July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_dptools.c:3831 Can't find user, different sip profiles
by Miha 17 Jul '12

17 Jul '12

HI, I have created two different sip_profiles due to NAT issue (5060 and 5070 port). FS also has multi tentat configuration. If I make call between two phones which are registered on same profile calls goes throught, If I make phone call on with phones, which are registered on two different profiles I get mod_dptools.c:3831 Can't find user. I have resolve this issue with presence but few members on IRC told me that this is not the bast way due to load. How can I deal with this issue? Thanks! Miha

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Sound problems in voicemail.
by Vbvbrj 17 Jul '12

17 Jul '12

Hello. Searching for a resolve. I've found in default dialplan from distro in file conf/dialplan/default.xml, line 270: <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> also on line 284 is: <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> Changing line 270 to: <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> and leaving line 284 as is, solves the problem with the sound when calling some extension which is not registered from 1000 to 1019. But problem with sounds persists when calling 1100 to 1119. Messages "Goodbay" is very distorted. Is this a bug or something else that "application="bridge" data="loopback/app=voicemail:default" distort sound and "application="voicemail" data="default" does not distort?

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Duplicate DTMF Events - Please Help!
by Joann R DeJesus 17 Jul '12

17 Jul '12

Hello All, I hope someone has experience this issue and can quickly assist me! We are supposed to be going live with a customer in a couple of days and need to find a resolution quickly in order to stay on track... We are using Freeswitch with Sangoma T1 cards to bridge TDM with our internal VoIP system. Call flow is as follows: PSTN --> T1/Sangoma cards --> Freeswitch --> IVR I have configured Freeswitch for RFC2833. When I make a call to the IVR from a landline or cell phone and press "5" at a prompt, FS will sometimes send two separate DTMF events to the IVR. This causes the IVR to receive "55" which then sends the call to the incorrect destination. I can verify with Wireshark that FS is sending two separate RFC2833 events. For some reason, this is exacerbated with heavy call volume (easy to reproduce during load testing). Has anyone experienced this symptom or have any ideas what I can check? Thanks in advance for any assistance!! - Joann ________________________________ NOTICE: The information contained in this electronic mail transmission is intended by Convergys Corporation for the use of the named individual or entity to which it is directed and may contain information that is privileged or otherwise confidential. If you have received this electronic mail transmission in error, please delete it from your system without copying or forwarding it, and notify the sender of the error by reply email or by telephone (collect), so that the sender's address records can be corrected.

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Re: [Freeswitch-users] Duplicate DTMF Events - Please Help!
by Joann R DeJesus 17 Jul '12

17 Jul '12

Hi Brian, Thank you for taking the time to respond. We are using Sangoma cards with Freeswitch as a gateway between the PSTN and our internal network which uses SIP. We prefer not to use inband with SIP due to numerous other potential issues, and are trying to determine why FS sometimes creates double DTMF events. Thanks for the tip though! Joann -----Original message----- From: Brian Foster <bdfoster(a)endigotech.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Sent: Mon, Jul 16, 2012 23:34:17 EDT Subject: Re: [Freeswitch-users] Duplicate DTMF Events - Please Help! Rfc2833 is for sip, not for analog. Other than that I don't have much to offer. Try using inband only. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 16, 2012 6:50 PM, "Joann R DeJesus" <Joann.DeJesus(a)convergys.com<mailto:Joann.DeJesus@convergys.com>> wrote: Hello All, I hope someone has experience this issue and can quickly assist me! We are supposed to be going live with a customer in a couple of days and need to find a resolution quickly in order to stay on track… We are using Freeswitch with Sangoma T1 cards to bridge TDM with our internal VoIP system. Call flow is as follows: PSTN --> T1/Sangoma cards --> Freeswitch --> IVR I have configured Freeswitch for RFC2833. When I make a call to the IVR from a landline or cell phone and press "5" at a prompt, FS will sometimes send two separate DTMF events to the IVR. This causes the IVR to receive "55" which then sends the call to the incorrect destination. I can verify with Wireshark that FS is sending two separate RFC2833 events. For some reason, this is exacerbated with heavy call volume (easy to reproduce during load testing). Has anyone experienced this symptom or have any ideas what I can check? Thanks in advance for any assistance!! - Joann ________________________________ NOTICE: The information contained in this electronic mail transmission is intended by Convergys Corporation for the use of the named individual or entity to which it is directed and may contain information that is privileged or otherwise confidential. If you have received this electronic mail transmission in error, please delete it from your system without copying or forwarding it, and notify the sender of the error by reply email or by telephone (collect), so that the sender's address records can be corrected. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ NOTICE: The information contained in this electronic mail transmission is intended by Convergys Corporation for the use of the named individual or entity to which it is directed and may contain information that is privileged or otherwise confidential. If you have received this electronic mail transmission in error, please delete it from your system without copying or forwarding it, and notify the sender of the error by reply email or by telephone (collect), so that the sender's address records can be corrected.

