July 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Less than one month to go before OpenSIPS Summit!
by Alex Goulis 09 Jul '14

09 Jul '14

*Less than one month to go before the OpenSIPS Summit!* Ever wonder how you can: * load balance your FreeSWITCH servers * provide fail over protection for your FreeSWITCH servers * distribute services to your FreeSWITCH server clusters If you have, then you need to attend the OpenSIPS Summit! For all of you attending ClueCon, we'll be holding an OpenSIPS summit on Aug 4 running most of the day before ClueCon officially kicks off. Receive a 25% discount for the summit by using "clucon2014" as your discount code. Last year's summit was a tremendous success. Everyone that attended enjoyed interesting topics and discussions. Some of the topics this year will include: * Redundancy and high availability for Cloud Solutions using gateways such FreeSWITCH * Advanced SIP Routing using OpenSIPS * Using OpenSIPS as a single entry point for a SIP network You can get more information about the summit at: http://www.opensips.org/Community/Summit-2014Chicago Registration is still open: http://www.opensips.org/Community/Summit-Registration Last year's summit highlights: http://www.opensips.org/Community/Summit-2013Chicago

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configure dialplan gsmopen
by D- Bamboo 09 Jul '14

09 Jul '14

Hi thanks for your answer when i try this command - make a call from the fs_cli and park it: originate gsmopen/RR/+6281313888001 &park() - make a call from the fs_cli and make sound test: originate gsmopen/RR/+6281313888001 &echo() i receive the call to my cell phone. but when i answer there is no sound to my server. and for this instruction - I can see from your logs that on incoming calls you use: bridge(user/${user_data(${destination_number}@${domain_name} attr id)}@${domain_name}) - which later is translated to: EXECUTE gsmopen/gsm01 bridge(user/(a)192.168.1.3) - basically the variable 'user_data' isn't set, which causes the problems on incoming calls sorry sir i don't know how to configure or setting this instruction. what i try to make is i can call to my cell phone from x-lite and i can received a call from cell phone to my x-lite. sorry again i really newbie. i just setting inbound route and outbound route from fusionpbx and i dont know how to set expression input. may you can help me with example xml file and where directory i must put in thank you very much

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configure dialplan gsmopen
by D- Bamboo 09 Jul '14

09 Jul '14

hi please help me to configure my dialplan outbound and inbound call. i use ms windows 7, freeswitch source that compile with VS 2010, mod_gsmopen, huawei e153u, x-lite 4. already follow this step https://wiki.freeswitch.org/wiki/Gsmopen but still can to establish calling to my cellular phone. this is my log when make a call from x-lite to my cell phone and other log when receive call from my cell phone to my modem number. https://jira.freeswitch.org/secure/attachment/21491/freeswitch%28calling%29… https://jira.freeswitch.org/secure/attachment/21492/freeswitch%28receive%20… thanks for your help

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Cannot heard sound on FS
by Nguyễn Văn Nghĩa Em 09 Jul '14

09 Jul '14

Dear All, I have a FS production deployment: - A FS server behind a NAT network - User can register and make calls via internet. It have worked well more than 2 years, but it took problem now: Assume I have following cases: 1. Phone number in PSTN call to Extension in my branch 2. Extensions from another branch call to Extension in my branch (via sip trunk) 3. An extension in my branch call to another extension in my branch 4. An extension in my branch call to a phone number in PSTN 5. An extension in my branch call to a an extension in another branch (via sip trunk) These are results: Cases 1. 2. and 3.: Caller heard well but sometime Calleed can't heard from caller. If i wait for a longer time (may be 1 minutes, 2 minutes...) Calleed might heard from Caller. Cases 4.5: Both Caller and Calleed heard well. I have captured PCAP file on FS server, I saw that there were no different between cases 1,2,3 and case 4,5. I have seen many packets that FS server send to calleed IP. There are only one change to my branch network. Network administrator make a NAT (80 and 8000) for camera service at public interface to a private IP in my branch network. My question is: why Calleed in case 1, 2, 3 can not heard from caller ? Another question: FS use RTP for carrying out voice/video but I can only see about 11 RTP packets (for one direction), but there are too many UDP packets in call session. I have attached the picture for that. In the picture: 113.161.90.103: public ip address in my branch 192.168.45.4: FS server private ip behind NAT 10.12.10.140: My AudioCodes Phone's private IP in my branch Thanks in advance for your help. -- Best regards, ------------------------------------------------------------------------------------------------------------- EM NGUYEN Website: http://freeswitch.vn Mobile: +84 949669075 Yahoo & Skype: nghiaembt --------------------------------------------------------------------------------------------------------------

