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OT: Snom phones reporting incorrect port in Contact/Via
by Jonas Gauffin 15 Apr '10

15 Apr '10

Hello, I got a problem with Snom phones where they report a different port than the one provisioned. I tried to talk with the snom support without success. REGISTER sip:phone.gateon.se:5070 SIP/2.0 Via: SIP/2.0/UDP 217.31.xxx.xxx:26062;branch=z9hG4bK-u1y0v7is9ix0;rport From: "Helene XXX" <sip:u1000028@phone.xxxx.se:5070>;tag=kjaa26j1qv To: "Helene xxx" <sip:u1000028@phone.xxxx.se:5070> Call-ID: 3c267019ee2d-ni5vq3v88za6 CSeq: 133 REGISTER Max-Forwards: 70 Contact: <sip:u1000028@217.31.xxx.xxx :26062;line=yhkplirb>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:1f2c528a-18f1-435d-ad61-2be3d1a88425>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom360/7.3.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.3.109 Expires: 3600 Content-Length: 0 The port that I provisioned (and I've doublechecked in the webinterface) is 5071 and not 26062. Have anyone else got this problem and solved it? Regards, Jonas

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mod_managed
by babak yakhchali 15 Apr '10

15 Apr '10

Hi can someone please explain, step by step, how a I can develop a module in .net? Thanks

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Multi-tenanting
by Kenneth Noisewater 15 Apr '10

15 Apr '10

Hi All, I'm just a few days into my FreeSwitch investigation, and so far I have to say, it seems almost too good to be true! Kudos to everyone involved. I've got quite a bit of experience with other systems, ranging from Asterisk to Mitel/Shoretel type systems, and FreeSwitch is really looking good. Anyway... So pouring over configs, it seems it would be really simple to set up a multi tenant system with the whole 'domain' concept. Does this domain model maintain good seperation throughout the system? Is there anything to be aware of in doing a multi tenant setup? Thanks, Kenny

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Problem with SIP INFO (caller display update) after connect
by Helmut Kuper 15 Apr '10

15 Apr '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have the following problem. A call comes in the following way: ISDN PRI->Sangoma->wanpipe->openzap->mod_openzap->freeswitch->B-Party I use a national numbering plan, so caller's number e.g 044112345 is transmitted as 4412345 on isdn level. My dialplan adds two zeros to the effective_caller_id_name and -number. So it is now 0044112345. The B-Party displays this correctly. But as soon as B-Party picks up the caller's name switched back to 44112345 which is quite bad. I tried to set caller's name and number via effective_caller_id_... , via origination_caller_id_... and via origination_callee_id_... but nothing worked. It looks like the sip info process after connecting two sessions is not effected by those channel variables mentioned above. I use FS tunk (17073M) Any ideas what is wrong here? kind regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFLxdcm4tZeNddg3dwRApgfAJ4mURR45QnHFqSfSP5moSpUdmo6QACeJ4zD xkuSy2yxE8UienOBJ+qgAKQ= =TqJ8 -----END PGP SIGNATURE-----

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Most appropriate mapping between Asterisk CDR and XML CDR
by Frank Church 15 Apr '10

15 Apr '10

I am trying to work out the most appropriate mapping between the Asterisk CDR and Freeswitch xml_cdr as an app of mine is based on the Asterisk CDR. The xml cdr is quite large and some of the values are repeated in the different sections. This is the nearest I have come to, I have omitted the full paths to the values, but would like to use the right paths if they are the most meaningful. The example displays t a call from extension 1001 to an external number 08451234567 calldate <start_stamp>2010-04-14%2023%3A08%3A10</start_stamp> or <start_epoch>1271282890</start_epoch> or <created_time>1271282890796875</created_time> clid <caller_id_name>Booth1</caller_id_name> + <caller_id_number>1001</caller_id_number> src <ani>1001</ani> or <username>1001</username> dst <destination_number>08451234567</destination_number> dcontext <context>default</context> channel <bridge_channel>sofia/internal/1001%40192.168.1.133</bridge_channel> or<chan_name>sofia/internal/1001(a)192.168.1.133</chan_name> dstchannel <chan_name>sofia/external/08451234567</chan_name> lastapp <application app_name="bridge" app_data="sofia/gateway/ provider.co.uk/08451234567"></application> lastdata <chan_name>sofia/external/08451234567</chan_name> duration <duration>46</duration> billsec <billsec>18</billsec> disposition <endpoint_disposition>ANSWER</endpoint_disposition>? amaflags bigint, accountcode - <accountcode>1001</accountcode> uniqueid character uuid=ce234b00-157a-456f-97be-62d15a53b0c9 Is there some guide to which values are more appropriate? I checked the templates for cdr_csv but the look rather simple to me, being as ever a sucker for complexity. -- Frank Church ======================= http://devblog.brahmancreations.com

