Re: [Freeswitch-users] schedule a DTMF tone into bridge
On Fri, Dec 12, 2008 at 8:21 PM, Michael Collins wrote: > Frank, > > I'm sure this is possible. Please give me a little bit to look into > this. I'm going to see if I can lab it up and give you a sample > dialplan. Also, thanks for the heads up on the wiki not having this > information. I will put that on my not-so-short wiki todo list. > > Thanks, > MC > > On Fri, Dec 12, 2008 at 7:37 PM, Frank @ Impact wrote: >> Not much written in the wiki on this. Also searched the list and not much >> on either sched_api or uuid_send_dtmf. >> >> So from an xml dialplan, can sched_api as an application? >> >> Is there any way to have the time offset reference the point at which the >> call started ? ie. When the called party answers? >> >> >> >> Ultimately, I was trying to insert some xml into my dial plan that would >> play a dtmf tone 10 seconds after the called party picked up the phone. But >> from the little that has been written so far that I can find, it is not >> clear to me how to piece this together. Am I being dense and missing >> anything that has already been written? >> >> >> >> /f >> >> -Original Message- >> From: freeswitch-users-boun...@lists.freeswitch.org >> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian >> West >> >> sched_api (hint uuid_send_dtmf) >> >> >> >> API CALL [sched_api()] output: >> >> -ERR Invalid syntax. USAGE: [...@] >> >> >> >> >> >> /b >> >> >> >> On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote: >> >> Is there a way to schedule a certain DTMF tone to be played into a bridge >> (both a and b legs) after a scheduled number of seconds into the call? >> >> ___ >> >> >> >> >> ___ >> 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] schedule a DTMF tone into bridge
Frank, I found a simple way to handle this scenario. I decided just to create a small Lua script that would do the job. It's committed in latest trunk. Look in src/scripts/contrib/mcollins for uuid_send_dtmf.lua. It has comments on how to call it, including a sample dp call. The way I would use this in your scenario is to setup a destination using the execute_on_answer variable. http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer Have the destination be an extension that does something like this: ...rest of diaplan... The 10 means ten seconds, the 123 means send the dtmf digits 1,2,3 in order. You can tinker with the settings as you see fit. Please let me know how it goes. BTW, be sure to put the Lua script in /usr/local/freeswitch/scripts or specify the complete path name when calling the lua app in the dialplan. -MC On Fri, Dec 12, 2008 at 7:37 PM, Frank @ Impact wrote: > Not much written in the wiki on this. Also searched the list and not much > on either sched_api or uuid_send_dtmf. > > So from an xml dialplan, can sched_api as an application? > > Is there any way to have the time offset reference the point at which the > call started ? ie. When the called party answers? > > > > Ultimately, I was trying to insert some xml into my dial plan that would > play a dtmf tone 10 seconds after the called party picked up the phone. But > from the little that has been written so far that I can find, it is not > clear to me how to piece this together. Am I being dense and missing > anything that has already been written? > > > > /f > > -Original Message- > From: freeswitch-users-boun...@lists.freeswitch.org > [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian > West > > sched_api (hint uuid_send_dtmf) > > > > API CALL [sched_api()] output: > > -ERR Invalid syntax. USAGE: [...@] > > > > > > /b > > > > On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote: > > Is there a way to schedule a certain DTMF tone to be played into a bridge > (both a and b legs) after a scheduled number of seconds into the call? > > ___ > > > > > ___ > 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] schedule a DTMF tone into bridge
Frank, I'm sure this is possible. Please give me a little bit to look into this. I'm going to see if I can lab it up and give you a sample dialplan. Also, thanks for the heads up on the wiki not having this information. I will put that on my not-so-short wiki todo list. Thanks, MC On Fri, Dec 12, 2008 at 7:37 PM, Frank @ Impact wrote: > Not much written in the wiki on this. Also searched the list and not much > on either sched_api or uuid_send_dtmf. > > So from an xml dialplan, can sched_api as an application? > > Is there any way to have the time offset reference the point at which the > call started ? ie. When the called party answers? > > > > Ultimately, I was trying to insert some xml into my dial plan that would > play a dtmf tone 10 seconds after the called party picked up the phone. But > from the little that has been written so far that I can find, it is not > clear to me how to piece this together. Am I being dense and missing > anything that has already been written? > > > > /f > > -Original Message- > From: freeswitch-users-boun...@lists.freeswitch.org > [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian > West > > sched_api (hint uuid_send_dtmf) > > > > API CALL [sched_api()] output: > > -ERR Invalid syntax. USAGE: [...@] > > > > > > /b > > > > On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote: > > Is there a way to schedule a certain DTMF tone to be played into a bridge > (both a and b legs) after a scheduled number of seconds into the call? > > ___ > > > > > ___ > 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] schedule a DTMF tone into bridge
Not much written in the wiki on this. Also searched the list and not much on either sched_api or uuid_send_dtmf. So from an xml dialplan, can sched_api as an application? Is there any way to have the time offset reference the point at which the call started ? ie. When the called party answers? Ultimately, I was trying to insert some xml into my dial plan that would play a dtmf tone 10 seconds after the called party picked up the phone. But from the little that has been written so far that I can find, it is not clear to me how to piece this together. Am I being dense and missing anything that has already been written? /f -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West sched_api (hint uuid_send_dtmf) API CALL [sched_api()] output: -ERR Invalid syntax. USAGE: [...@] /b On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote: Is there a way to schedule a certain DTMF tone to be played into a bridge (both a and b legs) after a scheduled number of seconds into the call? ___ ___ 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] Error loading portaudio module