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Re: [Freeswitch-users] mod_cdr_mongodb and not correct start_uepoch (*_uepoch) values
by Sergey Zhuravlov 09 Jul '14

09 Jul '14

Steven! Thank you! Will use 64-bit platform for mongodb! I'll tell you this in the google group CDR-Stats - they forgot to mention about it. > From: Steven Ayre <steveayre(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Jul 2014 14:28:28 +0100 > Subject: Re: [Freeswitch-users] mod_cdr_mongodb and not correct start_uepoch (*_uepoch) values > Reading the manual I'm guessing you're on a 32bit platform. Running on 64bit would be better as 32bit limits the size of the database. > > Integers are 64bit on 64bit platforms, but you can use bigint to get 64bit numbers on a 32bit platform. > http://api.mongodb.org/perl/current/MongoDB/DataTypes.html > > > On 8 July 2014 14:23, Steven Ayre <steveayre(a)gmail.com> wrote: >> >> I think the data type you're using in the database is the problem. 2147483647 is the maximum value you can store in a signed 32bit integer. You're trying to store a larger value so it's truncating it to the maximum possible value. Try a type that uses 64bits. >> >> >> On 8 July 2014 10:47, Sergey Zhuravlov <4orbit(a)gmail.com> wrote: >>> >>> hi! I tune cdr-stats now. >>> >>> Found that in the database mongodb, which stores data from freeswitch with mod_cdr_mongodb strange value >>> >>> start_epoch ": 1399643826," start_uepoch ": 2147483647 .... "start_stamp": "2014-05-09 13:57:06" >>> >>> The problem with the value "start_uepoch": 2147483647 >>> >>> This is not true value should be: >>> >>> <start_epoch> 1399643826 </ start_epoch> >>> <start_uepoch> 1399643826442976 </ start_uepoch> >>> >>> This is a file from cdr_xml >>> >>> Someone faced with such behavior mod_cdr_mongodb? >>> HOWTO correct it? >>> >>> FreeSWITCH Version 1.5.12b+git~20140419T062924Z~f408dae2af~32bit (git f408dae 2014-04-19 06:29:24Z 32bit) >>> >>> Ubuntu 12.04.4 LTS >>> >>> >>> >>> _________________________________________________________________________ >>> 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 >>> -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com

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Which 1.4 version to use?
by Yehavi Bourvine 08 Jul '14

08 Jul '14

Hi. I am running version 1.2.22 and once in a few days I have a problem of calls losing voice and fs_cli hung (in 1.2.23 it was worse). I would like to try the 1.4 version, and my question is: What is recommended: using 1.4.6 or doing a git clone of 1.4 branch? If the latter, then I guess it would be a good idea to update this once in while, right? Thanks! __Yehavi:

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mod_cdr_mongodb and not correct start_uepoch (*_uepoch) values
by Sergey Zhuravlov 08 Jul '14

08 Jul '14

hi! I tune cdr-stats now. Found that in the database mongodb, which stores data from freeswitch with mod_cdr_mongodb strange value start_epoch ": 1399643826," start_uepoch ": 2147483647 .... "start_stamp": "2014-05-09 13:57:06" The problem with the value "start_uepoch": 2147483647 This is not true value should be: <start_epoch> 1399643826 </ start_epoch> <start_uepoch> 1399643826442976 </ start_uepoch> This is a file from cdr_xml Someone faced with such behavior mod_cdr_mongodb? HOWTO correct it? FreeSWITCH Version 1.5.12b+git~20140419T062924Z~f408dae2af~32bit (git f408dae 2014-04-19 06:29:24Z 32bit) Ubuntu 12.04.4 LTS

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The Call hang up after about 60 seconds by sip.js
by liuyp2 08 Jul '14