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Profiling FreeSwitch?
by Tom C 16 Jul '12

16 Jul '12

Are there any tools already in use for profiling FreeSwitch? That is, getting statistics on how often various procedures are called, and how much processor time they take, etc. Being a linux noob, I spent hours looking at "gprof", and when I finally got it working, I learned that it doesn't handle multi-threaded apps like FS. Is there a recommended tool that people are already using? Or should I just add my own logging code and recompile?

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AVMD
by Ian Bonham 16 Jul '12

16 Jul '12

Hi Everyone, New member, so nice to 'meet' you all! I'm experimenting with FreeSwitch to see how it could replace Asterisk in a mass dialler system that I run in Europe, and I have been playing with the AVMD module. It seems to work perfectly, and does detect voicemail very well. What I am trying to do through is have the dialler generate calls and run as much as possible through FreeSwitch's core, rather than external scripts (I'm using PHP for most of my external functionality as thats what I use to write the agent interfaces too). So I generate individual calls and launch them in FreeSwitch, and it then does most of the call logic from XML dialplan commands, jumping to PHP shell scripts for quick MySQL functions. I'd like very much to be able to run AVMD from the second the call is answered, which is simple enough, but at the same time have my outgoing message playing. So we are assuming the call is a human, but if during the OGM playout AVMD hears a beep, it just terminates the call. Can anyone tell me if this is possible please, either using XML dialplan (I'm guessing not as the XML is sequential), or using a programming script like LUA. Many thanks for any advise anyone can offer, Ian

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FreeSWITCH Weekly News and Notes
by Michael Collins 16 Jul '12

16 Jul '12

One week closer to ClueCon! Things have been very busy for the FreeSWITCH team and community. In addition to gearing up for ClueCon <http://www.cluecon.com?cc12-0716> we've all been working on our day jobs and mixing in the occasional summer vacation. We've also had inclement weather play a role in scheduling one of our weekly conference<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12-0716>call presentations. Dave Kompel, one of our resident Windows gurus, had been scheduled to discuss a technique for handling the Windows firewall in a manner similar to how we use Fail2Ban<http://wiki.freeswitch.org/wiki/Fail2ban?cc12-0716>in a Linux environment, however thunderstorms in his area required his full attention on protecting and keeping his servers up and running. We will reschedule as soon as possible. This week we have scheduled Darren Schreiber from 2600hz<http://www.2600hz.com?cc12-0716>. He will be doing a follow up to his SIP 101 discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_20?cc12-0716>from several weeks ago. We will be diving a bit deeper into the SIP protocol. In the initial discussion we focused most of our attention on getting SIP endpoints registered and the setup/teardown process. This week we will look at some of the other things SIP can do, like presence. We invite you to join us this Wednesday at 1PM Eastern, 10AM Pacific. In ClueCon news we are very happy to report that 2600hz is back as a Silver sponsor! In fact, the aforementioned Darren Schreiber will be giving a presentation<http://www.cluecon.com/presentation/open-source-revamping-telecom-you-want-…>at ClueCon, discussing some of the hard-earned lessons that 2600hz has learned over the past few years. If you are interested in cloud telephony then definitely be there for Darren's talk. For those who've not registered yet please keep in mind that the "bits will shift again" this Wednesday and registrants will receive four entries instead of the current eight. Register now <http://www.cluecon.com/register?cc12-0716> to maximize your opportunity to win lots of great prizes. The ClueCon countdown is already down to 21 days! We are looking forward to seeing everyone together again in Chicago. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0716