Jason, Thanks for troubleshooting this! At the very least I will add a note to the PA section of the wiki. -MC On Fri, Dec 12, 2008 at 6:01 PM, Jason White wrote: > The problem is now solved. > > It turned out to be permissions: the freeswitch user wasn't added to the audio > group in /etc/group, hence didn't have permission to interrogate the audio > devices. > > Perhaps a future version of the Debian package could address this, or at least > it should be noted somewhere. > > > ___ > 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] Error loading portaudio module
The problem is now solved. It turned out to be permissions: the freeswitch user wasn't added to the audio group in /etc/group, hence didn't have permission to interrogate the audio devices. Perhaps a future version of the Debian package could address this, or at least it should be noted somewhere. ___ 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] missing 3 seconds of audio on bridge calls
Thanks again Anthony ! You fixed the issue with DTMF i had reported : http://jira.freeswitch.org/browse/FSCORE-251 Chris Danielson added to Wiki a nice page collecting these issues with Sonus : http://wiki.freeswitch.org/wiki/RTP_Issues Cheers, El mié, 10-12-2008 a las 03:10 +0100, Angel Carpintero escribió: > Thanks Anthony , you did a great work ! this is fixed in svn r10691. > > Some notes for people using Sonus and L3 as was my case : > > in var.xml in some scenario you may need : > > > > in sip_profiles/internal.xml : > > > > might help for some people with rtp issues : > > > > If you have issues with DTMF and timestamps add also : > > > > I've a little issues with DTMF from VOIP , i i'll figure out can could > be the issue , from PSTN all works like a charm :) > > Cheers, > > El jue, 04-12-2008 a las 09:34 -0600, Anthony Minessale escribió: > > most likely it's because during the time you are dong artificial > > ringback the other side is not doing RTP right. > > > > When the call is answered we flush the rtp buffer and your missing > > audio is probably flushed with it. > > so you can choose to have a 3 second delay or erase the 3 seconds as > > it does now. > > > > This is a known problem with sonus which has been proven to build up > > an audio delay during the time > > you are waiting for the call to answer. I'm sure you prefer the way > > it is to a large audio delay. > > > > > > > > On Wed, Dec 3, 2008 at 10:46 PM, Angel Carpintero > > wrote: > > No TDM , all is SIP : > > > > > > PSTN ---> Sip Proxy_A --> FS ( brigde ) > > ringback/transfer_ringback > > -> Sip Proxy_B --> PSTN > > > > > > In logfile i think you can get some details about Media > > Gateways > > ( Sonus ) PSTN inbound / outbound is provided by Level3. > > > > I can get a capture of a call if you want, in capture the > > audio is not > > missing, issue with : > > > > - rtp buffer ? > > - Sonus ? > > > > Let me know anything you need so i can provide a log or create > > a new > > scenario. > > > > > > Thanks, > > > > El mié, 03-12-2008 a las 22:12 -0600, Anthony Minessale > > escribió: > > > > > what does PSTN represent? > > > > > > I know what the PSTN is but how are you reaching it? > > > is it TDM, SIP etc... what gateway type other details. > > > > > > > > > On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero > > > > > wrote: > > > Hi guys, > > > > > > I've a strange issue with FS , version svn > > -r10584 , > > > when FS bridges a call first 3 seconds of audio are > > missing , > > > looks that > > > only happens on PSTN calls and using ringback or > > > transfer_ringback. This > > > only happens in calls from PSTN , not from VOIP. > > Some > > > scenarios i tried > > > to isolate this issue : > > > > > > > > > - Issue > > > > > > PSTN --> FS ( brigde ) ringback/transfer_ringback -> > > PSTN > > > > > > - Good setting bypass_media before run bridge but i > > need rtp > > > in FS path > > > > > > PSTN --> FS ( brigde ) ringback/transfer_ringback -> > > PSTN > > > > > > - Good > > > > > > PSTN --> FS ( brigde ) WITHOUT > > ringback/transfer_ringback -> > > > PSTN > > > > > > - Always good > > > > > > VOIP --> FS ( brigde ) -> PSTN > > > > > > > > > Dialplan has nothing wrong ( i guess ): > > > > > > > > > > > expression="^1??XX$"> > > > > > > > > > > > data="hangup_after_bridge=false"/> > > > > data="playback_terminators=#"/> > > > > > > > data="transfer_ringback= > > > $${hold_music}"/> > > > > data="effective_caller_id_name= > > > ${caller_id_name}"/> > > > > > data="effective_caller_id_number= > > > ${caller_id_number}"/> > > > > data="originate_timeout=30"/> > > > > data="call_timeout=30"/> > > > > > data="sofia/default/18008226...@pstn_gw"/> > > > > > >
Re: [Freeswitch-users] Error loading portaudio module
With apologies for the noise on the list, I just realized that FreeSWITCH is building its own version of PortAudio. I can confirm that Alsa is being detected and support for it included. So, there must be some difference between the version of PortAudio that comes with FreeSWITCH, and the version installed in my /usr/lib, such that the FreeSWITCH version fails to detect my Alsa devices. ___ 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] Error loading portaudio module
This is interesting... I wrote the following test. #include #include int main(int argc, char **argv) { Pa_Initialize(); printf("Number of devices: %d\n",Pa_GetDeviceCount()); Pa_Terminate(); } then I compiled and executed it: gcc -o pa_test pa_test.c -lportaudio ./portaudio Number of devices: 10 >From what I can see, the code in mod_portaudio.c is using exactly the same API call, but it seems to be returning 0 in that case. ___ 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] schedule a DTMF tone into bridge
sched_api (hint uuid_send_dtmf) API CALL [sched_api()] output: -ERR Invalid syntax. USAGE: [...@] /b On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote: Is there a way to schedule a certain DTMF tone to be played into a bridge (both a and b legs) after a scheduled number of seconds into the call? ___ 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] schedule a DTMF tone into bridge
Is there a way to schedule a certain DTMF tone to be played into a bridge (both a and b legs) after a scheduled number of seconds into the call? ___ 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] Error loading portaudio module
