- FreeSWITCH-users - FreeSWITCH Mailing Lists

Illegal character?
by Ben Holtsclaw 15 Aug '08

15 Aug '08

Earlier today, I was in the chat room trying to get help with a problem where one of my extensions didn't ring back when calling it. Any other extension would dial that extension, 305, and just hear dead air. 305 would actually ring, but when picked up, no sound was on the line. This is what the log looked like when trying to call 305: http://pastebin.freeswitch.org/5119 Here is the pastebin that shows a few of my extensions: http://pastebin.freeswitch.org/5126 You can see that extension 305 is registered with the caller id name Testing & Accountability. That is what I set in the caller id field on the Aastra 9143i phone. I finally figured out that if I remove the ampersand, the problem goes away. My question is... Is this a limitation (bug?) of the Aastra firmware, sofia, sip in general, or something in freeswitch? Ben Holtsclaw Network Engineer Avery County Schools Ph: 828.733.3567 x301

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ANNOUNCEMENT: Mod of the week - mod_enum
by Michael Collins 15 Aug '08

15 Aug '08

Greetings! The documentation group met on Wednesday and decided that the next mod to work on is mod_enum. We'd like to ask all those willing and capable to assist. We want to have this page be the central point for all things mod_enum: http://wiki.freeswitch.org/wiki/Mod_enum There are other pages that have enum info, but we'd like to get this organized. If you are familiar with how enum works and can clean up the wiki a bit that would be well-received by the community. Please note these pages: http://wiki.freeswitch.org/wiki/ENUM_support <== has some enum info here http://wiki.freeswitch.org/wiki/ISN <== ISN is related to enum If you feel confident then jump right in and start fixing the enum pages. If you have questions then please join #freeswitch-docs and ask away. Thanks for your assistance! -MC P.S. - Thanks to the community for jumping in and adding content to the mod_cdr_csv page over the past two weeks!

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Dialplan translation to FS.
by Henk Oegema 15 Aug '08

15 Aug '08

I still haven't been able to do this in FS: :-( exten => _9X.,1,Dial(SIP/2001/${EXTEN:1}) exten => _9X.,n,Congestion() Who can help please. Rgds Henk (p.s. I asked this earlier in: http://www.nabble.com/GSM-gateway-MV370-to18932091.html but no response)

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TLS but OPTIONS requests in clear text from port 5060
by Peter P GMX 15 Aug '08

15 Aug '08

Hello, I have successfully set up Snom phones for TLS and SRTP so the whole communication should be encrypted. However I see OPTIONS SIP requests in clear text coming from Freeswitch. Is there a chance to avoid this? Best regards Peter

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FreeSWITCH EC2 AMI
by Brian West 15 Aug '08

15 Aug '08

I'm currently putting together an EC2 image with FreeSWITCH on it. I'll be hosting a 1.0.1 image and try to load a weekly image of SVN trunk. I'll be incurring some costs associated with this. If anyone wants to donate to the cost of me hosting these images in my AWS account please contact me off list. Brian West sip:brian@freeswitch.org

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context public , dialplan public.xml relation ?
by Lee JJ 14 Aug '08

14 Aug '08

Hello : Under sip_profiles directory these files "internal.xml , external.xml , nat.xml" all have a setting ==> <param name="context" value="public"/> Does the "public context" relate to the dialplan directory "public.xml" file ? if do , I can route to another dialplan file under some condition, do I ? " <extension name="public_extensions"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="transfer" data="$1 XML *my-default*"/> " ; ^^^^^^^^^^^^^^^ if not , what should I do ? ps: Brian, thanks your effort here.

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help getting FXO inbound to bridge to extension
by Ian Henderson 14 Aug '08

14 Aug '08

Hi, I'm tearing my hair out trying to get an inbound FXO call to bridge to a SIP extension. My setup is: Zapmicro ZMA400P (TDM400P equivalent). Cisco 7940 with SIP firmware FS 1.0.1 + zaptel 1.4.11 I'm able to make calls out from various SIP extensions via the ZMA400P FXO, however I can't get inbound calls to cause the SIP extension to ring and acknowledge the call. In variations of the dialplan I can get a delay_echo to work for the inbound call; The sample config for Local Extension will appear to transfer the call to the extension, but the extension doesn't ring; however after 30 seconds, the call diverts to voicemail successfully. I've tried transfers, bridging; default and public profiles with no success. I've been through the wiki. Does anyone have a sample dialplan, or the tricks to get FXO-to-extensions to work? thanks in advance, Ian. -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com

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T1 RBS Support Revisited
by John Wehle 14 Aug '08

14 Aug '08

Okay, I have a T1 up and running between our System 25 PBX and FreeSWITCH by simply configuring the Sangoma A104 as an analog span as previously mentioned on this list. It appears that this works when using Loopstart / Groundstart / Koolstart RBS protocols. I need to use Winkstart when doing certain things with this PBX in order to avoid spilling digits ... the zaptel.conf indicates this is supported by configuring the device as "e&m" and then implementing Winkstart in the userspace library. Is this currently supported by FreeSWITCH? If so, how is it configured? If not, what needs to happen in order to implement it? I may be up for contributing code if someone points me in the right direction. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Dialplan Bridge
by Robert Clayton 14 Aug '08

14 Aug '08

Is it possible to bridge to other extensions in the dial plan such as below where extension 100 would ring 9000, 9001 and 9002. <?xml version="1.0" encoding="utf-8"?>  <include> <context name="default"> <extension name="Main Line"> <condition field="destination_number" expression="^100$"> <action application="bridge" data="sofia/sip/9000@192.168.1.222,sofia/sip/9001@192.168.1.222,sofia/sip/9002@192.168.1.222"/> </condition> </extension> <extension name="DictationLines"> <condition field="destination_number" expression="^900[0-2]$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> </context> </include>

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./freeswitch -vvvvv Verbose mode
by Sheeju Alex 14 Aug '08

14 Aug '08

Dear All, I am actually asterisk user, now I am wetting my hand in freeswitch and so far it look great for me. Is there any way where I can use freeswitch CLI with verbose mode similar to asterisk where I could see what exactly is happening action by action similar to asterisk. Looking for your reply.. Cheers Alex

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