[Freeswitch-users] Error causing freeswitch to crash
Hi, Every few days I'm getting this error which is causing Freeswitch to crash. Can anyone tell me what may be causing this or how to prevent it? 2009-05-25 10:37:38 [CRIT] switch_core_state_machine.c:263 handle_fatality() Caught signal 11 for unmapped thread! Many thanks Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] crash-protection and monit
Hi, Is there any reason why the crash-protection parameter in switch.conf.xml defaults to false and are there any downsides to setting it to true? The documentation says it helps with certain types of crashes, can anyone tell me what sort of crashes in particular it helps to prevent as my freeswitch install seems to crash every few days. Also, does anyone have an example of the monit setup for freeswitch to restart it when it fails? Many thanks Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Using recordFile with Icecast - looses the end of the call
Hi, I have mod_shout installed and I'm using session.recordFile to capture the audio in a call. When I specify a local file mp3 or wav the audio is captured fine. However, I'm using an icecast server to manage the audio for me and when I specify a remote mp3 location(shout://myserver.com/myaudio.mp3) the end of the call is missing off the resultant mp3 file. A wild shot in the dark I know but does anyone have any experience of this and how it might be resolved? Many thanks Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Calls being cut off while recording a message
Hi Brian,
Is NAT a known problem? Is there a work around? The messages on the lists
seem to imply other folks have this working ok behind NAT firewalls. What's
your recommendation for how I should proceed?
regards
Andy
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: 06 April 2009 17:39
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Calls being cut off while recording a
message
Three letters come to mind... N A T! ;) What is your network topo?
/b
On Apr 6, 2009, at 11:31 AM, Andy Ayers wrote:
Hi Brian,
Just doing some more testing, simplified the call by not even trying to
record the incoming audio and placing a while (session.ready()) {} loop in
the ivr code instead and the calls all now terminate with
RECOVERY_ON_TIMER_EXPIRE.
Does this shed any light on the subject at all?
regards
Andy
Brian West
br...@freeswitch.org
-- Meet us a ClueCon! http://www.cluecon.com <http://www.cluecon.com/>
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Re: [Freeswitch-users] Calls being cut off while recording a message
Hi Brian,
The freeswitch server is connect to the internet via a Cisico ASA firewall
currently running in NAT mode. I believe it's that simple but can't be sure
of the equipment between my firewall and the internet.
regards
Andy
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: 06 April 2009 17:39
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Calls being cut off while recording a
message
Three letters come to mind... N A T! ;) What is your network topo?
/b
On Apr 6, 2009, at 11:31 AM, Andy Ayers wrote:
Hi Brian,
Just doing some more testing, simplified the call by not even trying to
record the incoming audio and placing a while (session.ready()) {} loop in
the ivr code instead and the calls all now terminate with
RECOVERY_ON_TIMER_EXPIRE.
Does this shed any light on the subject at all?
regards
Andy
Brian West
br...@freeswitch.org
-- Meet us a ClueCon! http://www.cluecon.com <http://www.cluecon.com/>
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Re: [Freeswitch-users] Calls being cut off while recording a message
Hi Brian,
Just doing some more testing, simplified the call by not even trying to
record the incoming audio and placing a while (session.ready()) {} loop in
the ivr code instead and the calls all now terminate with
RECOVERY_ON_TIMER_EXPIRE.
Does this shed any light on the subject at all?
regards
Andy
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: 06 April 2009 14:24
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Calls being cut off while recording a
message
Please update... rebootstrap.. you caught SVN with the libtool patch which
kinda broken a few things linking.
/b
On Apr 6, 2009, at 4:07 AM, Andy Ayers wrote:
Hi Brian,
I've upgraded to svn trunk but am now getting errors on load which are
preventing it from working:
2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_shout.so
**/usr/local/freeswitch/mod/mod_shout.so: undefined symbol:
ogg_stream_pagein**
2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_spidermonkey.so
**/usr/local/freeswitch/lib/libjs.so.1: undefined symbol:
PR_LocalTimeParameters**
Sorry if this is obvious but what have I done wrong?
