[Freeswitch-users] Problem with Freeswitch capturing DTMF
Hi, I am wondering if I am the only one getting this problem or not. When sending in DTMF to freeswitch, freeswitch is not always capable of capturing all the DTMF being sent. For instance, sending 1000 to freeswitch may end up becoming 100 or 10003 becoming 1003. Am I the only one getting this strange issue? If anyone know how to fix this problem, I would greatly appreciate it. Regards, Keith ___ 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] Problem with Freeswitch capturing DTMF
Hi Keith, I was just writing a note along similar lines to Mike's. If you need a hand getting a packet capture or interpreting it, drop me a note off-list. Cheers -- Dave We generally are as good as possible on capturing dtmf reliably. If you are seeing dropouts like that I would have to guess that this is a very lossy line. Could you try and look at the packet capture of a call that is missing digits and see if you are indeed dropping a lot of packets. If this is the case you could try info dtmf although that method has it's own issues. Mike On Dec 2, 2008, at 6:23 AM, Keith Wood [EMAIL PROTECTED] wrote: Hi, I am wondering if I am the only one getting this problem or not. When sending in DTMF to freeswitch, freeswitch is not always capable of capturing all the DTMF being sent. For instance, sending 1000 to freeswitch may end up becoming 100 or 10003 becoming 1003. Am I the only one getting this strange issue? If anyone know how to fix this problem, I would greatly appreciate it. Regards, Keith ___ 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 -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk ___ 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] Question about wrapping libfreeswitch
Hi,
I am sorry again for sending another email to the group again. I am working
on embedding libfreeswitch to provide better monitoring. The first thing I
attempt to do is to run the sample code provided in the website:
#include switch.h
int main(int argc, char **argv)
{
switch_core_flag_t flags = SCF_USE_SQL;
int nc=0; /* this is for 'no console' mode, FALSE console is there, TRUE
it isnt */
const char **err = NULL; /* error value for return from freeswitch
initialization */
#define LOGFILE freeswitch.log
static char *lfile = LOGFILE; /* if NULL no logfile is generated */
switch_core_init_and_modload(*lfile,flags,err);
switch_core_runtime_loop(nc);
switch_core_destroy();
return (0); /* per C89 spec */
}
But this code gives me segmentation fault when executing it. This piece of
code is supposed to start up freeswitch and run it is a loop. Does anyone
see what is wrong with it? Does anyone have any working example that I can
refer to?
Thanks,
Woody
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[Freeswitch-users] Wrong # in voicemail
My dialplan is pretty simple. I have a single trunk with a vonage softphone
DID (1303... we'll call it main) and a virtual DID (1816...) which rings
the softphone DID. All incoming calls show up as from softphone DID but the
sip_to_user holds the actual number dialed so I can enter the dialplan
properly.
I have 2 extensions in my directory/extensions, one for each of the DID's.
The extensions check sip_to_user for match and that works great. I match on
([0,1]?)(10 digit did) and it enters the dialplans correctly, plays the
right music for each DID while the dial is occuring, so all that works. The
bridge to user/[EMAIL PROTECTED] also works fine.
The continue_on_fail is set properly so on no answer call_timeout hits (at
25 secs), and goes to voicemail... works also for both numbers.
transfer to voicemail is as follows
action application=answer/
action application=voicemail data=default $${domain} $2/
which should be pulling $2 from the condition check shown above, which it
does, cuz the bridge works...
When I call in on main DID, I get leave a message for 1303... The Main
DID..
When I call in on virtual, I get leave a message for 1303... The Main DID
rather than the 1816
How can I get voicemail to use the correct DID.HELP!! :D
--
View this message in context:
http://www.nabble.com/Wrong---in-voicemail-tp20791453p20791453.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
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Re: [Freeswitch-users] Listen to a file, while recording?
we configured mod_shout and are able to record mp3. but if we start to playback the file, it will only be played back to that point, which was recorded, when we started the player. we do this with api uuid_record uuid start /var/www/test.mp3. we are also able to playback a (radio-)stream to an uuid with shout://ip-adress:12345 but what do we have to do, to listen to the file/stream with a player? it seems, that fs has to stream to recording file to a streaming server (like icecast), right? but if we do api uuid_record uuid start shout://user:[EMAIL PROTECTED]:12345/ (and other combinations), we get an error: 2008-12-02 14:28:38 [ERR] mod_shout.c:730 shout_file_open() Invalid URL: x 2008-12-02 14:28:38 [ERR] switch_ivr_async.c:851 switch_ivr_record_session() Error opening shout:// are we on the right track? is there something else we have to do to make it work? thanks for your help. 2008/12/1 Anthony Minessale [EMAIL PROTECTED]: yes, mod_shout will broadcast calls as MP3 that you can listen to in itunes/winamp live. ___ 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] Linksys/Cisco SPA400 (ATA 4 Line FXO) Now Documented in Wiki
Folks; I've just taken the time to document the Sipura, err, Linksys, errr Cisco SPA400 4 line FXO Analog Telephone Adapter in the Wiki. http://wiki.freeswitch.org/wiki/SPA400_FreeSwitch_HowTo If anyone uses these ATA's and has questions about it let me know and I'll see if I can answer them in the Wiki. Of course, if you know something about it that I haven't documented, by all means, document it in the Wiki. ;-) **Note: MikeJ seems to think that I'm overlooking something pertaining to the registration of the SPA400 without a password. Evidently this pertains to accepting blind auth, but I'd need to know more about that before knowing how to piece it in. Best Regards, Karl J. Vesterling [EMAIL PROTECTED] 202-448-3009 x0 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
Re: [Freeswitch-users] Problem with Freeswitch capturing DTMF
We generally are as good as possible on capturing dtmf reliably. If you are seeing dropouts like that I would have to guess that this is a very lossy line. Could you try and look at the packet capture of a call that is missing digits and see if you are indeed dropping a lot of packets. If this is the case you could try info dtmf although that method has it's own issues. Mike On Dec 2, 2008, at 6:23 AM, Keith Wood [EMAIL PROTECTED] wrote: Hi, I am wondering if I am the only one getting this problem or not. When sending in DTMF to freeswitch, freeswitch is not always capable of capturing all the DTMF being sent. For instance, sending 1000 to freeswitch may end up becoming 100 or 10003 becoming 1003. Am I the only one getting this strange issue? If anyone know how to fix this problem, I would greatly appreciate it. Regards, Keith ___ 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] libfreeswitch question
On Dec 2, 2008, at 5:55 AM, Woody Dickson wrote: Hi, I am just having a dumb question and hoping someone can help me. I am trying to run a c program with libfreeswitch embedded so I can use some external mechanism to keep track of freeswitch, but I am having problem while compiling: [EMAIL PROTECTED] fs]# gcc switchnode.c -I/usr/local/freeswitch/ include -L/usr/local/freeswitch/lib -lfreeswitch -lpthread switchnode.c: In function 'main': switchnode.c:11: warning: passing argument 1 of 'switch_core_init_and_modload' makes integer from pointer without a cast switchnode.c:11: warning: passing argument 3 of 'switch_core_init_and_modload' from incompatible pointer type looks like you have the wrong var types you are passing here. /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `clock_gettime' -lrt /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `uuid_generate' -luuid /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `crypt_r' -lcrypt collect2: ld returned 1 exit status [EMAIL PROTECTED] fs]# Does anyone know which library is missing? ___ 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] libfreeswitch question
Hi, I am just having a dumb question and hoping someone can help me. I am trying to run a c program with libfreeswitch embedded so I can use some external mechanism to keep track of freeswitch, but I am having problem while compiling: [EMAIL PROTECTED] fs]# gcc switchnode.c -I/usr/local/freeswitch/include -L/usr/local/freeswitch/lib -lfreeswitch -lpthread switchnode.c: In function 'main': switchnode.c:11: warning: passing argument 1 of 'switch_core_init_and_modload' makes integer from pointer without a cast switchnode.c:11: warning: passing argument 3 of 'switch_core_init_and_modload' from incompatible pointer type /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `clock_gettime' /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `uuid_generate' /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `crypt_r' collect2: ld returned 1 exit status [EMAIL PROTECTED] fs]# Does anyone know which library is missing? Thanks, Woody ___ 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] Wrong # in voicemail
Can you show me the full XML for this extension including the regular
expression?