On Fri, Dec 12, 2008 at 09:30:16PM +0100, Giovanni Maruzzelli wrote: > But in this specific case, no device at all was found. > > So, maybe portaudio was not commpiled with ALSA support (do you have > the ALSA development library installed?). Yes, and in any case the version of PortAudio which is installed came from the Debian package. Does FreeSWITCH support PortAudio 19? If not, maybe there are API differences. > > Also, after recompiling portaudio and mod_portaudio, you can launch FS > giving it the PA_ALSA_PLUGHW=1 environment variable, so portaudio will > use the plughw devices (that are automatically converted to the > desired rate/format) and not the raw devices. I'll try that. To answer another question that arose in this thread, I have no other software currently using the audio devices. Alsa is known to work, as is other software that accesses the Alsa devices with PortAudio. ___ 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] Cepstral SDK
If you're on linux you need to go download and install any voice. If you're on windows I have to forward your request to Cepstral to get the SDK for windows. /b On Dec 12, 2008, at 3:47 PM, Pedro . wrote: > Hi, > > I'm trying to integrate Cepstral TTS I read in the wiki that I need > Ceptral's SDK to compile the mod_ceptral, can somebody tell me where > can I get the trial version of this SDK?. > > 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 ___ 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] Cepstral SDK
Hi, I'm trying to integrate Cepstral TTS I read in the wiki that I need Ceptral's SDK to compile the mod_ceptral, can somebody tell me where can I get the trial version of this SDK?. 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] CDR logs - adding a custom field
I don't know about "good" examples. I just hacked together a perl script to extract the very specific elements for my application. If anyone out there has a sample XML-to-db parser that would be very welcomed... -MC On Fri, Dec 12, 2008 at 12:28 PM, Shelby Ramsey wrote: > Are there any good examples floating around of XML parsers for this to dump > to MySQL? > > On Fri, Dec 12, 2008 at 2:22 PM, Michael Collins wrote: >> >> On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu >> wrote: >> > Thanks Michael, >> > >> > I'm going to use XML, since I don't really know what variables I want. >> > Another problem with CSV is that many people parse them with regular >> > expressions and scripts break when you add a new column. >> > >> >> This is true. If you build a proper parser for your XML it will easily >> be able to handle new channel variables. >> -MC >> >> ___ >> 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
Re: [Freeswitch-users] CDR logs - adding a custom field
On Fri, Dec 12, 2008 at 12:21 PM, Brian West wrote: > What I think would be neat is to have a perl script to parse the XML > cdr and spit out a graphic of the call path... now that would be neat. > /b I think that is a great idea. I was kicking that around as an add-on feature to a simple CDR database. For example, when browsing the db for calls, you could click a link that says "view call path" and it would print a nice purty graph/chart of the call flow. I'll put that on my rainy-day list... -MC > > On Dec 12, 2008, at 2:14 PM, Alexandru Nedelcu wrote: > >> Thanks Michael, >> >> I'm going to use XML, since I don't really know what variables I want. >> Another problem with CSV is that many people parse them with regular >> expressions and scripts break when you add a new column. > > > ___ > 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] Error loading portaudio module
Sorry, the previous one was sent by mistake. This one is complete: Hi there, you have to use the "default" ALSA audio device to share it, and to have it automatically format and rate converted. the "default" ALSA device is not the default portaudio device (not in the portaudio version used currently by FS). You have to find out what device id it has under portaudio. But in this specific case, no device at all was found. So, maybe portaudio was not commpiled with ALSA support (do you have the ALSA development library installed?). Also, after recompiling portaudio and mod_portaudio, you can launch FS giving it the PA_ALSA_PLUGHW=1 environment variable, so portaudio will use the plughw devices (that are automatically converted to the desired rate/format) and not the raw devices. Giovanni Maruzzelli = Company : Celliax Website: www.celliax.org Address : via Pierlombardo 9, 20135 Milano Country/Territory : Italy Business Email: gmaruzz at celliax dot org Cell : 39-347-2665618 Fax : 39-02-87390039 On Fri, Dec 12, 2008 at 9:25 PM, Giovanni Maruzzelli wrote: > Hi there, > > you have to use the "default" ALSA audio device to share it, and to > have it automatically format and rate converted. > > the "default" ALSA device is not the default portaudio device (not in > the portaudio version used currently by FS). > > You have to find out what device id it has under portaudio. > > But in this specific case, no device at all was found. > > So, maybe portaudio was not commpiled with ALSA support (do you have the ALSA > > > Sincerely, > > Giovanni Maruzzelli > = > Company : Celliax > Website: www.celliax.org > Address : via Pierlombardo 9, 20135 Milano > Country/Territory : Italy > Business Email: gmaruzz at celliax dot org > Cell : 39-347-2665618 > Fax : 39-02-87390039 > > > > > On Fri, Dec 12, 2008 at 8:58 PM, Michael Collins wrote: >> Jason, >> >> If I understand correctly software other than PA can lock up the sound >> card so that PA doesn't "see" it. That might explain why PA reports >> number of devices = 0. Could you check to see if possibly something >> else has control of your sound card, perhaps ALSA? Turn off anything >> that might use the sound card and then restart FS to see if PA can >> then detect your device. >> >> -MC >> >> On Fri, Dec 12, 2008 at 1:38 AM, Jason White wrote: >>> I am new to FreeSWITCH; hence this is the first of what will probably be a >>> number of questions as I learn. >>> >>> I've compiled the latest code from svn trunk under Debian Sid (Linux kernel >>> 2.6.27, x86_64 architecture), with the portaudio19-dev package installed. >>> >>> Whenever I try to load the portaudio module I get the following in the >>> logs. I >>> haven't changed anything in the default portaudio configuration that comes >>> with FreeSWITCH. >>> >>> PortAudio version number = 1899 >>> PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)' >>> Number of devices = 0 >>> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an >>> input >>> device! >>> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an >>> input >>> device! >>> 2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839 >>> switch_loadable_module_l >>> oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so >>> >>> Other software that uses portaudio is known to work. I would expect >>> FreeSWITCH >>> to detect my Alsa sound devices. >>> >>> Suggestions welcome. >>> >>> >>> ___ >>> 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
Re: [Freeswitch-users] Configuring FreeSwitch
