July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] From header rewrite on B2BUA scenario
by Peter Olsson 08 Jul '11

08 Jul '11

Make sure to set caller-id-in-from to true in the gateway config. /Peter ----- Reply message ----- Från: "Dave Horton" <dave(a)dchorton.com> Datum: fre, jul 8, 2011 06:53 Rubrik: [Freeswitch-users] From header rewrite on B2BUA scenario Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> I have a scenario where I have FS configured so as to take an incoming call and send it out through a configured gateway. Now this gateway does not require authentication, so in my gateway configuration I have the register param set to false: <gateway name="gwInternal"> <param name="username" value="foo"/> <param name="password" value="bar"/> <param name="realm" value="204.215.65.202"/> <param name="proxy" value="204.215.65.202"/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> Now, even though I am not registering to this gateway, FS requires me to provide a username and password or else I get an error when starting/loading this configuration. So you can see above that I have just put in a username of "foo", just to have something there. Now I come to the problem. The result of the above configuration is that the From header coming in looks like this: From: <sip:+15083084809@204.215.65.201:5060>;tag=3519088051-756100 and the one that FS creates to send out through this gateway looks like this: From: "+15083084809" <sip:foo@204.215.65.202;transport=udp>;tag=Sr6taB8rFgt1H You can see that it has inserted the username of 'foo' in the user part of the uri of the From header. The reason that this is a problem is that this INVITE is going out to another remote FS, as it turns out. And when the dialed remote extension off that FS PBX does not pick up it then goes into voicemail, and the username of 'foo' is what is stored as the calling number. So when the user on that remote PBX collects that voicemail and tries to do call return, the FS ends up trying to transfer a call to 'foo', which fails. So call return is not working in the remote FS due to this rewriting of the From header. Is there a way I can configure my FS to send calls out through a gateway and maintain the user part of the From URI that was received on the incoming call? Or, alternatively, can I configure the voicemail application to look for the caller id field from somewhere else than the user part of the From URI ? Dave _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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 !DSPAM:4e168bec32761053165832!

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From header rewrite on B2BUA scenario
by Dave Horton 08 Jul '11

08 Jul '11

I have a scenario where I have FS configured so as to take an incoming call and send it out through a configured gateway. Now this gateway does not require authentication, so in my gateway configuration I have the register param set to false: <gateway name="gwInternal"> <param name="username" value="foo"/> <param name="password" value="bar"/> <param name="realm" value="204.215.65.202"/> <param name="proxy" value="204.215.65.202"/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> Now, even though I am not registering to this gateway, FS requires me to provide a username and password or else I get an error when starting/loading this configuration. So you can see above that I have just put in a username of "foo", just to have something there. Now I come to the problem. The result of the above configuration is that the From header coming in looks like this: From: <sip:+15083084809@204.215.65.201:5060>;tag=3519088051-756100 and the one that FS creates to send out through this gateway looks like this: From: "+15083084809" <sip:foo@204.215.65.202;transport=udp>;tag=Sr6taB8rFgt1H You can see that it has inserted the username of 'foo' in the user part of the uri of the From header. The reason that this is a problem is that this INVITE is going out to another remote FS, as it turns out. And when the dialed remote extension off that FS PBX does not pick up it then goes into voicemail, and the username of 'foo' is what is stored as the calling number. So when the user on that remote PBX collects that voicemail and tries to do call return, the FS ends up trying to transfer a call to 'foo', which fails. So call return is not working in the remote FS due to this rewriting of the From header. Is there a way I can configure my FS to send calls out through a gateway and maintain the user part of the From URI that was received on the incoming call? Or, alternatively, can I configure the voicemail application to look for the caller id field from somewhere else than the user part of the From URI ? Dave

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Playing multiple files simultaneously
by Bryan Smart 07 Jul '11

