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Originate command
by Jon Bruel 11 Oct '08

11 Oct '08

I have tried to use the originate command from the API, from javascript apiExecute and from javascript using newsession.originate. In all cases I get the same behaviour: when the dialstring includes several destinations such as: user/100008888205@10.3.1.24,user/100008888206@10.3.1.24 it's only the first user dialled. Actually this user is dialled twice. If I add a third number, the first user is dialled three times. Is this as per design, or should I report a bug. And if so, which project should I use? /Jon

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Accepting SIP Calls from unregistered devices
by Klaus Teller 10 Oct '08

10 Oct '08

Hi, How do i configure my Freeswitch to accept SIP calls from peers/devices not registered with it? Thanks, Klaus. -- GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen! Jetzt dabei sein: http://www.shortview.de/wasistshortview.php?mc=sv_ext_mf@gmx

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Ringback with originate?
by Noah Silverman 10 Oct '08

10 Oct '08

Hi, I'm writing a script to connect one of my SIP phones with an external number via a web interface. (Like "click to call".) Right now, I'm using xml_rpc and after some help from the guys on IRC, I've got it working nicely. When I execute the script, first my local SIP phone rings. I answer it and hear silence until the person on the other end of the call answers their phone. It would be nice if I could simulate some kind of ringing sound so that It sounds like I'm hearing the other phone ring like a normally placed call. I THOUGHT that the "ringback" variable would do this, but no matter what I try, I don't get any sound. Here is what I'm sending now... <value><string>{$${us-ring},accountcode=1}sofia/internal/ 3235551212%111.111.111.111 &bridge(accountcode=1}sofia/internal/13237773456(a)outbound1.vitelity.net ) XML internal 3235551212 3235551212</string></value> Interestingly, (or not), I can control which phone works by the order I use in the originate command. If I set MY phone first and the outside number in the bridge, then I ring first. If I set the outside number first and my phone in the bridge, then they ring first. Any suggestions? -Noah

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Problem with first use.
by gary 10 Oct '08

10 Oct '08

Hello, I am new to freeswitch. Just installed freeswitch-1.0.1 with default configurations on CentOS 4.6 with OpenVZ. I registered two Cisco7960 phones using static IP with default users(1001.xml, 1007.xml). The registration are OK, but when I try to call from each other, it immedially send to voicemail. The following is the log info from console: (I replaced the real IP with 10.1.1.117 in the log) 2008-10-10 11:41:02 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/1001@10.1.1.117:51029;transport=udp;fs_nat=yes [bfe75310-638d-48df-8f3d-262143c15b22] 2008-10-10 11:41:02 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/internal/1001@10.1.1.117:51029;transport=udp;fs_nat=yes [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2008-10-10 11:41:02 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [NORMAL_TEMPORARY_FAILURE] 2008-10-10 11:41:02 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2008-10-10 11:41:02 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 6 (sofia/internal/1001@10.1.1.117:51029;transport=udp;fs_nat=yes) Ended 2008-10-10 11:41:02 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1001@10.1.1.117:51029;transport=udp;fs_nat=yes [CS_HANGUP] Is it possible the NAT causing problem? I am using static IP on both server and phones and do not really need NAT. If it is, how can I disable NAT on the FS server. Can anybody tell me the cause? Gary

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Newb question on dialplan config
by John Millican 10 Oct '08

10 Oct '08

Hello, I am a FreeSwitch newb but have been using asterisk for a while now. I have a project for which I think FreeSwitch will be the best answer, so I need to learn. Have been reading the docs and followed the example at: http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk when I call from a Polycom on the asterisk box to a polycom on the freeswitch box all is good. When id do the reverse I.E. call the ast polycom from the freeswitch polycom I get only the following in the freswitch CLI: 2008-10-10 13:33:24 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/internal/1002(a)192.168.100.16 [521c96a2-5205-bf46-9f9f-31124757b0ef] 2008-10-10 13:33:24 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing John Millican->2002 in context default 2008-10-10 13:33:24 [NOTICE] switch_ivr.c:1098 switch_ivr_session_transfer() Transfer sofia/internal/1002(a)192.168.100.16 to enum[2002@default] 2008-10-10 13:33:24 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-10-10 13:33:24 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/internal/1002(a)192.168.100.16 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-10-10 13:33:24 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 12 (sofia/internal/1002(a)192.168.100.16) Ended 2008-10-10 13:33:24 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel sofia/internal/1002(a)192.168.100.16 [CS_HANGUP] It would seem that the line: 2008-10-10 13:33:24 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting is telling me my problem but I do not yet know why freeswitch does not have a route. I am certain that I have not correctly set the dial plan but haven't a clue what to look at. Both machines are on the 192.168.100.0 net, firewall is off on both the freeswitch box which is running on a VMware installation of WinXP SP3 and the asterisk box. I am using the default configs with the additions per the above page. I did have to change the following from the defaults in vars.xml to get 2 way audio when I call from asterisk to freeswitch: <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.100.16"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=192.168.100.16"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=192.168.100.16"/> Any ideas? Is there something else I need to post to help decipher what I have done wrong or have not yet done? Thanks, JohnM

