- FreeSWITCH-users - FreeSWITCH Mailing Lists

Dialplan not executing on continue_on_fail=true
by Ashish gautam 24 May '13

24 May '13

Hi, I have a dialplan as follows: <include> <extension name="public_did"> <condition field="destination_number" expression="^(47673501)$"> <action application="answer"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="api_hangup_hook=perl hook.pl"/> <action application="set" data="session_in_hangup_hook=true"/> <action application="perl" data="perl/ash.pl"/> </condition> </extension> </include> when the called party does not pick up the phone or is busy, the dialplan does not proceed and hook.pl does not get executed. Please help -- Ashish Gautam IVR Developer Nucleus Microsystems (Pvt.) Ltd.

3 2

Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg
by Adam Raszynski 24 May '13

24 May '13

Hi, I have very strange problem with audio after upgrading of FS to the newest GIT version. Previous version was quite old, but worked without problems for a long time: FreeSWITCH version: 1.0.head (git-313b164 2011-11-26 08-53-01 -0600) Current version is: FreeSWITCH version: 1.5.1b () All configuration files are the same (exact copy) System: Debian Lenny 32-bit All calls are made over SIP Symptoms: - IVR prompt playback is OK - FAX sending/receiving is OK - Calling to remote servers via VoIP provider is OK - The only problem is calling to local user and it's very strange. When external user calls local user (SIP call, via IVR, then bridge), audio stream from external user is OK, but audio stream from local user sent to external user is very delayed. Delay increases during a call. At the beginning is about 1 second, all the time delay is increasing. Audio is reaching external user but after about 30 seconds of call stream is delayed for about 10 seconds. At the same time audio stream from external user is heard by local user with no delays. Propably something changed in FS core or config files and I need to adjust to the new version. As I stated before configuration is exact copy and worked perfectly on older version. Any hints what should I check or change to solve this problem? Kind Regards

4 9

problem bridging lots of dtmf in a session
by Michel Brabants 24 May '13

24 May '13

Hello, we have a problem with dtmf not being bridged by freeswitch when there is lots of dtmf in the call. Freeswitch resides between an asterisk gateway (which sends lot of duplicated dtmf) and an ivr. When there is a lot of dtmf being bridged in a single call, we can see in a wireshark-trace that it stops bridging the dtmf at a certain moment. I can still see the dtmf being received in the freeswitch-logs, bit I'm not seeing it in the wireshark-trace towards the ivr. We're using freeswitch 1.2.7. Thanks, Michel

1 1

Google Talk XMPP Support
by Andrew Cassidy 24 May '13

24 May '13

I've heard rumors that XMPP support in Google Talk is being killed. I'm guessing this means no more Google Talk in FreeSWITCH? -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F<info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

3 3

how to turn an ongoing call in to a conference
by Vincent Xia 24 May '13

24 May '13

Dear Support, my question for using freeswitch is: how to turn an ongoing call in to a conference so i can have the third party in without breaking up the call (no hangup, dial stuff)? something like call intrusion，A and B are already talking on the phone, then C wants to talk to both A and B. any response will be greatly appriciated. Jacob

3 5

why it hangs up after originate
by Vincent Xia 24 May '13

24 May '13

when im executing console commands like "originate user/1005 @eavesdrop(uuid)" to have 1005 listen to a call party, 1005 rings then i click answer and find out it hangs up immediately, the freeswitch log says "[NOTICE] switch_core_state_machine.c:262 sofia/internal/sip:1005@IPaddr:5088 has executed the last dialplan instruction, hanging up." does that mean i need to add something the to the dialplan? is there a simple solution by extending the originate command like "originate user/1005 @eavesdrop(uuid) &wait_for_hangup"?