07 Jul '11

Is it possible for Freeswitch to play more than one file to a channel at a time? What I've seen and tried from the dialplan and scripts either queues files to play, or will stop a currently playing file so that the newly requested file will play. This also seems to be the case in conferences. When I send multiple play commands to conferences, the files are queued. As for how this might be used, think of an IVR that plays queued prompts, yet continuously plays looping music or a Shoutcast stream in the background. I also want to be able to play short cue tones that start at the same time as a prompt (don't want to pre-mix them in to a single file, though). Is this currently possible through any means? Perhaps with the event socket? Bryan

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Freeswitch dynamic routing of all calls
by Mateusz Bartczak 07 Jul '11

07 Jul '11

Hi all I'm new to FS and I would like to know is it possible to implement following scenario: 1. User dials number 2. Routing script detects outgoing call event. Every call needs to be handled by routing script. 3. Routing script takes in input: user name, domain, dialed number. Than it query database to find best SIP trunk to route the call, it also checks destination price per minute and calculates maximum call time for prepaid user. 4. Routing script output is: SIP trunk to use, SIP call parameters (ie. callerid), maximum call duration 5. FS read output from routing script and make call using returned parameters Preferred routing implementation technology: background running unix deamon written in Java or PHP. Connection with FS via socket. Event routing script will be multi-threaded, must be able to deal with a lot of calls in parallel and processing of one call should not block processing of other calls (I have this problem with Yate voip server, and that's really big problem) Is it possible to do this using FS? Any advices where to search for additional info? I know that there is event handler but can it return "dialstring" for outgoing call events? Some code examples? I will really appreciate your help

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Having problem completing calls between two phones behind double NAT...
by Rob Hutton 07 Jul '11

07 Jul '11

The network layout it something like: FS and Ext 3 <- Local Net ->Inet Rtr(Static NAT) <== INET ==>Inet Rtr(Dynamic NAT)<- Local Net -> Ext 1 and Ext 2 where: The public IP of the Freeswitch box is static There are 2 sip profiles, one for the internal clients on 5060 and one for the NATd clients on 5070 with the external IP hard set and Freeswitch running with -nonat Ext 3 is on the same subnet as the FS server and is using sipinterface_1 to connect Ext 1 and Ext2 are on the same subnet on the remote side of the double nat and using sipinterfacec_2 I can call from Ext 1 to Ext 3 with no problem I can call from Ext 2 to Ext 3 with no problem When I try to call from Ext 1 to Ext 2 or Ext 2 to Ext 1, I get a 606 USER_NOT_REGISTERED show registrations seem to indicate that they are, and I can't find any difference in the call processing to the extensions that it works to, and those that it doesn't. Pastebin with registrations and failed call log is: http://pastebin.freeswitch.org/16705 Thanks in advance for any help...

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absolute_codec_string
by David Villasmil 07 Jul '11

07 Jul '11

Hello Guys, I might be getting it wrong, but i'm trying to send out for the B-leg ONLY g.729 with this in the dial plan: <variable name="absolute_codec_string" value="G729"/> FS is still offering what it gets on the A-leg... any ideas? Thanks David

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SIP 100 response and media
by Boris Kovalenko 07 Jul '11

