Re: [Freeswitch-users] Trunk @13273 - build on VS2008 fails - since switch_xml.h was checked in
Issues of this type must be reported to jira http://jira.freeswitch.org On Mon, May 11, 2009 at 3:54 AM, Richard Lamkin richard.lam...@mettoni.comwrote: Build on fails @ trunk =13273 My last working build @ trunk = 13231 Looks like the addition of the directives _Out_, _In_, _In_opt_, has caused the problem. Unfortunately I'm not familiar that with the code or these directives to suggest a solution, apart from the obvious - roll back. Regards Richard Lamkin richard.lam...@mettonigroup.com Error 787 error C2220: warning treated as error - no 'object' file generated d:\wip\freeswitch\trunk\src\include\switch_xml.h 107 mod_managed Error 816 error C2220: warning treated as error - no 'object' file generated d:\wip\freeswitch\trunk\src\include\switch_xml.h 107 mod_managed Error 846 error C2220: warning treated as error - no 'object' file generated d:\wip\freeswitch\trunk\src\include\switch_xml.h 107 mod_managed Error 1372error C2220: warning treated as error - no 'object' file generated d:\wip\freeswitch\trunk\src\include\switch_xml.h 107 FreeSwitchConsole 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning C6510: Invalid annotation: 'NullTerminated' property may only be used on buffers whose elements are of integral or pointer type 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(212) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(212) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 83d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : error C2220: warning treated as error - no 'object' file generated 83d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 83d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(256) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(256) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 83d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning C6510: Invalid annotation: 'NullTerminated' property may only be used on buffers whose elements are of integral or pointer type 83d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(309) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 83d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 82d:\wip\freeswitch\trunk\src\include\switch_xml.h(309) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 83d:\wip\freeswitch\trunk\src\include\switch_xml.h(212) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type
Re: [Freeswitch-users] FreeSWITCH under the Linux 2.6.29 kernel
Folks; Bear in mind that the frequency is (X)Hz * (num cores), hence saying 100Hz on a dual core winds up being 200Hz. My setup is 250Hz on a Dual-Core and the quality is perfect. Oh, btw folks, don't attempt to do anything involving QOS (be it TBF, CBQ, HTB, or whatnot) on anything less than kernel 2.6.28.4 I don't know why exactly that is, but extensive testing here in the lab showed that this was entirely FUBAR until 2.6.25.7 where it got better, but not perfect until 2.6.28.4 (I may not be exact on the revisions, but close enough...) There's also some options in the kernel that you must disable (not compile in) if you expect packet shaping to work. If there's interest in this, e-mail me directly and I'll see if I can toss it into the FS Wiki some time this upcoming weekend. Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On May 9, 2009, at 2:14 AM, Jason White wrote: Pablo Hernan Saro pablos...@gmail.com wrote: IMHO, it is related with the following kernel options: CONFIG_HZ, CONFIG_NO_HZ and CONFIG_HIGH_RES_TIMERS. Take a look at those options in your kernel and try modifying them until get the desired result. Google that options and you will find lots of discussions that will clarify your mind. Here you will find a simple explanation: http://www.smk.co.za/2007/07/21/a-tickless-kernel/ Thank you for the references. I think I'll modify my kernel parameters in the grub configuration, since I am using Debian kernels at the moment. (I do know how to compile my own, which I will gladly do if it becomes necessary). ___ 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 PGP.sig Description: This is a digitally signed message part ___ 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] Audio clicks between playback of audio files
I'm implementing an IVR solution for FreeSWITCH, but I have a little problem with audio playback. I'm just calling into application park, and then handle the flow using the event socket. All my audio phrases are .PCMA (8KHz a-law), and I play lots of files after eachother. Between each file I can sometimes here a little click, even though I'm 100% sure that this is not from my files. My guess is that it might be caused of the fact that no RTP is sent at all when the phrase is not playing. If I merge the files together before playing them it sounds just fine. Is it possible to make FreeSWITCH send silence frames, even when not needed? I know this is a waste of resources, but it will still make the solution sound much better. Regards Peter Olsson ___ 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] Audi record using uuid_record
