- FreeSWITCH-users - FreeSWITCH Mailing Lists

Mobile + FS + FXS = 487 ORIGINATOR_CANCEL
by Garrison Hoffman 05 Nov '10

05 Nov '10

I have this recurring issue when using an ATA with FreeSwitch. The specific ATA doesn't seem to matter, as I've tried two Grandstream's and currently a Patton SL4022. When voicemail is set to ring for 30 seconds, many calls — especially those from mobile networks — get dropped with 487 ORIGINATOR_CANCEL. Oddly enough. the issue never occurs when using an IP phone. Reducing the ring time to 20 sec allows calls to go to voicemail, but presumably doesn't actually fix anything. I've tried a variety of early media & ringback settings, but to little effect. I've searched the mail archives, wiki, and flipped through the book, but while I've found lots of things to try, I'm not really any closer to understanding precisely why the caller leg hangs up. sip trace: http://pastebin.freeswitch.org/14398 -gh

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Does Freeswitch and Digium TE220 support E1 in Macau ?
by Tony Tin 05 Nov '10

05 Nov '10

Hi, Does Freeswitch and Digium TE220 support E1 in Macau ? If no, any alternative? Thanks for all input. Regards, Tony

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Google voice incoming call doesn't work
by Jian Ren 05 Nov '10

05 Nov '10

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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no audio when originate to 2 PSTNs
by Tidiane Sy 04 Nov '10

04 Nov '10

Hi all, When I do an originate between two PSTNS, the two telephones ring. But when both answer, there is no audio. I have audio when the originate is between an sip client and a PSTN. My freeswitch is not natted my console logs here: http://pastebin.freeswitch.org/14351 my network capture here: https://rcpt.yousendit.com/978754413/751404552085a1a09ea574cb819a5f74 Your help will be really appreciated Tid

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FreeSwitch Registration Handling
by Daniel Eiland 04 Nov '10

04 Nov '10

Hi, I've got an issue with the way FreeSwitch is currently handling (un)Register requests. When my SIP client performs a registration to remove all its binding by using a Contact header with a value of "*", FS generates a corresponding 200 / OK response with a Contact header with a value of "*". My current SIP client does not like a Contact header being included the 200 / OK response in this particular scenario and flags a failure. Is there any way to configure FreeSwitch (or its embedded Sofia stack) to prevent it from adding a Contact header in this particular scenario? Thanks, Daniel

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CFP - Open Source Telephony Devroom - FOSDEM 2011
by Bryan M. Johns 04 Nov '10

04 Nov '10

Greetings, This is a call for presentations for the open source telephony devroom at FOSDEM 2011 - http://www.fosdem.org/. We will be holding a day full of presentations on development topics in the area of open source telephony on Sunday, February 6th. The schedule allows for presentations from 9:00 to 17:00. The room we have available will have a projector, wifi, and 59 seats. Please submit all proposals no later than 2010-12-10. Notification of accepted speakers will be provided by 2010-12-17. We will then work to have a schedule finalized by 2011-01-07. Talks should be submitted by fulling out the following form: http://goo.gl/9SWlz Please direct discussion about this devroom to the telephony-devroom mailing list hosted on http://lists.fosdem.org/. If you would like to contact the devroom organizer directly, please contact Russell Bryant <russell(a)digium.com>. Feel free to forward this along to any people or mailing lists that you think would be interested in this event. Thank you! Bryan M. Johns Digium, Inc. | Community Director 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6007 Check us out at : www.asterisk.org or www.digium.com

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Conferences with members using different languages
by Peter Olsson 04 Nov '10

04 Nov '10

Is this possible, or do I need to patch mod_conference to get it working? I want different users in the same conference to be able to get audio prompts in different languages. Since the audio paths is stored within the conference profile (which is the same) I'm not sure this can be done today - am I right? /Peter

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Freeswitch as STUN server?
by GillesToo 04 Nov '10

04 Nov '10

Hello To solve the issue of opening UDP ports dynamically on the firewall that protects a remote SIP client located elsewhere on the Net, the most used solution is to have the client use STUN to open UDP ports, learn its public IP address, and then re-use those UDP ports to register with the IP PBX and let SIP/RTP packets in through the firewall. While not actively handing calls, STUN-capable SIP clients send keep-alive datagrams regularly to keep the UDP ports open on their firewall. Unfortunately, symmetric NAT firewalls don't allow re-using UDP ports to work with another host. So I was wondering if Freeswitch could run as STUN server or if someone knew of a good open-source STUN server that I could install on the same host as the Freeswitch server? http://img545.imageshack.us/img545/3580/ippbxstun.png Would this solve the issue? If not, I guess the alternatives are trying UPnP, NAT-PMP, or IAX, and if all else fails, running OpenVPN and configure both end-points to pass all SIP/RTP packets through this TCP tunnel. Thank you for any feedback. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-as-STUN-server-tp5… Sent from the freeswitch-users mailing list archive at Nabble.com.

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problem with jingaling (may cause by TLS)
by joy this 04 Nov '10

04 Nov '10

Dear all: I am confused with dingaling for several days. I hope someone could help me please. Below are my config steps: 1. Download “gnutls-2.9.9.exe” from “ http://josefsson.org/gnutls4win/” and install. 2. Build “libgnutls-26.lib” (refer to “ http://wiki.freeswitch.org/wiki/Ixemel_MSVS_project_example”) I have rebuilt the FreeSwitch. 3. Copy the TLS dlls to “C:\FreeSWITCH\Debug”. 4. Set the conf in “\conf\jingle_profiles\client.xml” (refer to “ http://wiki.freeswitch.org/wiki/Mod_dingaling#TLS” and the letter “[Freeswitch-users] Problems making a receiveing calls with mod_jingling” ). 5. Start FreeSwitch. Below are my config in step 3: <include> <profile type="client"> <param name="name" value="thisjoy"/> <param name="login" value="account(a)gmail.com/gtalk"/> <param name="password" value="pwd"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <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="888"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include> I input the command “dingaling status” and found the account was “UNCONNECTED”. Besides, I saw a message in “freeswitch.log”, it is: I. 2010-10-28 18:46:56.468750 [DEBUG] libdingaling.c:1237 TLS NOT SUPPORTED IN THIS BUILD! II. 2010-10-28 18:47:16.453125 [DEBUG] libdingaling.c:1607 io error 2 7 retry in 20 second(s) My OS is Windows XP and the FreeSwitch is 1.0.head. Thank you for helping. Sincerely yours, Thisjoy.

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"nat-options-ping" timer
by Stas Khirman 04 Nov '10

04 Nov '10

Trying to test "nat-options-ping", but somehow see no OPTION message issued by FreeSWITCH to NATed clients . What is a default timer for this feature? How to set up a short one? Regards Stas

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