/b
On Dec 2, 2008, at 7:25 AM, ccav wrote:
transfer to voicemail is as follows
action application=answer/
action application=voicemail data=default $${domain} $2/
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Re: [Freeswitch-users] Listen to a file, while recording?
Are you on SVN trunk or what rev are you trying to use? /b On Dec 2, 2008, at 7:48 AM, Dennis wrote: it seems, that fs has to stream to recording file to a streaming server (like icecast), right? but if we do api uuid_record uuid start shout://user:[EMAIL PROTECTED]:12345/ (and other combinations), we get an error: 2008-12-02 14:28:38 [ERR] mod_shout.c:730 shout_file_open() Invalid URL: x 2008-12-02 14:28:38 [ERR] switch_ivr_async.c:851 switch_ivr_record_session() Error opening shout:// ___ 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] Dialing tone when placing a call with portaudio
What are you calling, sip I assume, this may be a case where the sip signaling is sending a 180 ringing instead of a 183 and we are not generating ringback in that case. Can you please confirm that and test if setting the ringback channel variable before bridge fixes this issue? Mike On Dec 2, 2008, at 4:12 AM, Rene Pankratz wrote: Hello, when using mod_portaudio for calling somebody I don't hear anything until the other party answers the call. Is it possible to play a dialing tone when the other party is ringing? ___ 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] Console Dialing in Freeswitch
What revision of freeswitch is this? Can you please test this with svn trunk? Mike On Dec 2, 2008, at 2:27 AM, Baskar wrote: Hi, I have updated all the above events you told .It's working fine but when i call extension 1002 from freeswitch console, call is connected to extension 1002, but FS is aborted but call is established in1002. what shall i do. what was the error. Full freeswitch get cut. output: [EMAIL PROTECTED] pa call 1002 2008-12-02 12:54:05 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel portaudio/1002 [20b1163a-29c7-4369-bdb5-27398dc1a263] 2008-12-02 12:54:07 [NOTICE] mod_portaudio.c:1555 place_call() Channel [portaudio/1002] has been answered API CALL [pa(call 1002)] output: SUCCESS:1:20b1163a-29c7-4369-bdb5-27398dc1a263 2008-12-02 12:54:07 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing FreeSWITCH-1002 in context default 2008-12-02 12:54:07 [WARNING] switch_ivr.c:1805 switch_ivr_set_user() can't find user [EMAIL PROTECTED] [EMAIL PROTECTED] 2008-12-02 12:54:07 [INFO] mod_dptools.c:902 info_function() CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [portaudio/1002] Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263] Call-Direction: [inbound] Answer-State: [answered] Channel-Read-Codec-Name: [L16] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [L16] Channel-Write-Codec-Rate: [8000] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [FreeSWITCH] Caller-Caller-ID-Number: [00] Caller-Network-Addr: [172.20.176.32] Caller-Destination-Number: [1002] Caller-Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263] Caller-Source: [mod_portaudio] Caller-Context: [default] Caller-Channel-Name: [portaudio/1002] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1228202645898038] Caller-Channel-Created-Time: [1228202645898038] Caller-Channel-Answered-Time: [1228202647630133] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_channel_name: [portaudio/1002] variable_endpoint_disposition: [ANSWER] variable_read_codec: [L16] variable_read_rate: [8000] variable_write_codec: [L16] variable_write_rate: [8000] variable_use_profile: [nat] variable_dialed_ext: [1002] variable_current_application: [info] 2008-12-02 12:54:07 [INFO] mod_dptools.c:888 log_function() Answer- State []n 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/ usr/local/freeswitch/recordings/00.2008-12-02-12-54-07.wav 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2008-12-02 12:54:07 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] :23878;rinstance=de482996ac747c8d [f7f80a05-be75-414b- bcea-4e5a34c3351e] freeswitch: src/switch_core_io.c:179: switch_core_session_read_frame: Assertion `(*frame)-codec != ((void *)0)' failed. Aborted (core dumped) [EMAIL PROTECTED] bin]# Thanks for the reply. Correct me were i am wrong. Warm Regards, N.Baskar ___ 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] Listen to a file, while recording?
i am using the latest svn trunk from today. 2008/12/2 Brian West [EMAIL PROTECTED]: Are you on SVN trunk or what rev are you trying to use? /b On Dec 2, 2008, at 7:48 AM, Dennis wrote: it seems, that fs has to stream to recording file to a streaming server (like icecast), right? but if we do api uuid_record uuid start shout://user:[EMAIL PROTECTED]:12345/ (and other combinations), we get an error: 2008-12-02 14:28:38 [ERR] mod_shout.c:730 shout_file_open() Invalid URL: x 2008-12-02 14:28:38 [ERR] switch_ivr_async.c:851 switch_ivr_record_session() Error opening shout:// ___ 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] Listen to a file, while recording?
And you have your shoutcast/icecast server set up and functional? /b On Dec 2, 2008, at 9:03 AM, Dennis wrote: i am using the latest svn trunk from today. ___ 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] Listen to a file, while recording?
no, not yet. i am still fiddling arround with icecast2. we tried it with someone, who offers radiostreams. perhaps this just works with icecast(2) and shoutcast? 2008/12/2 Brian West [EMAIL PROTECTED]: And you have your shoutcast/icecast server set up and functional? /b On Dec 2, 2008, at 9:03 AM, Dennis wrote: i am using the latest svn trunk from today. ___ 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] Listen to a file, while recording?
icecast2 is a known working server we have talked to before. /b On Dec 2, 2008, at 9:25 AM, Dennis wrote: no, not yet. i am still fiddling arround with icecast2. we tried it with someone, who offers radiostreams. perhaps this just works with icecast(2) and shoutcast? ___ 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] Dialing tone when placing a call with portaudio
Hello, when using mod_portaudio for calling somebody I don't hear anything until the other party answers the call. Is it possible to play a dialing tone when the other party is ringing? Best regards René Pankratz ___ 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] Question about wrapping libfreeswitch
I think the api changed a little bit for this. The easiest starting
point would be to just clone switch.c and chop out any of the stuff
you don't need, it's mostly argument handling code in there.