07 Jul '11

Hello! This thread again. There are two logs. First between Cisco<->Cisco call and second between Cisco<->FreeSwitch. Cisco<->Cisco works fine. As I can see the difference is that Cisco has a=rtpmap:8 PCMA/8000 and FreeSwitch not. May be this is a problem? Is this field mandatory? Jul 6 18:20:39.341: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:28032@192.168.35.15:5060 SIP/2.0 Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F Remote-Party-ID: "test" <sip:30030@10.240.14.15>;party=calling;screen=no;privacy=off From: "test" <sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To: <sip:28032@192.168.35.15> Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA(a)10.240.14.15 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2076058060-2805141984-2507440228-2131084746 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1309976439 Contact: <sip:30030@10.240.14.15:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 187 v=0 o=CiscoSystemsSIP-GW-UserAgent 0 703 IN IP4 10.240.14.15 s=SIP Call c=IN IP4 10.240.14.15 t=0 0 m=audio 18720 RTP/AVP 8 c=IN IP4 10.240.14.15 a=rtpmap:8 PCMA/8000 a=ptime:20 Jul 6 18:20:39.361: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F From: "test" <sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To: <sip:28032@192.168.35.15> Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA(a)10.240.14.15 Timestamp: 1309976439 CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Jul 6 18:20:39.369: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F From: "test" <sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To: <sip:28032@192.168.35.15>;tag=CBEFC0-16E4 Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA(a)10.240.14.15 Timestamp: 1309976439 CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: <sip:28032@192.168.35.15:5060> Supported: replaces Supported: sdp-anat Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 194 When call is between Cisco<->Freeswitch messages are Jul 6 18:21:11.966: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:73435230030@192.168.3.6:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp Max-Forwards: 67 From: "anonymous" <sip:0000000000@192.168.1.1>;tag=QFKH52Z9NaN2c To: <sip:73435230030@192.168.3.6:5060> Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac CSeq: 14652171 INVITE Contact: <sip:mod_sofia@192.168.1.1:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1c95ad9 2011-01-20 22-43-50 -0300 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 329 X-FS-Support: update_display Remote-Party-ID: "anonymous" <sip:0000000000@192.168.1.1>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1309956111 1309956112 IN IP4 192.168.1.1 s=FreeSWITCH c=IN IP4 192.168.1.1 t=0 0 m=audio 20360 RTP/AVP 8 0 18 101 13 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 19030 RTP/AVP 31 34 98 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:98 H264/90000 Jul 6 18:21:11.986: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp From: "anonymous" <sip:0000000000@192.168.1.1>;tag=QFKH52Z9NaN2c To: <sip:73435230030@192.168.3.6:5060> Date: Wed, 06 Jul 2011 18:21:11 GMT Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac CSeq: 14652171 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 -- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005

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Custom GUI for FS
by Alessandro 07 Jul '11

07 Jul '11

Hi, I'm going to develop a custom GUI web-based for freeswitch that allow only to add/remove extension and show the call active and the status of extensions configured. Where can I find in FS the information about actives call and extensions status? Thanks Alessandro Luppi -- Ing. Alessandro Luppi Software development Seletech srl Via Collodi 8, 20052 Monza (MI) - Italy Tel: +39.039.5962000 - Fax: +39.039.9716905 email: a.luppi(a)seletech.com - Web: www.seletech.com or www.seletech.eu

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Chinese sounds
by Seven Du 07 Jul '11

07 Jul '11

Hi, We got someone in the Chinese Community would like to record a Chinese version of sounds, here's some questions for the community - 1) Where can I find the original English scripts ? It would be hard to listen all sound files 2) What's base256/*.wav usage? 3) We'd like to share to world, so, how to upload to files.freeswitch.org when we are done? 4) I guess the raw recording should be 48K, 16bit, single channel, any other params should we set when recording? Thanks. -- Seven Du About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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question about continue_on_fail
by Boris Kovalenko 07 Jul '11

07 Jul '11

Hello! Here is an extension: <extension name="ext_domestic_e164"> <condition field="destination_number" expression="^(7[0-9]{10})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/>  <action application="lcr" data="$1 lcr_tagnet"/> <action application="set" data="call_timeout=60"/> <action application="set" data="bridge_answer_timeout=60"/> <action application="set" data="disable-transcoding=true"/> <action application="bridge" data="${lcr_auto_route}"/> </condition> </extension> lcr_auto_route returns 2 bridges to a same destination. [lcr_carrier=naukanet.ru,lcr_rate=1.45140,v_ats_dstport=50002]sofia/epbx/89979049898ZZZ@X.X.X.42:5061|[lcr_carrier=mtt.ru,lcr_rate=1.60480,v_ats_dstport=50006]sofia/epbx/79049898ZZZ@Y.Y.Y.132:5060 Here is a problem: when the reject from the remote is [CS_CONSUME_MEDIA] [USER_BUSY] the call is routed to the next bridge. When the reject is [CS_EXCHANGE_MEDIA] [USER_BUSY] the bridge hangs up. So, what is the right way to stop processing when USER_BUSY arrived? I tried continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION without success. -- Regards, Boris

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