Thanks for your help
Andy
Brian West
br...@freeswitch.org
-- Meet us a ClueCon! http://www.cluecon.com <http://www.cluecon.com/>
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Re: [Freeswitch-users] Calls being cut off while recording a message
Hi Brian, Ok, all up to date, the errors have gone and the software is basically working but the cut off problem still exists. I have an identical software install running on a machine that is not behind a firewall and the cut off doesn't seem to occur. This would seem to suggest it's firewall related. Any clues? regards Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 06 April 2009 14:24 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Please update... rebootstrap.. you caught SVN with the libtool patch which kinda broken a few things linking. /b On Apr 6, 2009, at 4:07 AM, Andy Ayers wrote: Hi Brian, I've upgraded to svn trunk but am now getting errors on load which are preventing it from working: 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_stream_pagein** 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/lib/libjs.so.1: undefined symbol: PR_LocalTimeParameters** Sorry if this is obvious but what have I done wrong? Thanks for your help Andy Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com <http://www.cluecon.com/> ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Calls being cut off while recording a message
Hi Brian, I've upgraded to svn trunk but am now getting errors on load which are preventing it from working: 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_stream_pagein** 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/lib/libjs.so.1: undefined symbol: PR_LocalTimeParameters** Sorry if this is obvious but what have I done wrong? Thanks for your help Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 31 March 2009 14:40 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Please try SVN trunk. /b On Mar 31, 2009, at 8:36 AM, Andy Ayers wrote: Hi Brian, 1.03 Thanks Andy Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com <http://www.cluecon.com/> ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Calls being cut off while recording a message
Hi Brian, I've upgraded to svn trunk but am now getting errors on load which are preventing it from working: 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_stream_pagein** 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/lib/libjs.so.1: undefined symbol: PR_LocalTimeParameters** Sorry if this is obvious but what have I done wrong? Thanks for your help Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 31 March 2009 14:40 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Please try SVN trunk. /b On Mar 31, 2009, at 8:36 AM, Andy Ayers wrote: Hi Brian, 1.03 Thanks Andy Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com <http://www.cluecon.com/> ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Calls being cut off while recording a message
Hi Brian, 1.03 Thanks Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 31 March 2009 14:27 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message I'm going to guess you're not on SVN trunk? what rev are you on? /b On Mar 31, 2009, at 8:04 AM, Andy Ayers wrote: Hi, I'm using freeswitch as a glorified answering machine. FS registers with a VOIP gateway and all calls into the gateway go through an ivr menu and are allowed to leave a message which gets recorded to a file. The FS box is behind a NAT firewall. Everything works fine except that intermittently, calls keep getting cut off after a number of seconds. I've attached a snapshot of the log at the point that the call gets cut off, can anyone suggest why this is happening or how I can prevent it? Many thanks Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com <http://www.cluecon.com/> ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Calls being cut off while recording a message
Hi, I'm using freeswitch as a glorified answering machine. FS registers with a VOIP gateway and all calls into the gateway go through an ivr menu and are allowed to leave a message which gets recorded to a file. The FS box is behind a NAT firewall. Everything works fine except that intermittently, calls keep getting cut off after a number of seconds. I've attached a snapshot of the log at the point that the call gets cut off, can anyone suggest why this is happening or how I can prevent it? Many thanks Andy 2009-03-30 11:14:43 [DEBUG] switch_ivr_play_say.c:272 switch_ivr_phrase_macro() Handle play-file:[7-mono-8kHz.wav] (en:en) 2009-03-30 11:14:43 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated l...@8000hz 1 channels 20ms 2009-03-30 11:14:43 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() sofia/external/x...@xxx.xxx.xxx.xxx receive message [TRANSCODING_NECESSARY] 2009-03-30 11:14:49 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-03-30 11:14:49 [INFO] ivrmenu.js:36 console_log() 2009-03-30 11:14:49 [INFO] ivrmenu.js:36 console_log() 2009-03-30 11:14:49 [INFO] ivrmenu.js:36 console_log() 2009-03-30 11:14:49 [INFO] ivrmenu.js:36 console_log() 2009-03-30 11:14:49 [INFO] ivrmenu.js:36 console_log() 2009-03-30 11:14:49 [INFO] ivrmenu.js:36 console_log() 2009-03-30 11:14:49 [WARNING] mod_shout.c:1088 shout_file_set_string() Value Ignored 2009-03-30 11:14:49 [DEBUG] switch_ivr_play_say.c:505 switch_ivr_record_file() Raw Codec Activated 2009-03-30 11:14:49 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/external/x...