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Registrations stop working for all NAT phones on FreeSWITCH Version 1.0.trunk (17177)
by Gavin Henry 06 Apr '10

06 Apr '10

Hi, We've just upgraded (switched back to building from svn in case this was a git migration issue) and all our phones behind various networks on NAT won't registered anymore, whereas on version dated 2010-02-22 20:06 (don't have the revision) all these phones were fine and handled the the FS NAT features. No NDLB features on enabled or rport. They weren't on the 2010-02-22 20:06 version either. freeswitch@internal> recv 524 bytes from udp/[external_ip]:43078 at 09:45:02.413013: ------------------------------------------------------------------------ REGISTER sip:pbx1.xxxx.co.uk SIP/2.0 Via: SIP/2.0/UDP internal_ip:5062;branch=z9hG4bK892744790 From: "Gavin Henry" <sip:609@pbx1.xxxx.co.uk>;tag=706763496 To: "Gavin Henry" <sip:609@pbx1.xxxx.co.uk> Call-ID: 1553007081@internal_ip CSeq: 24 REGISTER Contact: <sip:609@internal_ip:5062> Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T28P 2.43.0.50 Expires: 3600 Content-Length: 0 ------------------------------------------------------------------------ send 641 bytes to udp/[external_ip]:5062 at 09:45:02.413206: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP internal_ip:5062;branch=z9hG4bK892744790;received=external_ip From: "Gavin Henry" <sip:609@pbx1.xxxx.co.uk>;tag=706763496 To: "Gavin Henry" <sip:609@pbx1.xxxx.co.uk>;tag=gtDayy31FDN9K Call-ID: 1553007081@internal_ip CSeq: 24 REGISTER User-Agent: Our test PBX 1.0 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="pbx1.xxxx.co.uk", nonce="cf9aaa61-89bf-4d9d-adbb-51a068b23440", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ What's changed? Thanks, Gavin. -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com

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mod_limit
by Todd 06 Apr '10

06 Apr '10

Hey list, I have a question: Can mod_limit be used to limit the number of incoming call from a specific caller ID in a certain time interval? Limit the calls per second by source ip + destination number: <action application="limit_hash" data="${sip_received_ip} ${destination_number} ${calls_per_second}/1" /> I found this in the wiki and I was wondering how you would change the action to limit an incoming caller id to 5 calls per minute. Still pretty noo, so any help in the right direction would be great. thx

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What is the 99999_enum.xml file?
by Jim 06 Apr '10

06 Apr '10

It seems to be preventing FS from parsing any files beyond it. FS parses conf/dialplan/default.xml then each of the files in conf/dialplan/default/, but never sees any of the other files in conf/dialplan such as public.xml and the files in public/. Thanks, Jim.

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mod_xml_cdr DNS
by Justin B Newman 06 Apr '10

06 Apr '10

I have mod_xml_cdr set with a url to POST entries to. The DNS record is something along the lines of: cdr.mydomain.com. 600 IN CNAME mydomain.com. mydomain.com. 600 IN A 10.10.10.1 mydomain.com. 600 IN A 10.10.10.2 Only one of the two web servers is receiving any POSTs. I tried adding "curl_easy_setopt(curl_handle, CURLOPT_DNS_CACHE_TIMEOUT, 0);" to mod_xml_cdr.c, to no avail. Any ideas? -jbn

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cepstral PB
by Jean-Yves F. Barbier 06 Apr '10

06 Apr '10

FS svn-17188M Debian Lenny ================= Hi list, I've installed cepstral voices (in /opt/swift, and also changed its rights recursively to freeswitch:freeswitch) but I had a complain at FS start that some cepstral libraries were missing. Eventually, I was obliged to simlink /opt/swift/lib/lib* into /usr/local/freeswitch/lib to have it working correctly, is it normal, or did I miss something? (I followed the wiki, so all libs were added to ld.so.cache, path in mod_cepstral source Makefile are good, SWIFT_HOME is defined, module flite loading is commented, etc- so I said to myself: Nantidiou!) -- Sorry never means having your say to love.