Mike
On Dec 2, 2008, at 7:05 AM, Woody Dickson [EMAIL PROTECTED]
wrote:
Hi,
I am sorry again for sending another email to the group again. I am
working on embedding libfreeswitch to provide better monitoring.
The first thing I attempt to do is to run the sample code provided
in the website:
#include switch.h
int main(int argc, char **argv)
{
switch_core_flag_t flags = SCF_USE_SQL;
int nc=0; /* this is for 'no console' mode, FALSE console is
there, TRUE it isnt */
const char **err = NULL; /* error value for return from
freeswitch initialization */
#define LOGFILE freeswitch.log
static char *lfile = LOGFILE; /* if NULL no logfile is generated */
switch_core_init_and_modload(*lfile,flags,err);
switch_core_runtime_loop(nc);
switch_core_destroy();
return (0); /* per C89 spec */
}
But this code gives me segmentation fault when executing it. This
piece of code is supposed to start up freeswitch and run it is a
loop. Does anyone see what is wrong with it? Does anyone have any
working example that I can refer to?
Thanks,
Woody
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Re: [Freeswitch-users] Problems with Mod_openMRCP
If you can get it to break on linux I will ssh in and fix it for you. If you cannot, i can try to fix it for you over rdp but that won't be very fun. We can think about reinstating mod_lumenvox as well as another windows based asr alternative. I deleted it for the same reason we will probably delete mod_openmrcp because nobody was using it and there was no way to support it because our dev licenses had expired. Lumenvox has offered us some new dev licenses to bring it back but I would need someone to actually want it to work to put in charge of it. We will be clear about what is supported and what is not in the 1.0.2 release scheduled to be released in the near future. On Tue, Dec 2, 2008 at 12:11 AM, David Knell [EMAIL PROTECTED] wrote: Hi Anthony, mod_openmrcp was a contribution to the community by a 3rd party individual. As i have clearly stated in 2 previous emails, the man has decided to discontinue the openmrcp project. So now we are left with the remains of the module and discontinued code. This was not our decision it was his. I absolutely understand this but it's important, from a user point of view, to be able to know which bits of FS are current/supported and which aren't. Some people use it without issue which may mean that the crash you reported is windows specific and I do not have a working lab of any mrcp capbable system to try it against in unix for that matter. I have a list of work to do from here to the moon and back so on an issue like this, unless someone can hand me login credentials to some box and give me a phone number to dial to reporduce the issue, it will be a long time until we can deal with it. It's useful to know that there are people using mod_openmrcp without issue: I did ask here if anyone was a while back, and no-one fessed up. I'll give it a go on a Linux box and report back. And if you'd like a dev/test environment set up, then just tell me which one. And the question arises, should we bother working on it anymore if the lib has been abandoned and we cannot even get any support from it's author which is where the problem most likely lies. I try not to get too annoyed by these remarks about what we *ought to do* because I know people lose sight of how much of the work to support the project is done by a small group of 3 people and not the 2000 people it appears to be from the outside looking in. (I've been answering email for 4 hours now) Those guys who claim to have all that money in an offshore bank account are lying - you don't have to reply to them in future ;-) Seriously, though, I don't think it's too outrageous an idea to document what's supported and were you (for example) to have suggested that I get in touch with the contributors to the various modules, ask them what their view of its status is, condense the answers in to a list and report back, it's something I'd quite happily do. My suggestion is to pool some cash and pay the guy to make mod_unimrcp for FS that we can maintain in tree knowing the development can be supported by the original author. Quite happy to participate in that, too.. the problem is that I've a demo to do like yesterday and the timescale for mod_unimrcp is a bit on the long side for that. I'd rather not have to do it with Asterisk and Lumenvox..! Cheers -- Dave On Mon, Dec 1, 2008 at 12:51 PM, David Knell [EMAIL PROTECTED] wrote: Hi Mike, My experience is that it's somewhat broken - it took two trivial tweaks to get it to work with IBM's ASR and TTS, but there's a more intractable problem to do with memory getting overwritten (I assume that this is something to do with something being freed when it shouldn't be) which causes a segfault on the second or third session after the module being loaded. Without wishing to sound like a stuck record, one thing that you guys really ought to do is to decide what's supported and what isn't, and make this obvious - for example, move unsupported modules to a different place in the tree, don't have them built by default, etc. MRCP is in the specsheet on the Wiki. Otherwise folk like Mark and I spend time installing stuff, go round in circles a bit trying to make it work, and then find out (a) that it doesn't and (b) it's not going to be fixed because it's not supported. Cheers -- Dave I would not say it is totally broken, it is known to work in quite a few places, but we are unlikely to be doing any new fixes in it. Mike On Dec 1, 2008, at 1:19 PM, [EMAIL PROTECTED] wrote: Hi Anthony, Oh! OK. So is this module totally broken. I say this because I can't seem to get it to work at all with the example in that Mod_openmrcp wiki page but I thought it might because I'm not be using the right Cepstral software (freetrial download versus the paided for SDK) or that I'm not using the right port numbers or something else I didn't do. I used TcpView to look at local port associated with my
Re: [Freeswitch-users] Problems with Mod_openMRCP
FreeSWITCH has an enterprise scale SIP UA. Not only can it listen on other ports it can listen and work on as many ip:port combos as you want simultaneously each with it's own specific config. If you have an affinity for port 5060 you can always bring up 2 IP on the same box and give one to each application. You can essentially do whatever you want. It's your box and everything involved is configurable. On Tue, Dec 2, 2008 at 1:00 AM, [EMAIL PROTECTED] wrote: I need to barge in again and add to my last post with this email from Voxeo support. Here is their response to the port binding conflict and it brings up a possible problem if FreeSwitch will be looking for Prophecy at that port? I assumed it would if I set up the extension right but now I don't know and need your assistance with this issue ... as well. Thank you. MESSAGE: Hi Mark, You are correct in that having multiple applications binding to the same port can cause a bundle of problems. You can configure Prophecy to stay away from port 5060, but then the question is whether FreeSwitch will be looking for Prophecy at that port (if its assuming that it's residing on a different box). Port 5060 is the standard for SIP traffic. To get Prophecy off 5060 you will need to edit the config.xml and callrouting.xml files. You will need to search out all instances of 5060 and replace with, perhaps, port 5068. For instance: item name=ListenOnIP10.0.0.0:5068/item item name=ListenOnIP20.0.0.0:5061/item item name=ListenOnIP30.0.0.0:5067/item item name=ListenOnIP40.0.0.0:5063/item item name=ListenOnIP50.0.0.0:5064/item item name=ListenOnIP60.0.0.0:5065/item instead of this... item name=ListenOnIP10.0.0.0:5060/item item name=ListenOnIP20.0.0.0:5061/item item name=ListenOnIP30.0.0.0:5062/item item name=ListenOnIP40.0.0.0:5063/item item name=ListenOnIP50.0.0.0:5064/item item name=ListenOnIP60.0.0.0:5065/item Regards, Jeff Kustermann Voxeo Support -Original Message- From: [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Sent: Mon, 1 Dec 2008 10:40 pm Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP Just to follow up. Moshe Yudkowsky has an article on Routing calls from FreeSwitch to Prophecy: http://www.prophecy2006.com/node/145 My problem is that Freeswitch and Prophecy need to be on the same machine BUT both need to bind to port 5060 so I'm getting errors from one or the other depending who's running first. So can I change what port(s) FS uses and that way avoid this conflict? Maybe, this might let me bridge the call via FreeSwitch to Prophecy similar to what Moshe's article discusses??? -Original Message- From: [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Sent: Mon, 1 Dec 2008 4:44 pm Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP Does bridging a call from FS to Voxeo's