May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP NOTIFY for MWI with AVAYA 9630
by Bernard Murphy 08 May '12

08 May '12

On anything newer than 1.06 of freeswitch the SIP NOTIFY sent to an Avaya 9630 phone does not get accepted and trigger the MWI light. Works absolutely fine on 1.0.6. The only difference in the messages i can see is in the contact parameter, possibly to do with the angle bracket. Can anyone advise if the difference shown in the two messages below is acceptable SIP or if something is broken in freeswitch. Also, I'm happy to plough through code and try to fix if you can point me in the right direction as to which src file constructs the SIP Notify message ? WORKING VERSION 1.0.6 No. Time Source Destination Protocol Length Info 1169 19:32:47 192.168.2.41 192.168.2.30 SIP 996 Request: NOTIFY sip:1001@192.168.2.30;transport=udp Frame 1169: 996 bytes on wire (7968 bits), 996 bytes captured (7968 bits) Ethernet II, Src: Vmware_1b:7d:13 (00:0c:29:1b:7d:13), Dst: Tenovis_b6:ec:4f (00:07:3b:b6:ec:4f) Internet Protocol Version 4, Src: 192.168.2.41 (192.168.2.41), Dst: 192.168.2.30 (192.168.2.30) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: NOTIFY sip:1001@192.168.2.30;transport=udp SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.2.41;rport;branch=z9hG4bKQ656yDjS3mavS Transport: UDP Sent-by Address: 192.168.2.41 RPort: rport Branch: z9hG4bKQ656yDjS3mavS Max-Forwards: 70 From: <sip:1001@192.168.2.41>;tag=eQH9S8pjtvp6N To: <sip:1001@192.168.2.41 >;tag=286229214fa041cb4fa80fb2_F1001192.168.2.30 Call-ID: 1_7c8e554c7b5cb4fa80abf_S(a)192.168.2.30 CSeq: 27613481 NOTIFY Sequence Number: 27613481 Method: NOTIFY * Contact: <sip:1001@192.168.2.41:5060>* User-Agent: FreeSWITCH-mod_sofia/1.0.6-hacked-20120317T221523Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=21 Content-Type: application/simple-message-summary Content-Length: 91 Message Body Messages-Waiting: yes\r\n Message-Account: sip:1001@192.168.2.41\r\n Voice-Message: 1/0 (0/0)\r\n \r\n BROKEN VERSION LATEST GIT No. Time Source Destination Protocol Length Info 6824 18:33:12 192.168.2.41 192.168.2.30 SIP 1034 Request: NOTIFY sip:1001@192.168.2.30;transport=udp Frame 6824: 1034 bytes on wire (8272 bits), 1034 bytes captured (8272 bits) Ethernet II, Src: Vmware_1b:7d:13 (00:0c:29:1b:7d:13), Dst: Tenovis_b6:ec:4f (00:07:3b:b6:ec:4f) Internet Protocol Version 4, Src: 192.168.2.41 (192.168.2.41), Dst: 192.168.2.30 (192.168.2.30) User Datagram Protocol, Src Port: sip (5060), Dst Port: iad2 (1031) Session Initiation Protocol Request-Line: NOTIFY sip:1001@192.168.2.30;transport=udp SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.2.41;rport;branch=z9hG4bKyyreB17m4eS1c Transport: UDP Sent-by Address: 192.168.2.41 RPort: rport Branch: z9hG4bKyyreB17m4eS1c Max-Forwards: 70 From: <sip:1001@192.168.2.41>;tag=Oa5sgbB1o7sl SIP from address: sip:1001@192.168.2.41 SIP tag: Oa5sgbB1o7sl To: <sip:1001@192.168.2.41 >;tag=2b269bdd4fa034ed4fa2ffe4_F1001192.168.2.30 SIP to address: sip:1001@192.168.2.41 SIP to address User Part: 1001 SIP to address Host Part: 192.168.2.41 SIP tag: 2b269bdd4fa034ed4fa2ffe4_F1001192.168.2.30 Call-ID: 7_2c27465e595004fa2f77c_S(a)192.168.2.30 CSeq: 231978492 NOTIFY Sequence Number: 231978492 Method: NOTIFY * Contact: <sip:1001@192.168.2.41:5060;transport=udp> Contact-URI: sip:1001@192.168.2.41:5060;transport=udp Contactt-URI User Part: 1001 Contact-URI Host Part: 192.168.2.41 Contact-URI Host Port: 5060 Contact parameter: transport=udp>* User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-de019ab 2012-05-03 15-23-57 +0200 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=60 Content-Type: application/simple-message-summary Content-Length: 91 Message Body Messages-Waiting: yes\r\n Message-Account: sip:1001@192.168.2.41\r\n Voice-Message: 2/0 (0/0)\r\n \r\n thanks as ever ! Bernie

