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call forward question
by Tihomir Culjaga 15 Nov '10

15 Nov '10

hello, its just a stupid question ... but how do we handle network call forward in FS ? usually you we can activate call forward on the terminal itself and when a call hits the terminal it responds with a 302 message. This works but not with an unplugged terminal. So, is there any way to do a call network forward? Meaning if extension 1000 activates a CFU (by dialing some FAC - e.g. *21*<number>) to 1002 and we have an incoming call to 1000, FS should place the call to 1002 without even going to look at extension 1000. any chance without a mambo-jumbo hacking? T.

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problem registering with sip providers
by dnb＠bluewin.ch 15 Nov '10

15 Nov '10

hello, i'm running FreeSWITCH Version 1.0.head (svn-18355) on vmware behind nat. i have problems registering with sip providers: 2010-11-12 13:20:28.845507 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:20:34.866728 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:20:38.898868 [WARNING] sofia_reg.c:387 iptel Failed Registration, setting retry to 30 seconds. 2010-11-12 13:20:40.902922 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:20:46.953148 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:20:52.974365 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:20:58.998596 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:21:04.025773 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:21:09.049973 [NOTICE] sofia_reg.c:342 Registering iptel 2010-11-12 13:21:10.051987 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:21:16.081210 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:21:22.106424 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:21:28.150642 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:21:34.216910 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:21:40.281096 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:21:40.281096 [WARNING] sofia_reg.c:387 iptel Failed Registration, setting retry to 30 seconds. 2010-11-12 13:21:46.297302 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:21:52.318592 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:21:58.379775 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:22:04.411977 [NOTICE] sofia_reg.c:342 Registering sipcall 2010-11-12 13:22:10.426183 [WARNING] sofia_reg.c:387 sipcall Failed Registration, setting retry to 5 seconds. 2010-11-12 13:22:11.430227 [NOTICE] sofia_reg.c:342 Registering iptel 2010-11-12 13:22:16.441404 [NOTICE] sofia_reg.c:342 Registering sipcall after a while sipcall still gets registered, im able to receive inbound calls, but i cant get outbound working. iptel doesn't get registered at all, for that i followed the instruction in the FreeSWITCH-106- Book i bought. can someone please give me a hint. thanks

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Bad sound with crackle
by Loïc 15 Nov '10

15 Nov '10

Hello, I am a new user of FreeSwitch and I try to switch from Asterisk to FreeSwitch. For now, I just registered a SIP line and I made an extension in the dialplan to playback music on hold when I call : <context name="default"> <extension name="test"> <condition>  <action application="answer"/> <action application="playback" data="$${hold_music}"/> </condition> </extension> </context> When I call I have a problem with the sound, it is very bad, I hear very fast crackle. On the console I can see a warning during my call and that's all: 2010-11-12 13:02:45.119517 [NOTICE] switch_channel.c:784 New Channel sofia/external/anonymous(a)anonymous.invalid [c27034e2-ee54-11df-ad91-7fc19b304698] 2010-11-12 13:02:45.122548 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <anonymous>->test in context default 2010-11-12 13:02:45.125555 [NOTICE] mod_dptools.c:920 Channel [sofia/external/anonymous(a)anonymous.invalid] has been answered 2010-11-12 13:02:45.508790 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20 If I add <X-PRE-PROCESS cmd="set" data="timer_name=soft"/> in vars.xml, when I call the sound is good but I still get the warning : 2010-11-12 13:05:42.789161 [NOTICE] switch_channel.c:784 New Channel sofia/external/anonymous(a)anonymous.invalid [2c565760-ee55-11df-a0d9-0f91e325cd76] 2010-11-12 13:05:42.791179 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <anonymous>->test in context default 2010-11-12 13:05:42.794179 [NOTICE] mod_dptools.c:920 Channel [sofia/external/anonymous(a)anonymous.invalid] has been answered 2010-11-12 13:05:43.174443 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20 2010-11-12 13:05:43.175461 [INFO] sofia.c:709 sofia/external/anonymous(a)anonymous.invalid Update Callee ID to "anonymous" <anonymous> Is this normal? Why should I add this line in vars.xml for it to work ? What does this warning? How to solve it? Thank you in advance for your help. Loïc

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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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