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SVN read only mirror ???
by Phillip Jones 14 Apr '10

14 Apr '10

Hi there, I read previously on this list that the devs would try and maintain a read only mirror of FreeSWITCH on SVN. Was this possible? When I use the "Latest Build Version:" link from the front page I get: "moose penis NOT IT IS NOT HACKED... ITS CALLED A JOKE and some advice to use FreeSWITCH 1.0.6" Nice! So the question is, is the SVN mirror being maintained or are we using GIT exclusively now? Pj

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origination_caller_id_name doesn't seem to be working
by Mark Sobkow 14 Apr '10

14 Apr '10

OriginateCmd is "{call_timeout=15,originate_timeout=15,origination_caller_id_name='Mark Test',origination_caller_id_number=3065551212}sofia/external/533(a)rats.marketel '&erlang( pbx_called_cust pursuit@testsrv )'" However, I'm not getting the origination caller id name displayed on the soft phone, even after answering the call. Could this be a problem with the Asterisk gateway, or is there a problem with the implementation in Freeswitch? Unfortunately I have to route all my calls through the Asterisk gateway, so I can't tell whether it's a problem with Freeswitch or not. -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow(a)marketelsystems.com Web: http://www.marketelsystems.com

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Freeswitch memory issue/question
by Henry Huang 14 Apr '10

14 Apr '10

Dear developers: Our product is close to sipping stage and we are doing stress tests on FS servers. We have now hit a dead end with our test results and running out of ideas. And the result is pointing to memory issues. We have ran our test on both FS 1.0.4 and 1.0.6 trunk with SIPP to do simply invite tests. Here is the test senario: Send 10,000,000 invites to our test server running FS 1.0.6 with fixed amount of invites per second. The xml dialplan simply bridge the call to an unknown/unregistered user like so (user/500001) and therefore FS would hang up once it doesn't find the user. When running the test, FS is steady, but the memory usage slowly piles up over time. The memory keeps adding up till it hit the ceiling, then FS crash. I don't understand what could be holding the memory and not releasing it since we are not doing anything special and the calls are not being answered. And I forgot to mention, through out the whole test, the CPU usage stays at about 35% . Please help Besides FS native test, we have created our own application module to run the same test. The app is written in C and basically what we did is to point the xml dialplan to the custom app. Inside our app , we are simply doing the same kind of test as the xml dialplan would do. The only thing we did for the stress test is the following code: switch_core_session_execute_application(session, "bridge", "user/500001"); Like the xml dialplan test, we are bridging the call to a unregistered user, so the call would hang up right away if system doesn't find the user. But the bizarre thing is this - the memory usage is about 7 ~ 8 times as much as we are running only xml dialplan. This I don't understand. Our app is calling the native FS C function to bridge the call , why would it use more memory than the xml dialplan... your input would be very much appreciated. thanks -- Henry Huang aka bbhenry VoIP & Open Source software Consultant

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getting spammed from the list
by Tristan Mahé 14 Apr '10

14 Apr '10

Hi everyone, just a message because it's been 4 days now that I receive retransmission of mails sent to the ML ( not every mail, just a few, but I got them hundred of times ) Exemple of these mails: - [Freeswitch-users] phocketsphinx from lua sent by janvb(a)live.com on 10.04.2010 at 01:51 - Re: [Freeswitch-users] need help on variable and param sent by msc(a)freeswitch.org on 09.04.2010 at 19:18 - Re: [Freeswitch-users] Problem with IVR app starting too soon sent by bcxml(a)hotmail.com on 09.04.2010 at 19:45 Am I the only one impacted ? Regards, Gled.

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Problem with IVR app starting too soon
by bcxml 14 Apr '10

14 Apr '10

I have Freeswitch setup to pass incomming calls off to Microsoft Speech Server 2007. Everything works fine except for one thing. When the user calls, they hear ringing, the call is answered, then they hear the Speech Server app welcome message, which may or may not be complete. For example, the application is suppoose to say "Welcome to the IVR application" at the beginning. The caller may hear the entire prompt or they may only hear "to the IVR Application" or perhaps "IVR Application" Can anyone point me in the right direction ? Thanks Brian Campbell -- View this message in context: http://old.nabble.com/Problem-with-IVR-app-starting-too-soon-tp28146712p281… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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