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Seeking Advice on SIP Phones like Aastra 6739i or Snom 870 that have good interoperability with FreeSWITCH
by Darren C. 05 Feb '10

05 Feb '10

Tim, Many thanks for your response. My first response became an orphan/new post on the list...I think if subject line is too long reply looses thread. We are doing something similar to what you described. We’re developing a rather complex IVR/Switching application and it currently does all its database writes via our Web Service to an MS SQL database. We have a web site that is updated with call details via the web service’s backend database. From this web interface a user can see counts of voicemails, see call activity, play voicemails, see calls in progress, record calls, etc. It’s a specialized application so it doesn’t have every PBX feature but this is what we wanted to do with a high-end SIP Phone to replace our office PBX. Currently we have an ESI (Estech) E-Class PBX that uses normal digital phones as well as proprietary VOIP (non-SIP) phones. I think these really nice SIP phones with huge color touchscreens would be much better than even high-end proprietary digital phones + we’d get all the benefits of FS. We’ll just need to add all the basic PBX capabilities to the phone’s GUI to see how many voicemails are waiting, how many lines are in use, button for call transfer, etc. As you mentioned: “you can do a DB query to find out what voicemail messages a user has, or what calls are active, etc.” We are keeping track of all this ourselves via our web service’s backend database. I’m not sure I need to do this for everything but we are. It’s a multi-tenant system with hundreds of tenants so I’m guessing I might lose some needed relationships by querying FS but I’m going to re-visit this…I will make sure I can’t just query FS like you’re doing for some of these things…we’ve never turned on ODBC or even looked for the documentation as to what FS stores. We have to run FS in Linux for some Sangoma stuff but we’re Windows people so that is another reason we store via web services to an MS SQLbackend. I was worried I’d get one of these fancy phones and find out it doesn’t support important SIP/FS features rendering the color touchscreen useless. I’ve never owned one of these SIP phones, just used various softphones. But thanks to you I’ll get an Aastra 6739i and give it a try. I have a fulltime programmer working on this system on and off now for over a year but this has been mostly developing the IVR. We haven't made any attempt at using FS as a PBX. So based on your comments I think I’ll purchase an Aastra 6739i and develop a custom SIP Phone GUI interface with FS. If you or others would like to collaborate on an Aastra 6739i phone GUI for FS, feel free to contact me. We can try to make it extensible for other phones as well. My email is ustcor<remove this junk>poration at yah<remove this too>oo.com. Thanks, teldev

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Running a session asynchronously from a Lua script
by Adam Wilt 05 Feb '10

05 Feb '10

Hi all, I want to park an inbound call and play hold music while I simultaneously place another outbound call. But the hold music doesn't play while the lua script is placing the second call. When the lua script ends, the hold music finally starts. Here's my example code: #!/usr/local/bin/lua session:answer() api = freeswitch.API() api:executeString("bgapi uuid_park " .. tostring(session.uuid)) api:executeString("bgapi uuid_broadcast " .. tostring(session.uuid) .. " /freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav") local new_session = freeswitch.Session("sofia/gateway/myprovider/15555555555") So it seems like the script is blocking the original session, despite the fact that I'm using bgapi. I'd really appreciate if somebody could help me with this. By the way, I'm using FreeSWITCH 1.0.4 in Windows. Thanks, Adam

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about bgapi
by Chia-Yen Wu 05 Feb '10

05 Feb '10

I tried to run a javascript by background API (bgapi jsrun test.js) this javascript (test.js) wont automatically stop, I tried to kill it by "uuid_kill" command with Job-UUID but return "-ERR no such channel!", how can I kill this bgapi? btw my FS run in windows. thank you

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Stripping Leading Digits of 10-digit Inbound Number
by Jerry Richards 04 Feb '10

04 Feb '10

I am using a Sangoma PRI card. When an inbound call is received, where do the leading digits get stripped off? For example, if the inbound called number is 4257405381, I notice the call is routed to 5381 extension, but I don't know what is stripping off the 425740 digits. Best Regards, Jerry

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Fw: need some hints on Softswitch deployment of FreeSwitch
by Prabin Shrestha 04 Feb '10

04 Feb '10

--- On Tue, 2/2/10, Prabin Shrestha <nepaligas(a)yahoo.com> wrote: From: Prabin Shrestha <nepaligas(a)yahoo.com> Subject: need some hints on Softswitch deployment of FreeSwitch To: freeswitch-users(a)lists.freeswitch.org Date: Tuesday, 2 February, 2010, 11:27 PM Dear all, I had been browsing through all the wikis of freeswitch, googling more than 1 week and couldn't figure out where to start. I have been finding so many problems and IRC thing I don't understand. Basically, I am just a average linux user running Ubuntu, trying to build a softswitch. It there was some book on freeswitch it would have been much easier for newbie like me. Here are some problems I have been facing. After installation, I found freeswitch in /opt/freeswitch directory. only creating freeswitch user, I can access it's fs_cli, and I have yet to learn the power of it. doing ps -A, I found freeswitch is running in background. Now comes the hard part. I wanted to test it using SPA3000 device with fxo and fxs ports, which after following guides in net, is not working for me. My reqirement is, 1. to run freeswitch as a softswitch which can route calls from voip call providers to Quintum gateways. 2. to have complete CDR reports generated to sql database. Some light on this matter will be highly appreciated. Prabin. Your Mail works best with the New Yahoo Optimized IE8. Get it NOW!. The INTERNET now has a personality. YOURS! See your Yahoo! Homepage. http://in.yahoo.com/