@xxx.xxx.xxx.xxx receive message [TRANSCODING_NECESSARY] 2009-03-30 11:14:53 [INFO] mod_shout.c:280 log_msg() LAME 3.97 32bits (http://www.mp3dev.org/) 2009-03-30 11:14:53 [INFO] mod_shout.c:280 log_msg() polyphase lowpass filter disabled 2009-03-30 11:15:24 [DEBUG] switch_core_io.c:403 switch_core_session_read_frame() Engaging Read Buffer at 320 bytes vs 40 2009-03-30 11:15:24 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP RECV DTMF #:2000 2009-03-30 11:15:43 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/x...@xxx.xxx.xxx.xxx entering state [received] 2009-03-30 11:15:43 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP: v=0 o=root 3242 3244 IN IP4 194.145.190.143 s=session c=IN IP4 194.145.190.143 t=0 0 m=audio 10314 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 2009-03-30 11:15:43 [DEBUG] sofia_glue.c:2549 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] 2009-03-30 11:15:43 [DEBUG] sofia_glue.c:2549 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] 2009-03-30 11:15:43 [DEBUG] sofia_glue.c:1635 sofia_glue_tech_set_codec() Already using PCMA 2009-03-30 11:15:43 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-03-30 11:15:43 [DEBUG] sofia_glue.c:1901 sofia_glue_activate_rtp() Audio params changed for sofia/external/x...@xxx.xxx.xxx.xxx from 213.166.5.140:17856 to 194.145.190.143:10314 2009-03-30 11:15:43 [DEBUG] sofia_glue.c:1908 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/x...@xxx.xxx.xxx.xxx] 192.168.4.2 port 27496 -> 194.145.190.143 port 10314 codec: 8 ms: 20 2009-03-30 11:15:43 [DEBUG] sofia_glue.c:1927 sofia_glue_activate_rtp() AUDIO RTP CHANGING DEST TO: [194.145.190.143:10314] 2009-03-30 11:15:43 [DEBUG] sofia.c:3084 sofia_handle_sip_i_state() Processing Reinvite 2009-03-30 11:15:43 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/x...@xxx.xxx.xxx.xxx entering state [completed] 2009-03-30 11:15:43 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/x...@xxx.xxx.xxx.xxx entering state [terminated] 2009-03-30 11:15:43 [NOTICE] sofia.c:3279 sofia_handle_sip_i_state() Hangup sofia/external/x...@xxx.xxx.xxx.xxx [CS_EXECUTE] [NORMAL_CLEARING] 2009-03-30 11:15:43 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/external/x...@xxx.xxx.xxx.xxx [KILL] 2009-03-30 11:15:43 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/external/x...@xxx.xxx.xxx.xxx [BREAK] 2009-03-30 11:15:43 [DEBUG] switch_core_codec.c:122 switch_core_session_set_read_codec() Restore original codec. 2009-03-30 11:15:43 [INFO] ivrmenu.js:36 console_log() ##IVR Message[07703345353]: Dropped out of record! 2009-03-30 11:15:43 [INFO] ivrmenu.js:36 console_log() ##IVR Message[07703345353]: Session no longer active! 2009-03-30 11:15:43 [INFO] ivrmenu.js:36 console_log() 2009-03-30 11:15:44 [DEBUG] mod_shout.c:620 write_stream_thread() Thread Done 2009-03-30 11:16:43 [ERR] mod_spidermonkey.c:2406 fetch_url_callback() Data do not fit in the allocated buffer 2009-03-30 11:16:43 [ERR] ivrmenu.js:96 mod_spidermonkey() TypeError: st has no properties 2009-03-30 11:16:43 [DEBUG] switch_
Re: [Freeswitch-users] Losing Gateway registration
Thanks for your help folks, the ping parameter seems to have resolved the gateway connection issue but I now seem to be having a related issue with calls being cut off after a number of seconds. The freeswitch logs show a normal call clearing. I am indeed behind a NAT firewall which I'm assuming is the main issue. do you have any further tips to make this more stable and prevent the call cut off? Many thanks Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: 18 March 2009 14:46 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Losing Gateway registration if you are behind NAT it is possible that your router "forgot" the mapping betweeen FS and your provider, try adding to your gateway. Math On 18-Mar-09, at 10:07 AM, Brian West wrote: Upgrade to 1.03 or SVN Trunk /b On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: Hi, I've recently ugrade to version 1.02 of freeswitch and am having some problems with my gateway registrations. The gateway successfully registers with my voip provider when freeswitch first starts but if left running it seems to loose it's connection to my voip provider. I can get it to reconnect with a sofia restart. I'm using the same provider and user account as with the old version of the software. Can you suggest any reaosn why this may be happening and how I can prevent it? Many thanks Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Losing Gateway registration
