November 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ftmod_libpri - RDNIS patch
by lakshmanan ganapathy 15 Nov '10

15 Nov '10

Hi all, Please find the attached patch file which contain a small change to handle RDNIS in ftmod_libpri. Somebody can review it and find ok, do commit it. regards, Lakshmanan G.

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Re: [Freeswitch-users] Radius AAA
by Abid Saleem 15 Nov '10

15 Nov '10

Dear Tihomir, I pointed to these lines before in a previous message. Anyway we had already fixed the issue ourselves. It was due to dictionary not having proper values for the attributes. We found the following problems with the information provided on the Wiki. 1- The default dictionary mentioned as dictionary.all is not present in the mentioned path so we had to change the config to point to exact name of dictionary.2- The default dictionary does not contain these Cisco attributes like "h323-credit-amount", "h323-credit-time" etc. So we had to add these attributes.3- Even after adding the attributes, it does not work because the radiusclient does not recognize the vendor from its vendor id = 9 but it checks another field like "vendor=Cisco". So for this module to work properly, the entries in the dictionary for Cisco should be like this. VENDOR Cisco 9ATTRIBUTE h323-credit-amount 101 string vendor=CiscoATTRIBUTE h323-credit-time 102 string vendor=CiscoATTRIBUTE h323-return-code 103 string vendor=Cisco I did not test the dictionary you have sent but it might be working because it has vendor name as Cxss. So may I update the Wiki? Thank you for all the time I took from you and sorry for inconvenience. Regards--------Abid Saleem Date: Sat, 13 Nov 2010 22:38:31 +0100 From: tculjaga(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Radius AAA On Sat, Nov 13, 2010 at 9:04 AM, Abid Saleem <abid_freeswitch(a)live.com> wrote: Dear Tihomir, I think you did not understand what my problem is. Let me explain. I know that I can evaluate the response from auth_function and act accordingly (bridge or not bridge the call according to auth result). This part is ok and I can do it because I am getting OK or NOK in AUTH_RESULT. The issue is I need to disconnect the call after scheduled number of seconds returned by radius in h323-credit-time attribute whose value in this case is not being populated into credit_time parameter of FS. well, you could have pointed the finger on this before :P 2010-11-04 18:09:53.396212 [DEBUG] mod_rad_auth.c:491 sending radius packet ... 2010-11-04 18:09:54.571999 [DEBUG] mod_rad_auth.c:497 RADIUS Authentication OK2010-11-04 18:09:54.571999 [DEBUG] mod_rad_auth.c:519 Handle attribute: BILING_MODEL2010-11-04 18:09:54.571999 [DEBUG] mod_rad_auth.c:529 No found out attribute id: 109, pec:9, (BILING_MODEL) 2010-11-04 18:09:54.571999 [DEBUG] mod_rad_auth.c:519 Handle attribute: CREDIT_AMOUNT2010-11-04 18:09:54.571999 [DEBUG] mod_rad_auth.c:529 No found out attribute id: 101, pec:9, (CREDIT_AMOUNT) the attribute with id:109 and pec:9 is not found in the returning radius message. This can just mean 2 things: 1. really not present in the returning radius message 2. the module doesn't have an appropriate dictionary referencing pec 9 and id 102. well, here its the dictionary im positive of that... so find attached the one im using. Put this on your server and its gonna work. If you still do not understand, lets put it this way that even if I use the mod_rad_auth.conf.xml and dialplan configuration you sent, still the FS is not able to obtain the value of credit_amount and credit_time although radius response parameters are received by the server. as i said before ... check your dictionary. Also I have checked in mod_rad_auth.c file that all entries related to credit_amount, billing_model etc are commented and credit_time does not even exist in that file. you are missing the point of this module ... the VSAs are configurable via config file. _______________________________________________ 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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Register problems
by Ivan C Myrvold 15 Nov '10

15 Nov '10

I have a LINKSYS SPA942 that refuses to register to my FreeSWITCH server. I have a FreeSWITCH log of the register transaction here: http://pastebin.freeswitch.org/14475 For comparison, I have my iSip phone which have no problem registering to my FreeSWITCH server: http://pastebin.freeswitch.org/14476 I hope anyone can give me a hint of why the Linksys phone is not registering. My FreeSWITCH server is not behind NAT. Ivan

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mod managed installation
by budi wibowo 15 Nov '10