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SIP documentation, interop, etc
by Kristian Kielhofner 10 Oct '08

10 Oct '08

Hello Freeswitch users, I've started working on a wiki for SIP documentation for interop and other features. I've created a basic page for Freeswitch: http://www.submityoursip.com/wiki/Freeswitch Would someone with more Freeswitch SIP experience help me fill in some more details? Things like support for various specs, issues when working with other devices, etc. I'm still trying to work on a template for implementations and some sort of format but I figured I'd get some information up and see where it goes. Thanks, and let me know if you have any suggestions! P.S. - Obviously it's not just limited to Freeswitch... If you have experience with other SIP implementations, service providers, etc I'd love to read about it! ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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Openzap: Analog FXO disconnect supervision. Is it possible to disconnect on busy? is battery reversal the default? is it power denial?
by Luis F Urrea 10 Oct '08

10 Oct '08

Hi All, I have been working for quite a while in trying to get Openzap to recognize my country specific busy tone in order to hang up an analog FXO channel. Basically, I am having FXO disconnect supervision issues and I wanted to know if it's possible to hang up the FXO by monitoring call progress tones so that when there is a busy tone the FXO goes on hook. I have been tweaking the /etc/openzap/tones.conf based on the ITU.T recommendation E.180 for my country, but so far it doesn't even seem that the changes I make to the tones.conf file are considered when dealing with disconnects from the PSTN side. I restart FS after every change I make to /etc/openzap/tones.conf with no positive results. The Openzap Wiki states that the "loadzone" parameter in /etc/zaptel.conf is not considered when dealing with tones, which makes sense since Openzap seems to use libteletone for tone detection and generation. Here is what I use for busy tone on /etc/openzap/tones.conf: [us] generate-busy => v=-7;%(330,330,450) detect-busy => 450 Therefore I set openzap.conf.xml as follows: <param name="tonegroup" value="us"/> I was able to check that the busy tone can be detected using the following dial plan rule: 1. <extension name="Inbound"> 2. <condition field="destination_number" expression="^1$"> 3. <action application="answer"/> 4. <action application"sleep" data="1000"/> 5. <action application="tone_detect" data="busy 450,0 r +100 hangup normal_clearing"/> 6. </condition> 7. </extension> So that I answer the FXO and hang up when the busy tone is detected. It works fine and the FXO port does not get stuck. Here's the output from the log for the rule above: http://pastebin.com/f43500bd3 I do not see a reason why settings on tones.conf may not be working? Is disconnect on busy-detect the default for Openzap? Is it power denial or battery reversal? Does it matter if I select fxsks in /etc/zaptel.conf instead of fxsls? Your help will be greatly appreciated. Regards, Luis

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VOIP vs PSTN
by Alfred Richmond 10 Oct '08

10 Oct '08

Hello, I am attempting to generate a message to convert to speech and send it out to my users. I am a newbie but I am just not getting it after reading through the documentation. In testing it works fine when sending to my voip connected users using the following: bgapi originate {ignore_early_media=true,originate_timeout=60}sofia/internal/1001 &playback(/usr/local/freeswitch/sounds/warning.wav) however, when I dial a cell phone as below it rings the phone but immediately hangs up before playing the message. Is there something obvious I am doing wrong? bgapi originate {ignore_early_media=true,originate_timeout=60}sofia/gateway/sip.startec.com/14431112222 &playback(/usr/local/freeswitch/sounds/warning.wav)<http://sip.startec.com/14431112222&playback(/usr/local/freeswitch/sounds/wa…> and then the follow up question is do I need bridge the call in the dialplan like so?  <extension name="outbound"> <condition field="destination_number" expression="^(\d{11})$">    <action application="bridge" data="sofia/gateway/ sip.startec.com/$1@xx.xx.xx.xx:5061"/> </condition> </extension>

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gateway with TLS
by paulo leonardo 10 Oct '08

10 Oct '08

Hi, I can't access IRC, because its forbiden. I have freeswitch like endpoint (gateway); I have an IPphone with TLS; When i make a call using the IPphone with TLS to my freeswitch (9995; "delay_echo"), works all signaling with TLS; When i make a call using freeswitch to my ipphone, when i answer the call, HANGUP! Thanks!

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Re: [Freeswitch-users] dialplan_hunt taking too long
by Ineya Ineya 10 Oct '08

10 Oct '08

You\'re right, I disabled ipv6 and replaced \"auto\" with ipv4 addresses (just to be sure) and the lag is smaller now, it\'s 1-2 secs max, which is \'normal\' I guess. Thank you for help. ----- Originálna Správa ----- Od: \"Anthony Minessale\" Komu: Poslaná: 10.10.2008 15:04 Predmet: Re: [Freeswitch-users] dialplan_hunt taking too long > > >From the looks of your SDP on line 129 your system appears to be > misconfigured with IPv6 > Try disabling IPv6 on your interface or setting sofia up according to the > IPv6 instructions on the wiki. > __________ http://cestovanie.sme.sk/ - Celý svet na vašom monitore

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