Prophecy server require openMRCP? If not then the other issue I might have is a database look up that is part of the dialogue that maybe need as the person response to prompts from the asr. It's possible to run a php script for the database stuff that Prophecy might need or could that happen via Javascript in FS? Then after the dialogue has completed I go from Prophecy back to FS. -Original Message- From: Anthony Minessale [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Sent: Mon, 1 Dec 2008 11:17 am Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP mod_openmrcp was a contribution to the community by a 3rd party individual. As i have clearly stated in 2 previous emails, the man has decided to discontinue the openmrcp project. So now we are left with the remains of the module and discontinued code. This was not our decision it was his. Since the author of openmrcp has stated that he has a new unimrcp we are certainly going to work towards getting mod_unimrcp to replace mod_openmrcp. He had already commented on that previous thread to state he is willing to consider making a new module. Some people use it without issue which may mean that the crash you reported is windows specific and I do not have a working lab of any mrcp capbable system to try it against in unix for that matter. I have a list of work to do from here to the moon and back so on an issue like this, unless someone can hand me login credentials to some box and give me a phone number to dial to reporduce the issue, it will be a long time until we can deal with it. And the question arises, should we bother working on it anymore if the lib has been abandoned and we cannot even get any support from it's author which is where the problem most likely lies. I try not to get too annoyed by these remarks about what we *ought to do* because I know people lose sight of how much of the work to support the project is done by a small group of 3 people and not the 2000 people it appears to be from the
Re: [Freeswitch-users] Listen to a file, while recording?
sorry, problem solved :-) it works very good with icecast2. 2008/12/2 Brian West [EMAIL PROTECTED]: And you have your shoutcast/icecast server set up and functional? /b On Dec 2, 2008, at 9:03 AM, Dennis wrote: i am using the latest svn trunk from today. ___ 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] Console Dialing in Freeswitch
from the source tree of FS please type make current when it completes, retest the call. On Tue, Dec 2, 2008 at 5:07 AM, Baskar [EMAIL PROTECTED] wrote: *Hi, This is the svn version i have installed before a month FreeSWITCH Version 1.0.trunk (10130M) * -- *Warm Regards, N.Baskar* ___ 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:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] 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] Windows is slow?
Have you tried the latest msi build? It's based off svn 10564. Carlos On Sun, Nov 30, 2008 at 11:03 AM, Per Møller [EMAIL PROTECTED] wrote: I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1 on a Windows Vista machine using the precompiled .msi - actually the same machine). Using the default configuration files, and using 2 Snom 360 phones I dialed from extension 1000 to extension 1001. On the Mac, 1001 starts ringing instantly, but under Windows it takes 1-2 seconds before it starts ringing. It seems to be in the dialplan the time is spent. From the time I see this line on the console: [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000-1000 in context default Until the next thing happens it always takes at least 1 full second, but on the Mac it happens instantly. Why is the Windows build this much slower? Is it a known problem? I get the feeling that the majority of the FS community is Unix based, which is fine by me, but I would really like to know just how well supported and stable the Win32 build is and if this is currently a viable way to go, or if I should stick to Linux/BSD/Mac for production use? // Per ___ 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] Support for Junghanns duoBRI
All HFC-based cards supported by bristuffed Zaptel should work. Stefan Am Monday 01 December 2008 schrieb Michael Jerris: The bri support is still in development, basic calls on ptmp bri do appear to work, although I am not sure with what hardware. Mike On Dec 1, 2008, at 10:26 AM, Sergey Kirillov wrote: Greetings, Can somebody tell me, if it is possible to use duoBRI card (http://www.junghanns.net/en/duobri_express_produkt.html) from Junghanns.net together with Freeswitch? I've found that this card has Zaptel drivers, and Freeswitch has mod_openzap. On the other side, I saw somewhere in wiki that Freeswitch does not support BRI at all at the moment. Please confirm or allay my apprehensions. ___ 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 -- Stefan Knoblich Systemadministrator axsentis GmbH Eupener Strasse 74 50933 Köln Tel: 0180 - 506 705 521* Fax: 0180 - 506 705 529* E-Mail: [EMAIL PROTECTED] Web: www.axsentis.de Eingetragen beim AG Köln: HR B 56238 UST-ID: DE244977565 Gesellschafter-Geschäftsführer: Yan Lecomte, Eduard Schlein, Apostolos Varsamis *14ct/min aus dem Festnetz der T-Com | dtms ___ 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] Problems with Mod_openMRCP
Mark and David, I am willing to help some with testing here as well, if you need it. Ping me directly or we can get on the IRC. I am on Mac OS, but have readily available vm's with Debian, etc. I also have Prophecy. I have a general interest in an ASR solution as well. Voxeo is great, but using it as an MRCP proxy seems odd. As a full fledged VXML solution it is great, if you can afford it. But having a good ASR solution is good first step to trying to get something like OpenVXI working as well. That said, seems like a bounty or money to help FS is a better spend anyway. It is a one time cost, not a variable cost. And it goes straight to the guys doing the real work. I built unimrcp last night, it was quite straight forward. In theory, if I weren't old and my C/autoconf skills rather atrophied, it wouldn't seem like it would be that huge a deal to port/fix openmrcp to unimrcp. Finally, Anthony I was looking at the Lumenvox path as well, but got deterred by the licensing hassle. This seems to be a universal ASR issue. I would reason I can find the old module in SVN? Were they going to grant community dev licenses? Again - I am willing to volunteer to do some testing/doc at least. Andy On Dec 2, 2008, at 10:51 AM, Anthony Minessale wrote: If you can get it to break on linux I will ssh in and fix it for you. If you cannot, i can try to fix it for you over rdp but that won't be very fun. We can think about reinstating mod_lumenvox as well as another windows based asr alternative. I deleted it for the same reason we will probably delete mod_openmrcp because nobody was using it and there was no way to support it because our dev licenses had expired. Lumenvox has offered us some new dev licenses to bring it back but I would need someone to actually want it to work to put in charge of it. We will be clear about what is supported and what is not in the 1.0.2 release scheduled to be released in the near future. On Tue, Dec 2, 2008 at 12:11 AM, David Knell [EMAIL PROTECTED] wrote: Hi Anthony, mod_openmrcp was a contribution to the community by a 3rd party individual. As i have clearly stated in 2 previous emails, the man has decided to discontinue the openmrcp project. So now we are left with the remains of the module and discontinued code. This was not our decision it was his. I absolutely understand this but it's important, from a user point of view, to be able to know which bits of FS are current/supported and which aren't. Some people use it without issue which may mean that the crash you reported is windows specific and I do not have a working lab of any mrcp capbable system to try it against in unix for that matter. I have a list of work to do from here to the moon and back so on an issue like this, unless someone can hand me login credentials to some box and give me a phone number to dial to reporduce the issue, it will be a long time until we can deal with it. It's useful to know that there are people using mod_openmrcp without issue: I did ask here if anyone was a while back, and no-one fessed up. I'll give it a go on a Linux box and report back. And if you'd like a dev/test environment set up, then just tell me which one. And the question arises, should we bother working on it anymore if the lib has been abandoned and we cannot even get any support from it's author which is where the problem most likely lies. I try not to get too annoyed by these remarks about what we *ought to do* because I know people lose sight of how much of the work to support the project is done by a small group of 3 people and not the 2000 people it appears to be from the outside looking in. (I've been answering email for 4 hours now) Those guys who claim to have all that money in an offshore bank account are lying - you don't have to reply to them in future ;-) Seriously, though, I don't think it's too outrageous an idea to document what's supported and were you (for example) to have suggested that I get in touch with the contributors to the various modules, ask them what their view of its status is, condense the answers in to a list and report back, it's something I'd quite happily do. My suggestion is to pool some cash and pay the guy to make mod_unimrcp for FS that we can maintain in tree knowing the development can be supported by the original author. Quite happy to participate in that, too.. the problem is that I've a demo to do like yesterday and the timescale for mod_unimrcp is a bit on the long side for that. I'd rather not have to do it with Asterisk and Lumenvox..! Cheers -- Dave On Mon, Dec 1, 2008 at 12:51 PM, David Knell [EMAIL PROTECTED] wrote: Hi Mike, My experience is that it's somewhat broken - it took two trivial tweaks to get it to work with IBM's ASR and TTS, but there's a more intractable problem to do with memory getting overwritten (I assume that
