[Freeswitch-users] Adding headers to INFO messages for Advice of Charge on SNOM
Hi, I have tried maintaining charging information on a SNOM 300's display using 'display' - but found that the phone has some timer, whereby every 60 seconds it wipes out whatever happens to be on the display at that time and replaces is with the dialled number. So not a viable option as it impacts usability. Really annoying when the display was just updated with valuable information for the user and a split second later it gets replaced. [If somebody knows how to disable this behaviour - please do tell...] I see that SNOM supports a number of features for Advice of Charge. From their Wiki: http://wiki.snom.com/Advice_of_charge_%28AOC%29_in_SIP Example of an SIP-Info Message: - INFO sip:b...@snom.com SIP/2.0 From: bil...@snom.com;tag=5354n3 To: u...@snom.com;tag=33rfh3 CSeq: 23423 INFO Call-ID: 3452tw43dt354dm03 AOC: charging;state=active; charging-info=currency; currency=EUR; amount=2000; multiplier=0.001 Content-Length: 0 - So the question - Is there some method available today to add these additional 'new' headers to an INFO message I can send out to these phones? If not, I guess it's a matter of looking at enhancing the case SWITCH_MESSAGE_INDICATE_DISPLAY section in mod_sofia.c ? Best 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] Clarification about channel variables please.
both types of variables are mutable On Sun, Nov 22, 2009 at 2:25 PM, Lon Baker l...@kickasspixels.com wrote: Are either global or regular channel variable mutable during a call? Or can they only be set before and after? Any clarification would help, since the existing wiki doesn't make it clear. Lon ___ 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] Callback to the user in ESL
Hi, I'm using perl ESL to control the call in freeswitch. I'm having the following scenario, but not able to get it right. Dialplan: extension name=outbound_soc condition field=destination_number expression=^9097$ action application=set data=continue_on_fail=true/ action application=socket data=192.168.1.222:8447 async full/ /condition /extension 1. User A calls to an extention (1000). 2. My ESL program will be running, and it answers the call. 3. Then the program will get a number from the user. 4. It will hangup the call. 5. The program has to call to the number that was given by the user. In the above scenario, I was able to do until the 4th step. After hangup the call, if I say originate it is not working. Any ideas on how to do this in ESL. ___ 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] Execute on Answer with JavaScript
Hi, How can we send the answer to the caller only when the callee answers, in JavaScript?? Many thanks. -- View this message in context: http://old.nabble.com/Execute-on-Answer-with-JavaScript-tp26476532p26476532.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] Using odbc in FS core
Title: Re[2]: [Freeswitch-users] Using odbc in FS core Hello Anthony, Is clear, thanks, I'll test and will let you know. Should I add 'core-db-dsn' parameter description to Wiki? Maybe we need to add this parameter also to sample conf files? Saturday, November 21, 2009 6:14:59 PM, you wrote: we had the code slightly out of order, you should update to latest trunk for the right version. The test of 2 deletes is to see if your odbc driver will fail when trying to execute 2 statements at once so I can properly fail over to sqlite because transactions are mandatory for a database running core in odbc. On Sat, Nov 21, 2009 at 6:02 AM, Mike Tkachuk m...@yes.net.ua wrote: Hello, Looks like the issue is not in multi statements in one request. Manually creating DB schema helped and everything started up. I will continue testing Also in code I see such construction: switch_cache_db_execute_sql(dbh, "begin;delete from channels where hostname='';delete from channels where hostname='';commit;", err); Anyone can explain why to do such delete twice and in transaction? Thanks. Saturday, November 21, 2009 1:41:06 PM, you wrote: MT Hello Folks, MT I'm interesting in completely moving away from sqlite and use MT postgresql everywhere including core ( switch_core.c ) MT All other applications can use odbc without issues (sofia, limit, MT fifo etc), but as I see in core only sqlite3 supported. MT I correctly set 'core-db-dsn' parameter, but looks like the problem MT that latest psqlodbc_08_04_0100 don't support multiple statements in MT one request that is often used in switch_core_sqldb.c: sql = switch_mprintf( "update channels set uuid='%q' where uuid='%q' and hostname='%q';" "update calls set caller_uuid='%q' where caller_uuid='%q' and hostname='%q';" "update calls set callee_uuid='%q' where callee_uuid='%q' and hostname='%q'", switch_event_get_header_nil(event, "unique-id"), ... SKIP ... MT So, does anyone have any clue how to us postgresql in the FS core? MT Thanks. MT -- MT Mike Tkachuk -- Mike Tkachuk ___ 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 FreeSWITCHhttp://www.freeswitch.org/ ClueConhttp://www.cluecon.com/ Twitter:http://twitter.com/FreeSWITCH_wire 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 -- Mike Tkachuk ___ 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] [This is a repost. I'm not sure if my message was delivered.] How to pick up someone's phone remotely.
Hello again.
This is a repost. I'm having difficulties communicating with this list
(I'm getting reports from the list saying something about excessive
bounces...), so I'm not sure anybody got this message.
I'm trying to mimic behavior of my analogue PBX with FS. I want to be
able to answer any incomming/transfered (from IVR or a person) call
remotely, and to cancel the possibility of intercepting this call
afterwards.
Greetings
Peter
-- Original message --
My problems evolve, because I didn't know all these functions in FS are
so much dependent on each other. But I'm learning fast...
The scenario I written about before appears to be too much simplified
version of what I need to achieve.
In fact, below scenario and solution works OK only one time - when
someone calls and there's no person on the called extension, and someone
manually answers that phone on other extension. Then any other person
can't intercept this call. Thats is correct and needed behavior. But if
the same person who answered the phone transfers this call - everything
goes back to normal and below solution does not work because the call
has been answered already and execute_on_answer does not execute ever
again during this call/channel. The same happens if there's IVR on the
external extension answering calls and then forwarding to extensions -
everyone can intercept last call even if it's already answered because
IVR answers all call on start (and execute_on_answer doesn't get executed).
So I think I need similar solution but working everywhere: on calls and
transfers. Is there some variable or some other thing that I could set
to block and unblock intercept when needed to get wanted behavior.
Any hints?
Greetings
Peter
Piotr Zurek pisze:
Thank You for such an elegant and simple solution that I have not
thought about.
With an exception that I'm using FS 1.0.4 right now and it appears
that something changed in time and following line should use hash
instead of db (when using default 1.0.4 FS config):
action application=set data=execute_on_answer=hash
delete/${domain_name}-last_dial/${called_party_callgroup}/${uuid}/.
After a few hours of experimenting everything works as planned.
Thank You very much.
Peter
Ognjen Seslija pisze:
Add the following:
action application=set data=execute_on_answer=db
delete/${domain_name}-last_dial/${called_party_callgroup}/${uuid}/.
after
action application=db
data=insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}/
in local extensions default example, or change it globally previously
than this extension. You can join us on IRC if you can any more
questions (sekil).
Regards,
Ognjen
On Tue, Nov 10, 2009 at 4:01 PM, Piotr Żurek
piotr_zu...@biprotech.com mailto:piotr_zu...@biprotech.com wrote:
Hello.
Thank You developers for Freeswitch.
I have installed it lately and it's working quite nicely, but I
have one problem:
I need to mimic behavior of my current analogue PBX installation
using Freeswitch.
This is the scenario:
In the office with a few desks (extensions 1000-1010) and only
one person behind one of desks (whatever extension - in example
1000).
1. There's incoming call on _one_ of extensions 1001-1010
2. The person on extension 1000 wants to answer this call on his
phone so dials #37 and this call is redirected to his phone.
That's how it works on my office on analogue PBX system. Anyone
can answer a call from any other phone as long as it hasn't been
answered already.
I tried to use the intercept action (with global example in
default config) but it's not what I need because it intercepts
the call even if it's already answered. I need to intercept all
but only unanswered calls. I tried to use Redirect but it does
not work on other's extensions call's (or does it?).
Please help.
Peter Żurek
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[Freeswitch-users] Question about rtp-timeout-sec variable
Hello, I have 2 instances of FS: one controlled by my application (making calls with TCP commands, recording sessions, listening to events etc) and one acting as a remote gateway to which all users register. When I leave the default values of rtp-timeout-sec and brutally kill x-lite during conversation, the 'hangup' event with 'media_timeout' cause is obviously sent after the default 5 minutes (and until then, the other leg is still connected to a 'dead' channel). The question is: which FS instance is responsible for terminating the connection after timeout? Only the 'remote' FS instance config seems to work. I thought that the shortest configured value should cause the timeout, but it's not the case. Am I missing something, or is this the correct behavior? Regards, Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Question-about-rtp-timeout-sec-variable-tp4050650p4050650.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] Using odbc in FS core
Title: Re[2]: [Freeswitch-users] Using odbc in FS core Hello, Database schema creation is OK now. Cheers. Monday, November 23, 2009 1:45:28 PM, you wrote: Hello Anthony, Is clear, thanks, I'll test and will let you know. Should I add 'core-db-dsn' parameter description to Wiki? Maybe we need to add this parameter also to sample conf files? Saturday, November 21, 2009 6:14:59 PM, you wrote: we had the code slightly out of order, you should update to latest trunk for the right version. The test of 2 deletes is to see if your odbc driver will fail when trying to execute 2 statements at once so I can properly fail over to sqlite because transactions are mandatory for a database running core in odbc. On Sat, Nov 21, 2009 at 6:02 AM, Mike Tkachuk m...@yes.net.ua wrote: Hello, Looks like the issue is not in multi statements in one request. Manually creating DB schema helped and everything started up. I will continue testing Also in code I see such construction: switch_cache_db_execute_sql(dbh, "begin;delete from channels where hostname='';delete from channels where hostname='';commit;", err); Anyone can explain why to do such delete twice and in transaction? Thanks. Saturday, November 21, 2009 1:41:06 PM, you wrote: MT Hello Folks, MT I'm interesting in completely moving away from sqlite and use MT postgresql everywhere including core ( switch_core.c ) MT All other applications can use odbc without issues (sofia, limit, MT fifo etc), but as I see in core only sqlite3 supported. MT I correctly set 'core-db-dsn' parameter, but looks like the problem MT that latest psqlodbc_08_04_0100 don't support multiple statements in MT one request that is often used in switch_core_sqldb.c: sql = switch_mprintf( "update channels set uuid='%q' where uuid='%q' and hostname='%q';" "update calls set caller_uuid='%q' where caller_uuid='%q' and hostname='%q';" "update calls set callee_uuid='%q' where callee_uuid='%q' and hostname='%q'", switch_event_get_header_nil(event, "unique-id"), ... SKIP ... MT So, does anyone have any clue how to us postgresql in the FS core? MT Thanks. MT -- MT Mike Tkachuk -- Mike Tkachuk ___ 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 FreeSWITCHhttp://www.freeswitch.org/ ClueConhttp://www.cluecon.com/ Twitter:http://twitter.com/FreeSWITCH_wire 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 -- Mike Tkachuk -- Mike Tkachuk ___ 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] SIP Digest nonce (stale=true)
Hi Anthony, I'm having an issue with a gateway after the nonce-ttl expires we are sending stale=true, the cpe some how only likes stale=true without the . I see on rev 15441 http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/endpoints/mod_sofia/sofia_reg.c?r=15441#l687 you made a change and marked it out. So my question is who is correct on this is it the CPE or are we sticking with the quoted (true). Thanks Nameer Kazzaz ___ 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] [This is a repost. I'm not sure if my message was delivered.] How to pick up someone's phone remotely.
