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404 error when adding new <extension> element to dialplan.
by Jian Yuan Peng 09 Jan '09

09 Jan '09

FYI, I have post message yesterday, and I did a quick tried on following step to add enum. I tried to use java sip client to make outbound call using gafachi or qwest. So, I added <extension name="matchpoint_caller"> <condition field="caller_id_number" expression="^16508101200$"/> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="effective_caller_id_number=16508101200"/> <action application="set" data="effective_caller_id_name=Matchpoint Inc."/> <action application="bridge" data="sofia/gateway/sip.gafachi.com/$1"/> </condition> </extension> First, created new file matchpoint_caller.xml, and put under conf/dialplan, (i also tried it at conf/dialplan/default) directory. For some reason, It never pickup this enum (404 error), when java sip sent the invite. Then remove the matchpoint_caller.xml and add the <extension name="matchpoint_caller"> to conf/dialplan/default.xml. The trick thing is that if put this element above <X-PRE-PROCESS cmd="include" data="default/*.xml"/> then 404 is gone. If I put it below <X-PRE-PROCESS cmd="include" data="default/*.xml"/> then i still got 404 error.

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Re: [Freeswitch-users] Compile Mod_Managed on Centos 5.2
by Tim B 09 Jan '09

09 Jan '09

MC was using 1.0.2. I don't have access to the machine right now. I can post the errors later tonight or try to build with latest build. Which do you recommend? Tim > > Are you using 1.0.2 or the latest SVN? Could you post the content of> the errors that you are getting?> > Thanks,> MC> > > I want to use C# with freeswitch. How do I go about compiling the> > mod_managed package on Centos 5.2? The wiki just shows how to compile it on> > windows. And I try doing a make in the package directory and get errors.> >> > Tim> > _________________________________________________________________ Windows Live™ Hotmail®: Chat. Store. Share. Do more with mail. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t1_hm_justgotbetter_how…

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SIP response code in Freeswitch
by Gopalakrishnan A.N 09 Jan '09

09 Jan '09

Hi, Is there any possibilities that Freeswitch may detect the SIP response code from the IP media gateway. -- Thank you with regards, Gopal,

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Re: [Freeswitch-users] Freeswitch-users Digest, Vol 31, Issue 44
by Adam Wilt 09 Jan '09

09 Jan '09

Thanks for the replies. I wrote a script in SpiderMonkey to place a call, and upon connct turn on vmd, play a "press 1" prompt to see if there is a human, and then play some more prompts. If a voicemail beep is heard it starts playing a different prompt. I tested this with almost 300 phone numbers, but I suspect a disproportionate number of these are cell phones. Among the calls not answered by a human (nobody pressed 1), about 60% of the calls failed to recognize a beep. My cell phone is T-Mobile, and it doesn't detect the beep for it. > On 1/8/09, Adam Wilt <wiltingtree(a)gmail.com> wrote: > > Hi, I have two issues I'd appreciate some help with. > > > > A) I'm testing VMD and I'm getting a success rate of well under 50%. I > know > > part of the reason is that some of the voicemail beeps it's encountering > are > > very short in length (I've noticed this for T-Mobile and Sprint > voicemails, > > and there may be others too), and it can't detect them. So my question > is > > about the notes in the vmd section of the wiki which states, "The > industry > > standard is 80% detection. This module if used properly should exceeds > the > > standard by a very wide margin". I'm curious about whether I'm using it > > properly, and what I can do to make it work better. Thanks for the help. > > > > B) When I place an outbound call and immediately play a prompt when the > call > > is answered, the prompt sounds garbled to the person answering the phone. > If > > I sleep for a second before playing the prompt, it sounds fine. Any idea > of > > what would cause this? > > > > Thanks, > > Adam > > > > Adam, > > Can I ask how you are "testing" vmd? Where? How? > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > >

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Compile Mod_Managed on Centos 5.2
by Tim B 09 Jan '09

09 Jan '09

I want to use C# with freeswitch. How do I go about compiling the mod_managed package on Centos 5.2? The wiki just shows how to compile it on windows. And I try doing a make in the package directory and get errors. Tim _________________________________________________________________ Windows Live™ Hotmail®: Chat. Store. Share. Do more with mail. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_hm_justgotbetter_explor…

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Trouble getting session.setInputCallback working.
by Erik Wickstrom 09 Jan '09

09 Jan '09

Hi all, I'm trying to get a setInputCallback function working with mod_python. I'm using a current svn checkout for my build and the hello world via call example from the wiki ( http://wiki.freeswitch.org/wiki/Mod_python#Hello_World_via_call ). I've tried repeatedly, but I can't get the callback function to execute. I keep pressing various digits on my phone, but none of them are being logged to the console. My understanding is that this callback is supposed to fire when a dtmf (123456789*#, right??) and also acts as an event handler for things like mod_vmd. Is this part of the wiki example working for anyone else? Any idea what I might be doing wrong? Thanks! Erik

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Hint for DTMF handling in sofia.c
by Helmut Kuper 09 Jan '09