08 Jul '14

Hi, Freeswitch, I used a web-ua that registered for 1001 by sip.js to call another web-ua that registered for 1002 by sip.js, all thing works ok, but the call hang up after about 60 seconds caused with 500 internal server error. here is the trace of the call from the freeswitch console with siptrace on after about 60s: ------------------------------------------------------------------------ send 3243 bytes to ws/[10.108.226.157]:5265 at 06:28:05.312440: ------------------------------------------------------------------------ INVITE sip:u8djvd17@952nbceag6b1.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WS 10.108.226.157:5066;branch=z9hG4bKXp48r89B9Q5Sp Max-Forwards: 70 From: <sip:1001@10.108.226.157>;tag=gQet6FS5ttN3r To: <sip:1002@10.108.226.157>;tag=hja4qr590s Call-ID: iee4mgmnj48j1gk8llrl CSeq: 62064378 INVITE Contact: <sip:1001@10.108.226.157:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.12b~32bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, RE FER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 2562 v=0 o=- 4843282287768807212 2 IN IP4 10.108.226.157 s=- t=0 0 a=group:BUNDLE audio video a=msid-semantic: WMS cUjDKZuVkeKIQ8S9wg68Ca8YGzKRAKF399Dt m=audio 37393 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 183.221.39.14 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=rtcp:1 IN IP4 0.0.0.0 a=candidate:1108746152 1 udp 2122260223 10.108.226.157 64558 typ host generat ion 0 a=candidate:2392531003 1 udp 1686052607 183.221.39.14 37393 typ srflx raddr 1 0.108.226.157 rport 64558 generation 0 a=candidate:211149656 1 tcp 1518280447 10.108.226.157 0 typ host generation 0 a=ice-ufrag:J8KtuA2R4XMBjEqi a=ice-pwd:+QrbfIxUKuVASsuqH3nLvSqW a=fingerprint:sha-256 A9:B5:DB:C4:40:2E:8A:C1:85:33:8F:CB:1D:49:CE:1C:17:84:9 C:A5:41:E9:DF:23:2B:84:6B:F0:2D:22:C2:60 a=setup:active a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=rtcp-mux a=maxptime:60 a=ssrc:4249320145 cname:+bv8SIkztDhAAFyt a=ssrc:4249320145 msid:cUjDKZuVkeKIQ8S9wg68Ca8YGzKRAKF399Dt 28cefa96-480d-441 f-bb56-4213896ef542 a=ssrc:4249320145 mslabel:cUjDKZuVkeKIQ8S9wg68Ca8YGzKRAKF399Dt a=ssrc:4249320145 label:28cefa96-480d-441f-bb56-4213896ef542 m=video 37393 RTP/SAVPF 100 116 117 c=IN IP4 183.221.39.14 a=rtpmap:100 VP8/90000 a=rtpmap:116 red/90000 a=rtpmap:117 ulpfec/90000 a=rtcp:1 IN IP4 0.0.0.0 a=candidate:1108746152 1 udp 2122260223 10.108.226.157 64558 typ host generat ion 0 a=candidate:2392531003 1 udp 1686052607 183.221.39.14 37393 typ srflx raddr 1 0.108.226.157 rport 64558 generation 0 a=candidate:211149656 1 tcp 1518280447 10.108.226.157 0 typ host generation 0 a=ice-ufrag:J8KtuA2R4XMBjEqi a=ice-pwd:+QrbfIxUKuVASsuqH3nLvSqW a=fingerprint:sha-256 A9:B5:DB:C4:40:2E:8A:C1:85:33:8F:CB:1D:49:CE:1C:17:84:9 C:A5:41:E9:DF:23:2B:84:6B:F0:2D:22:C2:60 a=setup:active a=mid:video a=extmap:2 urn:ietf:params:rtp-hdrext:toffset a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=rtcp-mux a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 nack pli a=rtcp-fb:100 goog-remb a=ssrc:2020563265 cname:+bv8SIkztDhAAFyt a=ssrc:2020563265 msid:cUjDKZuVkeKIQ8S9wg68Ca8YGzKRAKF399Dt 0f941cdb-811a-42f f-8c4c-dc88444b2e7b a=ssrc:2020563265 mslabel:cUjDKZuVkeKIQ8S9wg68Ca8YGzKRAKF399Dt a=ssrc:2020563265 label:0f941cdb-811a-42ff-8c4c-dc88444b2e7b ------------------------------------------------------------------------ recv 278 bytes from ws/[10.108.226.157]:5265 at 06:28:05.339441: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/WS 10.108.226.157:5066;branch=z9hG4bKXp48r89B9Q5Sp To: <sip:1002@10.108.226.157>;tag=hja4qr590s From: <sip:1001@10.108.226.157>;tag=gQet6FS5ttN3r Call-ID: iee4mgmnj48j1gk8llrl CSeq: 62064378 INVITE Supported: outbound Content-Length: 0 ------------------------------------------------------------------------ recv 293 bytes from ws/[10.108.226.157]:5265 at 06:28:05.378443: ------------------------------------------------------------------------ SIP/2.0 500 Server Internal Error Via: SIP/2.0/WS 10.108.226.157:5066;branch=z9hG4bKXp48r89B9Q5Sp To: <sip:1002@10.108.226.157>;tag=hja4qr590s From: <sip:1001@10.108.226.157>;tag=gQet6FS5ttN3r Call-ID: iee4mgmnj48j1gk8llrl CSeq: 62064378 INVITE Supported: outbound Content-Length: 0 ------------------------------------------------------------------------ send 315 bytes to ws/[10.108.226.157]:5265 at 06:28:05.378443: ------------------------------------------------------------------------ ACK sip:u8djvd17@952nbceag6b1.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WS 10.108.226.157:5066;branch=z9hG4bKXp48r89B9Q5Sp Max-Forwards: 70 From: <sip:1001@10.108.226.157>;tag=gQet6FS5ttN3r To: <sip:1002@10.108.226.157>;tag=hja4qr590s Call-ID: iee4mgmnj48j1gk8llrl CSeq: 62064378 ACK Content-Length: 0 ------------------------------------------------------------------------ send 562 bytes to ws/[10.108.226.157]:5265 at 06:28:05.378443: ------------------------------------------------------------------------ BYE sip:u8djvd17@952nbceag6b1.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WS 10.108.226.157:5066;branch=z9hG4bKyZX1t3tF60Ucj Max-Forwards: 70 From: <sip:1001@10.108.226.157>;tag=gQet6FS5ttN3r To: <sip:1002@10.108.226.157>;tag=hja4qr590s Call-ID: iee4mgmnj48j1gk8llrl CSeq: 62064379 BYE Contact: <sip:1001@10.108.226.157:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.12b~32bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, RE FER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2014-07-08 14:51:17 Tonny