Piotr Żurek piotr_zu...@biprotech.com said: This is a repost. I'm having difficulties communicating with this list=20 (I'm getting reports from the list saying something about excessive=20 bounces...), so I'm not sure anybody got this message. 1. http://lists.freeswitch.org/pipermail/freeswitch-users/ 2. Click the link for Thread for November 2009 3. Search for your topic -- Russell Mosemann Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! ___ 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] need help !! Problem with freeswitch uniMRCP
Hi Jeff, Your input would be very helpful, I just wanted to understand where the problem is and contribute the way I can. I see you're the assignee, so please go ahead and let me know if there is anything left I can help with. Arsen. From: Jeff Lenk jl...@frontiernet.net To: freeswitch-users@lists.freeswitch.org Sent: Mon, November 23, 2009 8:16:28 AM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP Hi Arsen, I would be happy to help with the FS integration if you want - please do put your patch in a Jira. Jeff Date: Sun, 22 Nov 2009 10:09:41 -0800 From: [hidden email] To: [hidden email] Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. So would you be interested in the patch? Where should I upload it? I thought I had a Jira account, but not sure it exists any more. -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org From: Jeff Lenk [hidden email] To: [hidden email] Sent: Fri, November 20, 2009 7:59:28 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: Hi Everyone, Please help freeswitch experts... !!! i have been working on freeswitch from last 2 days. i have downloaded freeswitch and unimrcp (server + client) for windows. I tested the unimrcp client and server, which is running fine with the command: run synth and run recog. I got both synth.pcm recog.pcm files. But my objective is to call Freeswitch through x-lite, where freeswitch should call unimrcp client and return the PCM files. I tried it alot, but unable to do it. after lots of reading i found that i do not have mod_unimrcp. i do not know from where to download it and how to merge it into freeswitch. I would be very thankful if you may help. Thanks, ss -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list [hidden email] 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 [hidden email] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org View message @ http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4047148.html To unsubscribe from Re: need help !! Problem with freeswitch uniMRCP, click here. Hotmail: Trusted email with powerful SPAM protection. Sign up now. View this message in context: RE: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP 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] Recording with Native File PCMU
If you're doing native file you DO NOT put an extension on the file name. /b On Nov 22, 2009, at 5:54 PM, Seven Du wrote: did you try without any .wav or .PCMU? ___ 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] GUI for Freeswitch -- wikiPBX
Hi Folks Is anyone using this on Fedora and is there a binary or installation script anywhere 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] GUI for Freeswitch -- wikiPBX
cd /usr/src wget http://www.freeswitch.org/eg/Makefile make /b On Nov 23, 2009, at 9:22 AM, Otis wrote: Hi Folks Is anyone using this on Fedora and is there a binary or installation script anywhere 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
Re: [Freeswitch-users] SIP Digest nonce (stale=true)
The quoted true is the correct way from my research. The commented line was to test a device, a grandstream, they apparently do not accept it with quotes and I was using the unquoted version it to gather evidence to issue a bug report to them. They told me it will be fixed in the next firmware, was this the brand of device you have as well? On Mon, Nov 23, 2009 at 6:23 AM, Nameer Kazzaz nameer.kaz...@gmail.comwrote: Hi Anthony, I'm having an issue with a gateway after the nonce-ttl expires we are sending stale=true, the cpe some how only likes stale=true without the . I see on rev 15441 http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/endpoints/mod_sofia/sofia_reg.c?r=15441#l687 you made a change and marked it out. So my question is who is correct on this is it the CPE or are we sticking with the quoted (true). Thanks Nameer Kazzaz ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@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] Fwd: Problem while playing more than 10 voice files using playback
Maybe it is a race condition, I can't tell you from just such a basic description the code is complicated and I would have to reproduce it myself, but I can tell you one more time for good measure that you should use execute_complete events to tell when a command you tried to execute has finished and not poll the channel for a variable to be set because FreeSWITCH is an asynchronous application in the mode you are describing and you can never be sure of the timing. On Sun, Nov 22, 2009 at 10:34 PM, Thangappan.M thangappan...@gmail.comwrote: I am waiting only for DTMF events. That's why I am setting freeswitch variable for knowing whether the playback has done. My question is why this freeswitch variable is not setting properly when I play back more than 10 files using playback_delimiter option?. When I play back lesser than ten voice files the variable has been set properly. What could be the reason? -- Forwarded message -- From: Thangappan.M thangappan...@gmail.com Date: Sat, Nov 21, 2009 at 2:52 PM Subject: Problem while playing more than 10 voice files using playback To: freeswitch-users freeswitch-users@lists.freeswitch.org Dear all, I am in the process of implementing IVR using event outbound socket (async mode). I have implemented using Perl language. I did the following steps: = Set the playback_delimiter variable = Set the playback_sleep_val variable = Set the event lock as true = Set the freeswitch ( my own) variable as zero = Wait in the loop until the variable is been set as zero = Playback the voice files ( Here I combined the voice files with the delimiter value if more than one voice files are there) = Set the freeswitch(my own) variable as true ( This is used to identify whether the voice files are played successfully). = Wait in the loop until the variable is been set as one. = Set the Event lock as false = Trying to get the DTMF digits ( Have a assurance that all the voice files are played). The problem is, The above steps are working fine when the voice file count is lesser than or equal to 10. After the voice files are played only the variable(my own freeswitch) is set. Based on the variable I am doing further things. But when I tried to give the voice files count of more than 10 the variable has been set while starting to play back the first voice file itself . Because of this I am not able to proceed further. *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* *NOTE*: I also referred mod_file_string documentation. In that they specified 128 files can be used to play back the voice files using playback_delimiter option. Please help me? Thanks in advance. -- Regards, Thangappan.M -- Regards, Thangappan.M ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@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] Problems with sighup and rotating csv files
did you really mean 1.0.4pre7 ? We are now on pre release of 1.0.5 so I cannot really debug such an old version so you may want to install one of the newer version first before you report an issue and when you do use our issue tracker not this mailing list http://jira.freeswitch.org be sure to answer all the questions carefully when filing the report. On Sun, Nov 22, 2009 at 12:46 PM, katarina djakovic kdjako...@hotmail.comwrote: Hi, I am using the Freeswitch 1.0.4pre7. Great application, but I encountered a problem wich I can not solve since I am very new to it. Two things are happening. 1) The mod_cdr_csv.c (line 122 do_rotate()) does not always respond to sighup signal to rotate the cdr-csv files. Some times it happens and some times it does not. I can not see any pattern in the behaviour. Seems that sometimes functions in the mod_cdr_csv.c catch the signal and some times they do not. 2) Playing with the kill -HUP fspid all of a sudden I started getting two freeswitch processes in the process list. One being parent of another. Then, when I send the sighup signal to the parent - the console dies off and the other freeswitch process stays (leaving the comment Hangup in the fsconsole). Freeswitch conitnues to work with the remaining process. In case when I send the sighup to the child, it will rotate the log files. However, it always rotates the freeswitch.log, but randomly rotates the cdr-csv files. 3) I have a feeling that above behaviours are somehow connected, but do not understand how. Anyone can help? Any comment or idea will be very very much apreciated. Cheers, Katarina -- Windows Live: Make it easier for your friends to see what you’re up to on Facebook.http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009 -- Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you.http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010 ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@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] Media got stuck after attended transfer...
I think that issue has been fixed in trunk re: proxy-mode and resume-media-on-hold On Sun, Nov 22, 2009 at 7:00 AM, Klaus Hochlehnert maili...@kh-dev.dewrote: For „only“ sending and receiving that’s true. But my customer wants 2 things: - Using HylaFAX as fax server, as there are a lot of client apps and other tools - Connecting “real” fax machines using a Linksys/Cisco SPA2102 (as this is certified by their SIP/ISDN gateway vendor) So I could really need t38 handling in FS to don’t make things more complicated as they already are… J Proxy mode doesn’t work for me because it gives an error when resume-media-on-hold is set. Klaus *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Tihomir Culjaga *Sent:* Sunday, November 22, 2009 1:15 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Media got stuck after attended transfer... it is better to enhance mod_fax with t.38 support... we have done sometihng and it is close to be work... T. On Sat, Nov 21, 2009 at 2:17 AM, Michael Jerris m...@jerris.com wrote: I think a better approach here is to use spandsp. We already have some groundwork done for this. If you are interested in contributing, please email consult...@freeswitch.org and we can discuss further. Mike On Nov 19, 2009, at 6:54 PM, Klaus Hochlehnert wrote: Hi, one of my customers is willing to contribute for t38 integration. The basic idea is to connect HylaFAX to FS: t38modem - FreeSWITCH - Media Gateway with t38 support All this without media proxy. Another idea might be to implement t38 origination/termination with a class 1 modem input/output for use with HylaFAX. Do you know how much money we need to collect for t38 support? How much time is needed for implementing this? Thanks, Klaus *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* Friday, October 16, 2009 2:10 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Media got stuck after attended transfer... On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga tculj...@gmail.com wrote: hi, any clue when can t38 be added? Eventually. :) Of course, if we could get more to add to the bounty it might grease the wheels of innovation. http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch Of course, I was listening to my A.M radio the other day and they said that there was this new invention called the Internet that would let people send documents to each other electronically. Maybe you should look into that. Next thing you know they'll come up with telephones that people don't have to plug into the wall and can take with them in the car. ;) -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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@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] mod_ldap
I'm new to Freeswitch and I'm looking for some help using mod_ldap to authenticate SIP endpoints to an LDAP database on registration. I have successfully installed the Freeswitch trunk and OpenLDAP 2.4.18 and I have compiled mod_ldap. I have setup a config file for mod_ldap. It is active and I can see its config in the compiled Freeswitch config file. I have turned on SIP traces, sofia logging and console logging in the Freeswitch command line. However, when I register a SIP phone to my Freeswitch server, I see no reference to mod_ldap in any logs. I have not set up the correct schema in ldap yet, so I would expect to see some indication that the module has searched my ldap server and found no useful information. My phone registers correctly and the normal tests (eg. music on hold) work. I am not sure if mod_ldap falls back on the usual xml config files if it fails to find information on the specified ldap server (as mod_xml_curl does), so I can't be sure if it is working or if my phone is simply registering in the normal way. Can anyone give me pointers on where to look next? If I can just get some feedback from the module I should be able to work out what to do. Thanks -- Grimsqueaker Even a fool, when he holdeth his peace, is counted wise. Proverbs 17:28a ___ 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] mod_flite sound profiles
Ok. I understand that. It would be great if someone can help me figure out: 1. Why mod_flite is not changing to the female voice even though I tried switching all 4 profiles it provides? 2. I would be alright for purchasing Cepstral for its quality. But FS doesn't come with it compiled I guess (it says swift.dll required when I enabled it in the config file). I asked Cepstral support but they say I have to purchase their SDK (no trial available) even though I just need it to compile it with FS. I understand I will be purchase the voices but how can I get Cepstral DLLs without purchasing the SDK. Thank you for help. Malay Thakershi From: Brian West [mailto:br...@freeswitch.org] Sent: Friday, November 20, 2009 5:33 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_flite sound profiles You pay top dollar for it. The free stuff just isn't as good as what you PAY good money for. I don't expect that to change anytime soon. /b On Nov 20, 2009, at 5:18 PM, Malay Thakershi wrote: Also, can someone tell me what is the best way to get TTS going with good quality? ___ 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] mod_flite sound profiles
If you're on linux the SDK comes with the voices. /b On Nov 23, 2009, at 10:07 AM, Malay Thakershi wrote: Ok. I understand that. It would be great if someone can help me figure out: 1. Why mod_flite is not changing to the female voice even though I tried switching all 4 profiles it provides? 2. I would be alright for purchasing Cepstral for its quality. But FS doesn’t come with it compiled I guess (it says swift.dll required when I enabled it in the config file). I asked Cepstral support but they say I have to purchase their SDK (no trial available) even though I just need it to compile it with FS. I understand I will be purchase the voices but how can I get Cepstral DLLs without purchasing the SDK. Thank you for help. Malay Thakershi ___ 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] FS compile error under Windows: error LNK2019
It sounds like the platform sdk is set up wrong. This used to be a problem with older versions of express edition. Double check that your compiler works at all with anything else. Mike On Nov 22, 2009, at 11:51 PM, 大泥人 wrote: All, I tried to compile FS source code under Windows while there are lots of errors: Error LNK2019, external _imp_sl...@4 can not be resolved, this function was referred by _tMCRTStartup. Some other more similiar errors detail information attached. Any ideas? Thanks Daniel Zeng ___ 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] SIP Digest nonce (stale=true)
Hey Anthony, Thanks for the quick response. No the device is a OneAccess so they are saying 'no quotes is the standard'. Thanks Nameer Anthony Minessale wrote: The quoted true is the correct way from my research. The commented line was to test a device, a grandstream, they apparently do not accept it with quotes and I was using the unquoted version it to gather evidence to issue a bug report to them. They told me it will be fixed in the next firmware, was this the brand of device you have as well? On Mon, Nov 23, 2009 at 6:23 AM, Nameer Kazzaz nameer.kaz...@gmail.com mailto:nameer.kaz...@gmail.com wrote: Hi Anthony, I'm having an issue with a gateway after the nonce-ttl expires we are sending stale=true, the cpe some how only likes stale=true without the . I see on rev 15441 http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/endpoints/mod_sofia/sofia_reg.c?r=15441#l687 you made a change and marked it out. So my question is who is correct on this is it the CPE or are we sticking with the quoted (true). Thanks Nameer Kazzaz ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@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 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] XML config file parsing
There is a formula to implement caching but it's very complicated and nobody has had time to work on it. You have to take every single input variable into account when caching because who is calling the extension, why they are calling it when they are calling it all make a difference. Web servers are designed to get thousands of hits per second so typically they can handle delivering custom xml instruction quite well. If you do not require such a dynamic setup, you could generate static files instead. On Sun, Nov 22, 2009 at 5:43 PM, Tim Uckun timuc...@gmail.com wrote: On Fri, Nov 20, 2009 at 3:03 AM, Rob Forman rob4manh...@gmail.com wrote: Hi Sam, Take a look at mod_xml_curl. Pretty sure it'll do everything you're looking for. Looking at that diagram it seems like mod_xml_curl makes a call for every SIP connection. That seems like overkill. Is there a way to set it up so that it caches the XML it got for a period of time? ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@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] Help Freeswitch with Voipuser Gateway
Hello
Could anyone point out what I have missed please ?
At the moment I configured a gateway voipuser as described here
http://www.onlinesolution.co.nz/viewtopic.php?p=119 :
Any suggestion as to what path I can take will be highly welcome
Thanks
.
Sam Abekah-Mensah wrote:
div class=moz-text-flowed style=font-family: -moz-fixedHi Michael
Thanks
I had set it to send incoming calls to extension 1001. This is in the
file abeka.xml in /usr/local/freeswitch/conf/dialplan/public directory.
The contents are :
extension name=inbound-*userna...@sip.voipuser.org]
condition field=destination_number expression=08444846450
action application=transfer data=1001 XML default/
/condition
/extension
Is there
anything wrong with this please ?
Thanks
Michal Bielicki wrote:
Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah:
I need help as I cannot receive calls through VOIPUSER. This is a
learning setup Attached are my conf files. What is wrong with them ?
When I dial from a landline I get a continuous beep.
Attached are my gateway and the conf file to transfer. Sopfia Status
is my screen message. I can see a FAIL and cannot make head or tail
of all that message. Hopefully anyone using voipuser or in fact any
of you clever folks can make sense of this.
Thanks for your time.
2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate Buggy
RFC2833 Mode!
