- FreeSWITCH-users - FreeSWITCH Mailing Lists

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>

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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

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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

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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

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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?

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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...

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how to manipulate sip headers in Freeswitch
by sam 15 Jan '12

15 Jan '12

hi, i am new to free switch and am trying to connect it to a call out system. But whenever i make the out call, the remote server says "too many Allow events" and the support representative says i need to remove the Allow-Events. does any one know how to manipulate or remove the "allow-event" part of the sip header ? My current allow events are Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer and the server has problem with Allow-Events. they only want talk and hold. thanks. -sam

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Commercial G729 codec - more calls than available licenses
by Vitalie Colosov 15 Jan '12

15 Jan '12

Hi, If we buy 5 commercial licenses of G729, and suddenly we get more users dialing, would the 6th, etc call fail, or FS will just propose one of the free codecs from the outgoing codecs list "G729,PCMA,PCMU"? Thank you, Vitalie

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Using Puppet on large scale setups
by Brian Foster 15 Jan '12

15 Jan '12

Has anyone had experience with Puppet and if so, have you used it in conjunction with Freeswitch? Just trying to see if this is a good avenue for server management. -BDF

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cdr minutes of use per did
by Rodney 14 Jan '12

14 Jan '12

does anyone have a cdr viewer or method that can help me determine the total minutes for all calls on a specified DID for the hour/day/month to date? I would like a method to generate such a report and maybe have it emailed on a 24 hour basis to the person in charge of the DID. the following I have seen in the past with any method to find the original program that would give an output like this. Calls by Day, DNIS -> DID ACCOUNT-> Friday, 8/19/11 (Eastern Standard Time) Daily Month to Date (Start of Month = Day 21) Day DNIS DNIS Group Total Calls Minutes of Use Average Hold Time (Minutes) Total Calls Minutes of Use Average Hold Time (Minutes) – No rows –

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