December 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

how to Ignore DTMF shorter than Fixed duration ?
by Regis M 07 Dec '12

07 Dec '12

Hi, We have a bugged provider that send us throw RTP wrong DTMF during call. It seems that wrong DTMF are shorter than 1200ms (about 90%) so I want to try to ignore them. For the moment, FS catch them and send it back to the bridged side, boring user. switch_rtp.c:3410 RTP RECV DTMF C:552 Does min-dtmf-duration will make FS ignore them and not RECV them ? I saw the DTMF in "normal" RTP packet, but no digit was pressed in user side. It's commented in my FS switch.conf.xml, how can I see the current value ?  Any other idea ? thanks

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sofia_reg.c
by Archana Venugopan 07 Dec '12

07 Dec '12

Hi, In sofia_reg.c I find variables like 'sip_auth_username','sip_auth_password',etc. Can anyone please tell me from where it will get value for all these variables? Is it from some query in lua script or is it from any sip URL? Regards, Archana

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Polycom IP 335 - 401 Unauthorized .... NAT ISSUE?
by Sean Devoy 07 Dec '12

07 Dec '12

HI All, I am still banging my head against the wall here try to get a Polycom 335 to register w/ FS. I have checked all the SERVER and USER/AUTH fields like 1000 times and 900 variations. I think my problem may be NAT related. I know on my CISCO 504G I had to enable several NAT features to work behind our firewall. I am totally new to Polycom, so some very basic help is needed. The server is remote but not behind a NAT there. The phones are NAT'ed to the internet. In the sofia sip trace I see this over and over: ------------------------------------------------------------------------ recv 552 bytes from udp/[71.127.152.57]:1026 at 16:26:07.358892: ------------------------------------------------------------------------ REGISTER sip:fs_bfis.bizfocused.com SIP/2.0 Via: SIP/2.0/UDP 10.10.40.47:5060;branch=z9hG4bKbf81dbdc8E687A5 From: "228 Sean" <sip:228@fs_bfis.bizfocused.com>;tag=3F42C046-B61A297 To: <sip:228@fs_bfis.bizfocused.com> CSeq: 1 REGISTER Call-ID: 2f482c2-2599cc43-1fb1a78(a)10.10.40.47 Contact: <sip:228@10.10.40.47:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069 Accept-Language: en Max-Forwards: 70 Expires: 600 Content-Length: 0 ------------------------------------------------------------------------ send 710 bytes to udp/[71.127.152.57]:5060 at 16:26:07.359067: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.40.47:5060;branch=z9hG4bKbf81dbdc8E687A5;received=71.127.152.57 From: "228 Sean" <sip:228@fs_bfis.bizfocused.com>;tag=3F42C046-B61A297 To: <sip:228@fs_bfis.bizfocused.com>;tag=t232me1NSD02S Call-ID: 2f482c2-2599cc43-1fb1a78(a)10.10.40.47 CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120 712T101002Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: precondition, path, replaces WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="b9583359-0163-4bf2-9818-788f64c34207", algorithm=MD5, qop="auth" Content-Length: 0 If I understand correctly, the server should be sending back this 401 message with the nonce so the phone can re-attempt the registration with an encrypted password. If NAT is failing, the phone is never seeing the 401 w/ the nonce. So what do I do in the WEB config interface to enable NAT on this phone? Thanks, Sean

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T.30 <---> T.38 howto ?!
by Tamer Higazi 07 Dec '12

07 Dec '12

Hi people! I am using hylafax with mod_spands and I have problems to setup the dialplan. Here is the case: Hylafax (T.30) <--> FS (T.38) <--> VoIP Provider (T.38) How is the diaplan successfully to setup to successfully make a T.30 <--> T.38 sending and receiving?! I am not getting smart, for any help I would kindly thank you all. Tamer

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FreeSWITCH Conference Call Subject: Windows debugging tools
by Michael Collins 07 Dec '12