On Thu, 2008-12-11 at 09:55 -0500, Raymond Chandler wrote: > > i think i answered all of this for you on irc yesterday > Yes you did, thanks for your help. I'm a total newbie, but the good news is that I'm almost finished with my setup. FS is great :) > use the bridge dialplan app to dial by ip similar to the following: > data="sofia/${use_profile}/num...@ip.address"/> I'm using "originate" initially. And I think I did something stupid. Is there anything wrong with the following code ... var new_session = new Session(); new_session.originate(session, URL); bridge(session, new_session); > http://wiki.freeswitch.org/wiki/Sofia#Syntax might also help you out > a > little It worked great. 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] CDR logs - adding a custom field
Are there any good examples floating around of XML parsers for this to dump to MySQL? On Fri, Dec 12, 2008 at 2:22 PM, Michael Collins wrote: > On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu > wrote: > > Thanks Michael, > > > > I'm going to use XML, since I don't really know what variables I want. > > Another problem with CSV is that many people parse them with regular > > expressions and scripts break when you add a new column. > > > > This is true. If you build a proper parser for your XML it will easily > be able to handle new channel variables. > -MC > > ___ > 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] Error loading portaudio module
Hi there, you have to use the "default" ALSA audio device to share it, and to have it automatically format and rate converted. the "default" ALSA device is not the default portaudio device (not in the portaudio version used currently by FS). You have to find out what device id it has under portaudio. But in this specific case, no device at all was found. So, maybe portaudio was not commpiled with ALSA support (do you have the ALSA Sincerely, Giovanni Maruzzelli = Company : Celliax Website: www.celliax.org Address : via Pierlombardo 9, 20135 Milano Country/Territory : Italy Business Email: gmaruzz at celliax dot org Cell : 39-347-2665618 Fax : 39-02-87390039 On Fri, Dec 12, 2008 at 8:58 PM, Michael Collins wrote: > Jason, > > If I understand correctly software other than PA can lock up the sound > card so that PA doesn't "see" it. That might explain why PA reports > number of devices = 0. Could you check to see if possibly something > else has control of your sound card, perhaps ALSA? Turn off anything > that might use the sound card and then restart FS to see if PA can > then detect your device. > > -MC > > On Fri, Dec 12, 2008 at 1:38 AM, Jason White wrote: >> I am new to FreeSWITCH; hence this is the first of what will probably be a >> number of questions as I learn. >> >> I've compiled the latest code from svn trunk under Debian Sid (Linux kernel >> 2.6.27, x86_64 architecture), with the portaudio19-dev package installed. >> >> Whenever I try to load the portaudio module I get the following in the logs. >> I >> haven't changed anything in the default portaudio configuration that comes >> with FreeSWITCH. >> >> PortAudio version number = 1899 >> PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)' >> Number of devices = 0 >> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an >> input >> device! >> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an >> input >> device! >> 2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839 >> switch_loadable_module_l >> oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so >> >> Other software that uses portaudio is known to work. I would expect >> FreeSWITCH >> to detect my Alsa sound devices. >> >> Suggestions welcome. >> >> >> ___ >> 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
Re: [Freeswitch-users] CDR logs - adding a custom field
On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu wrote: > Thanks Michael, > > I'm going to use XML, since I don't really know what variables I want. > Another problem with CSV is that many people parse them with regular > expressions and scripts break when you add a new column. > This is true. If you build a proper parser for your XML it will easily be able to handle new channel variables. -MC ___ 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] CDR logs - adding a custom field
What I think would be neat is to have a perl script to parse the XML cdr and spit out a graphic of the call path... now that would be neat. /b On Dec 12, 2008, at 2:14 PM, Alexandru Nedelcu wrote: > Thanks Michael, > > I'm going to use XML, since I don't really know what variables I want. > Another problem with CSV is that many people parse them with regular > expressions and scripts break when you add a new column. ___ 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] Freeswitch logging
On Sat, 2008-12-13 at 01:26 +1300, Hadley Rich wrote: > This was answered on IRC and a note added to the mod_cdr wiki page. Thanks Hadley, I'm a total newbie to FreeSwitch and voip in general, sorry for my persistence :) I'll try writing an article about my setup this weekend. ___ 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] CDR logs - adding a custom field
Thanks Michael, I'm going to use XML, since I don't really know what variables I want. Another problem with CSV is that many people parse them with regular expressions and scripts break when you add a new column. On Fri, 2008-12-12 at 11:50 -0800, Michael Collins wrote: > Are you using CSV or XML? The reason I ask is because I personally use > XML and I find that having lots of information (even too much) is > better than not enough. The only drawback to XML that I find is that > you have to know how to parse it properly. :) The level of detail in > the XML CDRs is unmatched by any telephony system I've ever > encountered. I highly recommend it. ___ 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] CDR logs - adding a custom field
On Fri, 2008-12-12 at 13:18 -0600, Anthony Minessale wrote: > Yes, I'm familiar with that since i invented that feature for Asterisk > =D > > > In FreeSWITCH, All variables are already available from the cdr > just set regular channel variables. > > for xml cdr they are all there right away > for csv cdr you can reference any channel variable in your template. > Thank you Anthony, In case someone wants to know how to set channel variables, there's a link on the wiki here: http://wiki.freeswitch.org/wiki/Channel_Variables ___ 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] Error loading portaudio module
