August 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_conference
by jay binks 24 Aug '10

24 Aug '10

is there a way ( ESL or API ) to list active conferences on an FS box. im playing with dynamic conferences, so they aren't pre-defined.. I COULD keep track using ESL but it would be nice to avoid tracking state if possible. -- Sincerely Jay

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fax ident and header variables
by François Delawarde 24 Aug '10

24 Aug '10

Anyone knows the meaning of the fax_ident and fax_header variables? Are those mandatory? What should go in there and how should it be formatted? The only thing I could find is that apparently in the USA, the fax header must contain the telephone number of the line where the fax is connected. Is this only in the USA? Thank you, François.

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Does anyone have a POTS phone connected to an openZAP supported card working?
by Jim 24 Aug '10

24 Aug '10

The phone can connect to other phones in the system but not to an outside line. Thanks, Jim.

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FreeSwitch - uuid find list dump
by Shamun toha md 24 Aug '10

24 Aug '10

Hi, i have few questions, don't know why its not working. Q. How do i get the uuid? To make a test from the CLI. freeswitch@example> sofia profile internal siptrace on Enabled sip debugging on internal freeswitch@example> uuid_exists 3f7f5dc2-af4f-11df-b704-69c5ea8291c5 false freeswitch@example> Q. When i register SIP phone, nonce = uuid? Authorization: Digest username="2001", realm="78.23.89.55", nonce="3f7f5dc2-af4f-11df-b704-69c5ea8291c5", uri="sip:78.23.89.55", algorithm=MD5, qop=auth, cnonce="339774495957948623487dc37177be3f", nc=00000001, response="8fa80654a0c7bfba71c81f5e9cbafee6"

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core.db table channels application field store wrong data when executing voicemail
by Antonio 24 Aug '10

24 Aug '10

Hi, during multiple selects in the table "channels" just realized that when a call is executing the app voicemail, to check for messages, during the navigation menu of the voicemail it changes to application "sleep". Is it normal? or is a bug for fs? The call: During the auth: sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101(a)192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| local|PCMU|8000|PCMU|8000||marces|101(a)192.168.10.25||EARLY|||| sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101(a)192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| local|PCMU|8000|PCMU|8000||marces|101(a)192.168.10.25||EARLY|||| sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101(a)192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| local|PCMU|8000|PCMU|8000||marces|101(a)192.168.10.25||EARLY|||| sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101(a)192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| local|PCMU|8000|PCMU|8000||marces|101(a)192.168.10.25||EARLY|||| sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101(a)192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| local|PCMU|8000|PCMU|8000||marces|101(a)192.168.10.25||EARLY|||| sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101(a)192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| local|PCMU|8000|PCMU|8000||marces|101(a)192.168.10.25||EARLY|||| in the navigation menu: sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101(a)192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|sleep|100|XML|local|PCMU|8000|PCMU|8000||marces| 101(a)192.168.10.25||EARLY|||| sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101(a)192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|sleep|100|XML|local|PCMU|8000|PCMU|8000||marces| 101(a)192.168.10.25||EARLY|||| Thanks, António

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scalability of FS vs yate
by jesse zhao 24 Aug '10

24 Aug '10

I am wondering whether someone did testing of FreeSwitch vs Yate under the same hardware system and testing profiles. Which one is more scalable? Any usage case of FS in large service providers or organization in terms of 100K subscribers? It is nice to list customers of FS in the wiki page. thanks! -jesse

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centos5.5 ready for FreeSWITCH?
by Seven Du 24 Aug '10

24 Aug '10

Hi, I see bluebox based on CentOS 5.5. Before that it is highly recommended on 5.3 over 5.4. So, is 5.5 safe for production use now? Thanks. -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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Phrase speak-text function return on first key press in phrase file on Windows
by Yitzchok 24 Aug '10

24 Aug '10

NOTE: I tested this on Centos and I wasn't able to reproduce it. So it looks like a windows problem or it might be that since I got one error when trying to compile FreeSWITCH on windows I tried to fix it with the fix on this jira bug report http://jira.freeswitch.org/browse/FSBUILD-296?focusedCommentId=20921&page=c… so maybe that is the problem. (I am using Flite for TTS) -- I have a macro (phrase) file that is set up like this. <macro name="test_tts"> <input pattern="(.*)"> <match> <action function="speak-text" data="*Please enter the extension then press the pound key.*"/> <action function="*play-file*" data="ivr/ivr-to_repeat_these_options.wav"/> <action function="play-file" data="ivr/ivr-please.wav"/> <action function="play-file" data="voicemail/vm-press.wav"/> <action function="play-file" data="digits/9.wav"/> <action function="play-file" data="silence_stream://2000"/> </match> </input> </macro> Then when I use this phrase file in the play_and_get_digits application, something like this. <action application="play_and_get_digits" data="1 4 3 7000 # phrase:test_tts phrase:test_play_file \d+"/> If I try to enter a number on the phone for example "234#" what happens here is that if it is in the middle of playing the *speak-text *part of the phrase file then it will just return "2" and stop right away (as if he is done) if "234#" is pressed when the *play-file* part of phrase file is playing then it will work as it should work (returning 234). Yitzchok

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performance between bridged call and conference
by Seven Du 24 Aug '10

24 Aug '10

Hi, Can someone explain the performance difference between bridged calls and 2-party conference? or just in the code point of view? Since in some scenarios third party may join into a bridged call, so we need to transfer a bridged call into a conference first. Make a conference anyway event for 2-parties will make logic simpler and clear. Thanks. -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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Newbie question
by Michael Scheidell 23 Aug '10

23 Aug '10

looked at FAQ's first, just want to clarify. Some background, looking at replacing a running sipx install due to limitations on ITSP's (conflicting ITSP and sipx limitations) Biggest limitation I see is that sipx without an external SBC forces the ITSP to send calls to port 5080. (Big Sip trunk providers like ATT, Level3 and Verizon don't want to do that. I actually understand why) in the firewall ports section it notes port 5060 for 'default' internal, 5070 for default nat and 5080 for default external. I suppose the biggest question is can I set up freeswitch to listen for SIP inbound trunk calls on port 5060 without loosing functionality (with sipx, if I do this, I can't forward calls: <http://list.sipfoundry.org/archive/sipx-users/msg23554.html> I don't want to interfere with normal sip: url inbound calls, so I would expect (in an 'askerisk like' setup) to be able to receive sip calls to internal extensions as well as SIP trunk calls to port 5060. (and be able to put both on hold, take them off hold, play moh to callers, transfer, blind and attended, and join a 3 way conference call) Next, looking for someone who has moved from sipx 4.2.0 to freeswitch/fusionpbx to quote me on a conversion (assuming I can use the above) POC would be to use someone like voip.ms and static registration (which uses port 5060). and have full regression testing. inbound calls, outbound calls, calls transfterd, (correct caller id shows up!) Currently, this only works on voip.ms user/password authentication since the invite/auth exchange will register port 5080. voip.ms has no way to specify the calling port, and even if they did, this is just the POC. Level3, Att and Verizon will ONLY send to port 5060. (ps, ISN dialing. seems to be tied in to e.164 dialing, but I was not able to actually tell if that is the case. on sipx, I added a two digit 'trunk' code '**' to make outbound ISN calls) -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 sip:michael.scheidell@secnap.com > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________

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