15 Nov '10

hi, i have mono 2.8 installed from source and want to build mod_managed. my server running on centos 5.5, but when i compile the module i got this error g++ -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp In file included from freeswitch_managed.cpp:35: freeswitch_managed.h:43:18: error: glib.h: No such file or directory freeswitch_managed.h:44:26: error: mono/jit/jit.h: No such file or directory freeswitch_managed.h:45:36: error: mono/metadata/assembly.h: No such file or directory freeswitch_managed.h:46:39: error: mono/metadata/environment.h: No such file or directory freeswitch_managed.h:47:39: error: mono/metadata/mono-config.h: No such file or directory freeswitch_managed.h:48:35: error: mono/metadata/threads.h: No such file or directory freeswitch_managed.h:49:41: error: mono/metadata/debug-helpers.h: No such file or directory freeswitch_managed.h:56: error: ISO C++ forbids declaration of âMonoDomainâ with no type freeswitch_managed.h:56: error: expected â;â before â*â token freeswitch_managed.h:57: error: ISO C++ forbids declaration of âMonoAssemblyâ with no type freeswitch_managed.h:57: error: expected â;â before â*â token freeswitch_managed.h:60: error: ISO C++ forbids declaration of âMonoMethodâ with no type freeswitch_managed.h:60: error: expected â;â before â*â token thx budi

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: Google voice incoming call doesn't work
by Jian Ren 14 Nov '10

14 Nov '10

Hi, I posted this issue last week. No one replied then the problem was gone after I recompiled and reinstalled everything. But from yesterday, it happened again. This time I couldn't get it work no matter how I tried. As I said, the outgoing works fine, only incoming doesn't. Now as a workaround I configured GV to forward calls to sipgate, then connect to sipgate as an SIP gateway to take the incoming calls. I am wondering whether anyone else got the same issue or I am the only unlucky one. Thanks! Jian On Thu, Nov 4, 2010 at 9:30 PM, Jian Ren <renjian(a)gmail.com> wrote: > mod_dingaling.c line 909: > > if (!(tech_pvt->rtp_session <http://docs.freeswitch.org/structprivate__object.html#d5829b7e4a473ccebac90…> = switch_rtp_new(tech_pvt->profile <http://docs.freeswitch.org/structprivate__object.html#6c15f3ddeb0b6b952e9a5…>->ip <http://docs.freeswitch.org/structmdl__profile.html#38dfeac1fbf9e8d3784dd7ee…>,00901 tech_pvt->local_port <http://docs.freeswitch.org/structprivate__object.html#63221941aab4357700497…>,00902 tech_pvt->remote_ip <http://docs.freeswitch.org/structprivate__object.html#e1cbf100db6a0a4fd7eea…>,00903 tech_pvt->remote_port <http://docs.freeswitch.org/structprivate__object.html#ca66c963cc0f4abf206ee…>,00904 tech_pvt->codec_num <http://docs.freeswitch.org/structprivate__object.html#d1e23e3cec68cdff6135e…>,00905 tech_pvt->read_codec <http://docs.freeswitch.org/structprivate__object.html#c696c2e7f6f43f4f24e14…>.implementation <http://docs.freeswitch.org/structswitch__codec.html#4e2eb0504df66a664f237ff…>->samples_per_packet <http://docs.freeswitch.org/structswitch__codec__implementation.html#f6c2d7e…>,00906 tech_pvt->read_codec <http://docs.freeswitch.org/structprivate__object.html#c696c2e7f6f43f4f24e14…>.implementation <http://docs.freeswitch.org/structswitch__codec.html#4e2eb0504df66a664f237ff…>->microseconds_per_packet <http://docs.freeswitch.org/structswitch__codec__implementation.html#f2bf615…>,00907 flags, tech_pvt->profile <http://docs.freeswitch.org/structprivate__object.html#6c15f3ddeb0b6b952e9a5…>->timer_name <http://docs.freeswitch.org/structmdl__profile.html#f34f270ada1c3c9c2203f4ca…>, &err, switch_core_session_get_pool(tech_pvt->session <http://docs.freeswitch.org/structprivate__object.html#ad69db0ca1011d01ada61…>)))) {00908 switch_log_printf(SWITCH_CHANNEL_SESSION_LOG <http://docs.freeswitch.org/switch__types_8h.html#bc899743e2a268974063fd82f6…>(tech_pvt->session <http://docs.freeswitch.org/structprivate__object.html#ad69db0ca1011d01ada61…>), SWITCH_LOG_DEBUG <http://docs.freeswitch.org/switch__types_8h.html#b74279dba7ee171b8b9d3decfc…>, "RTP ERROR %s\n", err);00909 switch_channel_hangup <http://docs.freeswitch.org/group__switch__channel.html#g2d4ca1c1e6bfea7ba35…>(channel <http://docs.freeswitch.org/group__switch__channel.html#g991acd084cbac25bf6d…>, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER <http://docs.freeswitch.org/switch__types_8h.html#1606cd34d8539e5da76ebdcb89…>);00910 return 0; > > > Something wrong with the rtp session, what's that? > > Thanks! > > On Thu, Nov 4, 2010 at 8:41 PM, Jian Ren <renjian(a)gmail.com> wrote: > >> Outcall works but the incoming doesn't. It always said >> "DESTINATION_OUT_OF_ORDER" >> >> 2010-11-04 20:32:31.797390 [NOTICE] switch_channel.c:784 New Channel >> dingaling/1001 [2d48674e-e874-11df-ab9a-9787aa669b3a] >> 2010-11-04 20:32:31.797390 [NOTICE] switch_channel.c:782 Rename Channel >> dingaling/1001->DingaLing/new [2d48674e-e874-11df-ab9a-9787aa669b3a] >> 2010-11-04 20:32:31.797390 [NOTICE] mod_dingaling.c:1104 Ring-Ready >> DingaLing/new! >> 2010-11-04 20:32:32.164739 [NOTICE] mod_dingaling.c:909 Hangup >> DingaLing/new [CS_INIT] [DESTINATION_OUT_OF_ORDER] >> 2010-11-04 20:32:32.165726 [NOTICE] switch_core_session.c:1242 Session 1 >> (DingaLing/new) Ended >> 2010-11-04 20:32:32.165726 [NOTICE] switch_core_session.c:1244 Close >> Channel DingaLing/new [CS_DESTROY] >> >> I had skypopen loaded but pointed to another extension, not sure whether >> it's related. Now I commented out mod_skypopen but still got the same >> problem. >> Please help! >> >> My jingling client.xml: >> >> <include> >>  >>  >> <param name="auto-login" value="true"/> >>  >> <param name="sasl" value="plain"/> >>  >> <param name="server" value="talk.google.com"/> >>  >> <param name="tls" value="true"/> >>  >> <param name="use-rtp-timer" value="true"/> >>  >> <param name="exten" value="1001"/> >>  >>  >>  >> <param name="vad" value="both"/> >>  >> <param name="candidate-acl" value="wan.auto"/> >> <param name="local-network-acl" value="localnet.auto"/> >> </profile> >> </include> >> >> Thanks! >> Jian >> > >