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Setting caller_ton doesn't work
by mbo 16 Jul '12

16 Jul '12

In the sangoma wiki I can find a sample how to configure the caller_ton per call (A200 E1 card): http://wiki.sangoma.com/wanpipe-freeswitch-config-appendix#ton-npi-per-call Based on that, I've configured the following test extension: <extension name="Caller Ton Test"> <condition field="destination_number" expression="^(0123456789)$"> <action application="set" data="caller_ton=1" /> <action application="set" data= "effective_caller_id_number=49111111111111111" /> <action application="bridge" data="freetdm/1/a/4922222222222"/> </condition> </extension> I'm dialing out from sip extension 1001 to the pstn. If I look in the ISDN Trace, i can see ... Calling Party Number:49111111111111111(l:17) plan:isdn(1) type:national(2)scr:user, passed(1) pres:allowed(0) ... that caller_ton is still set to 2 (national). You can find the full freeswitch trace here: http://pastebin.freeswitch.org/19525 It seems it's not overwriting the default parameter outbound-calling-ton from freetdm.conf.xml. Here my freetdm.conf.xml: <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="1000"/> <param name="crash-on-assert" value="no"/> <param name="early-media-override" value="on-alert" /> </settings> <config_profiles> <profile name="telco_1"> <param name="switchtype" value="euroisdn"/> <param name="signalling" value="cpe"/> <param name="overlap" value="yes"/> </profile> </config_profiles> <sangoma_pri_spans> <span name="PRI_1" cfgprofile="telco_1"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="outbound-called-ton" value="international"/> <param name="outbound-calling-ton" value="national"/> <param name="outbound-rdnis-ton" value="international"/> <param name="outbound-called-npi" value="ISDN"/> <param name="outbound-calling-npi" value="ISDN"/> <param name="outbound-rdnis-npi" value="ISDN"/> </span> </sangoma_pri_spans> </configuration> I'm pretty sure that was working in the past. Has something changed here?

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Music on Hold overly loud and distorted.
by Todd Bailey 16 Jul '12

16 Jul '12

Hi All, I'm still setting up FS and got a complaint from callers that music on hold is overly loud and distorted. But from an internal line, no issues. Could this be a setting issue with FS or perhaps the spa-31002 ata adapter? thanks

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programming/config Polycom ip501
by Todd Bailey 16 Jul '12

16 Jul '12

Hi All, I'm running a vanilla install of Fs and current have a hard phone attached - the ip501. tftp config files. The address of the phone is 192.168.1.23 & 00:04:F2:04:79:68 If I read the setup wiki correctly, I need to have created 3 different files 0004F2047968.cfg 0004F2047968-settings.cfg 0004F2047968-directory.cfg these don't appear to have any effect on phone settings or functionality, did I name these properly? and what if anything should I do with the files from the firmware zip file I downloaded: 000000000000.cfg 000000000000-directory~.xml phone1_318.cfg sip_318.cfg Speed dial (stored contacts) ? I was able to config the phone using the web interface but I'm running into several issues in getting various functions to work. how do I setup and store a contact to get to an outside line I have to dial 9 then send then dial the rest of the number. Needless to say , speed dial doesn't work, What sections do I need to study to get 9 + ac + phone number to work either manually or via a saved contact? Messages? how do I program the phone access voice mail? when I press "messages", I can select 1 of the 3 extensions after doing so it dials that extension instead of going to the voice mail. sorry for being verbose, it couldn't be helped.

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