- FreeSWITCH-users - FreeSWITCH Mailing Lists

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!

1 0

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

2 2

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

1 1

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

4 11

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.

1 0

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 > >

2 1

Call recovery - Multi-primary / multi-backup scenario
by Carlo Dimaggio 26 Jan '12

26 Jan '12

Hi all, I would like to know if the track-calls feature could be used in a multi-primary - multi-backup scenario. What I think is an environment with Kamailio dispatcher to N freeswitch boxes with a (multi or) single freeswitch backup. In case of failure, one primary should be replaced with a backup (spare) box that will take its IP and active calls. If all active freeswitch track call state in the same table/database, how the backup box will know what calls should be recovered? Anyone could explain the call recovery algorithm? Best Regards,

3 5

FreeSWITCH load CDR CSV to Mysql, theoretical problem?
by Henrik Aagaard Sørensen 26 Jan '12

26 Jan '12

I'm using the script on: http://convergence.pk/freeswitch-load-cdr-csv-to-mysql.html which also can be found on the http://wiki.freeswitch.org/wiki/Mod_cdr_csv. I'm having in running every 5 minute. And everything works. But I do have some concern. What if the job suddenly takes more than 5 minutes to complete and another job start, doing the same thing. Couldn't that be a problem and can it be fixed?

3 5

Registration Failed with status DNS Error
by Antonio 26 Jan '12

26 Jan '12

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? if you need more information, please let me know. Btw i just compile the last git head. Thanks, António ******** Log register. 2012-01-25 17:20:42.637192 [NOTICE] sofia_reg.c:414 Registering sol 2012-01-25 17:20:42.737189 [ERR] sofia_reg.c:2004 sol Registration Failed with status DNS Error [503]. failure #1 2012-01-25 17:20:43.637195 [WARNING] sofia_reg.c:473 sol Failed Registration [0], setting retry to 30 seconds. ******** sofia status gateway sol: freeswitch@internal> sofia status gateway sol ================================================================================================= Name sol Profile 192.168.10.25 Scheme Digest Realm mobile.commsmundi.com:5060 Username 2203 Password yes From <sip:2203@mobile.commsmundi.com:5060;transport=udp> Contact <sip:gw+sol@192.168.10.25:5060;transport=udp;gw=sol> Exten 2203 To sip:2203@mobile.commsmundi.com:5060 Proxy sip:mobile.commsmundi.com:5060 Context incoming Expires 1800 Freq 1800 Ping 1327508466 PingFreq 25 PingState -1/0/1 State FAIL_WAIT Status DOWN CallsIN 0 CallsOUT 0 FailedCallsIN 0 FailedCallsOUT 0 ================================================================================================= ******** sofia_dig: freeswitch@internal> sofia_dig mobile.commsmundi.com Preference Weight Transport Port Address ================================================================================ 1 0.200 udp 5060 192.168.10.1

2 2

Gateway Configuration with No Authentication
by Vishal Kakkar 26 Jan '12

26 Jan '12

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

3 2