- FreeSWITCH-users - FreeSWITCH Mailing Lists

how to response to message OPTIONS
by mashudi 31 Oct '08

31 Oct '08

Dear Folks, i have a softswitch and i will connect this softswitch to freeswitch as media server. My softswitch send OPTIONS message periodically to freeswitch but i don'tknow how to response to this message. please help how to config freeswitch in order to make response from softswitch OPTIONS message ,and how to accept every message comming from softswitch ip address to freeswitch without authentication. thanks you for helps. best regards, mashudi ***************************************** Flexi-Gratis bicara sepanjang waktu se-Jawa Barat, Banten dan DKI Jakarta. Speedy-Gratis Internetan unlimited dari Pkl. 20.00 s/d 8.00 se- Jabodetabek, Banten, Karawang dan Purwakarta *****************************************

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Call Forwarding from phone
by Noah Silverman 31 Oct '08

31 Oct '08

Hello, I have a configuration (dialplan?) question. One of my users used the "call forward" button on his sip phone. The phone is directly registered to FS. The forwarded calls fail. It looks like the phone is sending a "redirect" message to FS, but then the call is then not getting routed correctly through FS. So, the call is dropped. Is there something specific I need to configure to account for possible call forwarding from phones?? Thanks, -Noah

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XML cdr postgres
by Chav Paskov 31 Oct '08

31 Oct '08

Hi Everybody, i have written few scripts to process the xml.cdrs into postgres from there basically you can do whatever you want with them. Pls let me know if you're interested in it. I'll be happy to submit them. Chav

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mod_cdr revival (or new module maybe)
by regs＠kinetix.gr 30 Oct '08

30 Oct '08

Hi, I saw in the wiki that the mod_cdr module is now unsupported. There is also a note about a revival of the module. I would like to ask the following : What is the current state of the revival process? (should we expect something in the near future?) Will it have the same functionality as before (DB support for instance)? Are there any plans for a brand new database specific event handler module? It would be great if there was one so that developers (especially those who develop billing applications) would not have to create their own hacks (cron scripts etc.) Thank you for your time, -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com Support Center

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Limit number of outbound channels
by Chav Paskov 30 Oct '08

30 Oct '08

Hi, Everybody, i was wondering if there is an option or command that allows to limit the number of outbound channels per gateway under external profile. Regards Chav

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Status of 1.02?
by Andy Spitzer 30 Oct '08

30 Oct '08

Woof! Is there any updated projection of when 1.0.2 will be released? Last I can find mention of it is over a month ago (http://freeswitch.org/node/143) Thanks, --Woof!

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Re: [Freeswitch-users] Clustering FreeSWITCH
by Yuval Hertzog 30 Oct '08

30 Oct '08

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Authorizing Anonynous Devices
by Klaus Teller 29 Oct '08

29 Oct '08

Hi Folks, I need some additional help with this issue. I already had some from Brian i'm but still not able to move forward. I want a non-registered device to be able to call extension 56900 in my Freeswitch in such a way that i can manage the call using the socket interface. I believe the issue i'm having is that of permissions. Based on what was suggested sofar, here is what i did. 1) Added following extension in dialplan/default.xml <extension name="check_auth" continue="true"> <condition field="${sip_authorized}" expression="^true$" break="never"> <anti-action application="respond" data="407"/> </condition> </extension> 2) Set auth-calls to false in internal.xml. That is <param name="auth-calls" value="$${internal_auth_calls}"/> was replaced with: <param name="auth-calls" value="false"/> 3) Changed acl.com.xml by replacing <list name="domains" default="DENY"> <node type="allow" domain="$${domain}"/> </list> with <list name="domains" default="allow"> <node type="allow" domain="$${domain}"/> </list> Now here is the result i get after these changes: a) Anonymous non-registered device can call registered soft phone at extension 1003 b) Anonymous non-registered device cannot call 56900 that needs to be managed via socket interface (error message 480). Also 9000 cannot be called. c) Registered soft phone (extension 1003) cannot call 56900 d) Registered soft phone (ext 1003) can call registered soft phone (ext 1000). If i perform only step 1 and 3 (i.e. auth-calls not set to false), a) become impossible, b) remains wrong, c) is now possible (i.e. socket interface being notified about call at 56900), while d) remains. valid. Disabling any of 1) or 3) would result into calls by non-registered device being rejected. Any idea what else can be tried? Thanks, Klaus. -- Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! Ideal für Modem und ISDN: http://www.gmx.net/de/go/smartsurfer

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Using Python to open a channel to outgoing leg and play a prompt
by fgastiaz 29 Oct '08

29 Oct '08

Hi guys, I what I want to do is based on what one of the register extension dials, contact another registered user but before bridging both legs i want to play a file into the egress side, if possible using python Example: call Flow: SIP Phone A --> FreeSwitch --> SIP Phone B. sequence: a) SIP PHONE A dials "777" b) a dialplan is executed based on the user-context, as a condition is matched, then we starts a script ( python is the one i chose ) c) The scripts starts to run and it need to contact the SIP Phone B. when the channel to B is open, I need to play a prompt "test.wav" to SIP PHONE B, when the prompt is over i do the bridging between SIP PHONE A and SIP PHONE B. If the SIP PHONE B hungs up during the prompt, both channels have to end. I tried with something like: newsession = Session("sofia/internal/221144(a)201.216.201.133") if ( newsession.ready() ): newsession.execute("playback","test1.wav") but it doesn't work... Do you have some ideas for this... or it is impossible to achieve... -- View this message in context: http://www.nabble.com/Using-Python-to-open-a-channel-to-outgoing-leg-and-pl… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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dptools "read" command doesn't appear to work properly
by Luke Graybill 29 Oct '08

29 Oct '08

I am having a hard time getting this command to work properly. Here is the c output from my test session, where I dial in to ext 500 with my sip client (Ekiga): ---- listening on localhost:8084 IPv4Address(TCP, '127.0.0.1', 36685): Connected >> 'connect' Caller: Killarny - >> 'myevents' result was: Events Enabled >> 'sendmsg\ncall-command: execute\nexecute-app-name: answer' OK >> 'sendmsg\ncall-command: execute\nexecute-app-name: read\nexecute-app-arg: 0 4 conference/8000/conf-pin.wav digits 10000 #' OK DTMF: 1 DTMF: 2 DTMF: 3 DTMF: 4 DTMF: 5 DTMF: 6 DTMF: 7 DTMF: 8 DTMF: 9 DTMF: 0 DTMF: * DTMF: # IPv4Address(TCP, '127.0.0.1', 36685): Connection was closed cleanly. ---- I couldn't figure out how to get read to function without passing a wav file (I plan to use TTS for the voice prompts, and I don't need a wav file to play here) so I just plugged in a random wav that comes with freeswitch for testing. As you can see, freeswitch reports that the command executed properly, illustrated by the OK (which was a reply-text: +OK) but the read application doesn't seem to be functioning beyond playing the given wav file (and truncating the first half of it as well). Pressing digits in the sip client results in DTMF events being sent to the socket. Note that the DTMF events stop at # because I manually shut down the connection, not because of anything that freeswitch does. Am I doing something wrong here? I feel like I'm following the available documentation given at http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read but I am having no luck. Is the read application basically a dead command, and should I be instead just using DTMF event output? Any help that can be provided would be appreciated, thanks. Killarny

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