Jason, If I understand correctly software other than PA can lock up the sound card so that PA doesn't "see" it. That might explain why PA reports number of devices = 0. Could you check to see if possibly something else has control of your sound card, perhaps ALSA? Turn off anything that might use the sound card and then restart FS to see if PA can then detect your device. -MC On Fri, Dec 12, 2008 at 1:38 AM, Jason White wrote: > I am new to FreeSWITCH; hence this is the first of what will probably be a > number of questions as I learn. > > I've compiled the latest code from svn trunk under Debian Sid (Linux kernel > 2.6.27, x86_64 architecture), with the portaudio19-dev package installed. > > Whenever I try to load the portaudio module I get the following in the logs. I > haven't changed anything in the default portaudio configuration that comes > with FreeSWITCH. > > PortAudio version number = 1899 > PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)' > Number of devices = 0 > 2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an > input > device! > 2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an > input > device! > 2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839 > switch_loadable_module_l > oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so > > Other software that uses portaudio is known to work. I would expect FreeSWITCH > to detect my Alsa sound devices. > > Suggestions welcome. > > > ___ > 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] CDR logs - adding a custom field
Are you using CSV or XML? The reason I ask is because I personally use XML and I find that having lots of information (even too much) is better than not enough. The only drawback to XML that I find is that you have to know how to parse it properly. :) The level of detail in the XML CDRs is unmatched by any telephony system I've ever encountered. I highly recommend it. Also, check out this wiki page if you haven't already: http://wiki.freeswitch.org/wiki/Mod_xml_cdr -MC On Fri, Dec 12, 2008 at 10:37 AM, Alexandru Nedelcu wrote: > In Asterisk I was able to set a custom CDR field by doing something > like: > Set(CDR(userfield)=${SOMETHING}) > > I need to set a custom field in FreeSwitch, and preferably I want to > have control over its value from Javascript. > > Can someone tell me how? :) > > Thanks, > > -- > Alexandru Nedelcu > Software Developer, Sinapticode > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CDR logs - adding a custom field
Yes, I'm familiar with that since i invented that feature for Asterisk =D In FreeSWITCH, All variables are already available from the cdr just set regular channel variables. for xml cdr they are all there right away for csv cdr you can reference any channel variable in your template. On Fri, Dec 12, 2008 at 12:37 PM, Alexandru Nedelcu wrote: > In Asterisk I was able to set a custom CDR field by doing something > like: > Set(CDR(userfield)=${SOMETHING}) > > I need to set a custom field in FreeSwitch, and preferably I want to > have control over its value from Javascript. > > Can someone tell me how? :) > > Thanks, > > -- > Alexandru Nedelcu > Software Developer, Sinapticode > > > ___ > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ 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] CDR logs - adding a custom field
In Asterisk I was able to set a custom CDR field by doing something like: Set(CDR(userfield)=${SOMETHING}) I need to set a custom field in FreeSwitch, and preferably I want to have control over its value from Javascript. Can someone tell me how? :) Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode ___ 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] Error loading portaudio module
I am new to FreeSWITCH; hence this is the first of what will probably be a number of questions as I learn. I've compiled the latest code from svn trunk under Debian Sid (Linux kernel 2.6.27, x86_64 architecture), with the portaudio19-dev package installed. Whenever I try to load the portaudio module I get the following in the logs. I haven't changed anything in the default portaudio configuration that comes with FreeSWITCH. PortAudio version number = 1899 PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)' Number of devices = 0 2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an input device! 2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an input device! 2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839 switch_loadable_module_l oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so Other software that uses portaudio is known to work. I would expect FreeSWITCH to detect my Alsa sound devices. Suggestions welcome. ___ 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] call transfer question
Thank you, that is exactly what I need. On Fri, Dec 12, 2008 at 9:14 AM, Brian West wrote: > You can use deflect to accomplish this.. it will do a refer to the > other FS box. > > /b > > On Dec 12, 2008, at 11:08 AM, jonathan augenstine wrote: > > > I have a call scenario that involves transferring the call and > > dropping out of the SIP/RTP stream. I need to accept the SIP call, > > play a prompt, and retrieve a pin code. After a database lookup, I > > need to transfer the call to another FS server and drop out of the > > SIP path. I have done this with the RTP media stream previously. I > > am not sure what I need to do to drop out of the SIP path. Is this > > possible on FS? > > > > Jonathan > > > ___ > 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] call transfer question
You can use deflect to accomplish this.. it will do a refer to the other FS box. /b On Dec 12, 2008, at 11:08 AM, jonathan augenstine wrote: > I have a call scenario that involves transferring the call and > dropping out of the SIP/RTP stream. I need to accept the SIP call, > play a prompt, and retrieve a pin code. After a database lookup, I > need to transfer the call to another FS server and drop out of the > SIP path. I have done this with the RTP media stream previously. I > am not sure what I need to do to drop out of the SIP path. Is this > possible on FS? > > Jonathan ___ 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] call transfer question
I have a call scenario that involves transferring the call and dropping out of the SIP/RTP stream. I need to accept the SIP call, play a prompt, and retrieve a pin code. After a database lookup, I need to transfer the call to another FS server and drop out of the SIP path. I have done this with the RTP media stream previously. I am not sure what I need to do to drop out of the SIP path. Is this possible on FS? Jonathan ___ 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] conference_auto_outcall_announce
