January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Line echo cancellation for voice
by georg＠riseup.net 27 Jan '12

27 Jan '12

Hello all, Could someone tell me, how to enable this: http://docs.freeswitch.org/echo_can_page.html ? Thanks, Georg

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sipXecs Conference March 5-6th
by Douglas Hubler 27 Jan '12

27 Jan '12

I wanted to invite everyone interested in learning about the upcoming release of sipXecs v4.6 to come to SIPfoundry's Co-Lab conference on March 5-6th in Fort Collins, CO USA. For those that do not know about sipXecs, it's an open source project that combines SIP based services together with FreeSWITCH, OpenFire and OpenACD into a complete unified communications system. New to the upcoming release 4.6, is sipXecs's tremendous platform capabilities to build and launch your own communication applications. This platform allows you to create services that are secure, redundant, load balancing and easily managed in any cluster. Customize the cluster to run whatever services you want. sipXecs takes a different approach to accomplishing this by letting you chose what technology stack and language to build your apps on. sipXecs just gives you a platform to compile, install and configure your services and integrate them other services down to the finest level of control. Once you're done building your component, you can share it with a greater SIPfoundry community all while maintaining your own source code repository. I've worked on sipXecs since version sipXecs v1.0 and I've never been more excited about a release than sipXecs v4.6. If I've just peeked your interest, please email me at dhubler(a)ezuce.com and I can provide more details. You can also email me on the sipxecs user's mailing list : sipx-users(a)list.sipfoundry.org http://www.sipfoundry.org/sipx-colab Douglas Hubler sipXecs developer

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Re: [Freeswitch-users] Sendmail segfaulting
by Rob Morin 27 Jan '12

27 Jan '12

I saw that, and tried everything I could. Still no luck. My stack limit is 8192, so that shouldn't be the issue. I'm running CentOS x_64. I tried creating a script as the wiki suggests, so that I could make changes without having to restart FS. But that didn't work either (I'm still a noob). My goal was to add a sleep delay, in the case that FS is calling the sendmail command before it's completely written and released the email it's sending. Any idea how to make that work? Thank you, Rob From: Michael Jerris [mailto:mike@jerris.com] Sent: Wednesday, January 25, 2012 1:57 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting before second message is sent http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings On Jan 25, 2012, at 1:31 PM, Rob Morin wrote: For some extensions, I've got FreeSWITCH configured to send both the voicemail to an email address and a notification to a cell phone that there is a message. Sendmail is segfaulting (error 6) before the second email message is sent. Sometimes the notification message is going out, sometimes the email message goes. If I am just sending an email, it is *generally* successful. Although sometimes it fails, so I really need to fix this. Does anyone have any suggestions as to how I can troubleshoot this? (Running CentOS 5.7, sendmail 8.13).

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Options ping
by Elliott Vogel 27 Jan '12

27 Jan '12

Hello, how can I make FS send Option pings to users that are registered? I would like to send a ping every 120 seconds because we are having problems with remote users that are behind firewalls using polycom phones, their firewalls are closing ports if there is no activity and calls fail. Also if anyone has better way to keep alive a firewall port I would love to hear it. Elliott

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FreeSWITCH Viking
by David Villasmil 27 Jan '12

27 Jan '12

Hello Guys, I updated the wiki with the following: CLI-based/Retail I have now made a retail version for FreeSWITCH-Billing. It works like this: - A specific context is created for this application. - Any calls which comes in on said context is handled by the retail script. - The script Looks up the incomming caller ID Number in a DB table. - The CLI is associated with a "Master" acount via the Master account's ID. - This Master account has a balance, a rate table and a username/password (for later use, this user will be able to check his traffic, balance, etc via web) - If the account has a positive balance, the rate table is used to apply a price per minute. - The Max talk time is calculated and set by the script and the call is sent out using the route tables (already existing in the wholesale side). - Once the call is finished, the CDR is posted using mod_xml_cdr on a web server and the receiving php inserts the cdr into the cdr table, also calculating and setting the cost of the call. TO DO: - web pages for the client to check out his balance, clis, etc. I will upload this into the github this weekend! Have fun! note: any suggestion is VERY welcome!