Thanks Brian, I've upgraded to 1.0.3 and things seem a little better but I'm still loosing the gateway connection intermittently. I rebuilt the config on upgrade, is there any possibility I've missed something? Is there a keep-alive setting for a gateway or a re-connect after x or something. Many thanks for your help. Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 18 March 2009 14:08 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Losing Gateway registration Upgrade to 1.03 or SVN Trunk /b On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: Hi, I've recently ugrade to version 1.02 of freeswitch and am having some problems with my gateway registrations. The gateway successfully registers with my voip provider when freeswitch first starts but if left running it seems to loose it's connection to my voip provider. I can get it to reconnect with a sofia restart. I'm using the same provider and user account as with the old version of the software. Can you suggest any reaosn why this may be happening and how I can prevent it? Many thanks Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Losing Gateway registration
Hi, I've recently ugrade to version 1.02 of freeswitch and am having some problems with my gateway registrations. The gateway successfully registers with my voip provider when freeswitch first starts but if left running it seems to loose it's connection to my voip provider. I can get it to reconnect with a sofia restart. I'm using the same provider and user account as with the old version of the software. Can you suggest any reaosn why this may be happening and how I can prevent it? Many thanks Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] recordFile bitrate
Thanks Brian, am I correct in saying therefore that all mp3 streams generated by the recordFile command(with mod_shout installed) will be 64Kbps? -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 08 January 2009 14:51 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] recordFile bitrate bitrate nor sample rate are configurable. The format depends on the extension of the filename. The sample rate is recorded at the channels native rate. /b On Jan 8, 2009, at 8:37 AM, Andy Ayers wrote: Hi, Is the bitrate, sample rate or format of the audio stream created by session.recordFile configurable at all? Apologies if I've missed something in the docs. cheers Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] recordFile bitrate
Hi, Is the bitrate, sample rate or format of the audio stream created by session.recordFile configurable at all? Apologies if I've missed something in the docs. cheers Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF and firewall
Hi, I'm using freeswitch to receive incoming calls from a sip provider namely AQL. When my freeswitch box is connected directly to the internet everything works fine. When I place a firewall/router inbetween the box and the internet, the software registers with the sip provider ok and answers calls but fails to respond to in call dtmf tones. AQL advised me to make sure I was using RFC2833 which I believe I have done by setting dtmf-type in my sip profile xml to 'RFC2833'. Can anyone advise me as to what other settings I should change to make the dtmf work correctly across the firewall/router? The router is currently set to allow all traffic. Many thanks for any help you can give. regards Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Recomended VOIP Providers?
Many thanks for the advice, Sorry should have said I'm based in the UK. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter P GMX Sent: 12 November 2008 16:13 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Recomended VOIP Providers? Please have a look at the wiki http://wiki.freeswitch.org/wiki/SIP_Provider_Examples In which country do you need a provider? So far I got every provider I tried (5) to work with freeswitch; even with FS behind NAT. Andy Ayers schrieb: > Hi, > > Can anyone recommend any good VOIP providers that integrate well with > Freeswitch. In particular I need one that can cope with Freeswitch > being behind a firewall/router. I've tried Voiptalk.org and > voipon.co.uk but neither seem to register correctly via the router. > > Any help much appreciated. > > regards > Andy > > -- > -- > > ___ > Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Recomended VOIP Providers?
Hi, Can anyone recommend any good VOIP providers that integrate well with Freeswitch. In particular I need one that can cope with Freeswitch being behind a firewall/router. I've tried Voiptalk.org and voipon.co.uk but neither seem to register correctly via the router. Any help much appreciated. regards Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] VoipTalk NAT
Hi, This is my first post so my apologies if I get the protocol wrong or if I'm posting to the wrong place. Has anyone any experience of setting up Freeswitch to accept incoming calls via VoipTalk (http://www.voiptalk.org) through a firewall/router. I have tried every possible combination of domain/realm and proxy settings but can't get it working. Everything works fine when the freeswitch machine is connected directly to the modem and a software voip-phone runs ok on the same machine through the router so the firewall is letting the traffic through. Any help greatly appreciated. Many thanks Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