Re: [Freeswitch-users] TLS receiving calls
Naturally, either way is stupid. The whole idea of putting the transport in a uri param is equally stupid to using 2 different protocol names but since SIP is the descendant of http it they decided to stick with the stupidity of http/https and have sip/sips which is almost as if it was designed to break all software trying to keep up with url syntax. If they are going to insist on using text params you'd think something like transport=foo;security=tls would be even *more* flexable in case alternate methods to encrypt crop up. This is, of course, the first step into a lengthy 12 hour discussion on how stupid SIP and url/text based protocols are. I dare someone to crank up the pcap on a box doing SIP presence for 20 phones and read the 1200 byte messages with all kinds of hyeroglyphic url syntax and embedded xml payloads and write up a paper on how much sense it makes to have it be readable. PS supposedly sofia can support sctp, someone should try it. On Mon, Dec 1, 2008 at 9:43 PM, Kristian Kielhofner [EMAIL PROTECTED] wrote: On 12/1/08, Thomas Troy [EMAIL PROTECTED] wrote: ..snip.. Out of interest do you have any links to anywhere this is discussed in terms of general sip implementations? Uh oh, here we go again... http://www.iana.org/assignments/sip-parameters http://tools.ietf.org/html/rfc3969 https://lists.cs.columbia.edu/pipermail/sip-implementors/2005-August/010047.html Implementation wise, most devices tend to use transport=tls: SIPFoundry - From what I've seen Snom SERs Asterisk (If you are using TLS) Cisco - I *believe* you can use either a SIPS URI or the transport=tls parameter for various SIP targets As the RFC (basically) states (RFC3261, section 12.1.x), transport=tls was deprecated in RFC 3261 because you should also be able to do TLS over SCTP (RFC3436), which makes transport=tls a bit ambiguous. sips:[EMAIL PROTECTED];transport=tcp or sips:[EMAIL PROTECTED];transport=sctp is a bit more flexible. I don't know if I've ever seen anything default to SIPS URIs. I also don't think I've ever specifically tried using them. However, my experience with TLS is admittedly somewhat limited so this shouldn't be taken as gospel. As you can see from the discussions on sip-implementors, this gets interesting when different devices are traversing a proxy using different URI schemes... However, I suspect this won't become an issue until most SIP implementations support SCTP. That should be exciting! ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] 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] Problems with Mod_openMRCP
from build root: svn co -r8809 http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/asr_tts/mod_lumenvoxsrc/mod/asr_tts/mod_lumenvox They did seem to express an interest in granting some dev licenses when they realized we took the code out of tree but I have not actually dealt with the issue yet because I have been overwhelmed. I don't know if this code works anymore with the latest revision of the api but there it is. On Tue, Dec 2, 2008 at 10:24 AM, Andrew Gilbert [EMAIL PROTECTED]wrote: Mark and David, I am willing to help some with testing here as well, if you need it. Ping me directly or we can get on the IRC. I am on Mac OS, but have readily available vm's with Debian, etc. I also have Prophecy. I have a general interest in an ASR solution as well. Voxeo is great, but using it as an MRCP proxy seems odd. As a full fledged VXML solution it is great, if you can afford it. But having a good ASR solution is good first step to trying to get something like OpenVXI working as well. That said, seems like a bounty or money to help FS is a better spend anyway. It is a one time cost, not a variable cost. And it goes straight to the guys doing the real work. I built unimrcp last night, it was quite straight forward. In theory, if I weren't old and my C/autoconf skills rather atrophied, it wouldn't seem like it would be that huge a deal to port/fix openmrcp to unimrcp. Finally, Anthony I was looking at the Lumenvox path as well, but got deterred by the licensing hassle. This seems to be a universal ASR issue. I would reason I can find the old module in SVN? Were they going to grant community dev licenses? Again - I am willing to volunteer to do some testing/doc at least. Andy On Dec 2, 2008, at 10:51 AM, Anthony Minessale wrote: If you can get it to break on linux I will ssh in and fix it for you. If you cannot, i can try to fix it for you over rdp but that won't be very fun. We can think about reinstating mod_lumenvox as well as another windows based asr alternative. I deleted it for the same reason we will probably delete mod_openmrcp because nobody was using it and there was no way to support it because our dev licenses had expired. Lumenvox has offered us some new dev licenses to bring it back but I would need someone to actually want it to work to put in charge of it. We will be clear about what is supported and what is not in the 1.0.2 release scheduled to be released in the near future. On Tue, Dec 2, 2008 at 12:11 AM, David Knell [EMAIL PROTECTED] wrote: Hi Anthony, mod_openmrcp was a contribution to the community by a 3rd party individual. As i have clearly stated in 2 previous emails, the man has decided to discontinue the openmrcp project. So now we are left with the remains of the module and discontinued code. This was not our decision it was his. I absolutely understand this but it's important, from a user point of view, to be able to know which bits of FS are current/supported and which aren't. Some people use it without issue which may mean that the crash you reported is windows specific and I do not have a working lab of any mrcp capbable system to try it against in unix for that matter. I have a list of work to do from here to the moon and back so on an issue like this, unless someone can hand me login credentials to some box and give me a phone number to dial to reporduce the issue, it will be a long time until we can deal with it. It's useful to know that there are people using mod_openmrcp without issue: I did ask here if anyone was a while back, and no-one fessed up. I'll give it a go on a Linux box and report back. And if you'd like a dev/test environment set up, then just tell me which one. And the question arises, should we bother working on it anymore if the lib has been abandoned and we cannot even get any support from it's author which is where the problem most likely lies. I try not to get too annoyed by these remarks about what we *ought to do* because I know people lose sight of how much of the work to support the project is done by a small group of 3 people and not the 2000 people it appears to be from the outside looking in. (I've been answering email for 4 hours now) Those guys who claim to have all that money in an offshore bank account are lying - you don't have to reply to them in future ;-) Seriously, though, I don't think it's too outrageous an idea to document what's supported and were you (for example) to have suggested that I get in touch with the contributors to the various modules, ask them what their view of its status is, condense the answers in to a list and report back, it's something I'd quite happily do. My suggestion is to pool some cash and pay the guy to make mod_unimrcp for FS that we can maintain in tree knowing the development can be supported by the original author. Quite happy to participate in that, too.. the problem is that I've a demo
[Freeswitch-users] Fax and Freeswitch: What is the status, what works?
