January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Voicemail overriding leg_delay_start ? (more info)
by Paul Cupis 16 Jan '12

16 Jan '12

Try setting continue_on_fail to true before the bridge command. Regards,

1 0

Bsuy, Wait, Callback - lua script..?
by Avi Marcus 15 Jan '12

15 Jan '12

Hi, I'm looking to implement a callback functionality. I get an incoming invite which has the user who is making the call and his destination. I'm supposed to 486 that and then after 2 second call them back, and connect the call. Should I do a hangup hook to a lua script which sleeps 2 seconds and does an originate? or zombie exec? And if I do the originate within the lua script, won't that leave the lua script running for the whole calls? What's the best way.. or even just a good way to do this - I don't expect more than 10 simultaneous calls as of now on this particular method. Thanks, -Avi

1 0

Voicemail overriding leg_delay_start ? (more info)
by Bob Smith 15 Jan '12

15 Jan '12

Should add that I also tried the following syntax, in this instance, the second bridge number rings, but then on expiry, it never gets to voicemail, it just appears to drop: <action application="bind_meta_app" data="3 b s execute_extension::att_xfer XML features"/> <action application="set" data="domain_name=mydmain"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="{ignore_early_media=true} [leg_timeout=30]group/mygroup@mydmain,[leg_delay_start=12,leg_timeout=30]sofia/gateway/mygate/1234"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default mydmain 1000"/> </condition> </extension> </context> </section> Has anyone got this sort of setup running ?

1 0

sched_hangup with playback
by Rodney 15 Jan '12

15 Jan '12

I would like to schedule a hangup at a set call time max but play a recording letting the person know the max time is up. is this possible in the dialplan? I have read http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup and see a reason code to set but no mention of playback warning the party. In addition, I would also like a 5 minute or so warning, like on a calling card platform. like "this call will end in ____ minutes" kind of thing. thanks.

2 1

Voicemail overriding leg_delay_start ?
by Bob Smith 15 Jan '12

15 Jan '12

Hi, I am trying to implement a follow-me using the XML syntax below. However instead of the second number ringing after 12 seconds, the voicemail picks up instead ? (For those of your reading this thread and remembering my earlier bridging problems, I've fixed those, so that's not the issue here ;-) <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="CURL Dialplan Result"> <context name="public"> <extension name="curl_dialplan"> <condition field="destination_number" expression="^(876543210)$"> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="3 b s execute_extension::att_xfer XML features"/> <action application="set" data="domain_name=mydomain"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="call_timeout=25"/> <action application="bridge" data="{ignore_early_media=true} group/mygroup@mydomain,[leg_delay_start=12]sofia/gateway/mygate/1234"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default mydomain 1000" </condition> </extension> </context> </section> </document>

1 0

undefined symbol: snd_config in mod_portaudio.so
by Frank Rosengart 15 Jan '12

15 Jan '12

Hi, after updating and compiling FS and mod_portaudio, I can not load the module anymore: Error Loading module /home/frank/fs_celt/mod/mod_portaudio.so **/home/frank/fs_celt/mod/mod_portaudio.so: undefined symbol: snd_config** It's the latest git checkout, and I did upgrade from Ubuntu natty to oneiric since it was working last time. I found a similar issue on a mailinglist in 2009, which seems to be a libtool update problem. My mod_portaudio.so is not linked to the libasound2, which could be the reason for the missing snd_config I did ran bootstrap. Any help appreciated Frank

1 0

Critical error buffer size
by Valery Kalinin 15 Jan '12

15 Jan '12

What is error: 2012-01-15 10:12:36.123103 [CRIT] switch_core_codec.c:660 Buffer size sanity check failed! What should I do? FreeSWITCH Version 1.0.head (git-187abe0 2012-01-13 15-55-54 -0600) OS: CentOS 5.5

2 1

Freeswitch with Bluebox on Mac OS X 10.6.8
by Daniel Melnechuk 15 Jan '12

15 Jan '12

Dear Freeswitch Users, I am just setting up for the first time, a freeswitch with bluebox on a mac mini running 10.6.8 snow leopard. I am trying to move off of a callweaver setup i have had going for a couple of years. Everything seemed to be going right along. Started with one "device" and could get and receive calls. Then added a second device and could call from one to the other. Then after doing something which at this point i confess i don't remember, one of the devices won't get a call anymore, but can still place one. What i keep getting in the log file is: 2012-01-15 00:47:28.529909 [DEBUG] switch_channel.c:1097 EXPORT (export_vars) [sip_invite_req_uri]=[sip:2001error/user_not_registered] But in fact the device (iSoftPhone on another Mac) says it is connected and Bluebox sip registrations shows it as registered. The device is on the LAN. And it can call out. I tried XLite instead of iSoftphone and the same thing happens. So now i'm starting to think the config has gotten messed up under the hood. In case it was a config issue for the device that wasn't receiving calls i created a new device and configured iSoftphone to use that device auth but same thing persists. No firewalls on. The device that works both directions is Bria for iPhone. Since i'm working with two new packages, freeswitch and bluebox, not sure where to go next. Any clues as to what i could look for would be most appreciated. I know i haven't given much config info but main thing to know that both devices are "Internal" numbers, they are configed the same, only one has a problem receiving calls. I haven't done much so far so i'm thinking to wipe clean the config and start from scratch if nothing shakes out soon. Peace, Dan

2 1

cdr records /destinations total minutes
by Rodney 15 Jan '12

15 Jan '12

I have a situation now where transfer to conferences etc. are not part of the total per call, it shows up as a total for a different destination. (the one they hung up from). is there a way to get total from the start of the call no matter what destination they hang up from? ie. keep destination DID the same no matter what destination the end up at so the call total will be intact?

2 1

T.38 Faxing - at a loss...
by Paige Sullivan 15 Jan '12

15 Jan '12

I've been trying in vein to get T.38 faxing to work properly. My provider is VOIP Innovations and my ATA is an EdgeWater 200EW (also tried a Grandstream HT-502) all of which support T.38. I have read the wiki about a thousand times and tried every combination of parameters in my dialplan that I can think of. I have the latest version of Freeswitch (built from git repo just today) and everything else works like a charm. What info should I post so maybe someone out there can give me some ideas? Like I said, I've tried about everything I can think of - even enabling t38_passthru = true on each individual call. Any ideas at all? Anyone using this provider and having T.38 work? I'm pretty desperate here and would even be willing to pay for an answer...

4 3