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FreeSWITCH vs Yate
by Mike Fedyk 23 Jul '08

23 Jul '08

Hi, I've been using asterisk for a while now and just love running into assorted deadlocks, crashes, quirks and SIP misimplementations (ie, new Call-ID when forking/branching calls). I'm currently looking very closely at Freeswitch and another project that has been around longer, Yate. Yate seems to have a good architecture, and looks like it may be a good alternative to Freeswitch (if there was ever a need for it). Looking from the outside, it makes me wonder why Yate didn't just get a few more developers instead of another project being started instead. The Freeswitch devs may have looked at it before starting a new project and I'm curious as to why it was passed over. Could it be because Yate is written in C++? Personally since even now Freeswitch has a bigger community I'd probably only look at Yate to make sure I have a somewhat generic event model for Vicidial (somewhere on my to-do list). This message is by no means meant to be taken as an attack on Freeswitch. I'm just curious. Mike

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NAT traversal, SIP replies don't go to originating port
by Leon de Rooij 22 Jul '08

22 Jul '08

Hi all, I've set up Freeswitch, but my ATA can't register properly when it's behind NAT. Apparently, SIP replies aren't sent back to the port on the external IP of the NAT gw, where the initial request originated from. The situation is as follows: UA (172.31.0.55) ---- (172.31.0.1) NAT GW (1.1.210.76) ----- (1.1.232.18) FS The UA registers to a profile with these settings: <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="dialplan" value="XML"/> <param name="context" value="cust"/> <param name="dtmf-duration" value="100"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="1.1.232.18"/> <param name="sip-ip" value="1.1.232.18"/> <param name="sip-port" value="5060"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="inbound-late-negotiation" value="true"/> <param name="accept-blind-reg" value="false"/> <param name="disable-transcoding" value="true"/> <param name="manage-presence" value="true"/> <param name="auth-calls" value="true"/> <param name="auth-all-packets" value="false"/> <param name="disable-transfer" value="true"/> <param name="disable-register" value="false"/> <param name="tls" value="false"/> <param name="odbc-dsn" value="freeswitch:freeswitch:freeswitch"/> </settings> First I tried without STUN, and got the following trace: U 2008/07/22 21:03:48.983041 1.1.210.76:57501 -> 1.1.232.18:5060 REGISTER sip:test.nl:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.0.55:5060;branch=z9hG4bK57489228aa596f71. Max-Forwards: 70. To: <sip:ldr-line1@test.nl>. From: <sip:ldr-line1@test.nl>;tag=xETN4EDMxED. Call-ID: 928BD3011481111(a)172.31.0.55. CSeq: 2 REGISTER. Contact: <sip:ldr-line1@172.31.0.55:5060>. Content-Length: 0. Expires: 3600. . U 2008/07/22 21:03:48.983421 1.1.232.18:5060 -> 1.1.210.76:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 172.31.0.55:5060;branch=z9hG4bK57489228aa596f71;received=1.1.210.76. From: <sip:ldr-line1@test.nl>;tag=xETN4EDMxED. To: <sip:ldr-line1@test.nl>;tag=02F7cjtcp4cvS. Call-ID: 928BD3011481111(a)172.31.0.55. CSeq: 2 REGISTER. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9117M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: 100rel, timer, precondition, path, replaces. WWW-Authenticate: Digest realm="test.nl", nonce="d7ddacbe-5820-11dd- a225-93e83ac49152", algorithm=MD5, qop="auth". Content-Length: 0. . So, the reply sent from FS to the NAT gw, is sent to port 5060, while it originated from port 57501. Result is, that the '401 Unauthorized' never arrives at the ATA. Then I tried enabling STUN in the ATA, and got another result: U 2008/07/22 21:09:54.015734 1.1.210.76:61341 -> 1.1.232.18:5060 REGISTER sip:test.nl:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.210.76:59173;branch=z9hG4bK482f2934fda524ff. Max-Forwards: 70. To: <sip:ldr-line1@test.nl>. From: <sip:ldr-line1@test.nl>;tag=xIzM4EDMxID. Call-ID: 799E6DDE35D1111(a)1.1.210.76. CSeq: 1 REGISTER. Contact: <sip:ldr-line1@1.1.210.76:59173>. Content-Length: 0. Expires: 3600. . U 2008/07/22 21:09:54.015995 1.1.232.18:5060 -> 1.1.210.76:59173 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 1.1.210.76:59173;branch=z9hG4bK482f2934fda524ff. From: <sip:ldr-line1@test.nl>;tag=xIzM4EDMxID. To: <sip:ldr-line1@test.nl>;tag=2m2rg8UKgpS1g. Call-ID: 799E6DDE35D1111(a)1.1.210.76. CSeq: 1 REGISTER. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9117M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: 100rel, timer, precondition, path, replaces. WWW-Authenticate: Digest realm="test.nl", nonce="bfbe5042-5821-11dd- a225-93e83ac49152", algorithm=MD5, qop="auth". Content-Length: 0. . Now the reply is sent back to port 59173. That's the same as in the Contact as it's sent by the ATA. Does this mean STUN doesn't function properly ? I am using stun.fwdnet.net:3478 (ATA is a Zyxel P2002). Can I force FS to reply always to the port where the original message originated from ? Or should I fix this differently ? Thanks in advance, Leon de Rooij