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Stuck at revision 17188
by afshin afzali 06 Apr '10

06 Apr '10

Hi, After some issues about svn update in openzap and compile, my FreeSWITCH version stuck at revision 17188 ! I'm in trouble? -- afshin

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How to configure g729 pass though on Freeswitch
by Frank Church 06 Apr '10

06 Apr '10

I am just trialling Freeswitch with Linksys adapters, whose default codec I have set to G729 with 'Use Pref Codec Only:' set to no. When I change that setting to 'yes' the calls don't go through. I am using the latest Windows SVN. What configuration changes do I need to allow freeswitch-codec-passthru-g729. -- Frank Church ======================= http://devblog.brahmancreations.com

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Re: [Freeswitch-users] Some question about mod_fifo ??
by Nguyễn Mạnh Hùng 06 Apr '10

06 Apr '10

Hi Anthony. What is version of FS has this feature of mod_fifo. I have update the latest version in svn: FreeSWITCH Version 1.0.trunk (16972M). I capture packets in an agent of mod_fifo with Wireshark, but i can't see any SIP UPDATE or SIP INFO packet. Best Regards. Anthony Minessale [anthony.minessale(a)gmail.com] its done by SIP UPDATE on polycom/aastra or sip INFO packets on snom when the call is bridged. X-lite does not update anything when it receives them. That's about it. Hi Anthony, Can you discuss some details in how polycom or snom can do this and x-lite not. If can, I want to edit some open source soffphone like officeSIP to do this. Best Regards. Anthony Minessale [ anthony.minessale(a)gmail.com ] We already do it. X-Lite does not support it. If you try it with a phone like snom or polycom you will see it works just like that. Hi Seven Du. Thanks to yours suggetion. I have an ideal, it is: when the call between caller and agent is set, the caller_id is determined. So, i want to edit code to sent the agent information (the call_id and call_id_number) which will be displayed againt in the agent's softphone (as Xlite..) when the call is happening. I read some documents but i still can't determine: It's maybe yes or maybe to do this and where to do this. Can you give me some comments. Best Regard. Seven Du [ dujinfang(a)gmail.com ] ��As discussed in the list, it's not a freeswitch problem but a reality of life. Think about customer A and B calls in one after another, then if FreeSWITCH call agent X with caller id A and Y with caller id B, and angent Y answers before X, then 1) if bridge Y with A with the FIFO rule, then the caller id is wrong 2) if bridge Y with B, the caller id is right but it breaks the rule of FIFO - A should be served before B!! And what even worse is that if X never answer A then A never can be served which is really unfair!! Of course you don't want 1), and you don't need mod_fifo if you want behavior 2), you just need some dialplan trick or some simple Lua script I think. Also FreeSWITCH is designed to be easily extended with almost any languages so feel free to implement anything. > Hi Mike and Seven Du. > Thanks to yours help. > I known the mechanism of mod_fifo. >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175… . > What a pity, It can't solve this problem. I can't use freeswitch for my call > center. > Hope new version can solve this !!!

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Re: [Freeswitch-users] Freeswitch for Windows - command history too short?
by Jeff Lenk 06 Apr '10

06 Apr '10

FS uses its own history logic for its console which are fixed length at this time. -----Original Message----- From: Jan Berger Sent: 4/4/2010 7:43:27 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch for Windows - command history too short? hi, This is not FreeSWITCH, but sounds like your window console settings? Once you open a FreeSWITCH console left click on the system icon 8topmost left icon in window) and select options/preferences at the bottom of the pop-up menu - there you can alter the buffer length and window appearance. Jan > Date: Sun, 4 Apr 2010 14:16:34 +0100 > From: vfclists(a)googlemail.com > To: freeswitch-users(a)lists.freeswitch.org > Subject: [Freeswitch-users] Freeswitch for Windows - command history too short? > > I am using svn 17048 for Windows and the command history is too short > and doesn't persist between restarts. > > Are there some configuration settings to fix that? > > -- > Frank Church > > ======================= > http://devblog.brahmancreations.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ________________________________ Hotmail: Trusted email with powerful SPAM protection. Sign up now.<https://signup.live.com/signup.aspx?id=60969>

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how do I get my DID number from lua script with session:getvariable
by Uzo Uzo 06 Apr '10

06 Apr '10

I probably am doing this wrong, but here is what I have. I have a bunch of DID's. I want my lua script to intercept all calls then handle them based on which DID was called. What I have done is create my lua script foo.lua then in the dialplan/default.xml I have <extension name="mylua"> <condition field="destination_number" "expression="^1002$"> <action application="lua" data="/usr/local/freeswitch/scripts/foo.lua"/> </condition> </extension> then i forward all my DIDs to extension 1002@myfreeswitchbox when I get a call to the DID the lua script intercepts it, is this the best way? from the lua script, I can do session:getVariable("caller_id_number") but that gives me the caller id of the incoming number, which is cool. which variable can i examine to get the DID number? thanks.

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