February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_com_g729 - moving licences to a new server
by Massimiliano Ravelli 04 Feb '11

04 Feb '11

I need to move our current pbx to a new hardware. How can I disable the g729 licences on the old server and enable them on the new one ? The current production server has an old installation of mod_com_g729: can I upgrade it to a more recent one without upgrading freeswitch ? Does mod_com_g729 upgrade need to stop running calls ? Thanks in advance Massimiliano

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mod_curl: how to parse xml response?
by james.royer＠gmail.com 04 Feb '11

04 Feb '11

Hi All, (This is my second post - are posts delayed? I never received an acknowledgment message.) I want to use mod_curl to query a web service and then extract information from the XML response. <response> <value1>test</value1> <value2>blah</value2> </response> I then want to use TTS to speak the values to the caller. How can I do this? I tried searches for "freeswitch parse xml" and other variations but all just talk about how FreeSWITCH parses config files. I see that people have created libraries for Lua proper but looking at the doc, there doesn't appear to be any functions to parse XML from within Lua. Thanks, James

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Seems att_xfer doesn't work at all?
by Dmitry Sytchev 04 Feb '11

04 Feb '11

Hi all! In latest git i'm unable to use att_xfer at all. Channels don't get bridged on transferor hangup, no moh to transferee, and transferee is locked in no moh state with CS_RESET after transferor and transfer target is already hangup Does anybody have working example of using att_xfer with bind_meta_app? -- Best regards, Dmitry Sytchev, IT Engineer

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curl: parse XML response?
by james.royer＠gmail.com 04 Feb '11

04 Feb '11

Hi All, I want to use mod_curl to query a web service and then extract information from the XML response. <response> <value1>test</value1> <value2>blah</value2> </response> I then want to use TTS to speak the values to the caller. How can I do this? I tried searches for "freeswitch parse xml" and other variations but all just talk about how FreeSWITCH parses config files. Thanks, James

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Polycom Issues
by DJB International 04 Feb '11

04 Feb '11

Has anyone ever seen Polycom 650 send REGISTER without Authorization even though FS responds with 401? The weird thing was that it only happened to this particular Polycom phone only. Please see below: ------------------------------------------------------------------------ recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:22.100024: ------------------------------------------------------------------------ REGISTER sip:199.87.44.19:5060 SIP/2.0 Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B From: "Mark" <sip:7025143416@199.87.44.19 <sip%3A7025143416(a)199.87.44.19> >;tag=EED068F-AC7EFF94 To: <sip:7025143416@199.87.44.19 <sip%3A7025143416(a)199.87.44.19>> CSeq: 1 REGISTER Call-ID: 37fb35c0-279ee281-fd8694be(a)67.232.144.163 Contact: <sip:7025143416@67.232.144.163:10003>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244 Accept-Language: en Max-Forwards: 70 Expires: 3600 Content-Length: 0 ------------------------------------------------------------------------ 2011-02-01 18:24:28.858417 [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia profile 'internal' for [7025143416(a)199.87.44.19] from ip 67.232.144.163 send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:22.174807: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 67.232.144.163:10003 ;branch=z9hG4bK85aaf1325070992B;rport=10058 From: "Mark" <sip:7025143416@199.87.44.19 <sip%3A7025143416(a)199.87.44.19> >;tag=EED068F-AC7EFF94 To: <sip:7025143416@199.87.44.19 <sip%3A7025143416(a)199.87.44.19> >;tag=5Byyp046mycjS Call-ID: 37fb35c0-279ee281-fd8694be(a)67.232.144.163 CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26 17-24-25 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="199.87.44.19", nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:22.628250: ------------------------------------------------------------------------ REGISTER sip:199.87.44.19:5060 SIP/2.0 Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B From: "Mark" <sip:7025143416@199.87.44.19 <sip%3A7025143416(a)199.87.44.19> >;tag=EED068F-AC7EFF94 To: <sip:7025143416@199.87.44.19 <sip%3A7025143416(a)199.87.44.19>> CSeq: 1 REGISTER Call-ID: 37fb35c0-279ee281-fd8694be(a)67.232.144.163 Contact: <sip:7025143416@67.232.144.163:10003>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244 Accept-Language: en Max-Forwards: 70 Expires: 3600 Content-Length: 0 ------------------------------------------------------------------------ send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:22.628449: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 67.232.144.163:10003 ;branch=z9hG4bK85aaf1325070992B;rport=10058 From: "Mark" <sip:7025143416@199.87.44.19 <sip%3A7025143416(a)199.87.44.19> >;tag=EED068F-AC7EFF94 To: <sip:7025143416@199.87.44.19 <sip%3A7025143416(a)199.87.44.19> >;tag=5Byyp046mycjS Call-ID: 37fb35c0-279ee281-fd8694be(a)67.232.144.163 CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26 17-24-25 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="199.87.44.19", nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 559 bytes from udp/[67.232.144.163]:10058 at 02:24:23.628660: ------------------------------------------------------------------------ REGISTER sip:199.87.44.19:5060 SIP/2.0 Via: SIP/2.0/UDP 67.232.144.163:10003;branch=z9hG4bK85aaf1325070992B From: "Mark" <sip:7025143416@199.87.44.19 <sip%3A7025143416(a)199.87.44.19> >;tag=EED068F-AC7EFF94 To: <sip:7025143416@199.87.44.19 <sip%3A7025143416(a)199.87.44.19>> CSeq: 1 REGISTER Call-ID: 37fb35c0-279ee281-fd8694be(a)67.232.144.163 Contact: <sip:7025143416@67.232.144.163:10003>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.4.0244 Accept-Language: en Max-Forwards: 70 Expires: 3600 Content-Length: 0 ------------------------------------------------------------------------ send 676 bytes to udp/[67.232.144.163]:10058 at 02:24:23.628895: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 67.232.144.163:10003 ;branch=z9hG4bK85aaf1325070992B;rport=10058 From: "Mark" <sip:7025143416@199.87.44.19 <sip%3A7025143416(a)199.87.44.19> >;tag=EED068F-AC7EFF94 To: <sip:7025143416@199.87.44.19 <sip%3A7025143416(a)199.87.44.19> >;tag=5Byyp046mycjS Call-ID: 37fb35c0-279ee281-fd8694be(a)67.232.144.163 CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9ffca05 2011-01-26 17-24-25 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="199.87.44.19", nonce="4b7a72c9-47ad-4d84-b1a1-fe0f68194efd", algorithm=MD5, qop="auth" Content-Length: 0 Thank you.

