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

04 Feb '10

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! http://downloads.yahoo.com/in/internetexplorer/

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Re: [Freeswitch-users] skypiax dtmf detection issue on Freeswitch
by Scott Torr 04 Feb '10

04 Feb '10

Hi Majdi, It is good to know that once the audio stream is in the right 'format', Freeswitch is able to decode the DTMF's to digits. (as best it can given the potential jitter and lossy quality of the signal) I'm sure someone on the list has the answer and it will quite simple. So here we go: How do we convert the audio stream from mod_skypiax into to a suitable format that <action application="start_dtmf" /> can attempt to decode the 'Inband' DTMF tones into digits? At present it seems that the decoder will not work on a 16Kbps stream. Perhaps this has been explained and not understood? Ideally Skype would decode the DTMF tones ingress to the Skype network at the PSTN-Skype gateway and pass on as out of band signaling. Either through bad design or intentionally they do not. It seems though at some locations (New Zealand) they did, then did not? Perhaps on their paid for (approx $9 AUD per month) SIP trunks they do? Or is this just as hit and miss as well? Does any body have any experience with those? -Scott On Wed, 03 Feb 2010 20:42 -0600, "BSOUL Majdi" <Majdi.Bsoul(a)alcatel-lucent.com> wrote: > I just saw your JIRA, so definitely by your comment date so close you > might still blocked on that resolution. > > > > I used SiptoSis with a single channel and because it passes the stream > to Freeswitch as PCMU, it allows FS to detect correctly. And this proves > that the inband DTMF quality coming from Skype is good enough. > > > > I am trying to see if I can use stsTrunkBuilder to support multiple > interfaces, but looks that it requires different skype account for each > instance, rather than allowing me to use single instance for all > instances as skypiax does. > > > > Regards, > > > > Majdi Bsoul > > Mobile NGN R&D > > Alcatel-Lucent > > 3400 W. Plano Parkway > > M/S 601-NGNRD > > Plano, TX 75075 > > Office: +1 972 477 0065 > > Fax: +1 972 519 3600 > Email: majdi.bsoul(a)alcatel-lucent.com > > From: BSOUL Majdi > Sent: Wednesday, February 03, 2010 8:26 PM > To: 'scott.torr.fs(a)letterboxes.org' > Cc: 'mbsoul(a)hotmail.com' > Subject: skypiax dtmf detection issue on Freeswitch > > > > Hello Scott, > > > > I thought I catch up with you and see if you had found a solution for > the skypiax dtmf detection in FS issue that you reported in Dec? > > I am running with the same issue. > > > > Thanks, > > > > Majdi Bsoul > > Mobile NGN R&D > > Alcatel-Lucent > > 3400 W. Plano Parkway > > M/S 601-NGNRD > > Plano, TX 75075 > > Office: +1 972 477 0065 > > Fax: +1 972 519 3600 > Email: majdi.bsoul(a)alcatel-lucent.com >

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adding sip header without X-
by Christian Löschenkohl 04 Feb '10

04 Feb '10

hello do anybody know a way to add "Alert-Info: <http://www.notused.com>;info=alert-group;x-line-id=0" as a custom sip header to the invite message? <action application="set" data="sip_h_X-Alert-Info=<http://www.notused.com>\;info=alert-internal\;x-line-id=0"/> works but "X-Alert-Info" as a header isn't usefull at all the purpose is to distinguish internal and external calls in a pbx. it is also described here http://wiki.snom.com/Web_Interface/V7/Preferences/Alert-Info_Ringer there was also a posting on the list by Kristian Kielhofner who suggested a very flexible solution ----- mail from: 2009-10-09 21:10 2) Make the behavior configurable with a channel variable and/or sofia config option: {sip_pass_headers=all|none|X-MyCustomHeaderByName} ----- 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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PAP2T issue
by John 04 Feb '10

04 Feb '10

I can register SNOM phones fine, one the same network, I am trying to get a Linksys PAP2T-NA registered. Anyone have a configuration example? Thanks

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Has anyone had Freeswitch crash on loading mod_voicemail?
by Mark Sobkow 03 Feb '10

03 Feb '10

I tried a "load mod_voicemail" in fs_cli, hoping to see what configuration section it requested from Erlang, but instead of loading the module, Freeswitch crashed without any error messages. SVN 15188 built on Ubuntu Hardy 32-bit.

