANGUP] I'm not sure if is a codec thing, this is what I have in vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,PCMU,PCMA,GSM"/> This is how I'm originating the call: freeswitch@Maitei> originate sofia/gateway/teliax/011595982343565 &playback('/home/diego/foo.gsm') The cellphone rings but I don't hear anything, I tried the same thing with Asterisk and worked fine... any ideas? Thanks, Diego On Sun, Jul 6, 2008 at 1:17 PM, Diego Viola <diego.viola@gmail.com> wrote:
Sure, here I send you the file... all I hear is static noise when I do:
originate sofia/internal/diego &playback('/home/diego/welcometorac.gsm')
My softphone rings but I hear the static noise... from the XML IVR (ivr.conf.xml) with playback works well...
Thanks,
Diego
On Sun, Jul 6, 2008 at 1:04 PM, Brian West <brian@freeswitch.org> wrote:
What kind of file is it and can you send me the file?
/b
On Jul 6, 2008, at 11:41 AM, Diego Viola wrote:
What about the originate thing, do you know why is not playing the sound file with playback?
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the one with gsm sounding like static can be addressed by reversing the order you load mod_voipcodecs and mod_sndfile On Sun, Jul 6, 2008 at 3:29 PM, Diego Viola <diego.viola@gmail.com> wrote:
Uhm, I just originated a call to my landline and it works great... but when I originate to my cellphone it doesn't work. I get this message:
freeswitch@Maitei> 2008-07-06 20:26:57 [NOTICE] sofia.c:2085 sofia_handle_sip_i_state() Hangup sofia/internal/011595982343565 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2008-07-06 20:26:57 [WARNING] mod_sndfile.c:212 sndfile_file_seek() File is not seekable 2008-07-06 20:26:57 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 5 (sofia/internal/011595982343565) Ended 2008-07-06 20:26:57 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/internal/011595982343565 [CS_HANGUP]
I'm not sure if is a codec thing, this is what I have in vars.xml:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,PCMU,PCMA,GSM"/>
This is how I'm originating the call:
freeswitch@Maitei> originate sofia/gateway/teliax/011595982343565 &playback('/home/diego/foo.gsm')
The cellphone rings but I don't hear anything, I tried the same thing with Asterisk and worked fine... any ideas?
Thanks,
Diego
On Sun, Jul 6, 2008 at 1:17 PM, Diego Viola <diego.viola@gmail.com> wrote:
Sure, here I send you the file... all I hear is static noise when I do:
originate sofia/internal/diego &playback('/home/diego/welcometorac.gsm')
My softphone rings but I hear the static noise... from the XML IVR (ivr.conf.xml) with playback works well...
Thanks,
Diego
On Sun, Jul 6, 2008 at 1:04 PM, Brian West <brian@freeswitch.org> wrote:
What kind of file is it and can you send me the file?
/b
On Jul 6, 2008, at 11:41 AM, Diego Viola wrote:
What about the originate thing, do you know why is not playing the sound file with playback?
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (3)
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Anthony Minessale
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Brian West
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Diego Viola