October 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to send sip MESSAGE from script
by Khue Nguyen Minh 16 Oct '12

16 Oct '12

Hi all, I want send message "MESSAGE" to caller from javascript. How I can do it, please guide me? Brs, Khue.

2 13

How to set destination_number_ton when originating a call with bgapi
by mbo 16 Oct '12

16 Oct '12

We are running freeswitch connected with a sangoma A101 card to the PSTN. We have an application that is connected to the event socket to control incoming calls. When bridging a call to an external number, I set the profile variable "destination_number_ton" and put it also as option to the bridge command to have the variable filled in all channel events. sendmsg call-command: execute execute-app-name: set_profile_var execute-app-arg: destination_number_ton=1 event-lock: true sendmsg call-command: execute execute-app-name: bridge execute-app-arg: {calltype=FORWARD,destination_number_ton=1}freetdm/1/a/XXXXXX event-lock: true In this scenario everything works fine, the type for the Called Party Number is set correctly to "international". Prot Disc:Q.931/I.451 (0x08) Call Ref:007C (Originating side) Type:SETUP (0x5) Sending complete: Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64 Kbit/s(16) L1Prot:G.711 A-Law(3) Channel Id:No:1 Type:B-chans(3) Exclusive/Implicit Calling Party Number:XXXXX(l:11) plan:isdn(1) type:national(2)scr:user, passed(1) pres:allowed(0) Called Party Number:XXXX(l:15) plan:isdn(1) type:international(1) <<<----------------- OK [ 08 02 00 7c 05 a1 04 03 80 90 a3 18 03 a9 83 81 6c 0d 21 81 32 32 31 35 35 34 30 30 33 31 38 70 10 91 34 39 32 32 31 34 30 30 36 37 38 33 31 30 33 ] For another scenario I need to originate calls with bgapi function like: bgapi originate {origination_caller_id_number=XXXX,originate_timeout=20,origination_uuid=7a1093b5-4225-4a99-b3dc-bb4254217e4,destination_number_ton=1}freetdm/1/a/XXXX &park event if I set the profile_var destination_number_ton, it's not "transported" to the ISDN layer. In the ISDN trace I can see that the number type of the Called Party Number is always "unknown". Prot Disc:Q.931/I.451 (0x08) Call Ref:007A (Originating side) Type:SETUP (0x5) Sending complete: Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64 Kbit/s(16) L1Prot:G.711 A-Law(3) Channel Id:No:1 Type:B-chans(3) Exclusive/Implicit Calling Party Number:XXXXX(l:12) plan:isdn(1) type:national(2)scr:user, passed(1) pres:allowed(0) Called Party Number:XXXXX(l:15) plan:isdn(1) type:unknown(0) <<<----------------- NOT OK [ 08 02 00 7a 05 a1 04 03 80 90 a3 18 03 a9 83 81 6c 0e 21 81 30 32 32 31 35 35 34 30 30 33 31 38 70 10 81 34 39 32 32 31 34 30 30 36 37 38 33 31 30 31 ] How to can the destination_number_ton be set when originating a call via bgapi? Thanks Markus

2 2

SLA/BLF causing 482 loop detected errors on powerpc build
by KEN KELLY-TURNER 16 Oct '12

16 Oct '12

Hi all, I've cross compiled freeswitch 1.2.3 for ppc and I'm trying to get SLA and BLF working on my polycom phones. I tested both features on a Ubuntu build of freeswitch 1.2.3 and they work perfectly. If I use the same configuration and phones on my powerpc version I get the following error: send 1243 bytes to udp/[10.10.8.81]:5060 at 00:28:17.493372: ------------------------------------------------------------------------ NOTIFY sip:1003@10.10.8.81 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.1;rport;branch=z9hG4bKX2crS8D8HQtaB Max-Forwards: 70 From: <sip:1002@10.10.11.1>;tag=0KH1Bj6SVsLd To: "free3" <sip:1003@10.10.11.1>;tag=739B1DFC-3CEB89D3 Call-ID: 4a84c5e6-67b90e1d-2a96ef64(a)10.10.8.81 CSeq: 4052937957 NOTIFY Contact: <sip:1002@10.10.11.1:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.2.3 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: presence Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3572 Content-Type: application/xpidf+xml Content-Length: 430 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd"> <presence> <status> <note>Registered(UDP)</note> </status> <presentity uri="sip:1002@10.10.11.1;method=SUBSCRIBE" /> <atom id="sip:1002@10.10.11.1"> <address uri="sip:mod_sofia@10.10.11.1:5060;user=ip" priority="0.800000"> <status status="open" /> <msnsubstatus substatus="online" /> </address> </atom> </presence> ------------------------------------------------------------------------ recv 395 bytes from udp/[10.10.8.81]:5060 at 00:28:17.513912: ------------------------------------------------------------------------ SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 10.10.11.1;rport;branch=z9hG4bKX2crS8D8HQtaB From: <sip:1002@10.10.11.1>;tag=0KH1Bj6SVsLd To: "free3" <sip:1003@10.10.11.1>;tag=739B1DFC-3CEB89D3 CSeq: 2147483647 NOTIFY Call-ID: 4a84c5e6-67b90e1d-2a96ef64(a)10.10.8.81 Event: presence User-Agent: PolycomSoundPointIP-SPIP_550-UA/4.0.2.11307_0004f2149566 Accept-Language: en Content-Length: 0 ------------------------------------------------------------------------ Anybody else encounter this or have any suggestions on how to resolve? Thanks, Ken

2 1

★ Freeswitch Users, Ian Bonham left a message for you
by Badoo 16 Oct '12

16 Oct '12

Ian Bonham left a message for you Only you can see the sender and content of your message, and you can delete it anytime. You can instantly reply using our message exchange system: http://eu1.badoo.com/bon5/in/RFmzm3kNWOM/?lang_id=3&m=21&mid=507d26d5000000… If the link in this message does not work, try copying and pasting it into your browser. This email is part of our delivery procedure for the message sent by Ian Bonham. If you have received this email by mistake, please ignore it. The message will be deleted soon. Have fun! The Badoo Team You have received this email from Badoo Trading Limited (postal address below). http://eu1.badoo.com/impersonation.phtml?lang_id=3&email=freeswitch-users%4… Badoo Trading Limited is a limited company registered in England and Wales under CRN 7540255 with its registered office at 44a The Green, Warlingham, Surrey, CR6 9NA.