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productivity of FS
by Phil Quesinberry 07 May '12

07 May '12

I must echo the sentiments of the other more experienced users here. A couple of quick questions do come to mind though. Is FS proxying the media between the endpoints? If so, is there any transcoding going on? If that's the case, you may want to enable late negotiation and tweak the codecs to avoid transcoding whenever possible, that can potentially take up a lot of system resources. Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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Recording all calls with good quality.
by brandon.mcginty＠gmail.com 07 May '12

07 May '12

Good day. I'm trying to record all calls sent through my Freeswitch PBX. The problem is that the quality is horrible. I've tried wav and GSM formats, and both are les than satisfactory. I'm looking for something around the quality of the recent recordings of the Freeswitch conference. (I checked on the wiki, and couldn't find anything through search.) Any information on the FS conference format or my configuration would be appreciated. I've got:  <condition field="destination_number" expression="^(.+)$" break="never"> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> </condition> Sincerely, Brandon McGinty-Carroll

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Javascript Outbound Event Socket - Linger Command
by Sara Higfler 07 May '12

07 May '12

Hi, I'm a newbie to developing an outbound event handler for Freeswitch. I'm looking to use the Javascript ESL implemention (using node.js) and have managed to get basic scenarios working, including digit collection and checking against a MySQL database. One problem I have is the capture of call termination events in my script. Having read around a lot, I know that I'm meant to use the Linger command, but cannot find any examples of how to do this with a Javascript outbound handler. I've included the rough structure of my code below (details removed for brevity) - I would really appreciate if someone could help show me how I would implement the linger command to ensure I capture all call termination events. Kind regards. (function() { var server, esl; esl = require('esl'); util = require('util'); server = esl.createCallServer(); server.on('CONNECT', function(req, res) { var uri, channel_data, unique_id; channel_data = req.body; unique_id = channel_data['Unique-ID']; req.execute('answer'); req.execute('playback', 'hello.wav'); req.on('DTMF', function(req) { var digit; var channel_data; channel_data = req.body; digit = channel_data['DTMF-Digit']; console.log('DTMF Received=' + digit); return util.log('DTMF Received'); }); req.on('CHANNEL_ANSWER', function(req) { return util.log('Call was answered'); }); req.on('CHANNEL_HANGUP', function(req) { console.log('CHANNEL_HANGUP'); return util.log('CHANNEL_HANGUP'); }); req.on('CHANNEL_HANGUP_COMPLETE', function(req) { console.log('CHANNEL_HANGUP_COMPLETE'); return util.log('CHANNEL_HANGUP_COMPLETE'); }); req.on('DISCONNECT', function(req) { console.log('DISCONNECT'); return util.log('DISCONNECT'); }) }); return util.log('CONNECT received'); }); server.listen(9123); }).call(this);

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News and Notes: ClueCon Hotel Gets a New Name, FreeSWITCH Gets a New Feature
by Michael Collins 07 May '12