hi, because we do not get tired of testing and playing a lot with the beloved fs, we now arrived at the fax feature :-) i am not sure if the docs are up to date or if there was a lot of development in the meantime. therefore i would like to ask, what is possible and what will come in the near future. we are using fs, socket outbound and php and would like to make something like fax to mail as an additional service. is t38 supported? can i pass incoming faxes over the same socket as calls? can i convert faxes into pdf? is fax over sip reliable (as far as i have heard, under asterisk fax is nothing one should use)? and so on, and so on i would be very happy to hear some user experiences with fs and fax. if it seems, that we can use fax with over socket outbound, we will do hardcore testing ;-) thanks, dennis ___ 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] Problems with Mod_openMRCP
They contacted us shortly thereafter and asked if we want to have them sell you the license for 50 bucks. hmm, i wonder why i deleted the module. I will tell them that if they give you a developer license you will work on getting it back into trunk. On Tue, Dec 2, 2008 at 11:27 AM, Andrew Gilbert [EMAIL PROTECTED]wrote: Ok I have a ping in with Lumenvox about dev licensing, and pulled the mod. Not sure where this will go, but will take a peek at things. Balancing the effort against something like getting unimcrp going and/or openmrcp tested and stable. Thanks. Andy On Dec 2, 2008, at 11:43 AM, Anthony Minessale wrote: from build root: svn co -r8809 http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/asr_tts/mod_lumenvoxsrc/mod/asr_tts/mod_lumenvox They did seem to express an interest in granting some dev licenses when they realized we took the code out of tree but I have not actually dealt with the issue yet because I have been overwhelmed. I don't know if this code works anymore with the latest revision of the api but there it is. On Tue, Dec 2, 2008 at 10:24 AM, Andrew Gilbert [EMAIL PROTECTED]wrote: Mark and David, I am willing to help some with testing here as well, if you need it. Ping me directly or we can get on the IRC. I am on Mac OS, but have readily available vm's with Debian, etc. I also have Prophecy. I have a general interest in an ASR solution as well. Voxeo is great, but using it as an MRCP proxy seems odd. As a full fledged VXML solution it is great, if you can afford it. But having a good ASR solution is good first step to trying to get something like OpenVXI working as well. That said, seems like a bounty or money to help FS is a better spend anyway. It is a one time cost, not a variable cost. And it goes straight to the guys doing the real work. I built unimrcp last night, it was quite straight forward. In theory, if I weren't old and my C/autoconf skills rather atrophied, it wouldn't seem like it would be that huge a deal to port/fix openmrcp to unimrcp. Finally, Anthony I was looking at the Lumenvox path as well, but got deterred by the licensing hassle. This seems to be a universal ASR issue. I would reason I can find the old module in SVN? Were they going to grant community dev licenses? Again - I am willing to volunteer to do some testing/doc at least. Andy On Dec 2, 2008, at 10:51 AM, Anthony Minessale wrote: If you can get it to break on linux I will ssh in and fix it for you. If you cannot, i can try to fix it for you over rdp but that won't be very fun. We can think about reinstating mod_lumenvox as well as another windows based asr alternative. I deleted it for the same reason we will probably delete mod_openmrcp because nobody was using it and there was no way to support it because our dev licenses had expired. Lumenvox has offered us some new dev licenses to bring it back but I would need someone to actually want it to work to put in charge of it. We will be clear about what is supported and what is not in the 1.0.2 release scheduled to be released in the near future. On Tue, Dec 2, 2008 at 12:11 AM, David Knell [EMAIL PROTECTED] wrote: Hi Anthony, mod_openmrcp was a contribution to the community by a 3rd party individual. As i have clearly stated in 2 previous emails, the man has decided to discontinue the openmrcp project. So now we are left with the remains of the module and discontinued code. This was not our decision it was his. I absolutely understand this but it's important, from a user point of view, to be able to know which bits of FS are current/supported and which aren't. Some people use it without issue which may mean that the crash you reported is windows specific and I do not have a working lab of any mrcp capbable system to try it against in unix for that matter. I have a list of work to do from here to the moon and back so on an issue like this, unless someone can hand me login credentials to some box and give me a phone number to dial to reporduce the issue, it will be a long time until we can deal with it. It's useful to know that there are people using mod_openmrcp without issue: I did ask here if anyone was a while back, and no-one fessed up. I'll give it a go on a Linux box and report back. And if you'd like a dev/test environment set up, then just tell me which one. And the question arises, should we bother working on it anymore if the lib has been abandoned and we cannot even get any support from it's author which is where the problem most likely lies. I try not to get too annoyed by these remarks about what we *ought to do* because I know people lose sight of how much of the work to support the project is done by a small group of 3 people and not the 2000 people it appears to be from the outside looking in. (I've been answering email for 4 hours now) Those guys who claim to have all that money in an offshore
Re: [Freeswitch-users] Support for Junghanns duoBRI
Cool. Thanks for the answer. All HFC-based cards supported by bristuffed Zaptel should work. Stefan ___ 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] TLS receiving calls
On 12/2/08, Anthony Minessale [EMAIL PROTECTED] wrote: Naturally, either way is stupid. Word. The whole idea of putting the transport in a uri param is equally stupid to using 2 different protocol names but since SIP is the descendant of http it they decided to stick with the stupidity of http/https and have sip/sips which is almost as if it was designed to break all software trying to keep up with url syntax. Too late now. If they are going to insist on using text params you'd think something like transport=foo;security=tls would be even *more* flexable in case alternate methods to encrypt crop up. I can agree with you here... URI parameters in SIP have come to be the catch all for random junk that doesn't seem to fit anywhere else. Note that random junk includes everything from transport, to number portability, to CICs, to ISUP-OLI and on. Even in my world setting up proxies, UAs, etc to parse out the various crap people put in SIP URI params is a hassle. A big one. What a mess!!! This is, of course, the first step into a lengthy 12 hour discussion on how stupid SIP and url/text based protocols are. I like them but I'm weird. I dare someone to crank up the pcap on a box doing SIP presence for 20 phones and read the 1200 byte messages with all kinds of hyeroglyphic url syntax and embedded xml payloads and write up a paper on how much sense it makes to have it be readable. I do it all the time. I think it's quite usable. ngrep provides a small enough binary and the ability to match on text. Certainly easier to use, especially on embedded systems without the luxury of dedicated protocol decoders. With a simple ngrep binary I can debug any text based protocol I understand. Of course, turn on TLS and see how useful *any* of these tools are... The core SIP spec and authors can't be blamed for the various junk people have been putting in SIP bodies. If what's going on in the real world is any indication, that ship sailed long ago. At this point as long as implementations can at least handle multi-part sensibly and everyone specifies the correct MIME type I don't really care. Even nastier examples abound - embedded, encapsulated ISUP! How about GTD? What about Linksys phones using SIP INFO to serve directories? Man I could go on and on... I'm not going to write a paper about it but I don't think it's that bad. Maybe I'm not just weird; maybe I'm a masochist! :) PS supposedly sofia can support sctp, someone should try it. That would be cool. For anyone wanting to try, various SERs support SCTP. Cisco gateways do too. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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] TLS receiving calls