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ClueCon Hotel Reminder
by Ken Rice 07 Jul '14

07 Jul '14

Hey Ya¹ll, Just a reminder we¹ve got most of the block the hotel has set aside for ClueCon members used up at this point. If you have not made your reservations for the hotel, I would do so ASAP. You can reserve them via http://goo.gl/biKxyy (complete with all the discount codes for ClueCon reservations in there). If you run into any problems booking your room, call Brian West @ +1 877-742-CLUE so we can get it addressed ASAP! See you @ ClueCon! ClueCon.com -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Re: [Freeswitch-users] condition
by Andrew 07 Jul '14

07 Jul '14

What I am trying to do is wait for the variable to be set. But it seems that the variable is never set in this code. Anyone have an idea on how to wait for a variable to be set? <extension name="public_extensions"> <condition field="destination_number" expression="^.*$"> <action application="set" data="hangup_after_bridge=true"/> <condition field="${my_dialstring}" expression="^$"> <action application="sleep" data="20"/> </condition> <condition field="${my_dialstring}" expression="^$"> <action application="sleep" data="500"/> </condition> <action application="bridge" data="${my_dialstring}"/> </condition> </extension> From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of aademattia(a)comcast.net Sent: Saturday, June 28, 2014 2:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] condition I am trying to find out if the variable is not set. The problem is this condition fires if the variable is set or not set. If the variable is not set, I want it to sleep. Sent from my HTC ----- Reply message ----- From: "Ken Rice" <krice(a)freeswitch.org <mailto:krice@freeswitch.org> > To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> > Subject: [Freeswitch-users] condition Date: Sat, Jun 28, 2014 1:24 PM thats tesing for an empty string... test to see if what you expect to be there and if not use anti-action to sleep Ken Sent from my iPad On Jun 28, 2014, at 11:15, "Andrew" <aademattia(a)comcast.net <mailto:aademattia@comcast.net> > wrote: Hi, I want to see if the variable is set. If its not I want to sleep for 500 mill sec. This does not seem to work. <condition field="${my_dialstring}" expression="^$"> <action application="sleep" data="500"/> </condition> _________________________________________________________________________ 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 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://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org

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