No, there is currently no way. On Fri, Dec 12, 2008 at 8:26 AM, Carole O. wrote: > > Hello, > > First, I would like to apologize for a mistake I have made: by adding the > following line in the profile > < param name="enter-sound" value="path/to/file.wav" / > > the enter sound is played. > I am sorry for this. I did not hear it because in the case I have been > analyzing the members of the conference the caller automatically invites > are > VoIP speakers which beep before playing anything and apparently miss the > enter sound. (both the beep and the enter-sound have about the same > length). > > I still have the following questions: > 1- Is it possible to introduce a delay so that the enter sound is played > only after 2s? > > 2- I have noticed that if the caller of the conference talks or makes some > noises at the very beginning when he is entering the conference and the > enter sound is played, we can hear it through the VoIP speakers. Is there > any way to prevent from this? I would like to mute the caller during the > enter-sound and I would need this to be done statically, I mean in the xml > files, and not from the shell. > > Thanks!! > Carole > > > > Carole O. wrote: > > > > Hello, > > > > Actually, I have already tried it but nothing happens: the file is not > > played and there is no error. > > There is still a difference: if I configure it as you said, I can not be > > listening anymore, there is simply nothing. > > > > Would you have an idea? I have checked the path and the syntax 1 million > > times so I do not think I make mistake there. > > > > Thanks, > > Carole > > > > > > > > Brian West-3 wrote: > >> > >> Don't have play: in there and it should be fine. Also if you want the > >> absolute path you start it with /path/to/file.wav > >> > >> > >> /b > >> > >> On Dec 11, 2008, at 7:13 AM, Carole O. wrote: > >> > >>> [/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav] > >>> [System error : no such file or directory] > >> > >> > >> ___ > >> 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 > >> > >> > > > > > > -- > View this message in context: > http://www.nabble.com/conference_auto_outcall_announce-tp20955216p20976612.html > Sent from the Freeswitch-users mailing list archive at Nabble.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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ 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] fifo.conf.xml usage
the entries are standard originate strings so all of the {} variables apply. On Fri, Dec 12, 2008 at 7:30 AM, Jon Bruel wrote: > I'm happy to see that you can add consumers to queues using the > fifo.conf.xml configuration file. I have made some tests and I hope it may > lead to a more universal way of setting up queues for small organisations > than the one I have described in the wiki, and which includes (too) many > javascripts. I have some questions to clarify my understanding. Using the > fifo.conf.xml, I find: > > - That the consumers continue to ring after the caller has abandoned the > queue. Is there a way to avoid this? > > Further: > > - Is there a way to control the caller_id_name/number presented to the > consumer? > > - Is there a way to control the ringing tone in the consumers like the one > which can be used in the dialplan? > > - Can the fifo.conf.xml refer to an ODBC connection in order to get the > members from a database? > > Finally, thanks for all the good work everybody in the FS community has put > into FS, I truly believe in the possibilities of this product. Checking the > hits on Google certainly indicates you moving into the right direction. /Jon > > > > ___ > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ 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] Bug in Q921.c? AVAYA-PBX issue
if you open a jira issue on it we can probably add your patch and/or the config option. the users-list is a tough place to manage TDM issues. On Fri, Dec 12, 2008 at 9:01 AM, Helmut Kuper wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Hello, > > I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an > AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu 8.04 > server in a non-root environment. > > We experienced a timer problem which led to this FS console error message: > > [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries > > > During anylizing this we found that q921 T203 is never reset when link > is in state "Multiple Frame Mode Established" and SABME frames are > received by FS. So it must timeout regardless if SABME frames are > received or not. > Additionally we found that the default T203 value (10 sec) was too short > for AVAYA (it has to be >=19 sec) > > To fix the problem we changed two things in q921.c: > > Change T203 default value from 10 sec to 2 sec > Line 406: trunk->T203Timeout = 2; > > Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each > received SABME frame > Line 1996: Q921T203TimerReset(trunk, tei); > > After recompiling FS the Error disapeared. Next week we will do some > calls over the link to make sure there are no other side effects. > > Is it planned to make the q921 timeouts configurable in openzap.conf or > in openzap.conf.xml? > > best regards > Helmut > > > PS: My openzap configs: > > openzap.conf > > [span wanpipe PRI_1] > trunk_type => E1 > b-channel => 1:1-15 > d-channel => 1:16 > b-channel => 1:17-31 > > > > > openzap.conf.xml > > > > > > > > > > > > > > > > > > > > > > Very interesting here is, that the "dialect" parameter doesn't seem to > have an effect on FS. I use that one above without any errors or warning > and I guess that was not intended. > > > > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p > BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB > =geXp > -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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ 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] LDAP Integration
does anyone have a sample of the config file for mod_xml_ldap? and know where to put it? Vinicius Kobashi escreveu: i did it... i still had some problems with sasl, but i managed to fix them. now the module is up and running but i still dunno where to put the mod_xml_ldap configuration file. does anyone have a sample of the config file? and know where to put it? Michael Collins escreveu: Please confirm your svn rev - I believe this was fixed recently. Do "make current" in your source directory. -MC On Thu, Dec 11, 2008 at 1:35 PM, Vinicius Kobashi wrote: ok ill try that i found another module thats mod_xml_ldap but when i try to load it, during compiling i get the 404 error http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz file not found ill try to download it myself and then try to compile freeswitch again and test =D thankz for the fast answer Hadley Rich escreveu: On Friday 12 December 2008 09:16:56 Vinicius Kobashi wrote: i found another module called mod_xml_curl and loaded it to freeswitch too... but still it shows me the following error: 2008-12-11 17:04:04 [WARNING] sofia_reg.c:1501 sofia_reg_parse_auth() Can't find user [usern...