Chances are if both legs are NOT already alaw you'll need to record it with .wav or .al files. .PCMU or .PCMA are native file formats if any transcoding is taking place you probably can't get way with .PCMA. /b On May 11, 2009, at 8:39 AM, Peter Olsson wrote: Hello again, I also have a problem when I try to record messages. I record to .PCMA-files, and the file is created perfectly. But it’s just distorted audio in it. It sounds to me that there might be a codec issue. The media stream is PCMA all the way from the phone to FreeSWITCH, and to start recording I simply call “uuid_record UUID start c:\test.PCMA”. According to the docs the file should automatically be recorded as PCMA when the file is named .PCMA. Any ideas what I can be doing wrong? Regards, Peter ___ 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 at ClueCon! 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] Audi record using uuid_record
I record them to file.wav and they play perfectly. I think it's recorded in a raw-format here. See: http://www.nabble.com/Recording-ULAW-files-td21587474.html Best regards Peter Peter Olsson schrieb: Hello again, I also have a problem when I try to record messages. I record to .PCMA-files, and the file is created perfectly. But it’s just distorted audio in it. It sounds to me that there might be a codec issue. The media stream is PCMA all the way from the phone to FreeSWITCH, and to start recording I simply call “uuid_record UUID start c:\test.PCMA”. According to the docs the file should automatically be recorded as PCMA when the file is named .PCMA. Any ideas what I can be doing wrong? Regards, Peter ___ 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] Audio clicks between playback of audio files
Have you tried wav files? /b On May 11, 2009, at 8:34 AM, Peter Olsson wrote: I’m implementing an IVR solution for FreeSWITCH, but I have a little problem with audio playback. I’m just calling into application park, and then handle the flow using the event socket. All my audio phrases are .PCMA (8KHz a-law), and I play lots of files after eachother. Between each file I can sometimes here a little “click”, even though I’m 100% sure that this is not from my files. My guess is that it might be caused of the fact that no RTP is sent at all when the phrase is not playing. If I merge the files together before playing them it sounds just fine. Is it possible to make FreeSWITCH send silence frames, even when not needed? I know this is a waste of resources, but it will still make the solution sound much better. Regards Peter Olsson Brian West br...@freeswitch.org -- Meet us at ClueCon! 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] Audi record using uuid_record
In this case it is pure PCMA, all the way from the phone. I just dial in to number 2100 (using SIP, codec PCMA), and then I have a event socket connected that sees the ivr-test flag. I then play some files (PCMA), and then start a recording. extension name=ivr-test2 condition field=destination_number expression=^2100$ action application=set data=ivr-test=true/ action application=answer/ action application=park/ /condition /extension I understand that it works with wav, however, the application I'm working on already exists, and it makes lots of trouble for me to change the file format during the recording - I have lots of other parts of the software that needs to be changed as well. I've used yate for this application before this - so everything does exist, I'm just porting it to FreeSWITCH. //Peter Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West Skickat: den 11 maj 2009 15:58 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Audi record using uuid_record Chances are if both legs are NOT already alaw you'll need to record it with .wav or .al files. .PCMU or .PCMA are native file formats if any transcoding is taking place you probably can't get way with .PCMA. /b On May 11, 2009, at 8:39 AM, Peter Olsson wrote: Hello again, I also have a problem when I try to record messages. I record to .PCMA-files, and the file is created perfectly. But it's just distorted audio in it. It sounds to me that there might be a codec issue. The media stream is PCMA all the way from the phone to FreeSWITCH, and to start recording I simply call uuid_record UUID start c:\test.PCMA. According to the docs the file should automatically be recorded as PCMA when the file is named .PCMA. Any ideas what I can be doing wrong? Regards, Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.orgmailto: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.orgmailto:br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.comhttp://www.cluecon.com/ !DSPAM:4a0830b432933740610192! ___ 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] Audio clicks between playback of audio files