2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec
Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec
Compare [PCMA:8:8000:0]/[PCMA:8:8000:20]
2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec
sofia/external/nob...@213.166.5.133 PCMA/8000 20 ms 160 samples
2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf
payload to 101
2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455
(sofia/external/nob...@213.166.5.133) State Change CS_NEW - CS_INIT
2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398
(sofia/external/nob...@213.166.5.133) Running State Change CS_INIT
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481
(sofia/external/nob...@213.166.5.133) State INIT
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83
sofia/external/nob...@213.166.5.133 SOFIA INIT
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111
(sofia/external/nob...@213.166.5.133) State Change CS_INIT -
CS_ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481
(sofia/external/nob...@213.166.5.133) State INIT going to sleep
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398
(sofia/external/nob...@213.166.5.133) Running State Change CS_ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484
(sofia/external/nob...@213.166.5.133) State ROUTING
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130
sofia/external/nob...@213.166.5.133 SOFIA ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78
sofia/external/nob...@213.166.5.133 Standard ROUTING
2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 Processing
anonymous-abeka in context public
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-unloop] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-outside_call] continue=true
Dialplan: sofia/external/nob...@213.166.5.133 Absolute Condition
[outside_call]
Dialplan: sofia/external/nob...@213.166.5.133 Action
set(outside_call=true)
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-call_debug] continue=true
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-public_extensions] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[public_extensions] destination_number(abeka) =~ /^(10[01][0-9])$/
break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-public_did] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[public_did] destination_number(abeka) =~ /^(5551212)$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-s...@sip.voipuser.org] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[...@sip.voipuser.org] destination_number(abeka) =~ /08715042951/
break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing
Re: [Freeswitch-users] XML config file parsing
Or, you can use something like Smarty to cache your generated XML on your web server and only invalidate those cached results when you change something that will impact them. On Mon, Nov 23, 2009 at 11:38 AM, Anthony Minessale anthony.miness...@gmail.com wrote: There is a formula to implement caching but it's very complicated and nobody has had time to work on it. You have to take every single input variable into account when caching because who is calling the extension, why they are calling it when they are calling it all make a difference. Web servers are designed to get thousands of hits per second so typically they can handle delivering custom xml instruction quite well. If you do not require such a dynamic setup, you could generate static files instead. On Sun, Nov 22, 2009 at 5:43 PM, Tim Uckun timuc...@gmail.com wrote: On Fri, Nov 20, 2009 at 3:03 AM, Rob Forman rob4manh...@gmail.com wrote: Hi Sam, Take a look at mod_xml_curl. Pretty sure it'll do everything you're looking for. Looking at that diagram it seems like mod_xml_curl makes a call for every SIP connection. That seems like overkill. Is there a way to set it up so that it caches the XML it got for a period of time? ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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 -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ 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] SIP Digest nonce (stale=true)
Tell you what, I don't have the patience for it, i'm sure most stuff does it either way and I'm sure nobody insists you have them so I will take them out so I can have some peace. On Mon, Nov 23, 2009 at 10:35 AM, Nameer Kazzaz nameer.kaz...@gmail.comwrote: Hey Anthony, Thanks for the quick response. No the device is a OneAccess so they are saying 'no quotes is the standard'. Thanks Nameer Anthony Minessale wrote: The quoted true is the correct way from my research. The commented line was to test a device, a grandstream, they apparently do not accept it with quotes and I was using the unquoted version it to gather evidence to issue a bug report to them. They told me it will be fixed in the next firmware, was this the brand of device you have as well? On Mon, Nov 23, 2009 at 6:23 AM, Nameer Kazzaz nameer.kaz...@gmail.com mailto:nameer.kaz...@gmail.com wrote: Hi Anthony, I'm having an issue with a gateway after the nonce-ttl expires we are sending stale=true, the cpe some how only likes stale=true without the . I see on rev 15441 http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/endpoints/mod_sofia/sofia_reg.c?r=15441#l687 you made a change and marked it out. So my question is who is correct on this is it the CPE or are we sticking with the quoted (true). Thanks Nameer Kazzaz ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.commsn%253aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.compaypal%253aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.orgsip%253a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.orggoogletalk%253aconf%252b...@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 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@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] need help !! Problem with freeswitch uniMRCP
Hi Arsen, I have merged your changes in now - thank you. Would you perhaps be able to look at the x64 changes I made to the projects and merge them back into your code to ease the future updating. Thanks Jeff Arsen Chaloyan wrote: Hi Jeff, Your input would be very helpful, I just wanted to understand where the problem is and contribute the way I can. I see you're the assignee, so please go ahead and let me know if there is anything left I can help with. Arsen. From: Jeff Lenk jl...@frontiernet.net To: freeswitch-users@lists.freeswitch.org Sent: Mon, November 23, 2009 8:16:28 AM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP Hi Arsen, I would be happy to help with the FS integration if you want - please do put your patch in a Jira. Jeff Date: Sun, 22 Nov 2009 10:09:41 -0800 From: [hidden email] To: [hidden email] Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. So would you be interested in the patch? Where should I upload it? I thought I had a Jira account, but not sure it exists any more. -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org From: Jeff Lenk [hidden email] To: [hidden email] Sent: Fri, November 20, 2009 7:59:28 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: Hi Everyone, Please help freeswitch experts... !!! i have been working on freeswitch from last 2 days. i have downloaded freeswitch and unimrcp (server + client) for windows. I tested the unimrcp client and server, which is running fine with the command: run synth and run recog. I got both synth.pcm recog.pcm files. But my objective is to call Freeswitch through x-lite, where freeswitch should call unimrcp client and return the PCM files. I tried it alot, but unable to do it. after lots of reading i found that i do not have mod_unimrcp. i do not know from where to download it and how to merge it into freeswitch. I would be very thankful if you may help. Thanks, ss -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list [hidden email] 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 [hidden email] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org View message @ http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4047148.html To unsubscribe from Re: need help !! Problem with freeswitch uniMRCP, click here. Hotmail: Trusted email with powerful SPAM protection. Sign up now. View this message in context: RE: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP 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 -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4052409.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
[Freeswitch-users] NAT problem
Hello I got the following setup: Phones - FreeSwitch - NAT - Internet - Gateway And I'm struggling to get NAT working properly. I'm running freeswitch with the -nonat option and have tried different ext-rtp-ip/ext-sip-ip combinations in external/internal profiles. The From header seems to be correct while contact header and SDP uses local ip? Please help me configure everything correctly. Currently I have this setup: API CALL [sofia(status profile external)] output: Nameexternal Domain Name N/A Context public Challenge Realm auto_to RTP-IP 192.168.1.110 Ext-RTP-IP 85.89.XX.XX SIP-IP 192.168.1.110 Ext-SIP-IP 85.89.XX.XX OUTBOUND-PROXY N/A PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDtrue STUN-AUTO-DISABLE false API CALL [sofia(status profile default)] output: Namedefault Domain Name N/A Alias Ofinternal Context public Challenge Realm auto_from RTP-IP 192.168.1.110 Ext-RTP-IP 85.89.XX.XX SIP-IP 192.168.1.110 OUTBOUND-PROXY N/A PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDfalse STUN-AUTO-DISABLE false Sample phone registration: Call-ID:xmbw9pyq5q6l2...@192.168.1.121 User: u1000...@default Contact:u109 sip:u1000...@192.168.1.121:6094 Agent: IP PHONE 3 V1.58.004 CFG0 Status: Registered(UDP)(unknown) EXP(2009-11-23 19:26:40) Host: jonas-PC IP: 192.168.1.121 Port: 6094 Auth-User: u109 Auth-Realm: default MWI-Account:u1000...@default Outbound INVITE: send 1122 bytes to udp/[62.80.XX.XX]:5060 at 17:05:01.74: INVITE sip:0706930...@sipgw2.x.se sip%3a0706930...@sipgw2.x.seSIP/2.0 Via: SIP/2.0/UDP 192.168.1.110;rport;branch=z9hG4bKB72B75aKmSyBp Max-Forwards: 69 From: Kundtjänst Arne sip:0500650...@85.89.xx.xx;tag=B7pve7F6eeH7c To: sip:0706930...@sipgw2.x.se sip%3a0706930...@sipgw2.x.se Call-ID: 2dcead20-52f5-122d-d3a1-77ca4f97ec23 CSeq: 123379614 INVITE Contact: sip:mod_so...@192.168.1.110:5060 Call-Info: answer-after=400 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 293 X-FS-Support: update_display Remote-Party-ID: Kundtjänst Arne sip:0500650...@85.89.xx.xx ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1258970915 1258970916 IN IP4 192.168.1.110 s=FreeSWITCH c=IN IP4 192.168.1.110 t=0 0 m=audio 24986 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 Many thanks, Jonas ___ 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] NAT problem
You set the ext-rtp-ip on the profile the phones talk too... but you shouldn't be doing that. /b On Nov 23, 2009, at 11:08 AM, Jonas Gauffin wrote: Hello I got the following setup: Phones - FreeSwitch - NAT - Internet - Gateway And I'm struggling to get NAT working properly. I'm running freeswitch with the -nonat option and have tried different ext-rtp- ip/ext-sip-ip combinations in external/internal profiles. The From header seems to be correct while contact header and SDP uses local ip? Please help me configure everything correctly. Currently I have this setup: ___ 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] NAT problem
Ok. Found the problem. I had started using sofia/outbound/ xxx...@sipgw2..se as bridge destination to try to get outbound_caller_id_name/outbound_caller_id_number working. It works if I use the correct profile name, sofia/internal/ xxx...@sipgw2..se When do FS use outbound_caller_id instead of effective_caller_id? On Mon, Nov 23, 2009 at 6:08 PM, Jonas Gauffin jonas.gauf...@gmail.comwrote: Hello I got the following setup: Phones - FreeSwitch - NAT - Internet - Gateway And I'm struggling to get NAT working properly. I'm running freeswitch with the -nonat option and have tried different ext-rtp-ip/ext-sip-ip combinations in external/internal profiles. The From header seems to be correct while contact header and SDP uses local ip? Please help me configure everything correctly. Currently I have this setup: API CALL [sofia(status profile external)] output: Nameexternal Domain Name N/A Context public Challenge Realm auto_to RTP-IP 192.168.1.110 Ext-RTP-IP 85.89.XX.XX SIP-IP 192.168.1.110 Ext-SIP-IP 85.89.XX.XX OUTBOUND-PROXY N/A PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDtrue STUN-AUTO-DISABLE false API CALL [sofia(status profile default)] output: Namedefault Domain Name N/A Alias Ofinternal Context public Challenge Realm auto_from RTP-IP 192.168.1.110 Ext-RTP-IP 85.89.XX.XX SIP-IP 192.168.1.110 OUTBOUND-PROXY N/A PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDfalse STUN-AUTO-DISABLE false Sample phone registration: Call-ID:xmbw9pyq5q6l2...@192.168.1.121 User: u1000...@default Contact:u109 sip:u1000...@192.168.1.121:6094 Agent: IP PHONE 3 V1.58.004 CFG0 Status: Registered(UDP)(unknown) EXP(2009-11-23 19:26:40) Host: jonas-PC IP: 192.168.1.121 Port: 6094 Auth-User: u109 Auth-Realm: default MWI-Account:u1000...@default Outbound INVITE: send 1122 bytes to udp/[62.80.XX.XX]:5060 at 17:05:01.74: INVITE sip:0706930...@sipgw2.x.sesip%3a0706930...@sipgw2.x.seSIP/2.0 Via: SIP/2.0/UDP 192.168.1.110;rport;branch=z9hG4bKB72B75aKmSyBp Max-Forwards: 69 From: Kundtjänst Arne sip:0500650...@85.89.xx.xx;tag=B7pve7F6eeH7c To: sip:0706930...@sipgw2.x.se sip%3a0706930...@sipgw2.x.se Call-ID: 2dcead20-52f5-122d-d3a1-77ca4f97ec23 CSeq: 123379614 INVITE Contact: sip:mod_so...@192.168.1.110:5060 Call-Info: answer-after=400 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 293 X-FS-Support: update_display Remote-Party-ID: Kundtjänst Arne sip:0500650...@85.89.xx.xx ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1258970915 1258970916 IN IP4 192.168.1.110 s=FreeSWITCH c=IN IP4 192.168.1.110 t=0 0 m=audio 24986 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 Many thanks, Jonas ___ 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] IP0010 SIP Phone
On Sun, Nov 22, 2009 at 6:09 AM, David V. Fansler dfans...@dv-fansler.comwrote: After the help of a couple of people from this list, I now have FreeSWITCH running - yeah! I have installed X-Lite on a couple of computers and they dial each other, play music on hold, etc. I have not yet connected to the outside world. I purchased an IP-0010 phone off eBay ($20 including shipping - docs at http://www.vanaccess.com/news/news_images/2007131_73_User%20Manual%20-%20IP0010.pdf) I cannot get this phone to work with the system. It gets an IP address, time/date, and a dial tone. After many tries with the http congifuration tool, I got the phone configured with the address of the SIP server, and a SIP User ID. When you dial an extension the FreeSWITCH window shows the following: sofica.c3844 Hanugup sofia/internal/1...@192.168.1.165 [CS_NEW] [INCOMPATIBLE_DESTINATION] switch_core_session.c1139 Session 20 (sofia/internal/1...@192.165.1.65) Ended switch_core_session.c1141 Close Channel sofia/internal/1...@192.168.1.165[cs_destroy] Has anyone else tried this phone, or does anyone have suggestions I could try. I have looked through the website but have not found anything to help. Thanks, David David, Time to do a little digging. First off, review this wiki page on reporting bugs - it has lots of useful information on how to gather information from your system and report it to the community: http://wiki.freeswitch.org/wiki/Reporting_Bugs I'd recommend that you get a debug log and a sip trace and post it to pastebin. Report back the pastebin URL here in this thread and we'll have a look. -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] NAT problem