@freeswitchserver.com] You must define a domain called 'freeswitchserver.com' in your directory and add a user with the id="username" attribute and you must configure your device to use the proper domain in it's authentication credentials. does anyone got an idea? Yes, you need to define a domain called 'freeswitchserver.com' in your directory and add a user with the id="username" just like the error message says. The directory files are in conf/directory/ If you would like to read up on mod_xml_curl there is a detailed page on the wiki; http://wiki.freeswitch.org/wiki/Mod_xml_curl hads -- Vinicius Kobashi Infra-Estrutura Ydea Desenvolvimento de Software LTDA. Av. Adolfo Pinheiro, 2338 - Alto da Boa Vista CEP.:04734-004 - São Paulo - SP Tel.: 55-11-5523-0333 ___ 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 -- Vinicius Kobashi Infra-Estrutura Ydea Desenvolvimento de Software LTDA. Av. Adolfo Pinheiro, 2338 - Alto da Boa Vista CEP.:04734-004 - São Paulo - SP Tel.: 55-11-5523-0333 ___ 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 -- Vinicius Kobashi Infra-Estrutura Ydea Desenvolvimento de Software LTDA. Av. Adolfo Pinheiro, 2338 - Alto da Boa Vista CEP.:04734-004 - São Paulo - SP Tel.: 55-11-5523-0333 ___ 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] Bug in Q921.c? AVAYA-PBX issue
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu 8.04 server in a non-root environment. We experienced a timer problem which led to this FS console error message: [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries During anylizing this we found that q921 T203 is never reset when link is in state "Multiple Frame Mode Established" and SABME frames are received by FS. So it must timeout regardless if SABME frames are received or not. Additionally we found that the default T203 value (10 sec) was too short for AVAYA (it has to be >=19 sec) To fix the problem we changed two things in q921.c: Change T203 default value from 10 sec to 2 sec Line 406: trunk->T203Timeout = 2; Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each received SABME frame Line 1996: Q921T203TimerReset(trunk, tei); After recompiling FS the Error disapeared. Next week we will do some calls over the link to make sure there are no other side effects. Is it planned to make the q921 timeouts configurable in openzap.conf or in openzap.conf.xml? best regards Helmut PS: My openzap configs: openzap.conf [span wanpipe PRI_1] trunk_type => E1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 openzap.conf.xml Very interesting here is, that the "dialect" parameter doesn't seem to have an effect on FS. I use that one above without any errors or warning and I guess that was not intended. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB =geXp -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] Sounds for pending 1.0.2/Hardware
FreeSWITCHers, I would like to thank everyone that donated. Enough was raised to cover the sound order. ;) Thanks, Brian West 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] conference_auto_outcall_announce
Hello, First, I would like to apologize for a mistake I have made: by adding the following line in the profile < param name="enter-sound" value="path/to/file.wav" / > the enter sound is played. I am sorry for this. I did not hear it because in the case I have been analyzing the members of the conference the caller automatically invites are VoIP speakers which beep before playing anything and apparently miss the enter sound. (both the beep and the enter-sound have about the same length). I still have the following questions: 1- Is it possible to introduce a delay so that the enter sound is played only after 2s? 2- I have noticed that if the caller of the conference talks or makes some noises at the very beginning when he is entering the conference and the enter sound is played, we can hear it through the VoIP speakers. Is there any way to prevent from this? I would like to mute the caller during the enter-sound and I would need this to be done statically, I mean in the xml files, and not from the shell. Thanks!! Carole Carole O. wrote: > > Hello, > > Actually, I have already tried it but nothing happens: the file is not > played and there is no error. > There is still a difference: if I configure it as you said, I can not be > listening anymore, there is simply nothing. > > Would you have an idea? I have checked the path and the syntax 1 million > times so I do not think I make mistake there. > > Thanks, > Carole > > > > Brian West-3 wrote: >> >> Don't have play: in there and it should be fine. Also if you want the >> absolute path you start it with /path/to/file.wav >> >> >> /b >> >> On Dec 11, 2008, at 7:13 AM, Carole O. wrote: >> >>> [/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav] >>> [System error : no such file or directory] >> >> >> ___ >> 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 >> >> > > -- View this message in context: http://www.nabble.com/conference_auto_outcall_announce-tp20955216p20976612.html Sent from the Freeswitch-users mailing list archive at Nabble.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] Freeswitch streamFile when the user answers
How are you originating calls? You probably need to add {ignore_early_media=true}. This tells FreeSWITCH not to return from origination when early media (progress/ringing) was received (I think anyway)... See http://wiki.freeswitch.org/wiki/Channel_Variables#ignore_early_media There is a sample of this in use with the originate command here: http://wiki.freeswitch.org/wiki/Mod_commands#originate (about halfway down) Setting channel variables before doing the originate originate {ignore_early_media=true}sofia/mydomain.com/18005551...@1.2.3.4 1551212 Since you are making a dialer, you may want to start the originations in the background and move on to the next call while tweaking the timeout value for originated calls. From the WIKI again: "You can originate a call in the background (asynchronously) and playback a message with a 60 second timeout. bgapi originate {ignore_early_media=true,originate_timeout=60}sofia/gateway/name/number &playback(message)" - Darren -Original Message- From: Alexandru Nedelcu [mailto:a...@sinapticode.ro] Sent: Friday, December 12, 2008 3:39 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch streamFile when the user answers Hi, I'm working on a simple dialer, and I have the following problem: the audio file starts playing before the user answeres the phone (while it's ringing). It only works when I introduce a delay, but that doesn't seem right. For instance in the asterisk context referred in the call files, I had: exten => s,4,Answer exten => s,n,Wait(2) exten => s,n,Background(${SOUNDFILE}) And indeed it played a soundfile 2 seconds after the called person picked up the phone In FS I currently initiate calls like this: session.waitForAnswer(1); if (session.ready()) { session.sleep(2000); session.streamFile(/*...*/); } Is this right? ___ 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] fifo.conf.xml usage