Brian, Thanks for the response. No, I didn't try wav files - and I'd prefer to keep the current codec if that's possible. But I could give it a try and see what happens. Do you think it might only be related to the native files in FS? //Peter Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West Skickat: den 11 maj 2009 16:09 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Audio clicks between playback of audio files Have you tried wav files? /b On May 11, 2009, at 8:34 AM, Peter Olsson wrote: I'm implementing an IVR solution for FreeSWITCH, but I have a little problem with audio playback. I'm just calling into application park, and then handle the flow using the event socket. All my audio phrases are .PCMA (8KHz a-law), and I play lots of files after eachother. Between each file I can sometimes here a little click, even though I'm 100% sure that this is not from my files. My guess is that it might be caused of the fact that no RTP is sent at all when the phrase is not playing. If I merge the files together before playing them it sounds just fine. Is it possible to make FreeSWITCH send silence frames, even when not needed? I know this is a waste of resources, but it will still make the solution sound much better. Regards Peter Olsson Brian West br...@freeswitch.orgmailto:br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.comhttp://www.cluecon.com/ !DSPAM:4a08332632931845617996! ___ 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] Audio clicks between playback of audio files
I'm not sure. Can you provide me a test file and a known case that you can produce this issue with? /b On May 11, 2009, at 9:20 AM, Peter Olsson wrote: Brian, Thanks for the response. No, I didn’t try wav files – and I’d prefer to keep the current codec if that’s possible. But I could give it a try and see what happens. Do you think it might only be related to the native files in FS? //Peter Brian West br...@freeswitch.org -- Meet us at ClueCon! 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] SRTP Error auth check failed
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, I can reproduce the error with Snom 370 and FW 7.3.11. An update to 7.3.20 (beta) seems to solve the problem - at least I can't reproduce it with the metioned scenario anymore. regards helmut -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKCDAH4tZeNddg3dwRAs29AKCxFN/193R1D+svIjB+Knzzbuhr1wCfQ4rq hMVg7svm2EoaOOexJS5HCMY= =jhJh -END PGP SIGNATURE- ___ 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] Audio clicks between playback of audio files
The problem is that it's not 100% reproducable. Sometimes I can play 2-3-4 files, and it sounds great, sometimes it clicks louder, somtimes not so much. I could get a Wireshark dump for you, could that help? Regards, Peter Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West Skickat: den 11 maj 2009 16:24 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Audio clicks between playback of audio files I'm not sure. Can you provide me a test file and a known case that you can produce this issue with? /b On May 11, 2009, at 9:20 AM, Peter Olsson wrote: Brian, Thanks for the response. No, I didn't try wav files - and I'd prefer to keep the current codec if that's possible. But I could give it a try and see what happens. Do you think it might only be related to the native files in FS? //Peter Brian West br...@freeswitch.orgmailto:br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.comhttp://www.cluecon.com/ !DSPAM:4a08362532934700715407! ___ 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] Audio clicks between playback of audio files
What phone are you using? /b On May 11, 2009, at 9:40 AM, Peter Olsson wrote: The problem is that it’s not 100% reproducable. Sometimes I can play 2-3-4 files, and it sounds great, sometimes it “clicks” louder, somtimes not so much. I could get a Wireshark dump for you, could that help? Regards, Peter Brian West br...@freeswitch.org -- Meet us at ClueCon! 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] Audi record using uuid_record