On 12/2/08, Anthony Minessale [EMAIL PROTECTED] wrote: We'll schedule a round table with the topic SIP OMFG STFU At the next ClueCon aug 4th-6th 2009 to stir things up a bit =D Heh. I've been trying to make it back these last couple of years. I just might make it in '09! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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] Wrong # in voicemail
Note: while reading up on regex, I see that the ',' in ([0,1]) is superflous, has been removed. regex is now: ^([01]?)(8162565804)$ Didn't fix the problem but I'm a perfectionist, had to be changed. :D -- View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20799146.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] Windows is slow?
I checked out the trunk version, and it's still slow. However I found one improvement - it does not crash on shutdown anymore. Could anymore give me some pointers on how to try to debug this on the Windows platform? // Per Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Carlos Talbot Sendt: 2. december 2008 17:13 Til: freeswitch-users@lists.freeswitch.org Emne: Re: [Freeswitch-users] Windows is slow? Have you tried the latest msi build? It's based off svn 10564. Carlos On Sun, Nov 30, 2008 at 11:03 AM, Per Møller [EMAIL PROTECTED] wrote: I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1 on a Windows Vista machine using the precompiled .msi - actually the same machine). Using the default configuration files, and using 2 Snom 360 phones I dialed from extension 1000 to extension 1001. On the Mac, 1001 starts ringing instantly, but under Windows it takes 1-2 seconds before it starts ringing. It seems to be in the dialplan the time is spent. From the time I see this line on the console: [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000-1000 in context default Until the next thing happens it always takes at least 1 full second, but on the Mac it happens instantly. Why is the Windows build this much slower? Is it a known problem? I get the feeling that the majority of the FS community is Unix based, which is fine by me, but I would really like to know just how well supported and stable the Win32 build is and if this is currently a viable way to go, or if I should stick to Linux/BSD/Mac for production use? // Per ___ 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] Windows is slow?
Can you do a console loglevel debug, then send all the output around that time? Apart from that, the quickest way might just to attach a debugger, then break all when it pauses and see where the threads are :). -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Per Møller Sent: Tuesday, December 02, 2008 12:32 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Windows is slow? I checked out the trunk version, and it's still slow. However I found one improvement - it does not crash on shutdown anymore. Could anymore give me some pointers on how to try to debug this on the Windows platform? // Per Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Carlos Talbot Sendt: 2. december 2008 17:13 Til: freeswitch-users@lists.freeswitch.org Emne: Re: [Freeswitch-users] Windows is slow? Have you tried the latest msi build? It's based off svn 10564. Carlos On Sun, Nov 30, 2008 at 11:03 AM, Per Møller [EMAIL PROTECTED] wrote: I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1 on a Windows Vista machine using the precompiled .msi - actually the same machine). Using the default configuration files, and using 2 Snom 360 phones I dialed from extension 1000 to extension 1001. On the Mac, 1001 starts ringing instantly, but under Windows it takes 1-2 seconds before it starts ringing. It seems to be in the dialplan the time is spent. From the time I see this line on the console: [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000-1000 in context default Until the next thing happens it always takes at least 1 full second, but on the Mac it happens instantly. Why is the Windows build this much slower? Is it a known problem? I get the feeling that the majority of the FS community is Unix based, which is fine by me, but I would really like to know just how well supported and stable the Win32 build is and if this is currently a viable way to go, or if I should stick to Linux/BSD/Mac for production use? // Per ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Javascript ODBC on Windows
Hi all, Is it possible to use mod_spidermonkey_odbc with a Windows installation of FreeSWITCH at the moment? If so does anyone have any pointers? I get: 2008-12-02 14:23:57 [DEBUG] switch_odbc.c:145 switch_odbc_handle_connect() Connecting ivr_test 2008-12-02 14:23:57 [ERR] switch_odbc.c:160 switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [Microsoft][ODBC Driver Manager] Data source name not found and no default driver specified when I try. Thanks in advance, Joe Bain ___ 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] Windows is slow?
is it stun timeout ? do you have one of the ip set to stun:foo ? On Tue, Dec 2, 2008 at 1:33 PM, Michael Giagnocavo [EMAIL PROTECTED]wrote: Can you do a console loglevel debug, then send all the output around that time? Apart from that, the quickest way might just to attach a debugger, then break all when it pauses and see where the threads are :). -Michael -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Per Møller Sent: Tuesday, December 02, 2008 12:32 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Windows is slow? I checked out the trunk version, and it's still slow. However I found one improvement - it does not crash on shutdown anymore. Could anymore give me some pointers on how to try to debug this on the Windows platform? // Per Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Carlos Talbot Sendt: 2. december 2008 17:13 Til: freeswitch-users@lists.freeswitch.org Emne: Re: [Freeswitch-users] Windows is slow? Have you tried the latest msi build? It's based off svn 10564. Carlos On Sun, Nov 30, 2008 at 11:03 AM, Per Møller [EMAIL PROTECTED] wrote: I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1 on a Windows Vista machine using the precompiled .msi - actually the same machine). Using the default configuration files, and using 2 Snom 360 phones I dialed from extension 1000 to extension 1001. On the Mac, 1001 starts ringing instantly, but under Windows it takes 1-2 seconds before it starts ringing. It seems to be in the dialplan the time is spent. From the time I see this line on the console: [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000-1000 in context default Until the next thing happens it always takes at least 1 full second, but on the Mac it happens instantly. Why is the Windows build this much slower? Is it a known problem? I get the feeling that the majority of the FS community is Unix based, which is fine by me, but I would really like to know just how well supported and stable the Win32 build is and if this is currently a viable way to go, or if I should stick to Linux/BSD/Mac for production use? // Per ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] 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] Fax and Freeswitch: What is the status, what works?