I'm happy to see that you can add consumers to queues using the fifo.conf.xml configuration file. I have made some tests and I hope it may lead to a more universal way of setting up queues for small organisations than the one I have described in the wiki, and which includes (too) many javascripts. I have some questions to clarify my understanding. Using the fifo.conf.xml, I find: - That the consumers continue to ring after the caller has abandoned the queue. Is there a way to avoid this? Further: - Is there a way to control the caller_id_name/number presented to the consumer? - Is there a way to control the ringing tone in the consumers like the one which can be used in the dialplan? - Can the fifo.conf.xml refer to an ODBC connection in order to get the members from a database? Finally, thanks for all the good work everybody in the FS community has put into FS, I truly believe in the possibilities of this product. Checking the hits on Google certainly indicates you moving into the right direction. /Jon ___ 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] Freeswitch logging
On Saturday 13 December 2008 01:10:29 Alexandru Nedelcu wrote: > Hi, > > I see that mod_cdr is marked as being non-functional on the wiki. I'm > working on a dialer and I need a way to log information about calls. > > What module should I use? This was answered on IRC and a note added to the mod_cdr wiki page. hads -- http://nicegear.co.nz New Zealands Open Source Hardware Supplier ___ 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] Freeswitch logging
Hi, I see that mod_cdr is marked as being non-functional on the wiki. I'm working on a dialer and I need a way to log information about calls. What module should I use? 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
[Freeswitch-users] Freeswitch streamFile when the user answers
Hi, I'm working on a simple dialer, and I have the following problem: the audio file starts playing before the user answeres the phone (while it's ringing). It only works when I introduce a delay, but that doesn't seem right. For instance in the asterisk context referred in the call files, I had: exten => s,4,Answer exten => s,n,Wait(2) exten => s,n,Background(${SOUNDFILE}) And indeed it played a soundfile 2 seconds after the called person picked up the phone In FS I currently initiate calls like this: session.waitForAnswer(1); if (session.ready()) { session.sleep(2000); session.streamFile(/*...*/); } Is this right? ___ 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] config help: openzap and T1 A102u
Did you try ./wanrouter start before starting FreeSWITCH ? - Original Message - From: To: Sent: Friday, December 12, 2008 2:51 AM Subject: [Freeswitch-users] config help: openzap and T1 A102u >I am stuck trying to bring up freeswitch with openzap on a Sangoma A102u T1 >card. > Works fine with asterisk. > > Please point out where I am failing to configure properly. > > Running Linux version 2.6.9-34.ELsmp on a Dell Celeron > > % wanrouter hwprobe verbose > > - > | Wanpipe Hardware Probe Info (verbose) | > - > 1 . AFT-A102u : SLOT=1 : BUS=2 : IRQ=145 : CPU=A : PORT=PRI : V=25 > +01:PMC4351:PCI > 2 . AFT-A102u : SLOT=1 : BUS=2 : IRQ=145 : CPU=B : PORT=PRI : V=25 > +01:PMC4351:PCI > > Card Cnt: S508=0 S514X=0 S518=0 A101-2=1 A104=0 A300=0 A200=0 > A108=0 > > % cat /usr/local/freeswitch/conf/autoload_configs/open > openmrcp.conf.xml openzap.conf.xml > [r...@pbxtra1466 freeswitch]# cat > /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > % cat /etc/openzap/openzap.conf > [span wanpipe] > trunk_type => t1 > b-channel => 1:1-23 > d-channel=> 1:24 > > [span wanpipe] > trunk_type => t1 > b-channel => 2:25-47 > d-channel=> 2:48 > > % cat /etc/openzap/wanpipe.conf > [defaults] > codec_ms => 20 > wink_ms => 150 > flash_ms => 750 > > % cat /etc/wanpipe/wanpipe1.conf > # > # WANPIPE1 Configuration File > # > # > # Date: Tue Dec 12 16:21:45 UTC 2006 > # > # Note: This file was generated automatically > # by /usr/sbin/wancfg program. > # > # If you want to edit this file, it is > # recommended that you use wancfg program > # to do so. > # > # Sangoma Technologies Inc. > # > > [devices] > wanpipe1 = WAN_AFT, Comment > > [interfaces] > w1g1 = wanpipe1, , TDM_VOICE, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort= PRI > AUTO_PCISLOT= NO > PCISLOT = 1 > PCIBUS = 2 > FE_MEDIA= T1 > FE_LCODE= B8ZS > FE_FRAME= ESF > FE_LINE = 1 > TE_CLOCK= NORMAL > TE_REF_CLOCK= 0 > TE_SIG_MODE = CCS > TE_HIGHIMPEDANCE= NO > LBO = 0DB > FE_TXTRISTATE = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_SPAN = 1 > TDMV_DCHAN = 24 > > [w1g1] > ACTIVE_CH = ALL > TDMV_ECHO_OFF = NO > TDMV_HWEC = NO > > % cat /etc/wanpipe/wanpipe2.conf > # > # WANPIPE1 Configuration File > # > # > # Date: Tue Dec 12 16:21:45 UTC 2006 > # > # Note: This file was generated automatically > # by /usr/sbin/wancfg program. > # > # If you want to edit this file, it is > # recommended that you use wancfg program > # to do so. > # > # Sangoma Technologies Inc. > # > > [devices] > wanpipe2 = WAN_AFT, Comment > > [interfaces] > w2g1 = wanpipe2, , TDM_VOICE, Comment > > [wanpipe2] > CARD_TYPE = AFT > S514CPU = B > CommPort= PRI > AUTO_PCISLOT= NO > PCISLOT = 1 > PCIBUS = 2 > FE_MEDIA= T1 > FE_LCODE= B8ZS > FE_FRAME= ESF > FE_LINE = 1 > TE_CLOCK= NORMAL > TE_REF_CLOCK= 0 > TE_SIG_MODE = CCS > TE_HIGHIMPEDANCE= NO > LBO = 0DB > FE_TXTRISTATE = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_SPAN = 2 > TDMV_DCHAN = 24 > > [w2g1] > ACTIVE_CH = ALL > TDMV_ECHO_OFF = NO > TDMV_HWEC = NO > > freeswi...@hostname-elided> load mod_openzap > 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure > configuring device s1c1 > 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure > configuring device s1c2 > 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure > configuring device s1c3 > 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure > configuring device s1c4 > 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure > configuring device s1c5 > 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure > configuring device s1c6 > 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure > configuring device s1c7 > 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure > confi