I've confirmed that recording to wav files works just fine. So it seems to be somthing strange with the native files in FS? /Peter Från: Peter Olsson Skickat: den 11 maj 2009 16:16 Till: 'freeswitch-users@lists.freeswitch.org' Ämne: RE: [Freeswitch-users] Audi record using uuid_record In this case it is pure PCMA, all the way from the phone. I just dial in to number 2100 (using SIP, codec PCMA), and then I have a event socket connected that sees the ivr-test flag. I then play some files (PCMA), and then start a recording. extension name=ivr-test2 condition field=destination_number expression=^2100$ action application=set data=ivr-test=true/ action application=answer/ action application=park/ /condition /extension I understand that it works with wav, however, the application I'm working on already exists, and it makes lots of trouble for me to change the file format during the recording - I have lots of other parts of the software that needs to be changed as well. I've used yate for this application before this - so everything does exist, I'm just porting it to FreeSWITCH. //Peter Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West Skickat: den 11 maj 2009 15:58 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Audi record using uuid_record Chances are if both legs are NOT already alaw you'll need to record it with .wav or .al files. .PCMU or .PCMA are native file formats if any transcoding is taking place you probably can't get way with .PCMA. /b On May 11, 2009, at 8:39 AM, Peter Olsson wrote: Hello again, I also have a problem when I try to record messages. I record to .PCMA-files, and the file is created perfectly. But it's just distorted audio in it. It sounds to me that there might be a codec issue. The media stream is PCMA all the way from the phone to FreeSWITCH, and to start recording I simply call uuid_record UUID start c:\test.PCMA. According to the docs the file should automatically be recorded as PCMA when the file is named .PCMA. Any ideas what I can be doing wrong? Regards, Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.orgmailto: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.orgmailto:br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.comhttp://www.cluecon.com/ !DSPAM:4a0830b432933740610192! ___ 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] Audio clicks between playback of audio files
In this case I'm using Avaya Phones connected to a Avaya CM PBX, which talks SIP to FreeSWITCH. But I'll try to connect a SIP phone directly as well - to see if it makes any difference. I Have a Polycom IP550 I can use for some testing. I've used the exact same setup using yate, and that worked fine, but I think one difference is that it sends rtp even when not playing files, so that's why I thought that could be an issue. /Peter Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West Skickat: den 11 maj 2009 16:43 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Audio clicks between playback of audio files What phone are you using? /b On May 11, 2009, at 9:40 AM, Peter Olsson wrote: The problem is that it's not 100% reproducable. Sometimes I can play 2-3-4 files, and it sounds great, sometimes it clicks louder, somtimes not so much. I could get a Wireshark dump for you, could that help? Regards, Peter Brian West br...@freeswitch.orgmailto:br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.comhttp://www.cluecon.com/ !DSPAM:4a083a8d32932061814324! ___ 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] Cluecon 2009 News
FYI, for those of you keeping up on ClueCon 2009 please visit the latest blog entry: http://cluecon.com/node/29 Thanks, Michael ___ 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] Can't configure outbound call
I am having difficulty making an outbound call. I have read the wiki many
times, but I am missing something.
First, after setting up a gateway xml file, should that gateway show in the
FS console after issuing the 'reloadxml' command? It does not.
Can anyone give me a push? Thanks, Lars
API CALL [sofia(status)] output:
Name Type Data
State
=
external profile sip:mod_so...@64.105.128.82:5080
RUNNING (0)
example.com gatewaysip:joeu...@example.com
NOREG
internal profile sip:mod_so...@192.168.10.29:5090
RUNNING (0)
192.168.10.29 alias internal
ALIASED
internal-ipv6 profile sip:mod_so...@[::1]:5090
RUNNING (0)
default alias internal
ALIASED
nat alias external
ALIASED
outbound alias external
ALIASED
in conf/directory/default/flowroute.com.xml:
include
gateway name=flowroute
param name=username value=flowrouteAccount/
param name=from-domain value=sip.flowroute.com/
param name=password value=password/
param name=extension value=flowrouteAccount/
param name=realm value=sip.flowroute.com/
param name=proxy value=sip.flowroute.com/
param name=expire-seconds value=3600/
param name=register value=true/
param name=retry-seconds value=3600/
param name=caller-id-in-from value=true/
/gateway
/include
in conf/dialplan/default/02_long_distance.xml:
!-- Dial any 10 digit number (222333) or 1+10 number (1222333)
here --
extension name=Long Distance - flowroute
condition field=destination_number expression=^(1{0,1}\d{10})$
action application=set