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mod_callcenter - ring tones when phone is ringing
by Weigel, Stefan 27 Jan '12

27 Jan '12

Hi all, we're using successfully mod_callcenter. If a caller enters the queue, he/she is getting default moh sound. The sound is also playing when the call is passed to an agent and the phone is ringing. Is it possible to play a normal ringtone when a call is passed to an agent ? Thanks and best regards Stefan

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Skypopen on debian squeeze - All sorts of fonts and alsa/oss driver and binary issues
by A E [Gmail] 27 Jan '12

27 Jan '12

Hello All, anyone tried to get Skypopen to work in Debian? I am not really sure what the real issue is as I get multiple errors when trying to start but the biggest one I guess is the one claiming that the skype binary cannot be executed. Will provide further info as requested as I really don't know what the issue really is. Does the Skype client need a real sound card/driver present? I doubt it. Does it really need a real screen attached to it? I doubt it since I see some magic being done with the Xserver to create fake displays and what not. Anyway, here's what I get when I try to start the start_skype_clients.sh *ERROR: Module snd_pcm_oss does not exist in /proc/modules* *ERROR: Module snd_mixer_oss does not exist in /proc/modules* *ERROR: Module snd_seq_oss does not exist in /proc/modules* *mknod: `/dev/dsp': File exists* *insmod: error inserting '/opt/freeswitch/skypopen/sound-drv/skypopen.ko': -1 File exists* *SELinux: Disabled on system, not enabling in X server* *[dix] Could not init font path element /var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!* *bash: /opt/freeswitch/skypopen/clientsymlnk/skype101: cannot execute binary file* Any ideas? Giovanni? :) Thanks in advance

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int/ext dial tones not synchronous
by georg＠riseup.net 27 Jan '12

27 Jan '12

Hello all, I made the first experiences with the dialplan, and after some while, I really got calls from the outside routed to one of my sip phones. However, my phone is ringing something like 1-3 seconds earlier than the dial tone the caller is hearing. I googled around, but found no solution. As a start, I just used this extension: <extension name="Local_Extension"> <condition field="destination_number" expression="^(4930XXXXXXXX)$"> <action application="bridge" data="user/12(a)192.168.1.93"/> </condition> </extension> Thanks, Georg

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mod_xml_cdr not saving file correctly
by Paul 27 Jan '12

27 Jan '12

Using latest build I have encountered a very odd behavior and want to know if it is a bug or not. If not - I would appreciate any ideas on how to work around it -- I have a lua script that processes my inbound calls, and uses mod_xml_cdr to do a http curl to a web api and also save a copy to the local /log/xml_cdr directory -- When the call comes in - sometimes the xml_cdr works as advertised (posts and saves). This is what the cdr looks like ..... <?xml version="1.0"?> <cdr> <channel_data> <state>CS_REPORTING</state> <direction>inbound</direction> <state_number>11</state_number> <flags>0=1;1=1;35=1;36=1;38=1;41=1;51=1</flags> <caps>1=1;2=1;3=1;4=1;5=1;6=1</caps> </channel_data> and sometimes it saves as follows, which breaks the http curl post. %3C%3Fxml%20version%3D%221.0%22%3F%3E%0A%3Ccdr%3E%0A%20%20%3Cchannel_data%3E%0A%20%20%20%20%3Cstate%3ECS_REPORTING%3C/state%3E%0A%20%20%20%20%3Cdirection%3Einbound%3C/direction%3E%0A%20%20%20%20%3Cstate_number%3E11%3C/state_number%3E%0A%20%20%20%20%3Cflags%3E0%3D1%3B1%3D1%3B35%3D1%3B36%3D1%3B38%3D1%3B41%3D1%3B51%3D1%3C/flags%3E%0A%20%20%20%20%3Ccaps%3E1%3D1%3B2%3D1%3B3%3D1%3B4%3D1%3B5%3D1%3B6%3D1%3C/caps%3E%0A%20%20%3C/channel_data%3E%0A%20%20%3Cvariables%3E%0A%20%20%20%20%3Cdirection%3Einbound%3C/direction%3E%0A%20%20%20%20%3Cuuid%3E8bc9c803-d459-4228-bf9d-87419e57c73b%3C/uuid%3E%0A%20%20%20%20%3Csession_id%3E66%3C/session_id%3E%0A%20%20%20%20%3Csip_local_network_addr%3E192.168.100.82%3C/sip_local_network_addr%3E%0A%20%20%20%20%3Csip_network_ip%3E72.249.14.242%3C/sip_network_ip%3E%0A%20%20%20%20%3Csip_network_port%3E5060%3C/sip_network_port%3E%0A%20%20%20%20%3Csip_received_ip%3E72.249.14.242%3C/sip_received_ip%3E%0A%20%20%20%20%3Csip_received_port%3E5060%3C/sip_received_port%3E%0A%20%20%20%20%3Csip_via_protocol%3Eudp%3C/sip_via_protocol%3E%0A%20%20%20%20%3Csip_from_user%3E4088924027%3C/sip_from_user%3E%0A%20%20%20%20%3Csip_from_uri I have taken the above snippets directly from the xml_cdr files that are saved both calls come in on same gateway and are answered with the same script THE only difference is the caller ID in the first case it is 10 digits NPA-NXX in the second i convert it to 11 digits before executing the bulk of the script with a 1NPA-NXX both numbers are used as strings (confirmed with the lua type(var)) Any ideas welcome - Paul -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-cdr-not-saving-file-c… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Gateway Configuration with No Authentication
by Vishal Kakkar 26 Jan '12