1 0

Can't use dtmf_type=rfc2833 in a dial plan
by peely 15 Oct '12

15 Oct '12

Hi, I recently moved up to FS 1.2 (FreeSWITCH Version 1.2.2+git~20120911T185917Z~3fd1a57902) and since then can't seem to be able to enable RFC2833 termination in the dialplan. I receive 2833 fine from the inbound: 2012-10-15 20:53:38.311883 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 0:1200 2012-10-15 20:53:40.131884 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 1:1200 2012-10-15 20:53:40.731885 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 2:1200 2012-10-15 20:53:41.031885 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 3:1200 But on the outbound I am always receiving DTMF as SIP INFO. I have tried various combinations in the dialplan of adding dtmf_type=rfc2833 in the bridge options, and setting dtmf_type=rfc2833, export dtmf_type=rfc2833, but I always receive DTMF as SIP INFO! The fact that I am receiving SIP INFO with the correct DTMF suggests that FS is identifying the RFC2833 from the inbound, but I just can't switch the DTMF mode to 2833 at will. I don't have any dtmf_type setting on the profile. Any ideas? I have currently had to fall back to pass-rfc2833=true which is OK but I would rather FS generated them so I can play with the minimum durations. Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-use-dtmf-type-rfc2833-i… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 1

SIP gateway - can't locate outgoing channel type
by Richard Weremiuk 15 Oct '12

15 Oct '12

Good afternoon I'm new to freeSWITCH and have setup my freeSWITCH successfully, from the latest pre-compiled version (windows). I have a successfully configured a SIP gateway (provider voipfone), but when I dial into the PSTN number, just to try and route to one of the default build extensions 1000 (I also tried the IVR) I get this error: 2012-10-12 15:41:16.997549 [ERR] switch_core_session.c:408 Could not locate channel type 1000 XML default 2012-10-12 15:41:16.997549 [NOTICE] switch_ivr_originate.c:2599 Cannot create outgoing channel of type [1000 XML default] cause: [CHAN_NOT_IMPLEMENTED] I've taken the code straight out of the book, and just added my SIP username and password, so I can't understand why it won't route through? Rather than past in code here, I have put the output from a clean startup in a word file, and also put the external sip_profile xml, and also the public directory xml file into the same document too. http://www.beyondmarathon.com/routes/freeswitch_output.docx voipfone was not a provider listed in the wiki, but given I have been able to get the gateway registered then I assume it is ok, and there must be a config issue somewhere? Any help would be greatly appreciated. Regards Richard Weremiuk

3 6

Debian Packages
by Andrew Cassidy 15 Oct '12

15 Oct '12

Hi all, I know there are people building nightlies with CI systems, but are there any repos of current stable FreeSWITCH builds? -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F<info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

3 2

Need git help - how to download a specific date version
by Mario G 15 Oct '12

15 Oct '12

I need to download a fresh HEAD using git, but I need to be able to specify the rev level or date (preferably). I am trying to solve a build issue that "goes away" on the second make which is why I need a fresh reload each time. A problem was introduced between and 9/26 and I am trying to narrow down the actual date/time it happened. I was not able to find out how to do this on the web. Thanks Mario G

2 2

FreeSWITCH Weekly News and Notes
by Michael Collins 15 Oct '12

15 Oct '12

Greetings everyone! As you may know the FreeSWITCH team is continuing to update the project's infrastructure. Among other things this includes getting ready for IPv6<http://en.wikipedia.org/wiki/IPv6>. Last week Brian West finished getting several of our servers all set up to handle IPv6 traffic. These includes www.freeswitch.org and conference.freeswitch.org. Thank you to all those who did testing and gave us valuable feedback. On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_10_10>we enjoyed our very own Ken Rice <http://www.cluecon.com/speaker/92/> giving us some great reminders on how to gather data for troubleshooting as well as tips on opening bug reports in Jira <http://jira.freeswitch.org>. We had a number of users comment on how useful it was to see examples of how to do this. The audio is up in the usual location<http://torrents.freeswitch.org/conf_call_2012-10-10.torrent>and we have a community member who is preparing a video which will be posted as soon as it is ready. This week we have Chad Engler from Patlive <http://www.patlive.com> coming to discuss with us his node-esl library. Chad has made the code available here <https://github.com/englercj/node-esl> on Github. He has included an interesting channel monitor example to give you an idea of what can be done by combining node.js with ESL. We look forward to hearing more about it on this week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_10_17> . Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

1 0

enhanced BLF on polycom
by KEN KELLY-TURNER 15 Oct '12

15 Oct '12

Hi all, I'm trying to get enhanced BLF behaviour working on my polycom 550 and 670 with firmware 4.0.2. I've set it up as described in http://wiki.freeswitch.org/wiki/Polycom_Presence_Setup but I'm finding the behaviour to be highly inconsistent. The phone that is listening doesn't detect ringing at all and only successfully picks up the first time the other phone is connected on another call. After that the only way to make it register that the line is busy is to toggle hold repeatedly until the listening phone recognises a change. Any idea what is causing these problems or a possible solution? Thanks, Ken

3 4