07 Dec '12

Hey all! I'm trying to gauge the interest level in this subject. We have an experienced Windows user who is willing to share with us a lot of his hard-earned knowledge when it comes to debugging crashes and such in a Windows environment. If you are interested in hearing about this subject please respond. (Only respond if you are interested - we don't need to hear from those who are not interested.) Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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FreeSWITCH Community Conference Call Today
by Michael Collins 07 Dec '12

07 Dec '12

Hello folks, We'll be having our conference call in just about one hour. The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_05 We'll be discussing some interesting facts about the FreeSWITCH XML pre-processor and also an object lesson in how to use the source code to answer a question and then put it on the wiki. Oh, and we have an update on the ClueCon videos! Talk to you soon. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Domains and profiles
by Lawrence Conroy 06 Dec '12

06 Dec '12

Hi Folks, I've started a new thread as it's not quite the same issue, and domains & profiles have confused the heck out of me every time I have developed a new setup for fS. I have sometimes had to hack/hard-doce the dialstring to make multiple domains in one profile work, had hours of fun with presence, db and force register settings, and have still had some odd gotchas that have required extensive meditation. [... and yes, I have read the 1.0.6 bridge book; I'm trying to abstract these elements ] Coming at this from standards/specs and rolling my own SIP stacks, sofia/fS seems to use the term "domain" differently from sipdomain, and alias seems to be tied to the directory (and thus to the profile listed in a directory file), but I'm not sure. so ... Before I capture to the sofia conf xml wiki page, I have a couple of questions on the sip-profile XML setup; Q: Is there a particular reason why there's a parameter called alias and an (entirely different) setting also called alias? The sofia conf xml wiki's comment on the setting "alias" shows I'm not alone. I agree that's what it appears to be doing, but can we nail this down please (and what happens if an external client uses this connection to register and call)? In the current sofia conf xml wiki page, the domain setting is not exactly well documented :). The current internal.xml vanilla example from git (as of time of writing) has the following lines: ------------------------- ...   <aliases>  </aliases>  <gateways> <X-PRE-PROCESS cmd="include" data="internal/*.xml"/> </gateways> <domains>     <domain name="all" alias="true" parse="false"/> </domains> ... ------------------------- This stuff is entirely missing from the sofia.conf.xml wiki page, and it IS really important. Q: what's the default value for the alias parameter in the domain element? -- it is missing from the first example. Q: if there is more than one profile, what's the impact of setting parse = true in one (or all) of the profiles' XML files? (or parse = false, or missing the parameter altogether)? AFAICT, the gateways get pulled in via the pre-process directive just fine, regardless of the value of the parse parameter -- it works for me, at least. Q: if there is more than one profile, what's the impact of putting domain name="all" into one (or all) of the profiles' XML files? Ideally, having more than one sipdomain tied to one profile "would be good", but aliases doesn't do that -- as the git file says, these are aliases for the profile name. Before I start scribbling, Answers on a postcard to this ML, please. all the best, Lawrence

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VIEW specific events for specific device (ip address)
by Sean Devoy 06 Dec '12

06 Dec '12

Hi, I am trying to debug a sip registration problem. To get the events I am currently using an SSH log feature with "sofia global siptrace on". Unfortunately, this server has many many many sip events and this produce a huge log very quickly. Is there another way to view/log SIP events and filter by device IP ADDRESS or maybe by extension? Thanks, Sean

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Using bypass_media or proxy_media selectively
by Emrah 06 Dec '12

06 Dec '12

Hi there, Can you please confirm that I need to either use proxy_media or bypass_media for a video call in H.264? If so, how do I enable proxy_media or bypass_media conditionally by examining the codec prefs of both SIP endpoints? For a setup where I have 2 devices registered as 1000@mydomain, one being audio only and the other video capable. Plus a video capable device registered as 1001@mydomain. How do I selectively use bypass_media or proxy_media for calls made by 1001 to 1000 and answered by the video capable terminal? Thanks a million, Emrah

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Missed call notification
by 王理 06 Dec '12

06 Dec '12

I heard that Freeswitch support missed call notification (i/o VM notification). But look around Wiki no document about this feature. Anyone knows where to setup it up? /brgds, Alex

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