outbound_caller_id is a made up variable that is used in the defaults that are used in the examples only. /b On Nov 23, 2009, at 11:24 AM, Jonas Gauffin wrote: Ok. Found the problem. I had started using sofia/outbound/xxx...@sipgw2..se as bridge destination to try to get outbound_caller_id_name/ outbound_caller_id_number working. It works if I use the correct profile name, sofia/internal/xxx...@sipgw2..se When do FS use outbound_caller_id instead of effective_caller_id? ___ 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] NAT problem
Ok. It would be a nice feature if outbound_caller_id was used by freeswitch. I do quite often bridge to both internal and external destinations in the same bridge command (as in sofia/internal/5530,sofia/internal/ 070123...@sipgw2..se). This forces me to always use complete phone numbers in the caller id since my gateway would reject the call otherwise. It would be really neat if FS could use effective_caller_id (5531) for the internal bridge and outbound_caller_id (+4681235531) for the external bridge. On Mon, Nov 23, 2009 at 6:31 PM, Brian West br...@freeswitch.org wrote: outbound_caller_id is a made up variable that is used in the defaults that are used in the examples only. /b On Nov 23, 2009, at 11:24 AM, Jonas Gauffin wrote: Ok. Found the problem. I had started using sofia/outbound/ xxx...@sipgw2..se as bridge destination to try to get outbound_caller_id_name/outbound_caller_id_number working. It works if I use the correct profile name, sofia/internal/ xxx...@sipgw2..se When do FS use outbound_caller_id instead of effective_caller_id? ___ 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] GUI for Freeswitch -- wikiPBX
Yeah a kind user (Innotel) took the time to write up Cent OS installation instructions for wikipbx and posted it to the wiki: http://wikipbx.subwiki.com/forum/t-115012/freeswitch-svn-1-0-2-wikipbx-svn-61-centos-5-1-installation-instructions If you have any problems please post in the forum: http://wikipbx.subwiki.com/forum:start On Mon, Nov 23, 2009 at 7:52 PM, Otis ab...@greatiam.com wrote: Hi Folks Is anyone using this on Fedora and is there a binary or installation script anywhere 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
Re: [Freeswitch-users] Callback to the user in ESL
On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy lakindi...@gmail.comwrote: Hi, I'm using perl ESL to control the call in freeswitch. I'm having the following scenario, but not able to get it right. Dialplan: extension name=outbound_soc condition field=destination_number expression=^9097$ action application=set data=continue_on_fail=true/ action application=socket data=192.168.1.222:8447 async full/ /condition /extension 1. User A calls to an extention (1000). 2. My ESL program will be running, and it answers the call. 3. Then the program will get a number from the user. 4. It will hangup the call. 5. The program has to call to the number that was given by the user. In the above scenario, I was able to do until the 4th step. After hangup the call, if I say originate it is not working. Any ideas on how to do this in ESL. I want to make sure I understand what the script is supposed to be doing. The caller will key in a phone number to your script and your script will collect those digits. The script will then hangup on the caller and originate a completely new call? Perhaps you could use sched_api to schedule a new originate command for a few seconds into the future and then hangup? -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] GUI for Freeswitch -- wikiPBX
s/i386/x86_64/ if you are 64bit /b On Nov 23, 2009, at 11:47 AM, Traun Leyden wrote: Yeah a kind user (Innotel) took the time to write up Cent OS installation instructions for wikipbx and posted it to the wiki: http://wikipbx.subwiki.com/forum/t-115012/freeswitch-svn-1-0-2-wikipbx-svn-61-centos-5-1-installation-instructions If you have any problems please post in the forum: http://wikipbx.subwiki.com/forum:start On Mon, Nov 23, 2009 at 7:52 PM, Otis ab...@greatiam.com wrote: Hi Folks Is anyone using this on Fedora and is there a binary or installation script anywhere 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 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] Building in a builddir using --srcdir option but modules still build in srcdir
I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? Thanks, Robert ___ 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] NAT problem
See default config it lsets you do that. Use the variables to store two versions of the callerid then set it depending on if its outside or inside... its rather easy to do. /b On Nov 23, 2009, at 11:50 AM, Jonas Gauffin wrote: Ok. It would be a nice feature if outbound_caller_id was used by freeswitch. I do quite often bridge to both internal and external destinations in the same bridge command (as in sofia/internal/5530,sofia/internal/070123...@sipgw2..se ). This forces me to always use complete phone numbers in the caller id since my gateway would reject the call otherwise. It would be really neat if FS could use effective_caller_id (5531) for the internal bridge and outbound_caller_id (+4681235531) for the external bridge. ___ 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] mod_flite sound profiles
I am not using Linux. I am using Windows 2008 server. Malay Thakershi From: Brian West [mailto:br...@freeswitch.org] Sent: Monday, November 23, 2009 10:23 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_flite sound profiles If you're on linux the SDK comes with the voices. /b On Nov 23, 2009, at 10:07 AM, Malay Thakershi wrote: Ok. I understand that. It would be great if someone can help me figure out: 1. Why mod_flite is not changing to the female voice even though I tried switching all 4 profiles it provides? 2. I would be alright for purchasing Cepstral for its quality. But FS doesn't come with it compiled I guess (it says swift.dll required when I enabled it in the config file). I asked Cepstral support but they say I have to purchase their SDK (no trial available) even though I just need it to compile it with FS. I understand I will be purchase the voices but how can I get Cepstral DLLs without purchasing the SDK. Thank you for help. Malay Thakershi ___ 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] mod_flite sound profiles
You don't have to buy the SDK... I have had it sent to everyone that has asked me for it... the address is on the wiki for who to contact. If you were using linux the SDK is included already. /b On Nov 23, 2009, at 12:09 PM, Malay Thakershi wrote: I am not using Linux. I am using Windows 2008 server. Malay Thakershi ___ 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] Git
Just wondering if anyone is keeping an update to date git repo of FreeSwitch? I been using git-svn to keep a copy on my machines but it can be quite time consuming due to the per revision fetching. If there was a repo to clone that would speed up the process considerably. -- W ___ 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] Git
William, Perhaps someone could setup one on github? It's free for open source project. Lon On Nov 23, 2009, at 10:22 AM, William Suffill william.suff...@gmail.com wrote: Just wondering if anyone is keeping an update to date git repo of FreeSwitch? I been using git-svn to keep a copy on my machines but it can be quite time consuming due to the per revision fetching. If there was a repo to clone that would speed up the process considerably. -- W ___ 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] GUI for Freeswitch -- wikiPBX
Thanks. I have to get a centos box I guess. Much appreciated Samuel 'Otis' Traun Leyden wrote: Yeah a kind user (Innotel) took the time to write up Cent OS installation instructions for wikipbx and posted it to the wiki: http://wikipbx.subwiki.com/forum/t-115012/freeswitch-svn-1-0-2-wikipbx-svn-61-centos-5-1-installation-instructions If you have any problems please post in the forum: http://wikipbx.subwiki.com/forum:start On Mon, Nov 23, 2009 at 7:52 PM, Otis ab...@greatiam.com mailto:ab...@greatiam.com wrote: Hi Folks Is anyone using this on Fedora and is there a binary or installation script anywhere Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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] mod_flite sound profiles
Thank you for your responses. I did follow that web link to ask them as instructed but they declined. They asked me where I want to use it. I told them I wanted it to build FreeSwitch so that I can use Cepstral voices (to be purchased from them with it). Their response was they do not provide trial of the SDK. They do not support FreeSwitch. Malay Thakershi From: Brian West [mailto:br...@freeswitch.org] Sent: Monday, November 23, 2009 12:14 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_flite sound profiles You don't have to buy the SDK... I have had it sent to everyone that has asked me for it... the address is on the wiki for who to contact. If you were using linux the SDK is included already. /b On Nov 23, 2009, at 12:09 PM, Malay Thakershi wrote: I am not using Linux. I am using Windows 2008 server. Malay Thakershi ___ 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] Git
I think this one is kept up to date, but we may re-do this at some point soon, so it may get re-built. http://svn.freeswitch.org/freeswitch.git/ Mike On Nov 23, 2009, at 1:22 PM, William Suffill wrote: Just wondering if anyone is keeping an update to date git repo of FreeSwitch? I been using git-svn to keep a copy on my machines but it can be quite time consuming due to the per revision fetching. If there was a repo to clone that would speed up the process considerably. ___ 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] tcp call misses sip message
This looks like a nat issue to me, please re-test this against latest svn trunk and if its still not working pastebin a full sip trace and report the link back here. Mike On Nov 21, 2009, at 6:23 PM, RobertT wrote: Yep, I use proxy media. First it started with 1.0.4 release, then I've updated a week or two ago with the latest svn trunk, not sure what was the rev number. ___ 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] How do I know the destination profile name?
Because if you dial local-u...@local-domain thats not the correct way this will usually trigger a call out and back in on the profile thus moving you one leg away from the actual user. If you're going to do that use sofia_contact and review how the defaults abstract this so you can just call user/x...@domain, You need to make sure the presence_id is set like the defaults have it. /b On Nov 22, 2009, at 1:39 AM, Yehavi Bourvine wrote: Thanks Mike! However, this doesn't fully solve my problem. When using sofia_contact() indeed it works ok with finding the destination's profile. However, it breaks the BLFs... When calling sofia/sip_profile/local-user%local-domain the BLF works ok. When calling sofia_contact(sofia/sip_profile/local-u...@local- domain) BLF doesn't work (nothing is sent to the watching phone). Any more clues??? Thanks! __Yehavi: ___ 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] Git
I'd rather it be a decision by the community as a whole and authorized. Sure there are ways to have anyone who wants to on their own. Thanks for the insight. -- W ___ 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] User who answer the bridge in a execute_answer
Try running the info app there and see if the info is anywhere in that output .
Mike
On Nov 23, 2009, at 5:36 AM, Albano Daniele Salvatore - Lavoro wrote:
Hi,
i'm writing some dialplan parts that get executed on execute_on_answer. In
this dialplan that get executed i need to make a directory to handle
recordings for record_session and my folder structure is:
USER/YEAR/MONTH/HOUR-MINUTE-SECOND-CALLER_NUMBER.wav
--
action application=system data=mkdir -p
$${base_dir}/recordings/${sip_from_user}/${strftime(%Y)}/${strftime(%m)}/ /
action application=bind_meta_app data=1 a s
record_session::$${base_dir}/recordings/${sip_from_user}/${strftime(%Y)}/${strftime(%m)}/${strftime(%H_%M_%S)}-${caller_id_number}.wav
/
--
The call flow is:
Call from external - IVR - Transfer to Group - Execute on Answer -
system/bind_meta_app
Pratically, i need the number (or better the user) that answered the call:
what variable should i check?
I tried with sip_from_user, callee_id_number and some other.
Thank for your help,
Best Regards,
Daniele
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Re: [Freeswitch-users] Execute on Answer with JavaScript
This is done automatically when you bridge 2 sessions together. Mike On Nov 23, 2009, at 6:45 AM, Oscav wrote: How can we send the answer to the caller only when the callee answers, in JavaScript?? ___ 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 rtp-timeout-sec variable
Take a look at a pcap of the traffic, I suspect the other side still has media flowing. On Nov 23, 2009, at 7:00 AM, Maciej Aniserowicz wrote: Hello, I have 2 instances of FS: one controlled by my application (making calls with TCP commands, recording sessions, listening to events etc) and one acting as a remote gateway to which all users register. When I leave the default values of rtp-timeout-sec and brutally kill x-lite during conversation, the 'hangup' event with 'media_timeout' cause is obviously sent after the default 5 minutes (and until then, the other leg is still connected to a 'dead' channel). The question is: which FS instance is responsible for terminating the connection after timeout? Only the 'remote' FS instance config seems to work. I thought that the shortest configured value should cause the timeout, but it's not the case. Am I missing something, or is this the correct behavior? ___ 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] Using odbc in FS core
Yes please On Nov 23, 2009, at 6:45 AM, Mike Tkachuk wrote: Hello Anthony, Is clear, thanks, I'll test and will let you know. Should I add 'core-db-dsn' parameter description to Wiki? Maybe we need to add this parameter also to sample conf files? ___ 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] Building in a builddir using --srcdir option but modules still build in srcdir
The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? ___ 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] mod_flite sound profiles
Sounds like they don't want your business that much. You can try using mrcp with them , not sure if they have that released on their side or not. I think the build integration for mrcp client just went into the windows build earlier today. To be honest we used to have a pretty good relationship with them but we have had basically no response at all to any technical problems we have had with them in quite some time, so maybe they have decided to move on and not work with open source any more. It would appear so from their actions at least. Mike On Nov 23, 2009, at 1:41 PM, Malay Thakershi wrote: Thank you for your responses. I did follow that web link to ask them as instructed but they declined. They asked me where I want to use it. I told them I wanted it to build FreeSwitch so that I can use Cepstral voices (to be purchased from them with it). Their response was they do not provide trial of the SDK. They do not support FreeSwitch. Malay Thakershi From: Brian West [mailto:br...@freeswitch.org] Sent: Monday, November 23, 2009 12:14 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_flite sound profiles You don't have to buy the SDK... I have had it sent to everyone that has asked me for it... the address is on the wiki for who to contact. If you were using linux the SDK is included already. /b ___ 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] Simplest of Conference Setup questions
Hi there,
I have created a simple conference that works great. The only problem is,
when a participant press # it exits the call. So when a user enters a
conference with a PIN, and by habit they enter 12345 followed by pound, it
puts them in and then straight out.