data=effective_caller_id_number=100/
!-- If your provider does not provide ringback (180 or 183) you may
simulate ringback by uncommenting the following line. --
!-- action application=ringback /--
action application=bridge
data=sofia/gateway/flowroute/flowrouteAccount#$1/
/condition
/extension
Dialplan: sofia/internal/1...@192.168.10.29:5090 Regex (PASS) [Long Distance
- flowroute] destination_number(3235551212) =~ /^(1{0,1}\d{10})$/
break=on-false
Dialplan: sofia/internal/1...@192.168.10.29:5090 Action
set(effective_caller_id_number=100)
Dialplan: sofia/internal/1...@192.168.10.29:5090 Action
set(effective_caller_id_name=${outbound_caller_id_name})
Dialplan: sofia/internal/1...@192.168.10.29:5090 Action
bridge(sofia/gateway/flowroute/flowrouteAccount#3235551212)
2009-05-11 14:05:41 [DEBUG] switch_core_state_machine.c:114
switch_core_standard_on_routing() (sofia/internal/1...@192.168.10.29:5090)
State Change CS_ROUTING - CS_EXECUTE
2009-05-11 14:05:41 [DEBUG] switch_core_session.c:933
switch_core_session_signal_state_change() Send signal
sofia/internal/1...@192.168.10.29:5090 [BREAK]
2009-05-11 14:05:41 [DEBUG] switch_core_state_machine.c:483
switch_core_session_run() (sofia/internal/1...@192.168.10.29:5090) State
ROUTING going to sleep
2009-05-11 14:05:41 [DEBUG] switch_core_state_machine.c:397
switch_core_session_run() (sofia/internal/1...@192.168.10.29:5090) Running
State Change CS_EXECUTE
2009-05-11 14:05:41 [DEBUG] switch_core_state_machine.c:490
switch_core_session_run() (sofia/internal/1...@192.168.10.29:5090) State
EXECUTE
2009-05-11 14:05:41 [DEBUG] mod_sofia.c:173 sofia_on_execute()
sofia/internal/1...@192.168.10.29:5090 SOFIA EXECUTE
2009-05-11 14:05:41 [DEBUG] switch_core_state_machine.c:151
switch_core_standard_on_execute() sofia/internal/1...@192.168.10.29:5090
Standard EXECUTE
EXECUTE sofia/internal/1...@192.168.10.29:5090 set(open=true)
2009-05-11 14:05:41 [DEBUG] mod_dptools.c:748 set_function()
sofia/internal/1...@192.168.10.29:5090 SET [open]=[true]
EXECUTE sofia/internal/1...@192.168.10.29:5090 set(use_profile=nat)
2009-05-11 14:05:41 [DEBUG] mod_dptools.c:748 set_function()
sofia/internal/1...@192.168.10.29:5090 SET [use_profile]=[nat]
EXECUTE sofia/internal/1...@192.168.10.29:5090
set_user(defa...@192.168.10.29)
EXECUTE sofia/internal/1...@192.168.10.29:5090
hash(insert/192.168.10.29-spymap/1000/7c7baffc-3e6f-11de-9b6b-7ba002f89c82)
EXECUTE sofia/internal/1...@192.168.10.29:5090
hash(insert/192.168.10.29-last_dial/1000/3235551212)
EXECUTE sofia/internal/1...@192.168.10.29:5090
hash(insert/192.168.10.29-last_dial/global/7c7baffc-3e6f-11de-9b6b-7ba002f89
c82)
EXECUTE sofia/internal/1...@192.168.10.29:5090
set(effective_caller_id_number=100)
2009-05-11 14:05:41 [DEBUG] mod_dptools.c:748 set_function()
sofia/internal/1...@192.168.10.29:5090 SET
[effective_caller_id_number]=[100]
Re: [Freeswitch-users] Can't configure outbound call
Try this ^1?(\d{10})$
and this sofia/gateway/flowroute/flowrouteAccount#1$1
/b
On May 11, 2009, at 4:38 PM, Lars Zeb wrote:
in conf/dialplan/default/02_long_distance.xml:
!-- Dial any 10 digit number (222333) or 1+10 number
(1222333) here --
extension name=Long Distance - flowroute
condition field=destination_number expression=^(1{0,1}\d{10})
$
action application=set
data=effective_caller_id_number=100/
!-- If your provider does not provide ringback (180 or 183)
you may simulate ringback by uncommenting the following line. --
!-- action application=ringback /--
action application=bridge data=sofia/gateway/flowroute/
flowrouteAccount#$1/
/condition
/extension
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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[Freeswitch-users] How to enable debug in dingaling?
Hi! I need to enable debug mode in dingaling as I can't see that freeswitch is coming online in gtalk. I have changed the following: changed the loglevel to debug in console.conf.xml changed the debug level to 1 in dingaling.conf to 1 I do not see any xmpp logs in the console or in freeswitch.log file. All I can see in the window is: 2009-05-12 17:56:35 [DEBUG] mod_dingaling.c:1854 init_profile() Started Thread for myfreeswitchn...@gmail.com/gt...@xml 2009-05-12 17:56:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_dingaling] 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'dingaling' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_debug' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_pres' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_logout' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_login' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:354 switch_loadable_module_process() Adding Chat interface 'jingle' Where is the XMPP traces? Thanks ___ 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] How to enable debug in dingaling?