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Trunk compilation error on Windows "Cannot open source file: '..\..\celt-0.7.0-1\libcelt\kfft_single.c'"
by Maciej Aniserowicz 04 Feb '10

04 Feb '10

Hi, I downloaded the latest bits from svn but have troubles compiling FS. I get 21 similar errors: Cannot open source file: '..\..\celt-0.7.0-1\libcelt\kfft_single.c' Error 40 fatal error C1083: Cannot open source file: '..\..\celt-0.7.0-1\libcelt\bands.c': No such file or directory File: c1 Project: libcelt I just opened Freeswitch.2008.sln in VS, changed configuration to Release and hit Build. What am I missing? Regards, Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Trunk-compilation-error-on-Windows-Cannot-open-source-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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max calls from a gateway
by FERNANDO VILLARROEL 04 Feb '10

04 Feb '10

Dear. How i can do for limits inbound calls from a gateway: My config is like this: ~sip_profiles/external/ gateway1.xml <include> <gateway name="foo"> <param name="username" value="foo"/> <param name="password" value="foo"/> <param name="realm" value="xx.xxx.xxx.xx"/> <param name="proxy" value="xx.xxx.xxx.xx/> <param name="register" value="false"/> <param name="context" value="public"/> <param name="max_calls" value="30"/>--> </gateway> </include> It´s fine? Fernando

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Event socket: filter delete isn't working
by Nagalenoj H. 04 Feb '10

04 Feb '10

Dear friends, I've tried to delete the filter which I applied for an unique id. But, it doesn't work. After executing 'filter delete', I am receiving the events from that uuid. I used the command as 'filter delete unique-id c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565'. I did the following operations. Made call to the event socket. Registered events for all. (events plain all). Applied filter for the uuid. (filter unique-id aa3cb8ea-0b2f-11df-9e84-fb15c3cd8565). I've got a new uuid by using create_uuid. Applied filter for this new uuid. (filter unique-id c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565) Originated a call with that uuid. Now, I could receive events from both uuids. (Tested by giving DTMFs in both end and checked unique-id in event header). Then, I wanted to delete a uuid from the filter. (filter delete unique-id c6d2e0e2-0b2f-11df-9e84-fb15c3cd8565). I thought, i won't receive the events from this deleted unique-id. But, I received the dtmfs from both unique-id. I'm using 'FreeSWITCH Version 1.0.trunk (15982)'. -- Regards, Nagalenoj H.

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presence with event socket
by Christian Löschenkohl 04 Feb '10

04 Feb '10

hello i have a little problem with setting the presence information manually (on and off) with an outbound socket script i do send a event like this sendevent PRESENCE_IN proto: sip from: xxxx(a)mydomain.com login: xxxx(a)mydomain.com event_type: presence alt_event_type: dialog Presence-Call-Direction: outbound answer-state: confirmed i monitor the estension on a snom phone with bfl (function key mode) the problem is now that the led on the monitoring snom gets lit very short an the is off again if i put a sleep in the outbound socket script the led lights up as long as i set the timeout. the expected behaviour is that the led lights up and can be switched off with another event (then answer-state: terminated) we do use trunk rev. 16456 in proxy mode 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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Seeking Advice on SIP Phones like Aastra 6739i or Snom 870 that have good interoperability with FreeSWITCH
by Darren C. 04 Feb '10

04 Feb '10

Hello, I'm working on a project and are interested in using one of the newer SIP Phones with color displays, perhaps touchscreen, etc. to implement PBX-like features like voice mail messages waiting, visual address book, conference call setup, and more. We want to send some information about a call to the SIP Phone either via FS or our own Web Service. These two phones have XML browsers that we may be able to utilize. I'm concerned that these phones may work OK with Asterisk but not sure about FreeSWITCH. There is a great guide for XML Services for Aastra 6739i but only mention Asterisk in the examples: http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-93A834D7/03/XML_Free… I prefer to use phones that work well with FS but these are too new to show up on interoperability page. Anyone have experience using these, any advise would be appreciated: Aastra 6739i http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-EAA51F83/03/6739i_pd… http://www.voipsupply.com/aastra-6739i SNOM 870 http://www.snom.com/uploads/docu/data_snom870_en.pdf http://www.snom.com/products/ip-phones/snom-870-touchscreen-voip-phone http://www.voipsupply.com/snom-870 Hoping someone has some positive or negative experiences working with these phones. Thanks, teldev

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