- FreeSWITCH-users - FreeSWITCH Mailing Lists

per domain vm templates
by Garrison Hoffman 16 May '10

16 May '10

Does mod_voicemail support per domain or per user templates? I'd like to do something like: <param name="web-template-file" value="templates/$${voicemail_domain}/web-vm.tpl"/> I've tried a couple variations with no success, is there a way to accomplish this without preprocessor variables? I guess plan 'B' would be to come at this via XML-RPC but that seems a bit like reinventing the wheel. -gh

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Configuring jbilling with freeswitch
by Shrouk Khan 16 May '10

16 May '10

hi , has anyone ever been able to use jbilling with freeswitch ? i have both of them installed , but can not figure where to start ? is there any guidlines or anything that anyone knows of? -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

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TDM400 & local echo
by Brian May 16 May '10

16 May '10

Hello, When either I receive a phone call on my TDM400 card, everything I say is echoed back to me, very loudly, with significant delay. This doesn't seem to match the usual echo problems though, as like I said the echo is only heard locally, not remotely. It seems to be with different callers too, so it seems unlikely the echo is coming from the caller. Telephone calls that don't use the TDM400 card don't have this issue. Any ideas? -- Brian May <brian(a)microcomaustralia.com.au>

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Hunting
by toqeer ali 15 May '10

15 May '10

Hi all, I am new to freeswitch and want to implement Hunting. For example if call come to my freeswitch box where DID is configured to an extension for example 1000 so if 1000 did not pick the phone then it should be transfered to other extension or landline. Please give me a clue... Is there is any special module for that? Any help will highly appreciated. Thanks -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

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Digium TDM400P
by Lars Zeb 15 May '10

15 May '10

I had a Digium TDM400P which I used successfully in another box running SwitchVox. I put it in my FreeSWITCH box. I haven't started installing the zaptel software because 'lspci' does not list the Digium card. Should it? I don't want to install the software if the board's dead. If it is dead, should I try to get it repaired or perhaps buy a Patton M-ATA? I have a Viking E-40 Entry Phone which I would like to use with FreeSWITCH. The Viking along with the Digium board did work on the SwitchVox. Thanks, Lars

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Mod directory question
by Sean Holt 15 May '10

15 May '10

Hey list, I had this working before but after upgrade it doesn¹t. Basically I setup the find by name feature for my ivr menu and it works great, but now after entering the name the voice wav file spells out the name. Before it would use the recorded user name that the user created on their phone. Is there a setting I need in my dailplan to make this work again. I see the user created recordings on the FS server. Thanks Sean

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openvox PRI card for freeswitch
by daqiang wang 15 May '10

15 May '10

hello: I test OpenVox Card : for 4X4E1 Pri Card. but when the calls> 60 . the channel answer slowly ,and some errs occured. do you test the PRI card. ? and why so many errors ? thanks .

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blind transfer and outbound socket
by Christian Löschenkohl 15 May '10

15 May '10

hello i have a problem with a freeswitch hosted pbx system. when i call in to a extension (e.g. 40) an answer the call and then do a blind transfer to extension 50 it doesn't work. the outbound socket script do not work or do not get called in this case (no mail from the script). on the other hand if i do a attended transfer it works as it should (sending mail in the script works). the same context, the same scripts. we use snom 320 phones, the sip refer looks normal. blind transfer also doesn't work with a softclient (sjphone) where could i start? br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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Skype SIP
by Jan Berger 14 May '10

14 May '10

Hi, Do anyone have experience using the Skype SIP trunking? Is this standard SIP and does it connect smoothly to FreeSWITCH? Jan

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Call for help: stuff to document
by Michael Collins 14 May '10

14 May '10

Hello all, The FS devs have been busy adding features (as usual) and we need help getting them all documented (as usual). I've added them all to next week's agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2010_05_19 There are a lot of new things listed under "Items needing documentation" - if you are able to document all of these please feel free to do so. Be sure to update the page by linking to any documentation you create. Thanks! -Michael

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