26 Jan '12

Hi, Not using username parameter in gateway is causing following OPTION message being published- OPTIONS sip:192.168.1.30;transport=udp SIP/2.0 Via: SIP/2.0/UDP 59.90.193.54:5080;rport;branch=z9hG4bKZ5D672rZgta9N Max-Forwards: 70 From: *<sip:(null)@192.168.1.30;*transport=udp>;tag=ZS9p1ZN164S4Q To: *<sip:(null)@192.168.1.30;*transport=udp> Call-ID: a78e7d46-c2c1-122f-d9a3-0013a9866b3c CSeq: 23455584 OPTIONS Contact: <sip:gw+192.168.1.50@59.92.192.53:5080 ;transport=udp;gw=192.168.1.50> User-Agent: FreeSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 On Thu, Jan 26, 2012 at 4:33 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Gateway Configuration with No Authentication (Vishal Kakkar) > 2. Re: Gateway Configuration with No Authentication > (georg(a)riseup.net) > 3. Re: Gateway Configuration with No Authentication (Avi Marcus) > 4. Re: Registration Failed with status DNS Error (Antonio) > 5. mod_callcenter - ring tones when phone is ringing (Weigel, Stefan) > > > ---------- Forwarded message ---------- > From: Vishal Kakkar <vishal.kakkar(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Thu, 26 Jan 2012 14:04:40 +0530 > Subject: [Freeswitch-users] Gateway Configuration with No Authentication > Hi all, > > We are trying to use FS to handle incoming call from SIP provider. > Provider doesnt have any authentication because its all inbound traffic. > They have configured their SIP trunk to point to our FreeSwitch Box. > > We have also configured one gateway in eternal/provider.xml with the IP of > provider and disable the registration as provider is not expecting the > same(they dont have any user configured for us, as they are using their box > in trunk mode). > > Issue is that when an incoming call reaches to provider, we are not > getting any INVITE from them and its being dropped at the IP that we have > added as gateway. Reason shared is that they are expecting us to have > heartbeat with their IP. > > When we turn ping option, it sends OPTION message but username is getting > appended to TO and FROM uri's so that cause 404 No User found response. > > Please help to configure FS to send option something like this- > > > *OPTIONS sip:10.129.43.154:5060;ttl=0 SIP/2.0* > > Via: SIP/2.0/UDP 10.129.43.23:5060; > rport;branch=z9hG4bK- > > 6df4-1156281802-19999-423 > > Call-ID: > 5aa9-1e61-7222006212322-chiloe-0(a)10.129.43.23 > > CSeq: 1 OPTIONS > > Max-Forwards: 70 > > *To: <sip:10.129.43.154;ttl=0>* > > *From: <sip:10.129.43.23>* > ;tag=95ffcd055e0f78f7d5d397020e8 > > 9288db5f2 > > User-Agent: Dialogic-SIP/10.3.2.57 chiloe 0 > > Contact: <sip:10.129.43.23:5060> > > Accept: application/ sdp > > Content-Length: 0 > > i.e. They want OPTION with only IP address not user@IP. So that on > recieving any call they can transfer it to our FS. > > Please Help. > Thanks > -Manav > > > ---------- Forwarded message ---------- > From: georg(a)riseup.net > To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 26 Jan 2012 10:08:12 +0100 > Subject: Re: [Freeswitch-users] Gateway Configuration with No > Authentication > He Vishal, > > > We are trying to use FS to handle incoming call from SIP provider. > > Provider > > doesnt have any authentication because its all inbound traffic. They have > > configured their SIP trunk to point to our FreeSwitch Box. > > I don't think you have to use a username. I've got a similar setup, my > external.xml config is looking like: > > <include> > <gateway name="xxx"> > <param name="proxy" value="THE.I.R.IP"/> > <param name="register" value="false"/> > <param name="extension" value="extension_that > _should_be_called_in_your_dialplan"/> > <param name="rtp-autofix-timing" value="false"/> > <param name="caller-id-in-from" value="true"/> > </gateway> > </include> > > HTH, > Georg > > > > > > ---------- Forwarded