So I edited conference.conf.xml so:
!--control action=hangup digits=#/--
and even assigned # to another function:
control action=energy up digits=#/
and the same occurs. Pressing # exits the conference.
What am I missing here?
tia - phil
Conf Setup:
extension name=conference.conf
condition field=destination_number expression=^(2125556625)$
action application=answer/
action application=conference data=$1-${domain_na...@default
+123/
/condition
/extension
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Re: [Freeswitch-users] Simplest of Conference Setup questions
issue
console loglevel debug
from the cli
then try again and see if there is any hint
On Mon, Nov 23, 2009 at 1:24 PM, Phillip Jones pjinthe...@gmail.com wrote:
Hi there,
I have created a simple conference that works great. The only problem is,
when a participant press # it exits the call. So when a user enters a
conference with a PIN, and by habit they enter 12345 followed by pound, it
puts them in and then straight out.
So I edited conference.conf.xml so:
!--control action=hangup digits=#/--
and even assigned # to another function:
control action=energy up digits=#/
and the same occurs. Pressing # exits the conference.
What am I missing here?
tia - phil
Conf Setup:
extension name=conference.conf
condition field=destination_number expression=^(2125556625)$
action application=answer/
action application=conference data=$1-${domain_na...@default
+123/
/condition
/extension
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--
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sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
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Re: [Freeswitch-users] Callback to the user in ESL
or open a new outbound connection at the end of your script so you can send your originate command. Since the channel hanging up will close your existing connection since it's only an outbound single session socket. On Mon, Nov 23, 2009 at 11:51 AM, Michael Collins m...@freeswitch.orgwrote: On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy lakindi...@gmail.com wrote: Hi, I'm using perl ESL to control the call in freeswitch. I'm having the following scenario, but not able to get it right. Dialplan: extension name=outbound_soc condition field=destination_number expression=^9097$ action application=set data=continue_on_fail=true/ action application=socket data=192.168.1.222:8447 async full/ /condition /extension 1. User A calls to an extention (1000). 2. My ESL program will be running, and it answers the call. 3. Then the program will get a number from the user. 4. It will hangup the call. 5. The program has to call to the number that was given by the user. In the above scenario, I was able to do until the 4th step. After hangup the call, if I say originate it is not working. Any ideas on how to do this in ESL. I want to make sure I understand what the script is supposed to be doing. The caller will key in a phone number to your script and your script will collect those digits. The script will then hangup on the caller and originate a completely new call? Perhaps you could use sched_api to schedule a new originate command for a few seconds into the future and then hangup? -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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@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] INCOMPATIBLE_DESTINATION on 180 Ringing
you need to provide a FS console trace of your problem from your FS source dir (build root) cd libs/esl make perlmod cd perl perl logger.pl -pb christian reproduce then hit ctl-c and tell me the url it posted to. 2009/11/23 Christian Löschenkohl christian.loeschenk...@xpirio.com hi our freeswitch server has to talk to a sonus ip-switch when we want to setup a call we do get a 100 Trying and then a 180 Ringing within the 180 Ringing we get a sdp with a=sendonly then our freeswitch quits with a CANCEL message. i simply don't get why our freeswitch aborts the session - i think it would work if no a=sendonly would be present in the sdp. my technical contact doesn't want to switch 180 to 183 on the sonus side - this would also work (i think). in fact he says that 180 ringing is vaild, he isn't that wrong in this case. our freeswitch works in proxy mode, we do use trunk 15396 see a ngrep trace under http://pastebin.freeswitch.org/11235 92.63.208.36 - freeswitch 38.105.229.100 - sonus br -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@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] INCOMPATIBLE_DESTINATION on 180 Ringing
Well its also G729 so I suspect you don't have G729 /b On Nov 23, 2009, at 1:17 PM, Christian Löschenkohl wrote: hi our freeswitch server has to talk to a sonus ip-switch when we want to setup a call we do get a 100 Trying and then a 180 Ringing within the 180 Ringing we get a sdp with a=sendonly then our freeswitch quits with a CANCEL message. i simply don't get why our freeswitch aborts the session - i think it would work if no a=sendonly would be present in the sdp. my technical contact doesn't want to switch 180 to 183 on the sonus side - this would also work (i think). in fact he says that 180 ringing is vaild, he isn't that wrong in this case. our freeswitch works in proxy mode, we do use trunk 15396 see a ngrep trace under http://pastebin.freeswitch.org/11235 92.63.208.36 - freeswitch 38.105.229.100 - sonus br -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.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] Building in a builddir using --srcdir optionbut modules still build in srcdir
Thanks Mike. modmake.rules is created in the $(switch_builddir)/build. What I see as the problem is in src/mod/Makefile.am There is a statement line 12 that points moddir to the source if test -d $(switch_srcdir)/src/mod/$$confmoddir ; then \ moddir = $(switch_srcdir)/src/mod/$$confmoddir ; And then the statements starting around line 22 that cd to moddir (in src) and fire off make if test -f $$moddir/Makefile ; then \-- Yep, this will be true cd $$moddir . $(MAKE) I'm not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). Does anybody have a better idea? Thanks, Robert _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Monday, November 23, 2009 11:16 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How do I know the destination profile name?
Let's just do this:
r15629 or higher
look for sip_profile_name
On Tue, Nov 17, 2009 at 3:03 AM, Eli Hayun eliha...@gmail.com wrote:
Hi
We have more then one profile. To make a call I have to enter : bridge
sofia/profile/num...@ip
The problem is when I use : ${use_profile} I am getting the caller
profile, and I need the destination profile.
How do I get this information?
Thanks
Eli
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Re: [Freeswitch-users] Simplest of Conference Setup questions
On Mon, Nov 23, 2009 at 11:24 AM, Phillip Jones pjinthe...@gmail.comwrote: Hi there, I have created a simple conference that works great. The only problem is, when a participant press # it exits the call. So when a user enters a conference with a PIN, and by habit they enter 12345 followed by pound, it puts them in and then straight out. So I edited conference.conf.xml so: !--control action=hangup digits=#/-- and even assigned # to another function: control action=energy up digits=#/ and the same occurs. Pressing # exits the conference. What am I missing here? tia - phil Phil, I recommend that you create a custom profile and a custom caller control group. Just copy the defaults and rename them to something meaningful. In conference.conf.xml you can add a new call control group like this: group name=custom !-- notice the new name -- control action=mute digits=0/ control action=deaf mute digits=*/ control action=energy up digits=9/ control action=energy equ digits=8/ control action=energy dn digits=7/ control action=vol talk up digits=3/ control action=vol talk zero digits=2/ control action=vol talk dn digits=1/ control action=vol listen up digits=6/ control action=vol listen zero digits=5/ control action=vol listen dn digits=4/ !-- Notice that I removed the hangup option; # digit is not bound to anything; you can bind it to something else if you wish -- /group Then make a copy of the default profile changing the profile name and the caller-controls parameter: profile name=custom !-- notice the new name -- !-- snip -- param name=caller-controls value=some name/ !-- snip -- /profile Give that a whirl and report back. :) -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] Simplest of Conference Setup questions
Thanks for replying.
Well in the log I see:
2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760
2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving
conference, cause: NONE
which make sense because just above I see:
009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default
caller control action 'hangup' bound to '#'.
The question I have - is how do I change that default caller control action
if it is not in conference.conf.xml ??
caller-controls
group name=default
...
*control action=energy up digits=#/*
On Mon, Nov 23, 2009 at 2:35 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
issue
console loglevel debug
from the cli
then try again and see if there is any hint
On Mon, Nov 23, 2009 at 1:24 PM, Phillip Jones pjinthe...@gmail.comwrote:
Hi there,
I have created a simple conference that works great. The only problem is,
when a participant press # it exits the call. So when a user enters a
conference with a PIN, and by habit they enter 12345 followed by pound, it
puts them in and then straight out.
So I edited conference.conf.xml so:
!--control action=hangup digits=#/--
and even assigned # to another function:
control action=energy up digits=#/
and the same occurs. Pressing # exits the conference.
What am I missing here?
tia - phil
Conf Setup:
extension name=conference.conf
condition field=destination_number expression=^(2125556625)$
action application=answer/
action application=conference data=$1-${domain_na...@default
+123/
/condition
/extension
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Re: [Freeswitch-users] Simplest of Conference Setup questions
see what happens if you set hangup to some other key or the word event
On Mon, Nov 23, 2009 at 2:17 PM, Phillip Jones pjinthe...@gmail.com wrote:
Thanks for replying.
Well in the log I see:
2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760
2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving
conference, cause: NONE
which make sense because just above I see:
009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default
caller control action 'hangup' bound to '#'.
The question I have - is how do I change that default caller control action
if it is not in conference.conf.xml ??
caller-controls
group name=default
...
*control action=energy up digits=#/*
On Mon, Nov 23, 2009 at 2:35 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
issue
console loglevel debug
from the cli
then try again and see if there is any hint
On Mon, Nov 23, 2009 at 1:24 PM, Phillip Jones pjinthe...@gmail.comwrote:
Hi there,
I have created a simple conference that works great. The only problem is,
when a participant press # it exits the call. So when a user enters a
conference with a PIN, and by habit they enter 12345 followed by pound, it
puts them in and then straight out.
So I edited conference.conf.xml so:
!--control action=hangup digits=#/--
and even assigned # to another function:
control action=energy up digits=#/
and the same occurs. Pressing # exits the conference.
What am I missing here?
tia - phil
Conf Setup:
extension name=conference.conf
condition field=destination_number expression=^(2125556625)$
action application=answer/
action application=conference data=$1-${domain_na...@default
+123/
/condition
/extension
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Re: [Freeswitch-users] Simplest of Conference Setup questions
On Mon, Nov 23, 2009 at 12:17 PM, Phillip Jones pjinthe...@gmail.comwrote: Thanks for replying. Well in the log I see: 2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760 2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving conference, cause: NONE which make sense because just above I see: 009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default caller control action 'hangup' bound to '#'. The question I have - is how do I change that default caller control action if it is not in conference.conf.xml ?? caller-controls group name=default ... *control action=energy up digits=#/* I believe that this is because the caller-controls param is commented out in the default profile config. I prefer not to mess w/ the default configs which is why I recommended the custom configs in my previous email... -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] FS dies after some minutes
Hi, I did a new installation with the trunk from Saturday (21. Nov.) and it always dies with a core after 5-10 minutes. It happened several times. After that I did a new installation of 1.0.4 and this runs without problems on the same host. I'm using Ubuntu 8.04 Server with all patches. Anyone else experiencing this problem? Thanks, Klaus Here's the bt: #0 0x7f7aa2eb22fc in sofia_reg_nonce_callback (pArg=0x40f3ca50, argc=value optimized out, argv=0x7f7a9c006758, columnNames=value optimized out) at ../../../../src/include/switch_utils.h:78 #1 0x7f7aa91f4a12 in sqlite3_exec (db=0x7f7a9c00a6a0, zSql=0x7f7a9c006cd0 select nonce from sip_authentication where nonce='b7ed6efa-d801-11de-a716-67cbb4a551f8', xCallback=0x7f7aa2eb22d0 sofia_reg_nonce_callback, pArg=0x40f3ca50, pzErrMsg=0x40f3c680) at ./src/legacy.c:95 #2 0x7f7aa917b98d in switch_core_db_exec (db=0x7f7a9c00a6a0, sql=0x7f7a9c006cd0 select nonce from sip_authentication where nonce='b7ed6efa-d801-11de-a716-67cbb4a551f8', callback=0x7f7aa2eb22d0 sofia_reg_nonce_callback, data=0x40f3ca50, errmsg=0x40f3c6e8) at src/switch_core_db.c:93 #3 0x7f7aa2e985b1 in sofia_glue_execute_sql_callback (profile=0x72e940, mutex=0x0, sql=0x7f7a9c006cd0 select nonce from sip_authentication where nonce='b7ed6efa-d801-11de-a716-67cbb4a551f8', callback=0x7f7aa2eb22d0 sofia_reg_nonce_callback, pdata=0x40f3ca50) at sofia_glue.c:4297 #4 0x7f7aa2ead8ec in sofia_reg_parse_auth (profile=0x72e940, authorization=0x7f7a9c078ad0, sip=0x7f7a9c0695d8, regstr=0x7f7aa2fe7137 REGISTER, np=0x40f3d940 b7ed6efa-d801-11de-a716-67cbb4a551f8, nplen=128, ip=0x40f3d840 10.134.38.59, v_event=0x40f3d930, exptime=3600, regtype=REG_REGISTER, to_user=0x7f7a9c0dd18e 29, auth_params=0x40f3cd60, reg_count=0x40f3cd58) at sofia_reg.c:1704 #5 0x7f7aa2eb004a in sofia_reg_handle_register (nua=0x7f7a9c006810, profile=0x72e940, nh=0x7f7a9c0cdb20, sip=0x7f7a9c0695d8, regtype=REG_REGISTER, key=0x40f3d940 b7ed6efa-d801-11de-a716-67cbb4a551f8, keylen=0, v_event=0x40f3d930, is_nat=0x0) at sofia_reg.c:888 #6 0x7f7aa2eb2f1c in sofia_reg_handle_sip_i_register (nua=0x7f7a9c006810, profile=0x72e940, nh=0x7f7a9c0cdb20, sofia_private=value optimized out, sip=0x7f7a9c0695d8, tags=value optimized out) at sofia_reg.c:1362 #7 0x7f7aa2e9371c in sofia_event_callback (event=value optimized out, status=100, phrase=0x7f7a9c071700 Trying, nua=0x7f7a9c006810, profile=0x72e940, nh=0x7f7a9c0cdb20, sofia_private=0x0, sip=0x7f7a9c0695d8, tags=0x7f7a9c0716f0) at sofia.c:672 #8 0x7f7aa2f1119e in nua_application_event (dummy=0x0, sumsg=0x40f3dd10, ee=0x7f7a9c0716c8) at nua_stack.c:393 #9 0x7f7aa2f7e2c1 in su_base_port_execute_msgs (queue=0x0) at su_base_port.c:280 #10 0x7f7aa2f7e039 in su_base_port_getmsgs (self=0x721320) at su_base_port.c:202 #11 0x7f7aa2f7e5dc in su_base_port_step (self=0x721320, tout=0) at su_base_port.c:473 #12 0x7f7aa2f7b68e in su_port_step (self=0x721320, tout=1000) at su_port.h:340 #13 0x7f7aa2f7b656 in su_root_step (self=0x723320, tout=1000) at su_root.c:858 #14 0x7f7aa2e8d40a in sofia_profile_thread_run (thread=value optimized out, obj=value optimized out) at sofia.c:1194 #15 0x7f7aa88b63f7 in start_thread () from /lib/libpthread.so.0 #16 0x7f7aa7e20b4d in clone () from /lib/libc.so.6 #17 0x in ?? () ___ 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] Simplest of Conference Setup questions