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gen_mc: create Linksys certificates for SRTP
by Patrick 22 Jul '08

22 Jul '08

Hi, If you would like to test Linksys SRTP against FreeSWITCH then you'll probably bump into the problem of how to generate the required certificates as that is pretty much not documented. Apparently Linksys (Sipura) have an app called gen_mc (generate mini certificate) but they have not released that. I found a replacement on the Net, also called gen_mc. Haven't tried it yet and I don't know if the code can be trusted but maybe it's useful to others. The site where you can download the replacement gen_mc app: http://www.megajournal.ru/user/booBot/ Direct link to the file: http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c… (after download, remove the mp3, it's just a tar.gz) Regards, Patrick

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ANNOUNCEMENT: Weekly docs conference call, Wednesday UTC 1800/1pm Central
by Michael Collins 22 Jul '08

22 Jul '08

FYI, We've decided to go back to the originally scheduled time of 1pm Central. My apologies to those who cannot attend or who find this time inconvenient. We tried to find that one perfect time of day and day of week and it simply does not exist, so we're going with what works for the majority of those who handle the wiki documentation. The agenda for tomorrow is quite thin at the moment so please add some stuff: http://wiki.freeswitch.org/wiki/Wiki_meet_2008_07_23 Also, even if you can't attend the meeting but there's stuff you want to see documented then please add your question/comment to the page. Thanks, Michael

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Question: PIKA cards in use?
by Michael Collins 22 Jul '08

22 Jul '08

FYI, Anthony needs to know if anyone has purchased a PIKA TDM card for use with FreeSWITCH. If you've bought one and are using it with FS then please let Anthony know about it. Thanks, MC

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(no subject)
by anthony kanlo 22 Jul '08

22 Jul '08

Hello freeswitch users. I have a serious problem of registering my softphones with freeswitch. I have jus installed the application and i was trying out the sample examples and the following error messages. 2008-05-11 23:49:23 [WARNING] sofia_reg.c:892 sofia_reg_parse_auth() can't find user [1005 at 10.3.2.34] 2008-05-11 23:49:23 [DEBUG] sofia_reg.c:482 sofia_reg_handle_register() send challange for [1005 at 10.3.2.34] 2008-05-11 23:50:16 [WARNING] sofia_reg.c:892 sofia_reg_parse_auth() can't find user [stanley at 10.3.2.34] 2008-05-11 23:50:16 [DEBUG] sofia_reg.c:482 sofia_reg_handle_register() send challange for [stanley at 10.3.2.34] 2008-05-11 23:50:55 [DEBUG] sofia_reg.c:482 sofia_reg_handle_register() send challange for [sye at 10.3.2.34] can you send me some sample solution to this problems.