07 May '12

Happy new week to all! We have some interesting news items to share. First off, we are happy to report that the Wyndham Hotel, where ClueCon 2012 is being held, has a new owner and will become a Hyatt property. As part of the transition, the Wyndham will be renamed in June. The new name will be Hyatt Chicago Miracle Mile. The ownership transfer and name change will not affect ClueCon attendees, and the ClueCon website <http://www.cluecon.com/hotels?cc12-0507>will continue to have the latest hotel information. Please feel free to call the Wyndham and make your ClueCon hotel reservations at your earliest convenience. Also in ClueCon news we'd like to let everyone know that we have two new sponsors: Vitelity and OpenVox. We are pleased to welcome them both to this year's event. Each will be giving a technical presentation this year and we look forward to seeing them in person. Please visit our site<http://www.cluecon.com?cc12-0507>for registration information. FreeSWITCH received a new feature last week: a much-improved group pickup implementation. The new pickup dialplan<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pickup?cc12-0507>application works in conjunction with the new "pickup" pseudo-channel. Together these allow for simple yet elegant call pickup features to be added to the dialplan. A number of FreeSWITCH community members are testing this new feature, and soon there will be an example added to the "Vanilla" dialplan. This Wednesday, May 9, we will be welcoming FreeSWITCH Community member Giovanni Maruzzelli to the community conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_05_09?cc12-0507>. He will be discussing how to implement a GSM channel in FreeSWITCH using mod_gsmopen and a GSM dongle. We invite everyone to join us this Wednesday at 1PM Eastern, 10AM Pacific (UTC 1700) for a very interesting discussion. Have a great week and we'll be in touch next Monday. -- Michael S Collins ClueCon Team www.ClueCon.com <http://www.ClueCon.com?cc12-0507> 877-7-4ACLUE

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Does Freeswitch replace Asterisk?
by Antonio Modesto 07 May '12

07 May '12

Hi, I Work at an ISP and we have an Asterisk PBX. Our PBX doesn't have anything special, it just do the common things like auto attendant, CDR, call transfer and such things. We have 2 digium FXO cards to connect to the PSTN. The problem is that asterisk is not working very well, sometimes some of the dahdi channels get stuck and we need to release it through the asterisk console, the detection of callerid doesn't work (We live in brazil, dtmf signaling), and other detais that if I list all of them here it's going to take some time. I read a lot about asterisk and its bad design, I think that it added a lot a features without worrying about making stable implementations. My question is, can freeswitch fully replace our Asterisk PBX? Or it's not its purpose? Regards.

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Max Call Limit
by Andrew Paul 07 May '12

07 May '12

Hai , How can i make freeswitch gateway(trunk) to handle maximum 'n' number of calls simutaneiously . Once if it crosses the limit i should get the response "503" response . I can make this one globally in switch.conf.xml by setting max-session . Thanks And Regards ANDREW

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Openvox tdma400p fail to Disconnect/Hangup detect (Dahdi/Freetdm)
by RrodolfoS . 07 May '12

07 May '12

Hi, Freeswitch can recive calls, but the call is end, openvox tdma400p, can't detect disconnection or hangup. Any idea? RrodolfoS

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New proof-of-concept module (mod_pickupgroup)
by João Mesquita 07 May '12

07 May '12

Hello folks, I've been working lately with pickup groups and I found some situations where the way the intercept application is presented along with the hash application on the default configs will not suffice the expected behavior on a PBX environment. Of course there are many solutions to this problem and I bet that there are even possible solutions without coding and just dial plan tricks. Nonetheless, a beginner could probably overlook some of those situations and this is the reason why I have started this module. The first problem I am trying to solve is that hash will only hold 1 UUID per group. If you have 2 consecutive calls and want to pick both up, you won't be able to. Simple way to test it. A calls B and leaves B ringing. C calls D and leaves D ringing. E intercepts (in which case it will intercept C). F intercepts same group. Expected behavior is to intercept A, but hash has forgotten about A and therefore you will get reorder tone. By "borrowing" (nicer than stealing) the idea from `eavesdrop all`, I have created mod_pickupgroup as a proof of concept and I would like the community not only to test the module but also to provide input as far as the code is concerned (I am not that proficient with C) and as far as functionality is concerned. The code is currently hosted on Github with the intent to move it somewhere more definitive if the module ends up being useful. The link is here: http://github.com/jmesquita/mod_pickupgroup To compile it, you need to adjust the BASE variable inside the Makefile by pointing it to your latest FreeSWITCH™ source. -- João Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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freetdm ring cadence
by Kevin Olinger 06 May '12

06 May '12

Is it possible to specify a ring cadence in the dialplan with freetdm? I have an analog extension (TDM400) that I would like to ring with a different cadence for internal and external calls. Thanks!

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