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FS Core Dump
by Jerry Richards 03 Feb '10

03 Feb '10

I spoke too soon. Using Freeswitch version Feb 01, 2010 and Wanpipe Driver ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.10.smg_pri.1.tgz, the FS core will dump if I call my external cell phone, answer and then hangup. See the pastebin: http://pastebin.freeswitch.org/12035. This happens every time. Best Regards, Jerry

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Need Help to setup freeswitch with sangoma card
by Thangappan.M 03 Feb '10

03 Feb '10

Dear all, I have successfully configured wanpipe with freeswitch. When I was the running wancfg_fs script the following files openzap.conf , autoload_confg/openzap.conf.xml , /etc/wanpipe/wanpipe1.xml, smg_pri.conf are created. I started the wanrouter command then executed the freeswitch. When I was executing freeswitch mod_openzap.c said the error as "Error for finding the span id. name:PRI_1". But in the openzap.conf and openzap.conf.xml files the span name is smg_prid. Why the freeswitch is referring the span name as PRI_1 ? Whether this has to configured in anywhere? In the freeswitch CLI using oz command I tried to dump the PRI_1 span id but it said te error as "PRI_1 is not found". When I was trying the command 'oz dump smg_prid' all the channel states and details shown. It seems that smg_prid span configured in openzap perfectly (Its my assumption). Then Why freeswitch is referring the span name as PRI_1. DID I MAKE ANY MISTAKE OR DID I MISS ANYTHING TO DO? Could anyone please help me? REFERENCE: openzap.conf [span wanpipe smg_prid] name => smg_prid trunk_type =>e1 b-channel => 1:1-15 b-channel => 1:17-31 openzap.conf.xml <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/>    </settings> <boost_spans> <span id="smg_prid">  <param name="dialplan" value="XML"/> <param name="context" value="default"/>     </span> </boost_spans> <analog_spans> </analog_spans> </configuration> -- Regards, Thangappan.M

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Let's buy the FreeSWITCH developers dinner!
by Michael Collins 03 Feb '10

03 Feb '10

Hello all! This is a reminder that next week the FreeSWITCH development team is gathering together in one location for the final push to get version 1.0.5 released. You can help facilitate the timely release of the latest version by helping to buy dinner for the FreeSWITCH developers. Remember, it's not just the three core developers who are meeting together. There will be eight FreeSWITCH team members gathering. Let's all pitch in a few dollars each and give them a nice dinner! It's a great way to say thanks for all the hard work they've done: building FreeSWITCH, answering questions on the mailing list, and spending many hours in the IRC channel. You can use the PayPal link on the main website. (http://www.freeswitch.org) Alternatively, if PayPal is not available to you then please contact Brian West (brian(a)freeswitch.org) to discuss alternate ways of donating. Let's really pull together and support the guys for all of their hard work. A nice meal paid for by the community would be greatly appreciated! -Michael

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Dialogic Board (JCT & DM3) Supported?
by Moiz Chinoy 03 Feb '10

03 Feb '10

Dear All, Can anyone please advise that whether Dialogic boards (JCT and DM3) are supported by FS. -- Regards, Moiz Chinoy.

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Ringback after before` Bridge
by Max Bridgewater 03 Feb '10

03 Feb '10

Hi, I'm trying to place a call to A and then bridge it to B. The problem I'm having right now is that after A answers and while dialing B is being dialed or rining, I want to send A a ringing tone. I don't succeed in doing this. No tone/ringback is being sent to A. Here is what i did using ESL: api originate {ringback=\'%(400,200,400,450);%(400,2200,400,450)\',transfer_ringback=\'%(400,200,400,450);%(400,2200,400,450)\',origination_caller_id_number=4156781020}sofia/gateway/voipms/4152309090 &park() To bridge, I then send the message: sendmsg e9dae14c-e473-466e-9d65-704e36a82e5f call-command: execute execute-app-name: bridge execute-app-arg: {{ringback=\'%(400,200,400,450);%(400,2200,400,450\'},origination_caller_id_number=4152309090 }sofia/gateway/voipms/4156781020 any idea? Max.

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