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FreeSWITCH uses old IP address
by Ivan C Myrvold 22 Jul '08

22 Jul '08

I have a NAT problem with my FreeSWITCH which shows up only when my public address have changed AFTER I have started FreeSWITCH. I don't know if this is a small bug in FreeSWITCH, therefore I want to get your opinion before I post a Jira ticket: I have an incoming SIP call from a registered SIP provider This trace is after I have forced my router to get a new public address. My routers old public address is 85.165.194.100 My routers new public address is 85.165.204.52 http://pastebin.freeswitch.org/4927 This trace is after I have restarted FreeSWITCH: My routers public address is still 85.165.204.52 http://pastebin.freeswitch.org/4928 The first trace is normal up to line 69, when FreeSWITCH is sending the old IP address in the Contact field. The same in lines 96, 98, 102 and 111, where the old IP address are used in the Via field, Contact field and Remote-Party-ID etc. FreeSWITCH is using the new IP address in the To field in all messages. Why is FreeSWITCH using both the old and new IP address? Ivan

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Current state of 1.0.1
by jay binks 22 Jul '08

22 Jul '08

I realize 1.0.1 is not a scheduled release, and that there is no set release date.. however Im hanging out for it to release on my production systems ( need some of the bug fixes ) I realize I COULD just use SVN head, but for a matter of a few days I decided to wait.. its now been a few weeks :) ( which is a bit of a pain, but I understand... and appreciate the reasons ) anyways... just wanting to get a feel for where things are at with 1.0.1 and see if I should bite the bullet and just install a SVN copy... ( but I just don't feel right about doing that in a production environment ) Thanks anyways. Jay

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originating a call from JS
by John Wehle 21 Jul '08

21 Jul '08

Still fighting with our System 25 voice mail ports. Turns out that sending DTMF to control the message waiting lights only works when done while receiving a dialtone from the PBX ... you can't just send the tones at the end of a voice mail call. I have some JavaScript which looks like: function mwi_on (ext) { mwi_session = new Session(); mwi_session.setCallerData ("caller_id_name", mwi_caller_name); mwi_session.setCallerData ("caller_id_number", mwi_caller_number); mwi_session.originate (session, mwi_device + "/#90" + ext, 15); mwi_session.execute("sleep", "1000"); mwi_session.hangup ("NORMAL_CLEARING"); } ... session.execute ("voicemail", "default " + domain + " " + to); mwi_on (to); ... which sometimes works fine and other times fails with a trace similar to: [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel OpenZAP/4:4/#90482 [9d2980b4-7557-dd11-8fb6-001fc6ab49e2] [WARNING] zap_zt.c:356 zt_open() Echo training not available for 4:4 [NOTICE] switch_ivr_originate.c:1263 switch_ivr_originate() Hangup OpenZAP/4:4/#90482 [CS_CONSUME_MEDIA] [NO_ANSWER] [WARNING] mod_spidermonkey.c:2824 session_originate() Cannot Create Outgoing Channel! [openzap/4/A/#90482] [ERR] inline:1 mod_spidermonkey() You must call the session.originate method before calling this method! [NOTICE] switch_core_state_machine.c:157 switch_core_standard_on_execute() Hangup OpenZAP/1:1/551 [CS_EXECUTE] [NORMAL_CLEARING] [NOTICE] switch_core_session.c:754 switch_core_session_thread() Session 4 (OpenZAP/1:1/551) Ended [NOTICE] switch_core_session.c:756 switch_core_session_thread() Close Channel OpenZAP/1:1/551 [CS_HANGUP] [NOTICE] switch_core_session.c:754 switch_core_session_thread() Session 5 (OpenZAP/4:4/#90482) Ended ... the result of handling a voice mail from openzap span 1 extension 551 (a System 25 voice mail port) and attempting to original a call using openzap span 4 (a System 25 loopstart line) to extension #90482 which, if successful, causes 482's message waiting light to go on (the DTMF signals from the System 25 indicated to the JavaScript program that the voice mail message was for ext 482). So what's the correct, reliable way to pick up an openzap line, wait for dial tone, and dial some digits / send some DTMF? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Weekly docs call time: survey
by Michael S Collins 21 Jul '08

21 Jul '08

FYI, It is painfully obvious that there is no perfect time to hold this call. I would like to survey those who are serious about regularly attending this call and get your input on when this call should take place. I can attend on Tuesdays, Wednesdays, or Thursdays. My preferred day is Thursday but Wednesday is pretty good too. I am on the west coast so I can't really attend if the call is before 7am my time. I can attend as late as 4pm (start) on thursday or as late as 2pm on Wednesday. Please let me know what works for you and we will pick a time that works for as many people as possible. Thanks, MC Sent from my iPhone

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