Try dl_debug on at the CLI /b On May 11, 2009, at 5:11 PM, Mark Campbell-Smith wrote: Where is the XMPP traces? Thanks Brian West br...@freeswitch.org -- Meet us at ClueCon! 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] Can't configure outbound call
I'm sorry, but do not understand what it is that I should try. Are you
saying to change the data attribute in the action command of the dialplan?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, May 11, 2009 2:58 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't configure outbound call
Try this ^1?(\d{10})$
and this sofia/gateway/flowroute/flowrouteAccount#1$1
/b
On May 11, 2009, at 4:38 PM, Lars Zeb wrote:
in conf/dialplan/default/02_long_distance.xml:
!-- Dial any 10 digit number (222333) or 1+10 number
(1222333) here --
extension name=Long Distance - flowroute
condition field=destination_number expression=^(1{0,1}\d{10})
$
action application=set
data=effective_caller_id_number=100/
!-- If your provider does not provide ringback (180 or 183)
you may simulate ringback by uncommenting the following line. --
!-- action application=ringback /--
action application=bridge data=sofia/gateway/flowroute/
flowrouteAccount#$1/
/condition
/extension
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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Re: [Freeswitch-users] Can't configure outbound call
You want to collect 11 or 10 digits and send it out flowroute which
requires 1+ number. The 1? makes the one optional. So now we only
collect 10 digits and append the on in the bridge line.
extension name=Long Distance - flowroute
condition field=destination_number expression=^1?(\d{10})$
action application=set
data=effective_caller_id_number=100/
!-- If your provider does not provide ringback (180 or 183)
you may simulate ringback by uncommenting the following line. --
!-- action application=ringback /--
action application=bridge data=sofia/gateway/flowroute/
flowrouteAccount#1$1/
/condition
/extension
/b
On May 11, 2009, at 5:23 PM, Lars Zeb wrote:
I'm sorry, but do not understand what it is that I should try. Are you
saying to change the data attribute in the action command of the
dialplan?
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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Re: [Freeswitch-users] get call durantion
Hi, I need get call duration after bridge application using mod_managed, my code: Session.Execute(bridge, sbNewOutBoundNum); Debug(billsec : + _Session.GetVariable(billsec)); Debug(duration : + _Session.GetVariable(duration)); The bridge is ok, but the variable value duration and billsec is zero (0). Diego --- On Sun, 5/10/09, Michael S Collins m...@freeswitch.org wrote: Do you mean from the CDR? I recommend XML CDRs because they give tons of information. If you are talking about gathering this stuff midcall then you'll need to supply more information about your setup. -MC Sent from my iPhone On May 10, 2009, at 6:17 PM, Diego Toro dft...@yahoo.com wrote: Hi, How can I get call durantion after bridge application ? I tried with billsec and duration but I don't get any value. Thank you ___ 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 -Inline Attachment Follows- ___ 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] Can't configure outbound call
On May 11, 2009, at 7:51 PM, Lars Zeb wrote: Thanks for the clarification. It made sense, but the results remain the same. The log still says ‘Invalid Gateway’ and ‘sofia status’ at the console does not show flowroute. It sounds like sofia hasn't picked up your new gateway yet. Have you tried something like the below yet? sofia profile external rescan reloadxml -Dale___ 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] rdnis variable from Lua
Hi, I can see the RDN in the log file, but don't know how to retrieve it from a Lua script. Regards, Cliff ___ 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] rdnis variable from Lua
I found a workaround, but it'd be nice to actually have the RDN easily accessible from Lua: calling_number = session:getVariable ( sip_h_Diversion ) _, _, calling_number = string.find ( calling_number, sip:(%d+)@ ) Cliff On Mon, 2009-05-11 at 17:22 -0700, Cliff Wells wrote: Hi, I can see the RDN in the log file, but don't know how to retrieve it from a Lua script. Regards, Cliff ___ 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] Can't configure outbound call
The one you emailed anthony about the invite... 487 repeat stuff.. can you give me more details on what might be going on? /b On May 11, 2009, at 5:23 PM, Lars Zeb wrote: I'm sorry, but do not understand what it is that I should try. Are you saying to change the data attribute in the action command of the dialplan? Brian West br...@freeswitch.org -- Meet us at ClueCon! 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] rdnis variable from Lua
Hehe its not really a work around... Its how you do it either way... but I did add the patch from http://jira.freeswitch.org/browse/MODSOFIA-7 which would require you to do similar. /b On May 11, 2009, at 8:20 PM, Cliff Wells wrote: I found a workaround, but it'd be nice to actually have the RDN easily accessible from Lua: calling_number = session:getVariable ( sip_h_Diversion ) _, _, calling_number = string.find ( calling_number, sip:(%d+)@ ) Cliff Brian West br...@freeswitch.org -- Meet us at ClueCon! 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] SDP Passthrough, INVITE messages.