Right now this page is up-to-date with the latest info: http://wiki.freeswitch.org/wiki/Mod_fax T.38 is not (yet) supported. -MC On Tue, Dec 2, 2008 at 9:40 AM, Dennis [EMAIL PROTECTED] wrote: hi, because we do not get tired of testing and playing a lot with the beloved fs, we now arrived at the fax feature :-) i am not sure if the docs are up to date or if there was a lot of development in the meantime. therefore i would like to ask, what is possible and what will come in the near future. we are using fs, socket outbound and php and would like to make something like fax to mail as an additional service. is t38 supported? can i pass incoming faxes over the same socket as calls? can i convert faxes into pdf? is fax over sip reliable (as far as i have heard, under asterisk fax is nothing one should use)? and so on, and so on i would be very happy to hear some user experiences with fs and fax. if it seems, that we can use fax with over socket outbound, we will do hardcore testing ;-) thanks, dennis ___ 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] Fax and Freeswitch: What is the status, what works?
T.38 passthrough IS supported, T.38 endpoint and gateway are not yet supported. Mike On Dec 2, 2008, at 4:28 PM, Kristian Kielhofner wrote: On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins [EMAIL PROTECTED] wrote: Right now this page is up-to-date with the latest info: http://wiki.freeswitch.org/wiki/Mod_fax T.38 is not (yet) supported. -MC Can you (or someone) elaborate on this? Maybe the answer really is no, but what about support for UDPTL, pass through, etc? It looks like Sofia should be good to go... ___ 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] Javascript ODBC on Windows
Yes, it should work fine. As the error message says it didn't find the data source name you specified. You need to setup your odbc data source on the system Mike On Dec 2, 2008, at 9:29 AM, Joe Bain wrote: Hi all, Is it possible to use mod_spidermonkey_odbc with a Windows installation of FreeSWITCH at the moment? If so does anyone have any pointers? I get: 2008-12-02 14:23:57 [DEBUG] switch_odbc.c:145 switch_odbc_handle_connect() Connecting ivr_test 2008-12-02 14:23:57 [ERR] switch_odbc.c:160 switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [Microsoft] [ODBC Driver Manager] Data source name not found and no default driver specified when I try. Thanks in advance, Joe Bain ___ 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] Wrong # in voicemail
After you set ${dialed_user}=$2 try using ${dialed_user} everywhere
instead of $2 just to test.
/b
On Dec 2, 2008, at 1:29 PM, ccav wrote:
Note: while reading up on regex, I see that the ',' in ([0,1]) is
superflous,
has been removed. regex is now:
^([01]?)(8162565804)$
Didn't fix the problem but I'm a perfectionist, had to be changed. :D
--
View this message in context:
http://www.nabble.com/Wrong---in-voicemail-tp20791453p20799146.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
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Re: [Freeswitch-users] Wrong # in voicemail
Made the change, no joy. Do I need to set sip_req_user to the updated DID? Also, I misspoke in my first post, apparently the bridge is NOT going through either. Is there some var/param I can set with $2 so I can see it in the info? -- View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20803931.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] Fax and Freeswitch: What is the status, what works?
On Tue, Dec 2, 2008 at 1:28 PM, Kristian Kielhofner [EMAIL PROTECTED] wrote: On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins [EMAIL PROTECTED] wrote: Right now this page is up-to-date with the latest info: http://wiki.freeswitch.org/wiki/Mod_fax T.38 is not (yet) supported. -MC Can you (or someone) elaborate on this? Maybe the answer really is no, but what about support for UDPTL, pass through, etc? Excellent questions! I will research and report back to the list... -MC It looks like Sofia should be good to go... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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
Re: [Freeswitch-users] Fax and Freeswitch: What is the status, what works?
Kristian, Are you on the IRC channel by any chance? -MC (IRC: mercutioviz) On Tue, Dec 2, 2008 at 1:28 PM, Kristian Kielhofner [EMAIL PROTECTED] wrote: On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins [EMAIL PROTECTED] wrote: Right now this page is up-to-date with the latest info: http://wiki.freeswitch.org/wiki/Mod_fax T.38 is not (yet) supported. -MC Can you (or someone) elaborate on this? Maybe the answer really is no, but what about support for UDPTL, pass through, etc? It looks like Sofia should be good to go... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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
[Freeswitch-users] Bridging from Event Socket API
Hi Folks, so far i could understand how to bridge calls with Javascript. I'm trying to do the same with Java via the Socket Interface. My first trials weren't successful. maybe you can help me understand what is goin on. What i want to do is to bridge an existing leg (Unique-ID is known) to a party that wasn't yet dialed (Unique-ID unknown). With javascript it is something like: session.bridge(sofia/internal/1002); How do i do this using the event socket interface? what application/command would i use with which arguments? One way i tried to do this is to orginate a call to 'sofia/internal/1002' and bridge the two existing legs using uuid_bridge. Unfortunately, it wasn't successful. The only message i had on the FS console is: 2008-12-02 16:57:34 [DEBUG] switch_core_session.c:693 switch_core_session_queue_private_event() Send signal sofia/internal/[EMAIL PROTECTED] [BREAK] Any idea what i'm missing? Thanks, Klaus. -- Pt! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger ___ 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] Bridging from Event Socket API
You probably have several options depending upon your needs. Could you elaborate a bit on what the big picture is? Also, what exactly were you doing when you established the second call leg? Did the second call let get created and a valid uuid assigned, etc.? Just checking. Let us know, MC On Tue, Dec 2, 2008 at 7:15 PM, Klaus Teller [EMAIL PROTECTED] wrote: Hi Folks, so far i could understand how to bridge calls with Javascript. I'm trying to do the same with Java via the Socket Interface. My first trials weren't successful. maybe you can help me understand what is goin on. What i want to do is to bridge an existing leg (Unique-ID is known) to a party that wasn't yet dialed (Unique-ID unknown). With javascript it is something like: session.bridge(sofia/internal/1002); How do i do this using the event socket interface? what application/command would i use with which arguments? One way i tried to do this is to orginate a call to 'sofia/internal/1002' and bridge the two existing legs using uuid_bridge. Unfortunately, it wasn't successful. The only message i had on the FS console is: 2008-12-02 16:57:34 [DEBUG] switch_core_session.c:693 switch_core_session_queue_private_event() Send signal sofia/internal/ [EMAIL PROTECTED] [BREAK] Any idea what i'm missing? Thanks, Klaus. -- Pt! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger ___ 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