2 2

Lua creating multiple session's
by Jon Schøpzinsky 23 May '13

23 May '13

Hello List, Im trying to do dial multiple destination through lua, on a single incoming call. I do know that i could do a simple session:execute("bridge", "dst1,dst2,dst3") but I need to do it in individual session, for processing I need to do in a later version of the lua script. I loop through the destinations that needs to be called, creating a new session for each destination, and storing that in an array. Firstly, it seems as freeswitch.Session doesn't reply right away, but waits for early-media. Thats ok though, but makes dialling mobile devices a rather long wait. The problem is, that when I do the second freeswitch.Session, it seems to hold up further lua processing, until the last created call is answered. Is this how its supposed to work. Heres my current code: local legs = {} for key, dev in pairs(dstDevices) do freeswitch.consoleLog("info", "Lets call " .. dev.username .. " with tech " .. dev.devicetech .. "\n") if dev.devicetech == "1" then freeswitch.consoleLog("info", "calling " .. dev.username .. " on uasbc\n") legs[key] = freeswitch.Session("sofia/gateway/uasbc01/" .. dev.username); freeswitch.consoleLog("info", "called " .. dev.username .. " on uasbc\n") elseif dev.devicetech == "2" then freeswitch.consoleLog("info", "calling " .. dev.username .. " on ccsbc, mvno\n") legs[key] = freeswitch.Session("{origination_caller_id_name=+xxxxxxxx,origination_calle r_id_number=+xxxxxxxxx}sofia/gateway/ccsbc01/+xx" .. string.match(dev.username, "^mvno_(.+)")) freeswitch.consoleLog("info", "called " .. dev.username .. " on ccsbc, mvno\n") end End freeswitch.consoleLog("info", "Enter loop now\n") It doesn't reach the last consoleLog until the last call is answered. Venlig hilsen/kind regards Jon Leren Schøpzinsky

2 1

Re: [Freeswitch-users] Conference questions
by Matthew Cordes 23 May '13

23 May '13

Hi Mike, Thanks for the quick response. Here are a few more questions / comments below: > 1. How do I translate conference cli commands (e.g. "conference xxx mute > yyy") into a dialplan xml actions? > > You don't really. Why exactly would you want to do this? > In some situations I'd like to mute all users other than the moderator from when they join for the duration of the conference. I thought the right way to do this would be via an action in my dialplan as new callers join my conference. What would you recommend? Thanks, Mod_xml_curl looks straightforward enough. 4. I'd like to have some private conferences that only certain callers can join. I'll know the callers' ids before hand. I'm assuming a reasonable way to handle is with user groups. How would I test if a user is a member of a particular group? If your doing dialplan via mod_xml_curl you can do all that validation in > your dialplan lookup. > Can you point me to a resource regarding how I would do this validation? I know how to define users and groups. It's not clear to me how to test for membership in a group or how that would be written (although I'm assuming it would be a condition in the dialplan, correct?). -Matt

2 1

Cannot ring extension from DID
by Mike Hendrie 23 May '13

23 May '13

*==========================* *Server:* Ubuntu 12.04 *Version:* FreeSWITCH Version 1.5.1 *Enabled:* Multi-Tenant *Extensions:* 1000 and 1001 *Connection:* freeswitch server NAT to provider *Connection: *extensions are on the same LAN as fs server *DID:* 1 DID ========================== *Can-Do: * - I can dial extension to extension (1000 to 1001) and the phone rings and can talk. - I can test with extension 5000 and listen to the echo test. - I can dial the DID to the Freeswitch server. The server sends the the call to voicemail, 1000, when it should ring the phone. *Cannot Do:* - I cannot call out of any extension. The phone does not ring. It is silent. - I cannot dial the DID and ring a phone. It only goes to VoiceMail. *Question:* - Why can I not dial the DID and have the extension ring? - Where should I check? Thank you, Mike

3 22

Conference questions
by Matthew Cordes 23 May '13

23 May '13

Hi, I'm new to FreeSwitch and starting to experiment with conferences. I've browsed the wiki and I'm half way through the ebook and I'm running into a few questions I'm hoping I might be able to get some help with. 1. How do I translate conference cli commands (e.g. "conference xxx mute yyy") into a dialplan xml actions? 2. The default behavior of the conference is to disable hold music and allow the participants to speak when there are two or more callers. How might I keep the conference on hold until a particular caller joins? Additionally when this special user leaves I'd like to place the conference back on hold. 3. Where might I find info regarding loading configuration information (particularly the user directory) from an external source (probably a database and probably via mod_python)? 4. I'd like to have some private conferences that only certain callers can join. I'll know the callers' ids before hand. I'm assuming a reasonable way to handle is with user groups. How would I test if a user is a member of a particular group? Thanks, -Matt

2 1