09 Jan '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today we found a DTMF problem in an older FS build. We weren't able to pass "*" and "#" to PSTN. FS sents those chars as "0". I fixed that on my own. After that I searched in build 11055M to if it was already fixed in trunk. I found it was fixed in parts. I found that in INFO messages of application type "dtmf-relay" the rfc2833 DTMF events "*#ABCD" are now handled correct, but "EF" are still ignored. In INFO messages of application type "dtmf" EVENTS 0-9 are considered. The others are still ignored resp. changed to "0". So I would recommend to replace the code beginning in line 3535 <<<<<<<<<<<<<<<<<< if (*signal_ptr && (*signal_ptr == '*' || *signal_ptr == '#' || *signal_ptr == 'A' || *signal_ptr == 'B' || *signal_ptr == 'C' || *signal_ptr == 'D')) { dtmf.digit = *signal_ptr; } else { tmp = atoi(signal_ptr); dtmf.digit = switch_rfc2833_to_char(tmp); } <<<<<<<<<<<<<<<<<< with this two lines: >>>>>>>>>>>>>> tmp=switch_char_to_rfc2833(*signal_ptr); dtmf.digit = switch_rfc2833_to_char(tmp); >>>>>>>>>>>>>> This will handle all rfc2833 dtmf events in dtmf-relay INFO messages correctly. This is tested by me with snom and AVM clients. Further replace the line 3556 <<<<<<<<<<<<<< int tmp = atoi(sip->sip_payload->pl_data); <<<<<<<<<<<<<< with this line: >>>>>>>>>>>>>> int tmp=switch_char_to_rfc2833(*(sip->sip_payload->pl_data)); >>>>>>>>>>>>>> That should handle all rfc2833 DTMF-Events in dtmf INFO messages. This last patch isn't tested by me, cause I've no phone which genereates INFO messages of type applicatin/dtmf... best regards and a late "happy new year" to all of u! Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkll54QACgkQ4tZeNddg3dxJgwCfULx8X9hw6xDd1q8r/Iih5unx c6EAn1T/7GK56pLLrMi140x50HLnO6i1 =3ENR -----END PGP SIGNATURE-----

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Idea/Suggestion required
by shehzad p 09 Jan '09

09 Jan '09

Hi All, Is there a way in freeswitch, such that we can play sound and receive a DTMF from the end user without actually answering a call like in session progress stage. Basically, the system should accept the call and play a progress tone during call progress (SIP 183). The system should be able to collect DTMF input. All this should happen while in SIP 183 (Progress) and the call shouldn't change its state to CONNECT, which means, no 200 OK should be issued to the use via the system. This is more like a IVR based system, however it all happens during call progress and not after connect. Can anyone suggest any idea about this situation? Thanks in advance. -- View this message in context: http://www.nabble.com/Idea-Suggestion-required-tp21291002p21291002.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Fax Tone Detect
by Baskar 09 Jan '09

09 Jan '09

*Hi, I want to get the events for the tone detect but i cant able to get any events, Procedures i follow to done detect:* Step 1:I have added the line in default.xml <extension name="Local_Extension"> <condition field="destination_number" expression="^(1[0-9][0-9][0-9])$"> <action application="set" data="dialed_ext=$1"/> * <action application="tone_detect" data="fax 1100 r +5000 transfer fax XML default" /> ===>I have add this line * <action application="answer"/> <action application="info"/> <action application="export" data="Answer-State"/> <action application="log" data="INFO Answer-State [${Answer-State}]\n"/> <action application="record_session" data="$${base_dir}/recordings/${destination_number}_${caller_id_number}.wav"/> <action application="export" data="dialed_ext=$1"/> </condition> * I have added the line to detect fax.* <context name="default"> <extension name="fax"> <condition field="destination_number" expression="^fax$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="set" data="fax_file_name=/tmp/fax/${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.tiff"/> <action application="set" data="fax_mode=recv"/> <action application="socket" data="127.0.0.1:8084 async"/> <action application="hangup"/> </condition> </extension> </context> Step 2:Then reloaded freeswitch console using reloadxml command reloadxml API CALL [reloadxml()] output: +OK [Success] 2009-01-09 17:49:51 [INFO] mod_enum.c:806 event_handler() ENUM Reloaded 2009-01-09 17:49:51 [INFO] switch_time.c:656 switch_load_timezones() Timezone reloaded 530 definitions Step3:Through x-lite i have dial the fax no *(43951333)* Step4:I have passed events through event socket * event plain DETECTED_TONE* Content-Type: command/reply Reply-Text: +OK event listener enabled plain ====>But i did not get any events in the event socket when i dial the fax no Step 5:I have pasted the freeswitch log in the link * http://pastebin.freeswitch.org/6713* *I need to get the fax tone detect but i cant able to detect the fax tone. If any thing is wrong correct me and help to solve the problem.* -- Warm Regards, N.Baskar

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How does location of FreeSWITCH server matter?
by Miles Keaton 09 Jan '09

09 Jan '09

I'm new to FreeSWITCH and hope I could ask the community's suggestion or experience with how the location of your FreeSWITCH server affects the quality of the VoIP call. I'm using JunctionNetworks (http://pstn.junctionnetworks.com/) for SIP trunking. Their servers are in NYC. I'm trying to decide between running FreeSWITCH on a Debian dedicated server in Virginia (http://www.serverbeach.com/) or a FreeBSD dedicated server in San Diego (http://www.m5hosting.com/) This is for a VERY small business without a lot of phone traffic, but our users are spread around the world, in LA, Toronto, NYC, Berlin, London, Bangalore, and Hong Kong. I've used both for a couple months and they both seem fine. Since I'm in LA, I get a better response-speed from the San Diego server, but people on the east coast might have the opposite experience. My recent FreeSWITCH calls through the Virginia server have sounded quite lo-fi. I also like FreeBSD better than Debian. What would you recommend? Server-location doesn't matter? Server-location does matter? Location doesn't matter much but the colocation facility matters more? Any suggestions appreciated. I'm trying to make a decision, but realized I didn't have all the facts. Thanks!

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