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Re: [Freeswitch-users] Auto test FS install - now necessary?
by Jerre Cope 04 Feb '11

04 Feb '11

I second Mr. Ayre. Thanks everyone. Especially the Rocket Scientists. On 02/04/2011 11:49 AM, freeswitch-users-request(a)lists.freeswitch.org wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > > Today's Topics: > > 1. Re: Auto test FS install - now necessary? (Steven Ayre) > 2. Re: Auto test FS install - now necessary? (Steven Ayre) > 3. Re: Auto test FS install - now necessary? (Phone) > 4. Re: Auto test FS install - now necessary? (mazilo) > 5. curl: parse XML response? (james.royer(a)gmail.com) > 6. mod_directory multi-tenant (Erick Baum) > > > _______________________________________________ > 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

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context
by Sam 04 Feb '11

04 Feb '11

I have defined a user as : <user id="2099"> <params> <param name="password" value="1234"/> <param name="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> <param name="mwi-account" value="2099@voicemail"/> <param name="vm-password" value="1234"/> <param name="vm-email-all-messages" value="true"/>   </params> <variables> <variable name="user_context" value="inter"/> <variable name="force_transfer_context" value="inter"/> <variable name="accountcode" value="2099"/> <variable name="internal_caller_id_name" value="2099"/> <variable name="internal_caller_id_number" value="2099"/> <variable name="callgroup" value="1"/> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> </variables> and in the dialplan i have <context name="inter"> <extension name="2099"> <condition field="destination_number" expression="^(2099)"> <action application="set" data="call_timeout=30"/> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="{sip_invite_domain=$${domain}}user/${dialed_extension}@ ${domain_name}"/> </condition> </extension> </context> Now extension 2075 is also made on above lines in the same context for user and dialplan, here the call when initiated from 2075 to 2099 searches for context public instead of context inter mod_dialplan_xml.c:331 Processing 2075 <2075>->2099 in context public any reason why ? Regards Sam

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Wrong codec used on outbound leg
by Marcin Wojtowicz 04 Feb '11

04 Feb '11

I have the following setup: voip.ms < (external profile) > freeswitch < (internal profile)> my devices. On my internal profile, the only inbound and outbound codec I allow is iLBC (I tried 20ms and 30ms), and on my external profile the only inbound and outbound codec I allow is PCMU. All my internal settings are compatible with the devices that I'm using, and everything works well when I call from outside in, and freeswitch now correctly transcodes from PCMU to iLBC. However, when I call from my internal devices to outside, the iLBC codec is offered on outbound leg B, not PCMU like I have on my external profile. Looking at wireshark voip call flow, my machine is sending iLBC to voip.msand voip.ms is sending me PCMU RTP.

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Re: [Freeswitch-users] Auto test FS install - now necessary?
by Peter Olsson 04 Feb '11

04 Feb '11

I agree, I always follow the commit log, and usually I get the big idea about what's going on. /Peter ----- Reply message ----- Från: "Steven Ayre" <steveayre(a)gmail.com> Datum: fre, feb 4, 2011 23:10 Rubrik: [Freeswitch-users] Auto test FS install - now necessary? Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Best source of that information is the commit log http://fisheye.freeswitch.org/changelog/freeswitch.git There's also a changelog file that's periodically brought up-to-date. -Steve On 4 February 2011 15:58, Tim St. Pierre <fs-list(a)communicatefreely.net<mailto:fs-list@communicatefreely.net>> wrote: > > Testing yourself is always a good idea before rolling out to a > production system. That'll show any obvious ones up that can be fixed > quickly. Not that this will always catch everything, but is there a "release notes" or some other simple document that outlines what parts of Freeswitch have recently been changed? We ended up building a sort of beta-test network, with a second server and some customers that are very benevolent about this sort of thing, and if I had a simple checklist of things that had been worked on, I can make sure that I test those areas more closely. I don't have any experience working with software development beyond my own little LUA and PHP scripts, so maybe this is already available and I just don't know where to look. Is there a tutorial somewhere on how to be a good user, from a developers perspective? -Tim _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4d4c248132761067889338!

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iLBC to PCMU transcoding broken on windows?
by Marcin Wojtowicz 04 Feb '11

04 Feb '11

I built freeswitch from the latest git on my windows machine, and it looks like transcoding from iLBC to PCMU is broken, and I hear only squeal on the line. I have only PCMU on my external profile and iLBC on my internal profiles, so no codec mismatch is possible, and all my codec settings are compatible with the devices that I'm using. I inspected the traffic flow with wireshark, and there is nothing incorrect there, SIP messages and SDP is ok, and traffic looks like my device <-> iLBC <-> freeswitch <-> PCMU <-> voip.ms. I've searched around and it looks like my issue is identical to the discussion here: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/063891.…. If I run PCMU on both ends, there are no problems and if I call into freeswitch voicemail on iLBC, there are no problems either.

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