message ---------- > From: Avi Marcus <avi(a)avimarcus.net> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 26 Jan 2012 11:58:56 +0200 > Subject: Re: [Freeswitch-users] Gateway Configuration with No > Authentication > I usually have my origination providers send to $number@$myserver:5080 so > it goes to external and hits the public context. > Or, if they won't, I create a user with their ACL and set the context on > the user to public. > > Hmm, perhaps external gateway that doesn't register would work and be > cleaner.. > -Avi > > > On Thu, Jan 26, 2012 at 11:08 AM, <georg(a)riseup.net> wrote: > >> He Vishal, >> >> > We are trying to use FS to handle incoming call from SIP provider. >> > Provider >> > doesnt have any authentication because its all inbound traffic. They >> have >> > configured their SIP trunk to point to our FreeSwitch Box. >> >> I don't think you have to use a username. I've got a similar setup, my >> external.xml config is looking like: >> >> <include> >> <gateway name="xxx"> >> <param name="proxy" value="THE.I.R.IP"/> >> <param name="register" value="false"/> >> <param name="extension" value="extension_that >> _should_be_called_in_your_dialplan"/> >> <param name="rtp-autofix-timing" value="false"/> >> <param name="caller-id-in-from" value="true"/> >> </gateway> >> </include> >> >> HTH, >> Georg >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > ---------- Forwarded message ---------- > From: Antonio <asilva(a)wirelessmundi.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 26 Jan 2012 11:05:06 +0100 > Subject: Re: [Freeswitch-users] Registration Failed with status DNS Error > ** > Mike, thanks for the answer. > > Actually the problem is the DNS server doesn't have an A entry for the > destination domain. When i add this entry to the domain, the register works > nice. > > Shouldn't freeswitch try to resolve first to the SRV entry and then if > none is found, try to resolve the A entry? > > you can find my logs in http://pastebin.freeswitch.org/18231 > > > I read something in the jira for some options to disable SRV in the > profile, i look at the code and it seams to be enabled by default. > Just to be sure i also try to put in the profile all the possibilities to > disable-srv and disable-srv503, without luck. > > > > Thanks, > António > > > > On Wed, 2012-01-25 at 13:53 -0500, Michael Jerris wrote: > > Is it even sending the request? try turning on nua and sresolv debug in sofia to see more about what is going on? > > Mike > > On Jan 25, 2012, at 11:37 AM, Antonio wrote: > > > > > Hi > > > > I have a problem registering a gateway, the return error is DNS error [503]. > > > > My DNS is OK, i can dig to the destination domain in question, and even when o do sofia_dig in fresswith it resolves the domain without problems. > > > > > > how does freeswitch resolves the destination domain? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.… > > FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… > > > > > ---------- Forwarded message ---------- > From: "Weigel, Stefan" <Stefan.Weigel(a)allianz-warranty.com> > To: "'freeswitch-users(a)lists.freeswitch.org'" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 26 Jan 2012 12:03:16 +0100 > Subject: [Freeswitch-users] mod_callcenter - ring tones when phone is > ringing > > Hi all,**** > > ** ** > > we’re using successfully mod_callcenter. If a caller enters the queue, > he/she is getting default moh sound. The sound is also playing when the > call is passed to an agent and the phone is ringing. Is it possible to play > a normal ringtone when a call is passed to an agent ?**** > > ** ** > > ** ** > > Thanks and best regards**** > > ** ** > > Stefan**** > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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