Michael that for the reply. I created a new group with # unbound and referenced it from the default profile: param name=caller-controls value=myConf/ And that worked fine. Strangely though, changing the default group and referencing that from the default profile does not. param name=caller-controls value=default/ Do you want me to test this on the latest trunk or is this as expected? Phil On Mon, Nov 23, 2009 at 3:12 PM, Michael Collins m...@freeswitch.org wrote: On Mon, Nov 23, 2009 at 11:24 AM, Phillip Jones pjinthe...@gmail.comwrote: Hi there, I have created a simple conference that works great. The only problem is, when a participant press # it exits the call. So when a user enters a conference with a PIN, and by habit they enter 12345 followed by pound, it puts them in and then straight out. So I edited conference.conf.xml so: !--control action=hangup digits=#/-- and even assigned # to another function: control action=energy up digits=#/ and the same occurs. Pressing # exits the conference. What am I missing here? tia - phil Phil, I recommend that you create a custom profile and a custom caller control group. Just copy the defaults and rename them to something meaningful. In conference.conf.xml you can add a new call control group like this: group name=custom !-- notice the new name -- control action=mute digits=0/ control action=deaf mute digits=*/ control action=energy up digits=9/ control action=energy equ digits=8/ control action=energy dn digits=7/ control action=vol talk up digits=3/ control action=vol talk zero digits=2/ control action=vol talk dn digits=1/ control action=vol listen up digits=6/ control action=vol listen zero digits=5/ control action=vol listen dn digits=4/ !-- Notice that I removed the hangup option; # digit is not bound to anything; you can bind it to something else if you wish -- /group Then make a copy of the default profile changing the profile name and the caller-controls parameter: profile name=custom !-- notice the new name -- !-- snip -- param name=caller-controls value=some name/ !-- snip -- /profile Give that a whirl and report back. :) -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] Simplest of Conference Setup questions
Anthony - setting control action=hangup digits=9/ or control action=hangup digits=event/ does not make a difference, even when the default profile has param name=caller-controls value=default/ un-commented. Looks to me like that default group is ignored even when specifically referred to? As Michael says though, creating a specific group: group name=myConf and adding param name=caller-controls value=myConf/ in the default profile works a charm. I am good - but let me know if you want me to try anything else. Phil On Mon, Nov 23, 2009 at 3:27 PM, Michael Collins m...@freeswitch.org wrote: On Mon, Nov 23, 2009 at 12:17 PM, Phillip Jones pjinthe...@gmail.comwrote: Thanks for replying. Well in the log I see: 2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760 2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving conference, cause: NONE which make sense because just above I see: 009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default caller control action 'hangup' bound to '#'. The question I have - is how do I change that default caller control action if it is not in conference.conf.xml ?? caller-controls group name=default ... *control action=energy up digits=#/* I believe that this is because the caller-controls param is commented out in the default profile config. I prefer not to mess w/ the default configs which is why I recommended the custom configs in my previous email... -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] INCOMPATIBLE_DESTINATION on 180 Ringing
thank you for your answer the relevant part of the log is 2009-11-23 21:46:49.625130 [NOTICE] sofia.c:3693 Pre-Answer sofia/interconnect/24785214448370...@38.105.229.100! 2009-11-23 21:46:49.625130 [INFO] sofia.c:3706 Sending early media 2009-11-23 21:46:49.625130 [ERR] sofia_glue.c:2029 No audio codec available 2009-11-23 21:46:49.625130 [NOTICE] switch_channel.c:2048 Hangup sofia/interconnect/nob...@81.94.55.100 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] it's the same with g729 and alaw (refering to brian) in my opinion the ringing here should be generated near end and no audio codec has to be used here (180 ringing) br On 2009-11-23 20:45, Anthony Minessale wrote: you need to provide a FS console trace of your problem from your FS source dir (build root) cd libs/esl make perlmod cd perl perl logger.pl http://logger.pl -pb christian reproduce then hit ctl-c and tell me the url it posted to. 2009/11/23 Christian Löschenkohl christian.loeschenk...@xpirio.com mailto:christian.loeschenk...@xpirio.com hi our freeswitch server has to talk to a sonus ip-switch when we want to setup a call we do get a 100 Trying and then a 180 Ringing within the 180 Ringing we get a sdp with a=sendonly then our freeswitch quits with a CANCEL message. i simply don't get why our freeswitch aborts the session - i think it would work if no a=sendonly would be present in the sdp. my technical contact doesn't want to switch 180 to 183 on the sonus side - this would also work (i think). in fact he says that 180 ringing is vaild, he isn't that wrong in this case. our freeswitch works in proxy mode, we do use trunk 15396 see a ngrep trace under http://pastebin.freeswitch.org/11235 92.63.208.36 - freeswitch 38.105.229.100 - sonus br -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com mailto:christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@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 -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.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] INCOMPATIBLE_DESTINATION on 180 Ringing
thany ou for your answer we use g729 on all our other connections in passthrough mode and it also doesn't work with alaw. so i don't think it's related to this. br On 2009-11-23 20:48, Brian West wrote: Well its also G729 so I suspect you don't have G729 /b On Nov 23, 2009, at 1:17 PM, Christian Löschenkohl wrote: hi our freeswitch server has to talk to a sonus ip-switch when we want to setup a call we do get a 100 Trying and then a 180 Ringing within the 180 Ringing we get a sdp with a=sendonly then our freeswitch quits with a CANCEL message. i simply don't get why our freeswitch aborts the session - i think it would work if no a=sendonly would be present in the sdp. my technical contact doesn't want to switch 180 to 183 on the sonus side - this would also work (i think). in fact he says that 180 ringing is vaild, he isn't that wrong in this case. our freeswitch works in proxy mode, we do use trunk 15396 see a ngrep trace under http://pastebin.freeswitch.org/11235 92.63.208.36 - freeswitch 38.105.229.100 - sonus br -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.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 -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.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] INCOMPATIBLE_DESTINATION on 180 Ringing
do you have the ringback variable set on the channel? if so it will cause 180 to attempt to play inband ringback indication I have nothing left to say because I asked for the whole log with the siptrace enables not just 5 lines of it. If you still want help, give me the log to examine and I will tell you what your problem is. 2009/11/23 Christian Löschenkohl christian.loeschenk...@xpirio.com thany ou for your answer we use g729 on all our other connections in passthrough mode and it also doesn't work with alaw. so i don't think it's related to this. br On 2009-11-23 20:48, Brian West wrote: Well its also G729 so I suspect you don't have G729 /b On Nov 23, 2009, at 1:17 PM, Christian Löschenkohl wrote: hi our freeswitch server has to talk to a sonus ip-switch when we want to setup a call we do get a 100 Trying and then a 180 Ringing within the 180 Ringing we get a sdp with a=sendonly then our freeswitch quits with a CANCEL message. i simply don't get why our freeswitch aborts the session - i think it would work if no a=sendonly would be present in the sdp. my technical contact doesn't want to switch 180 to 183 on the sonus side - this would also work (i think). in fact he says that 180 ringing is vaild, he isn't that wrong in this case. our freeswitch works in proxy mode, we do use trunk 15396 see a ngrep trace under http://pastebin.freeswitch.org/11235 92.63.208.36 - freeswitch 38.105.229.100 - sonus br -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.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 -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@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] Simplest of Conference Setup questions
The behavior of not being able to change the default caller controls are documented on the wiki: http://wiki.freeswitch.org/wiki/Mod_conference#.3Ccaller-controls.3E *Reserved Group Names* - none - Use this name to prevent installing caller-controls for callers of a conference. - default - Use this name to utilize the hard-coded set of controls built-in to mod_conference. Do NOT name a custom set of conference-controls default as they will be overridden with the hard-coded set. The behavior of the default group is defined below: On Mon, Nov 23, 2009 at 2:42 PM, Phillip Jones pjinthe...@gmail.com wrote: Anthony - setting control action=hangup digits=9/ or control action=hangup digits=event/ does not make a difference, even when the default profile has param name=caller-controls value=default/ un-commented. Looks to me like that default group is ignored even when specifically referred to? As Michael says though, creating a specific group: group name=myConf and adding param name=caller-controls value=myConf/ in the default profile works a charm. I am good - but let me know if you want me to try anything else. Phil On Mon, Nov 23, 2009 at 3:27 PM, Michael Collins m...@freeswitch.orgwrote: On Mon, Nov 23, 2009 at 12:17 PM, Phillip Jones pjinthe...@gmail.comwrote: Thanks for replying. Well in the log I see: 2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760 2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving conference, cause: NONE which make sense because just above I see: 009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default caller control action 'hangup' bound to '#'. The question I have - is how do I change that default caller control action if it is not in conference.conf.xml ?? caller-controls group name=default ... *control action=energy up digits=#/* I believe that this is because the caller-controls param is commented out in the default profile config. I prefer not to mess w/ the default configs which is why I recommended the custom configs in my previous email... -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 -- -Rupa ___ 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] Building in a builddir using --srcdir optionbut modules still build in srcdir
In these builds how is it supposed to work, do generated files like Makefiles get put it builddir or srcdir? Mike On Nov 23, 2009, at 2:54 PM, Robert Hadley wrote: Thanks Mike. modmake.rules is created in the $(switch_builddir)/build. What I see as the problem is in src/mod/Makefile.am There is a statement line 12 that points moddir to the source if test –d “$(switch_srcdir)/src/mod/$$confmoddir” ; then \ moddir = “$(switch_srcdir)/src/mod/$$confmoddir” ; And then the statements starting around line 22 that cd to moddir (in src) and fire off make if test –f “$$moddir/Makefile” ; then \ß Yep, this will be true cd $$moddir … $(MAKE) I’m not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). Does anybody have a better idea? Thanks, Robert From: Michael Jerris [mailto:m...@jerris.com] Sent: Monday, November 23, 2009 11:16 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? ___ 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] Simplest of Conference Setup questions
Default controls are hard coded. If you want to change them you must use a name other than default. Mike On Nov 23, 2009, at 3:42 PM, Phillip Jones wrote: Anthony - setting control action=hangup digits=9/ or control action=hangup digits=event/ does not make a difference, even when the default profile has param name=caller-controls value=default/ un-commented. Looks to me like that default group is ignored even when specifically referred to? As Michael says though, creating a specific group: group name=myConf and adding param name=caller-controls value=myConf/ in the default profile works a charm. I am good - but let me know if you want me to try anything else. Phil ___ 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] Building in a builddir using --srcdiroptionbut modules still build in srcdir
In typical automake builds the configure step takes the Makefile.am from the srcdir and generates the Makefile in the builddir. Most src/mod subdirs are not using automake and/or configure. They just have a simple Makefile in with the source. Robert _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Monday, November 23, 2009 1:09 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir In these builds how is it supposed to work, do generated files like Makefiles get put it builddir or srcdir? Mike On Nov 23, 2009, at 2:54 PM, Robert Hadley wrote: Thanks Mike. modmake.rules is created in the $(switch_builddir)/build. What I see as the problem is in src/mod/Makefile.am There is a statement line 12 that points moddir to the source if test -d $(switch_srcdir)/src/mod/$$confmoddir ; then \ moddir = $(switch_srcdir)/src/mod/$$confmoddir ; And then the statements starting around line 22 that cd to moddir (in src) and fire off make if test -f $$moddir/Makefile ; then \-- Yep, this will be true cd $$moddir . $(MAKE) I'm not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). Does anybody have a better idea? Thanks, Robert _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Monday, November 23, 2009 11:16 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? ___ 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] INCOMPATIBLE_DESTINATION on 180 Ringing
sorry about wasting your time (wasn't my intent) the log is at http://pastebin.freeswitch.org/11240 i called 5214448370068 (also other calls are in the log) they now have changed 180 to 183 on the sonus, but makes no difference here br On 2009-11-23 22:07, Anthony Minessale wrote: do you have the ringback variable set on the channel? if so it will cause 180 to attempt to play inband ringback indication I have nothing left to say because I asked for the whole log with the siptrace enables not just 5 lines of it. If you still want help, give me the log to examine and I will tell you what your problem is. 2009/11/23 Christian Löschenkohl christian.loeschenk...@xpirio.com mailto:christian.loeschenk...@xpirio.com thany ou for your answer we use g729 on all our other connections in passthrough mode and it also doesn't work with alaw. so i don't think it's related to this. br On 2009-11-23 20:48, Brian West wrote: Well its also G729 so I suspect you don't have G729 /b On Nov 23, 2009, at 1:17 PM, Christian Löschenkohl wrote: hi our freeswitch server has to talk to a sonus ip-switch when we want to setup a call we do get a 100 Trying and then a 180 Ringing within the 180 Ringing we get a sdp with a=sendonly then our freeswitch quits with a CANCEL message. i simply don't get why our freeswitch aborts the session - i think it would work if no a=sendonly would be present in the sdp. my technical contact doesn't want to switch 180 to 183 on the sonus side - this would also work (i think). in fact he says that 180 ringing is vaild, he isn't that wrong in this case. our freeswitch works in proxy mode, we do use trunk 15396 see a ngrep trace under http://pastebin.freeswitch.org/11235 92.63.208.36 - freeswitch 38.105.229.100 - sonus br -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com mailto:christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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 mailto: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 -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com mailto:christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@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 -- Ing. Christian Löschenkohl
