October 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

CELT Codec on Friday's meeting
by Brian West 30 Oct '09

30 Oct '09

Please update to 1.0.5pre4 if you plan on using the CELT codec on the conference tomorrow. I'm rolling the pre4 tarball right now. Thanks, Brian

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Error building rpm's from trunk
by Joseph L. Casale 30 Oct '09

30 Oct '09

I fetched a copy of trunk, tar and gzipped it up and placed it in my SOURCES dir, I edited the bzip extension in the spec file and tried to build rpms but got this error: Checking for unpackaged file(s): /usr/lib/rpm/check-files /var/tmp/freeswitch-1.0.4-1-root-rpm-builder error: Installed (but unpackaged) file(s) found: /opt/freeswitch/mod/ozmod_sangoma_boost.so RPM build errors: Installed (but unpackaged) file(s) found: /opt/freeswitch/mod/ozmod_sangoma_boost.so Anyone know if trunk successfully builds into rpms on CentOSx86 or is this an issue on my end? Thanks! jlc

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Error 488- Incompatible Destination
by Lars Zeb 30 Oct '09

30 Oct '09

I must have been playing with the settings on a SNOM 320 because suddenly it cannot connect to FreeSwitch. I have looked at similar posts and it seems like it must be a codec problem on the SNOM, but I cannot figure it out. Can someone point me in the right direction? http://pastebin.freeswitch.org/10887 Thanks, Lars

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Problem with inbound call answered but no sound
by Lars Zeb 30 Oct '09

30 Oct '09

I have tried to update (make current) twice since 15183. All inbound calls are picked up but the caller hears nothing but a couple of clicks. The most recent version I've tried is 15241. Any ideas on what may be causing this? http://pastebin.freeswitch.org/10843 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux Thanks Lars

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Gateway registration fails: IP used for realm in place of configured realm value
by Robert Wilczyński 29 Oct '09

29 Oct '09

Hi, We have two FS instances production (prod) and gateway (gw) when prod starts it tries to register a gateway: <include> <gateway name="fs_gateway"> <param name="realm" value="24ccdomain"/> <param name="proxy" value="$${local_ip_v4}:15060" /> <param name="username" value="24cc"/> <param name="password" value="24ccpass"/> <param name="expire-seconds" value="60"/> <param name="register" value="true"/> </gateway> </include> and fails: 2009-10-29 10:19:45.449220 [NOTICE] sofia_glue.c:3534 Reload XML [Success] 2009-10-29 10:19:45.450221 [INFO] mod_enum.c:808 ENUM Reloaded 2009-10-29 10:19:45.450221 [NOTICE] sofia_reg.c:2117 Added gateway 'fs_gateway' to profile 'external' 2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias [outbound] for profile [external] 2009-10-29 10:19:45.450221 [NOTICE] sofia.c:3093 Adding Alias [nat] for profile [external] 2009-10-29 10:19:45.451221 [NOTICE] sofia.c:3108 Started Profile external [sofia_reg_external] 2009-10-29 10:19:45.452221 [INFO] switch_time.c:661 Timezone reloaded 530 definitions 2009-10-29 10:19:45.628743 [NOTICE] sofia_reg.c:333 Registering fs_gateway 2009-10-29 10:19:45.996790 [NOTICE] sofia.c:1202 Waiting for worker thread 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:1729 Adding Alias [192.168.1.20] for profile [internal] 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3093 Adding Alias [default] for profile [internal] 2009-10-29 10:19:45.998790 [NOTICE] sofia.c:3108 Started Profile internal [sofia_reg_internal] 2009-10-29 10:19:53.354724 [ERR] sofia_reg.c:1425 fs_gateway Registration Failed with status Forbidden [403]. failure #1 2009-10-29 10:19:53.440735 [WARNING] sofia_reg.c:364 fs_gateway Failed Registration, setting retry to 60 seconds. 2009-10-29 10:20:53.108812 [NOTICE] sofia_reg.c:333 Registering fs_gateway 2009-10-29 10:20:53.116813 [ERR] sofia_reg.c:1425 fs_gateway Registration Failed with status Forbidden [403]. failure #2 2009-10-29 10:20:53.208825 [WARNING] sofia_reg.c:364 fs_gateway Failed Registration, setting retry to 90 seconds. gw is configured like that: <include> <domain name="24ccdomain"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <groups> <group name="default"> <users> <user id="24cc"> <params> <param name="password" value="24ccpass"/> </params> <variables> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="24 CallCenter"/> </variables> </user> </users> </group> </groups> </domain> </include> and outputs: 2009-10-29 10:11:29.165700 [WARNING] sofia_reg.c:1737 Can't find user [24cc(a)192.168.1.20] You must define a domain called '192.168.1.20' in your directory and add a user with the id="24cc" attribute and you must configure your device to use the proper domain in it's authentication credentials. on each registration attempt from prod. This worked perfectly fine with one of previous revisions but started failing on 15257. Using siptrace we were able to narrow down the issue to prod sending an IP address as realm instead of gateway configured realm (as it did in previous revision): Authorization: Digest username="24cc", realm="192.168.1.20", nonce="4d82d624-c884-6340-8f7f-a4575f2583bd", cnonce="Es23bj8MEi29T7UQ80i7sw", algorithm=MD5, uri="sip:192.168.1.20:15060", response="afea161876aad137657bf1f923d18659", qop=auth, nc=00000004 Is there any configuration setting we are missing that would bring back previous behavior? -- Best regards, Robert Wilczynski