Hi, I'm trying to use the Freeswitch as a proxy (I know that is not designed for that, but I really need to do it in this way), here is my config: Endpoint 1- FS A--FS B-FS A-Endpoint 2 * Both Endpoints are registered in FS A how is acting as a proxy and registrar. * FS B only sends back the Invite to FS A in order to reach Endpoint 2. * Both FS have a public IP * FS A Only handles SIP messages * FS B Handles RTP (Also SIP) My objetive is to keep the signaling in FS A and the RTP in FS B so basically FS A will work as a registrar. So far I've been able to succesfully do it if both endpoint are not nated, how ever I do need to do it in a Natted sceneario too, for what I have been sniffing the problem is that in the INVITE, the SDP is sending the internal IP instead of the external. I've tried to change the switch_r_sdp and switch_l_sdp but I'm not quite sure if I'm doing the correct config of the switch (late_codec_negotiation) If anyone could give a tip or a sample of how can I change the INVITE messages I will appreciate. Thanks in advance Regards ___ 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] SDP Passthrough, INVITE messages.
Juan, Can you explain your situation a little better you seem to have breezed over the critical details. Also you should enable STUN on your endpoints and not depend on your Registrar to overcome nat issues since its not its job. /b On May 11, 2009, at 10:03 PM, Juan Manuel Vicente wrote: So far I've been able to succesfully do it if both endpoint are not nated, how ever I do need to do it in a Natted sceneario too, for what I have been sniffing the problem is that in the INVITE, the SDP is sending the internal IP instead of the external. Brian West br...@freeswitch.org -- Meet us at ClueCon! 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] Can't configure outbound call
I believe you're talking about the FS sending out an ICMP in response to the client's invite, which resulted in 'ICMP Destination unreachable'. He told me to turn iptables off, which I did. Then the Eyebeam registered successfully. I don't know what the '487 repeat stuff' was. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, May 11, 2009 7:59 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't configure outbound call The one you emailed anthony about the invite... 487 repeat stuff.. can you give me more details on what might be going on? /b On May 11, 2009, at 5:23 PM, Lars Zeb wrote: I'm sorry, but do not understand what it is that I should try. Are you saying to change the data attribute in the action command of the dialplan? Brian West br...@freeswitch.org -- Meet us at 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] Can't configure outbound call
Haha that's ok I sent that email to you by mistake I wondered where that email went! /b Sent from my iPhone On May 11, 2009, at 11:29 PM, Lars Zeb larc...@yahoo.com wrote: I believe you’re talking about the FS sending out an ICMP in respons e to the client’s invite, which resulted in ‘ICMP Destination unreachable’. He told me to turn iptables off, which I did. Then th e Eyebeam registered successfully. I don’t know what the ‘487 repeat stuff’ was. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, May 11, 2009 7:59 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't configure outbound call The one you emailed anthony about the invite... 487 repeat stuff.. can you give me more details on what might be going on? /b On May 11, 2009, at 5:23 PM, Lars Zeb wrote: I'm sorry, but do not understand what it is that I should try. Are you saying to change the data attribute in the action command of the dialplan? Brian West br...@freeswitch.org -- Meet us at ClueCon! 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 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