Re: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir
I'll work on this, can you open me up a bug on http://jira.freeswitch.org in regards to this please. Mike On Nov 23, 2009, at 4:19 PM, Robert Hadley wrote: In typical automake builds the configure step takes the Makefile.am from the srcdir and generates the Makefile in the builddir. Most src/mod subdirs are not using automake and/or configure. They just have a simple Makefile in with the source. Robert From: Michael Jerris [mailto:m...@jerris.com] Sent: Monday, November 23, 2009 1:09 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir In these builds how is it supposed to work, do generated files like Makefiles get put it builddir or srcdir? Mike On Nov 23, 2009, at 2:54 PM, Robert Hadley wrote: Thanks Mike. modmake.rules is created in the $(switch_builddir)/build. What I see as the problem is in src/mod/Makefile.am There is a statement line 12 that points moddir to the source if test –d “$(switch_srcdir)/src/mod/$$confmoddir” ; then \ moddir = “$(switch_srcdir)/src/mod/$$confmoddir” ; And then the statements starting around line 22 that cd to moddir (in src) and fire off make if test –f “$$moddir/Makefile” ; then \ß Yep, this will be true cd $$moddir … $(MAKE) I’m not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). Does anybody have a better idea? Thanks, Robert From: Michael Jerris [mailto:m...@jerris.com] Sent: Monday, November 23, 2009 11:16 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? ___ 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] Memory leak with mod_local_stream
Hey guys, Having a problem with mod_local_stream. I recently did a make current from 15334 to the latest trunk (15630). After restarting, there now appears to be a memory leak. On a test system (CentOS 5.4, 64-bit) with no calls or registrations, Freeswitch gradually consumes all of the host memory (rate of about 200K/second), then swaps out, eventually rendering the system useless. I isolated it to mod_local_stream. If I unload mod_local_stream, the memory use stops climbing. If I re-load mod_local_stream, it starts again. I would submit the logs except they aren't any besides it starting. The system is just sitting there idle. Even valgrind didn't show much (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- check=full --leak-resolution=high --show-reachable=yes .libs/ freeswitch -vg Questions: * has anyone else seen this? * what is the best way I can assist troubleshooting this? I saw a patch to mod_local_stream (rev 15431) a few weeks back. Could that have anything to do with it? Rob ___ 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] Memory leak with mod_local_stream
if you suspect 15431 to have caused this, then revert to 15430 and see if the problem exists. if you can narrow do the bug to a specific svn revision, then you greatly assist in the resolution of the issue. apart from that im not much help sorry. maybe someone else can lab it up and see if they get the same result. ( Im on a train now, so not so easy :P ) J On 24/11/2009, at 7:53 AM, Rob Forman wrote: Hey guys, Having a problem with mod_local_stream. I recently did a make current from 15334 to the latest trunk (15630). After restarting, there now appears to be a memory leak. On a test system (CentOS 5.4, 64-bit) with no calls or registrations, Freeswitch gradually consumes all of the host memory (rate of about 200K/second), then swaps out, eventually rendering the system useless. I isolated it to mod_local_stream. If I unload mod_local_stream, the memory use stops climbing. If I re-load mod_local_stream, it starts again. I would submit the logs except they aren't any besides it starting. The system is just sitting there idle. Even valgrind didn't show much (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- check=full --leak-resolution=high --show-reachable=yes .libs/ freeswitch -vg Questions: * has anyone else seen this? * what is the best way I can assist troubleshooting this? I saw a patch to mod_local_stream (rev 15431) a few weeks back. Could that have anything to do with it? Rob ___ 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] Memory leak with mod_local_stream
That rev should have fixed that memory leak, could you test mod_local_stream.c from rev 15430 (http://fisheye.freeswitch.org/browse/~raw,r=15430/FreeSWITCH/src/mod/formats/mod_local_stream/mod_local_stream.c) with your current fs version to confirm this is the cause please? Mike On Nov 23, 2009, at 4:53 PM, Rob Forman wrote: Hey guys, Having a problem with mod_local_stream. I recently did a make current from 15334 to the latest trunk (15630). After restarting, there now appears to be a memory leak. On a test system (CentOS 5.4, 64-bit) with no calls or registrations, Freeswitch gradually consumes all of the host memory (rate of about 200K/second), then swaps out, eventually rendering the system useless. I isolated it to mod_local_stream. If I unload mod_local_stream, the memory use stops climbing. If I re-load mod_local_stream, it starts again. I would submit the logs except they aren't any besides it starting. The system is just sitting there idle. Even valgrind didn't show much (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- check=full --leak-resolution=high --show-reachable=yes .libs/ freeswitch -vg Questions: * has anyone else seen this? * what is the best way I can assist troubleshooting this? I saw a patch to mod_local_stream (rev 15431) a few weeks back. Could that have anything to do with it? Rob ___ 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] Building in a builddir using--srcdiroptionbut modules still build in srcdir
Thanks Mike, How is the easy way to give you the changes I found so far? There are around 10 changes in 30 files (all configure.gnu files need a fix). Robert _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Monday, November 23, 2009 1:42 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using--srcdiroptionbut modules still build in srcdir I'll work on this, can you open me up a bug on http://jira.freeswitch.org in regards to this please. Mike On Nov 23, 2009, at 4:19 PM, Robert Hadley wrote: In typical automake builds the configure step takes the Makefile.am from the srcdir and generates the Makefile in the builddir. Most src/mod subdirs are not using automake and/or configure. They just have a simple Makefile in with the source. Robert _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Monday, November 23, 2009 1:09 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir In these builds how is it supposed to work, do generated files like Makefiles get put it builddir or srcdir? Mike On Nov 23, 2009, at 2:54 PM, Robert Hadley wrote: Thanks Mike. modmake.rules is created in the $(switch_builddir)/build. What I see as the problem is in src/mod/Makefile.am There is a statement line 12 that points moddir to the source if test -d $(switch_srcdir)/src/mod/$$confmoddir ; then \ moddir = $(switch_srcdir)/src/mod/$$confmoddir ; And then the statements starting around line 22 that cd to moddir (in src) and fire off make if test -f $$moddir/Makefile ; then \-- Yep, this will be true cd $$moddir . $(MAKE) I'm not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). Does anybody have a better idea? Thanks, Robert _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Monday, November 23, 2009 11:16 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? ___ 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] Building in a builddir using--srcdiroptionbut modules still build in srcdir
go to the src root and type: svn diff patch.diff then open a jira and attach patch.diff /b On Nov 23, 2009, at 4:21 PM, Robert Hadley wrote: Thanks Mike, How is the easy way to give you the changes I found so far? There are around 10 changes in 30 files (all configure.gnu files need a fix). Robert ___ 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] register timeout / cisco 7960
hi there, I have set up some cisco 7960 up with fs. They work fine - but the only way I can keep them registered is to set the timer_register_expires in the Cisco cfg file to something really short like 10s. Does anyone know the default register timeout for fs? And where I might change this in fs? Thanks! Phil ___ 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 Orgination_caller_id
Is there any way to set the origination_caller_id for a FIFO outbound call to an on-hook agent? I can't find anything in the wiki about a FIFO or member variable to set this. It seems to be set to 'Queue' by default, and appears to be hardcoded in the module source. It would be nice to be able to change per FIFO queue. That way agents that handle multiple companies can more easily see which queue is calling and answer accordingly. It is not a big deal, since it does automatically set the origination_caller_id_number to 'fifo+fifo name'. However, depending on the phone, the caller ID number is not always readily shown, and must be looked for. Thanks to anyone who has some insight on this, -Adam ___ 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] tcp call misses sip message
OK, this is what I've got. First, I've updated FreeSwitch from trunk to version 15630 and deployed it to my server. Performed a tets and again no magic happened. The link to SIP trace is below. Then I've installed 1.0.4 version to another server (virtual hosting), and performed tha same. And everything went OK. This server's log is below as well. Not working - http://pastebin.com/m2e97985d Working - http://pastebin.com/m3c1e6bfe Also in both cases there is a strange detail - clients' SIP ports are configured to be 5060 and 5061, but what can be seen in trace differs from these values whereas stun resolution shows that there is no NAT (clients connect with ADSL modem). Regards, Robert ___ 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] Memory leak with mod_local_stream
I tried mod_local_stream.c from rev 15430, did a make clean make all make install-- but it didn't fix it so it wasn't that patch. I'll make current and try valgrind again unless someone has other ideas. Rob On Nov 23, 2009, at 4:15 PM, Michael Jerris wrote: That rev should have fixed that memory leak, could you test mod_local_stream.c from rev 15430 (http://fisheye.freeswitch.org/browse/ ~raw,r=15430/FreeSWITCH/src/mod/formats/mod_local_stream/ mod_local_stream.c) with your current fs version to confirm this is the cause please? Mike On Nov 23, 2009, at 4:53 PM, Rob Forman wrote: Hey guys, Having a problem with mod_local_stream. I recently did a make current from 15334 to the latest trunk (15630). After restarting, there now appears to be a memory leak. On a test system (CentOS 5.4, 64-bit) with no calls or registrations, Freeswitch gradually consumes all of the host memory (rate of about 200K/second), then swaps out, eventually rendering the system useless. I isolated it to mod_local_stream. If I unload mod_local_stream, the memory use stops climbing. If I re-load mod_local_stream, it starts again. I would submit the logs except they aren't any besides it starting. The system is just sitting there idle. Even valgrind didn't show much (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- check=full --leak-resolution=high --show-reachable=yes .libs/ freeswitch -vg Questions: * has anyone else seen this? * what is the best way I can assist troubleshooting this? I saw a patch to mod_local_stream (rev 15431) a few weeks back. Could that have anything to do with it? Rob ___ 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] tcp call misses sip message
You know what, guys? I've just made it working be opening ALL tcp trafic in and out from server by adding two match-all ip filters into local security policy. I can't say I like this solution... Why did this problem appeared with policy matching exact (sofia profiles) ports? Regards, Robert. ___ 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 Orgination_caller_id
if you add {origination_caller_id_name=foo,origination_caller_id_number=123}
before the static entries for the on hook agent it will prevail over the
default one.
If you are using 1.0.4, this feature is only available in trunk or one of
the 1.0.5 pre releases.
On Mon, Nov 23, 2009 at 4:49 PM, Adam Ford li...@redbonez.net wrote:
Is there any way to set the origination_caller_id for a FIFO outbound
call to an on-hook agent? I can’t find anything in the wiki about a FIFO or
member variable to set this. It seems to be set to ‘Queue’ by default, and
appears to be hardcoded in the module source. It would be nice to be able
to change per FIFO queue. That way agents that handle multiple companies
can more easily see which queue is calling and answer accordingly.
It is not a big deal, since it does automatically set the
origination_caller_id_number to ‘fifo+fifo name’. However, depending on
the phone, the caller ID number is not always readily shown, and must be
looked for.
Thanks to anyone who has some insight on this,
-Adam
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
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MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
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IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] conference digits and conference control
On Thu, Oct 15, 2009 at 3:44 AM, god.nirvana god.nirv...@gmail.com wrote: hi all how can i get the digits when users in the conference?? and,in conference.conf.xml control action=mute digits=0/ the action will set another value?e.g:transfer? thanks I'm not sure I understand your question, but the wiki covers actions on keystrokes. If you need the user to dial other digits after the caller control then route the call to an extension that asks the user for input, like with play_and_get_digits, and handle the call accordingly. As far as the question about about setting another value - can you expound upon that a bit? I'm not sure what you're trying to accomplish. -MC P.S. - http://wiki.freeswitch.org/wiki/Mod_conference#.3Ccaller-controls.3E ___ 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] Help Freeswitch with Voipuser Gateway
Has anyone got any suggestion how I can set up a gateway to receive
incoming call on extension 1001 please.
Any generic conf file will do. my username with my gateway is s=say
qwerty and password ytrewq
I have used the intruction from the link below without success.
Thanks.
Otis wrote:
Hello
Could anyone point out what I have missed please ?
At the moment I configured a gateway voipuser as described here
http://www.onlinesolution.co.nz/viewtopic.php?p=119 :
Any suggestion as to what path I can take will be highly welcome
Thanks
.
Sam Abekah-Mensah wrote:
div class=moz-text-flowed style=font-family: -moz-fixedHi Michael
Thanks
I had set it to send incoming calls to extension 1001. This is in the
file abeka.xml in /usr/local/freeswitch/conf/dialplan/public directory.
The contents are :
extension name=inbound-*userna...@sip.voipuser.org]
condition field=destination_number expression=08444846450
action application=transfer data=1001 XML default/
/condition
/extension
Is there
anything wrong with this please ?