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Register and non register in same profile
by dome＠tel.co.th 14 Nov '10

14 Nov '10

Dear All, Now i use internal and external profile for my solution. my billing check network address when user send call to external profile (port 5080). So i think about how to use all in same profile Is posible todo ? BG Dome C.

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Compile error on the latest git
by Frank Park 13 Nov '10

13 Nov '10

I was trying to setup another FS box today and had a compile error on ./configure After looking at the git log, I saw that freetdm was modified today and wanted to know if that affected any. Here's the last few lines of output: <<>> Sangoma SS7 stack checking for sng_isup_init in -lsng_ss7... no checking whether to build ftmod_sangoma_ss7... no ./configure: line 11623: syntax error near unexpected token `libisdn,' ./configure: line 11623: ` PKG_CHECK_MODULES(libisdn,' configure: error: ./configure.gnu failed for libs/freetdm Thanks, Frank ----=======================---- Frank Park Telonium Communications, LLC frank(a)telonium.com http://www.telonium.com Follow Us on Twitter: @GetTelonium <http://www.doubleshotwebdev.com/>404-566-8888 x1001 Office 404-939-4242 Cell ----=======================----

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bypass_media=true & using continue_on_fail
by Matthew Fong 13 Nov '10

13 Nov '10

I am trying to write an application that bypasses media between 2 endpoints. If I set both legs to bypass_media=true then bypass media works GREAT, but if leg B (a SIP provider) returns an error code that is found inside continue_on_fail, then the leg A will immediately hangup instead of continuing on with the dial plan. ignore_early_media is set to true both on leg A (FS) and this program, which is suppose to do switching amongst various voip providers with a fail-over to the next provider if the first provider throws an error inside continue_on_fail I tried programming around this by instead setting bypass_media=false initially and listening to the to the CHANNEL_BRIDGE event and issuing a uuid_media off command but this causes the first half second or so of audio not to be transmitted between the endpoints. Thanks.

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Call a registered user
by Christian Hiller 13 Nov '10

13 Nov '10

How can i call a registered User in Freeswitch, but send any destination number, like i am sending calls to a gateway. It is working in asterisk without any problem, but i wasnt able to bridge or originate a call like this in FS Thanks in advance Chris

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FreeSWITCH Snapshot Information
by Michael Collins 13 Nov '10

13 Nov '10

Hello all, It was recently brought to my attention that the freeswitch-snapshot files were not getting updated properly. It turns out that the cron process was still using the SVN repo! Our bad. I have fixed the snapshot to use the git repo. I manually ran the script and it created new snapshot files which can be downloaded now. I will monitor the cron job this weekend to confirm if it is properly generating new files each night. Note: it can take up to 24 hours from the time the cron job runs to the time you see the files over at files.freeswitch.org so be patient. The new files are: freeswitch-snapshot.tar.bz2 freeswitch-snapshot.tar.bz2.md5 freeswitch-snapshot.tar.bz2.sha1 freeswitch-snapshot.tar.gz freeswitch-snapshot.tar.gz.md5 freeswitch-snapshot.tar.gz.sha1 The files should be dated 2010-Nov-12. Any files earlier than this should be considered unreliable. Feel free to try the snapshot on your system. Remember that the snapshots are pre-bootstrapped so you can go straight to the ./configure && make install process. Email me off list if you have any questions. Thanks, Michael

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