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Employment Opportunities
by Anthony Minessale 29 Oct '09

29 Oct '09

We're looking for a few individuals to work with us on the CudaTel Product powered by FreeSWITCH. AJAX Developer: Must have strong skill in HTTP/CGI/Javascript preferably some exp in jquery. We would prefer if are not overly obsessed with any standards but respect them enough to try to follow them in a well-balanced way. Best if you live near Silicon Valley, Ann Arbor Michigan, or Milwaukee WI but not mandatory. Must be self-motivated. OCD a plus. Willing to work late if necessary to get things done right. Working knowledge of Perl, DBI-X Catalyst and TT2 would make your life easier. Knowing a little about FS and Telephony would be nice. Must be able to work diligently from a ssh terminal. Back-end Developer: Good knowledge of 2 or more Perl, DBI-X Catalyst and TT2. Attention to detail, good debugging skills. Best if you live near Silicon Valley, Ann Arbor Michigan, or Milwaukee WI but not mandatory. Must be self-motivated. OCD a plus. Willing to work late if necessary to get things done right. Ability to work well with the guy in the other job description above. Knowing a little about FS and Telephony would be nice. Must be able to work diligently from a ssh terminal. Both positions involve a hands-on "extreme programming" type environment with shared screens and a permanent line of communication to your peers. Send CR to jobs(a)freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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Determining Frame Size
by Matthew Fong 29 Oct '09

29 Oct '09

I noticed that the wait_for_silence's <silence_hits> <listen_hits> use number of frames as base values (rather than seconds or milliseconds). I'd like to wait for 5 seconds of consistent silence, but I'm wondering how I can determine the sample rate (and thus frames per second) for an existing call. Is there a variable I can read? Thanks. --matt

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Re: [Freeswitch-users] FreeSWITCH on FreeBSD
by Mikhail T. 29 Oct '09

29 Oct '09

Richard E Neese написав(ла): >> * What hardware is known to work well on FreeBSD? >> * What else -- other than a computer with a phone card -- will I >> need? >> * Do end-user phones have to be regular analog phones (all >> plugged-in to the computer), or IP-phones, such as made by Avaya >> (connecting via an Ethernet switch)? >> * How is the computer connecting to the world? Do I plug-in the >> incoming analog phone line(s) somewhere, or is it going to go >> through the Internet? >> * If the connection to the world is over the Internet, don't we >> need >> an account with someone else, or is Verizon known to work with >> FreeSWITCH-based PBX-systems directly? >> * Should I not simply buy a used Avaya-system like this: >> >> http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=110450339950&_trksid=p27… >> >> What will I gain and what will I lose (other than personal >> satisfaction, of course) by going that route instead of building >> my own box? Thank you very much, Richard, for you answering some of my questions. A few remain, however: > using freeswitch on bsd is free. nd all you kneed is phones like > polycom soundpoint or aastra or snome or cisco . Sip phones. So, the phones connect -- via CAT6 cables -- to my network switch, and talk to my FreeBSD server running FreeSWITCH. Ok. But how does the server connect to the world? Somewhere between me talking into a Polycom handset in New Jersey and my friend in hearing me in Ukraine, the audio has to hit the telephone network... Where does this happen? Do I plug-in a regular phone line into the same server? If so, what hardware do I need inside it? Or do I use a VOIP-provider -- connecting my server to them via the Internet? And if so, which providers should I consider -- would Verizon themselves qualify? I ask, because the phone-service provided with Verizon's FIOS is already a VOIP of some kind -- there is no copper line any more, it is all done through optic cable... Verizon-provided device connects to that fiber on one end, and gives you a phone-line on the other... So it would be rather silly for my SIP-phones to convert the analog speech into packets, then for the FreeSWITCH server to convert them back into analog audio, and then for Verizon's equipment right in my house to digitize back... Thanks! Yours, -mi

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pass arguments into javascript
by Erwin Davis 29 Oct '09

29 Oct '09

Hi, new to javascript. I tried to pass two arguments into javascript, <action application="javascript" data="test.js $1 $2"/> In test.js, I tried to use argv[1] to retrieve $1 and argv[2] to retrieve $2, however, the javascript test.js complained about argv[] as undefined variables. How to retrieve the passing arguments in a javascript same as the case above. Thanks,

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error in loading spidermonkey
by Erwin Davis 29 Oct '09

29 Oct '09

Hi, I got an error in loading mod_spidermonkey. my fs in ver 1.04 runs on fedora 8 VM. Any clue? thanks, [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**

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