Thanks
Michal Bielicki wrote:
Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah:
I need help as I cannot receive calls through VOIPUSER. This is a
learning setup Attached are my conf files. What is wrong with them
? When I dial from a landline I get a continuous beep.
Attached are my gateway and the conf file to transfer. Sopfia
Status is my screen message. I can see a FAIL and cannot make head
or tail of all that message. Hopefully anyone using voipuser or in
fact any of you clever folks can make sense of this.
Thanks for your time.
2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate Buggy
RFC2833 Mode!
2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec
Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec
Compare [PCMA:8:8000:0]/[PCMA:8:8000:20]
2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec
sofia/external/nob...@213.166.5.133 PCMA/8000 20 ms 160 samples
2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf
payload to 101
2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455
(sofia/external/nob...@213.166.5.133) State Change CS_NEW - CS_INIT
2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398
(sofia/external/nob...@213.166.5.133) Running State Change CS_INIT
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481
(sofia/external/nob...@213.166.5.133) State INIT
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83
sofia/external/nob...@213.166.5.133 SOFIA INIT
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111
(sofia/external/nob...@213.166.5.133) State Change CS_INIT -
CS_ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481
(sofia/external/nob...@213.166.5.133) State INIT going to sleep
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398
(sofia/external/nob...@213.166.5.133) Running State Change CS_ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484
(sofia/external/nob...@213.166.5.133) State ROUTING
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130
sofia/external/nob...@213.166.5.133 SOFIA ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78
sofia/external/nob...@213.166.5.133 Standard ROUTING
2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 Processing
anonymous-abeka in context public
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-unloop] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-outside_call] continue=true
Dialplan: sofia/external/nob...@213.166.5.133 Absolute Condition
[outside_call]
Dialplan: sofia/external/nob...@213.166.5.133 Action
set(outside_call=true)
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-call_debug] continue=true
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-public_extensions] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[public_extensions] destination_number(abeka) =~ /^(10[01][0-9])$/
break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-public_did] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[public_did] destination_number(abeka) =~ /^(5551212)$/ break=on-false
Dialplan:
[Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH
I actually checked out the latest version of FreeSWITCH in the SVN repository.
I have the following configured in
/usr/local/freeswitch/conf/dialplan/default.xml:
extension name=setup_media continue=true
condition field=${sip_nat_detected} expression=true
action application=set data=proxy_media=true /
action application=set data=bypass_media=false /
anti-action application=set data=proxy_media=false /
anti-action application=set data=bypass_media=true /
/condition
/extension
I have the following configured in /usr/local/freeswitch/conf/vars.xml:
X-PRE-PROCESS cmd=set
data=global_codec_prefs=G729,i...@20i,G722,PCMU,PCMA/
X-PRE-PROCESS cmd=set
data=outbound_codec_prefs=G729,i...@20i,G722,PCMU,PCMA/
Here is the SIP trace for the failing call:
Nov 23 17:55:05.245 CST: //-1//SIP/Msg/ccsipDisplayMsg:
Received:
INVITE
sip:+19725357...@ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246
SIP/2.0
v: SIP/2.0/UDP
65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24
Record-Route: sip:65.211.120.237:5060;lr
v: SIP/2.0/UDP
63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236
record-route: sip:63.77.76.236;lr
f:
sip:+19729831...@199.173.101.208:5060;user=phone;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a
t: sip:+19725357...@63.77.76.236:5060;user=phone
i: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585
CSeq: 1 INVITE
Max-Forwards: 16
k: 100rel, replaces
allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
v: SIP/2.0/UDP
DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208
m: sip:199.173.101.208:5060;transport=UDP
c: application/SDP
l: 210
P-Asserted-Identity: sip:9729831...@63.77.76.236;user=phone
Privacy: none
v=0
o=- 641026559 641026559 IN IP4 199.173.111.147
s=-
c=IN IP4 199.173.111.147
t=0 0
m=audio 33344 RTP/AVP 18 0 8 101
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 23 17:55:05.257 CST: //-1//SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24,SIP/2.0/UDP
63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236,SIP/2.0/UDP
DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208
From:
sip:+19729831...@199.173.101.208:5060;user=phone;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a
To: sip:+19725357...@63.77.76.236:5060;user=phone
Date: Mon, 23 Nov 2009 23:55:05 GMT
Call-ID: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Nov 23 17:55:05.257 CST: //-1//SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:19725357...@168.75.202.212:5062 SIP/2.0
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3
From: sip:19729831...@168.75.202.246;tag=105BD148-201C
To: sip:19725357...@168.75.202.212
Date: Mon, 23 Nov 2009 23:55:05 GMT
Call-ID: 74e5b003-d7c211de-a29ad9df-3419a...@168.75.202.246
Supported: timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1961129755-3619819998-2727664095-874095366
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1259020505
Contact: sip:19729831...@168.75.202.246:5060
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 15
P-Asserted-Identity: sip:19729831...@168.75.202.246
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 314
v=0
o=CiscoSystemsSIP-GW-UserAgent 5041 5861 IN IP4 168.75.202.246
s=SIP Call
c=IN IP4 199.173.111.147
t=0 0
m=audio 33344 RTP/AVP 18 0 8 101
c=IN IP4 199.173.111.147
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 23 17:55:05.261 CST: //-1//SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3
From: sip:19729831...@168.75.202.246;tag=105BD148-201C
To: sip:19725357...@168.75.202.212
Call-ID: 74e5b003-d7c211de-a29ad9df-3419a...@168.75.202.246
CSeq: 101 INVITE
Timestamp: 1259020505 0.000345
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M
Content-Length: 0
Nov 23 17:55:05.309 CST: //-1//SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3
From: sip:19729831...@168.75.202.246;tag=105BD148-201C
To: sip:19725357...@168.75.202.212;tag=DFKSy9Q5DK1Na
Call-ID: 74e5b003-d7c211de-a29ad9df-3419a...@168.75.202.246
CSeq: 101 INVITE
Contact: sip:19725357...@168.75.202.212:5062;transport=udp
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE,
Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH
What rev exactly?
/b
On Nov 23, 2009, at 6:19 PM, John Platts wrote:
I actually checked out the latest version of FreeSWITCH in the SVN
repository.
I have the following configured in /usr/local/freeswitch/conf/
dialplan/default.xml:
extension name=setup_media continue=true
condition field=${sip_nat_detected} expression=true
action application=set data=proxy_media=true /
action application=set data=bypass_media=false /
anti-action application=set data=proxy_media=false /
anti-action application=set data=bypass_media=true /
/condition
/extension
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Re: [Freeswitch-users] INCOMPATIBLE_DESTINATION on 180 Ringing
You forgot to set freeswitch to debug loglevel You need both of the following: console loglevel debug sofia profile internal siptrace on 2009/11/23 Christian Löschenkohl christian.loeschenk...@xpirio.com sorry about wasting your time (wasn't my intent) the log is at http://pastebin.freeswitch.org/11240 i called 5214448370068 (also other calls are in the log) they now have changed 180 to 183 on the sonus, but makes no difference here br On 2009-11-23 22:07, Anthony Minessale wrote: do you have the ringback variable set on the channel? if so it will cause 180 to attempt to play inband ringback indication I have nothing left to say because I asked for the whole log with the siptrace enables not just 5 lines of it. If you still want help, give me the log to examine and I will tell you what your problem is. 2009/11/23 Christian Löschenkohl christian.loeschenk...@xpirio.com mailto:christian.loeschenk...@xpirio.com thany ou for your answer we use g729 on all our other connections in passthrough mode and it also doesn't work with alaw. so i don't think it's related to this. br On 2009-11-23 20:48, Brian West wrote: Well its also G729 so I suspect you don't have G729 /b On Nov 23, 2009, at 1:17 PM, Christian Löschenkohl wrote: hi our freeswitch server has to talk to a sonus ip-switch when we want to setup a call we do get a 100 Trying and then a 180 Ringing within the 180 Ringing we get a sdp with a=sendonly then our freeswitch quits with a CANCEL message. i simply don't get why our freeswitch aborts the session - i think it would work if no a=sendonly would be present in the sdp. my technical contact doesn't want to switch 180 to 183 on the sonus side - this would also work (i think). in fact he says that 180 ringing is vaild, he isn't that wrong in this case. our freeswitch works in proxy mode, we do use trunk 15396 see a ngrep trace under http://pastebin.freeswitch.org/11235 92.63.208.36 - freeswitch 38.105.229.100 - sonus br -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com mailto:christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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 mailto: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 -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com mailto:christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.commsn%253aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.compaypal%253aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.orgsip%253a...@conference.freeswitch.org
Re: [Freeswitch-users] FIFO Orgination_caller_id
And because it's static string for on-hook members, it's hard to set
dynamically. For now, I'm using a callback way - whenever the sip client
answered the call, it fetch the real connected number from a http server.
That's not ideal because not only it add the complexity but also the callee
have no idea what the number is before answer.
The problem for on-hook agent is that it call the agent first, and then get
one customer from the fifo queue, so it is not possible to let the agent
know the real caller-id before answer. Ideas?
2009/11/24 Anthony Minessale anthony.miness...@gmail.com
if you add
{origination_caller_id_name=foo,origination_caller_id_number=123} before the
static entries for the on hook agent it will prevail over the default one.
If you are using 1.0.4, this feature is only available in trunk or one of
the 1.0.5 pre releases.
On Mon, Nov 23, 2009 at 4:49 PM, Adam Ford li...@redbonez.net wrote:
Is there any way to set the origination_caller_id for a FIFO outbound
call to an on-hook agent? I can’t find anything in the wiki about a FIFO or
member variable to set this. It seems to be set to ‘Queue’ by default, and
appears to be hardcoded in the module source. It would be nice to be able
to change per FIFO queue. That way agents that handle multiple companies
can more easily see which queue is calling and answer accordingly.
It is not a big deal, since it does automatically set the
origination_caller_id_number to ‘fifo+fifo name’. However, depending on
the phone, the caller ID number is not always readily shown, and must be
looked for.
Thanks to anyone who has some insight on this,
-Adam
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Re: [Freeswitch-users] FIFO Orgination_caller_id
On Mon, Nov 23, 2009 at 4:43 PM, Adam Ford li...@redbonez.net wrote: I actually tried that, as a guess, based on the configuration output of fifo list. However I am running a tarball release of 1.0.4, which would explain why it did not work for me. I appreciate the feedback, and will make a note to implement this when I update my installation. Are the svn-trunk updates pretty solid? I have not attempted an update yet, as it is a production system. Trunk has been very solid with a few minor exceptions. Best bet is to back up everything and do the upgrade during down time. If you have a test system that you can use as a sandbox that would be even better... -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] FIFO Orgination_caller_id
On Mon, Nov 23, 2009 at 4:58 PM, Seven Du dujinf...@gmail.com wrote: And because it's static string for on-hook members, it's hard to set dynamically. For now, I'm using a callback way - whenever the sip client answered the call, it fetch the real connected number from a http server. That's not ideal because not only it add the complexity but also the callee have no idea what the number is before answer. The problem for on-hook agent is that it call the agent first, and then get one customer from the fifo queue, so it is not possible to let the agent know the real caller-id before answer. Ideas? Tony and Brian were discussing this today. They bring up a really good point: do you want to risk having calls remain on hold as they bounce around looking for an agent? This can happen if you pre-determine which caller goes to which agent and the agent doesn't answer. I do understand why this feature matters to many people - it's how old school ACD systems work. However, mod_fifo is more efficient. It's hard to justify decreasing call routing efficiency in order to display the caller's info to the on-hook agent prior to answering. -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] FIFO Orgination_caller_id
You do realize that the whole concept is OLD skewl. You should be popping this info via external resources when the agent is bridged to the caller and the info is there before they are done saying thanks for calling spacely sprockets, this is George how may I help you /b On Nov 23, 2009, at 7:07 PM, Michael Collins wrote: Tony and Brian were discussing this today. They bring up a really good point: do you want to risk having calls remain on hold as they bounce around looking for an agent? This can happen if you pre- determine which caller goes to which agent and the agent doesn't answer. I do understand why this feature matters to many people - it's how old school ACD systems work. However, mod_fifo is more efficient. It's hard to justify decreasing call routing efficiency in order to display the caller's info to the on-hook agent prior to answering. -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] FIFO Orgination_caller_id
On Mon, Nov 23, 2009 at 5:12 PM, Brian West br...@freeswitch.org wrote: You do realize that the whole concept is OLD skewl. You should be popping this info via external resources when the agent is bridged to the caller and the info is there before they are done saying thanks for calling spacely sprockets, this is George how may I help you /b Agreed! Screen pop should be easy in the 21st Century. If it's not then you've got MUCH bigger problems than caller ID being delivered to your FIFO agents... -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] FIFO Orgination_caller_id
Yes, that's what we are doing. 2009/11/24 Brian West br...@freeswitch.org You do realize that the whole concept is OLD skewl. You should be popping this info via external resources when the agent is bridged to the caller and the info is there before they are done saying thanks for calling spacely sprockets, this is George how may I help you /b On Nov 23, 2009, at 7:07 PM, Michael Collins wrote: Tony and Brian were discussing this today. They bring up a really good point: do you want to risk having calls remain on hold as they bounce around looking for an agent? This can happen if you pre- determine which caller goes to which agent and the agent doesn't answer. I do understand why this feature matters to many people - it's how old school ACD systems work. However, mod_fifo is more efficient. It's hard to justify decreasing call routing efficiency in order to display the caller's info to the on-hook agent prior to answering. -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
