August 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

something related to channel variables
by Vincent Xia 18 Aug '13

18 Aug '13

how to get the contact field of the sip message "invite" or "200 ok" in the dialplan xml? how to get the registering ip address of an endpoint in the dialplan xml (as we have sofia_contact in the command line)?

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Codec order flipped in proxy-media mode re-INVITE
by Jeff Pyle 16 Aug '13

16 Aug '13

Hello, FreeSWITCH Version 1.3.17+git~20130318T211211Z~2dc3b47db1. The config is a basic SBC to handle NAT traversal between public and private interfaces. Proxy-media mode is configured. Inbound call from carrier hits us with G711u and G729 in the SDP, in that order. Inside PBX chooses G711u and call is established. 30 minutes later the carrier does a session refresh with a re-INVITE with an identical SDP as the initial INVITE. When FreeSWITCH the re-INVITE leaves FreeSWITCH towards the inside PBX, the attached SDP has G711u and G729 are flipped such that G729 is now preferred. The PBX accepts the G729 request and the session switches to G729. There are no SDP smarts in the dialplan. Codecs are not handled at all by design. Is this something that could be caused by a misconfiguration? Or, is my next step a jira? - Jeff

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Re: [Freeswitch-users] SIP TLS Issues
by Lappe, Adam 16 Aug '13

16 Aug '13

Hi all, Some more things I tried so far: openssl x509 -noout -modulus -in agent.pem | openssl md5 (stdin)= ebdfb317206ba89d07217c06e1f0d6eb openssl rsa -noout -modulus -in agent.pem | openssl md5 (stdin)= ebdfb317206ba89d07217c06e1f0d6eb At least the certificate and private key in the agent.pem are correct. There is no output on the cli when I try to register a phone. My guess is that the content of agent.pem and/or cafile.pem is wrong. Can someone please confirm this? Best regards, Adam Wed, 14 Aug, 2013 at 16:07 PM, Adam <ala(a)qsc.de>: Hi all, i am trying to configure FreeSWITCH to speak TLS with all Clients. I followed the tutorial on http://wiki.freeswitch.com/wiki/SIP_TLS but I am still not sure what key / cert belongs in which file. I have a SSL123 Thawte Wildcard Certificate. Am I supposed to cat this cert + priv. key into agent.pem and the primary and secondary intermediate into the cafile.pem? I did this and set the right permissions. The internal sofia profile on port 5061 (TLS) is RUNNING. But no client (for example Polycom VVX1500) can register now. If I set it TCP and Port 5060 (which is RUNNING as well) everything works fine. Wireshark shows me the following Client -> FS Client Hello FS -> Client Alert (Level Fatal, Description: Handshake Failure) I also tested openssl s_client -connect (IP):5061 -showcerts but it only says: CONNECTED(00000003) 139847050823328:error:14077410:SSL routines:SSL23_GET_SERVER_HELLO:sslv3 alert handshake failure:s23_clnt.c:724: --- no peer certificate available --- No client certificate CA names sent --- SSL handshake has read 7 bytes and written 225 bytes --- New, (NONE), Cipher is (NONE) Secure Renegotiation IS NOT supported Compression: NONE Expansion: NONE --- I guess the problem is the agent.pem and/or cafile.pem agent.pem looks like this -----BEGIN CERTIFICATE----- (Thawte SSL123 Wildcard Web Certificate) -----END CERTIFICATE----- -----BEGIN RSA PRIVATE KEY----- (Unencrypted Private Key) -----END RSA PRIVATE KEY----- cafile.pem like that: -----BEGIN CERTIFICATE----- (Thawte Primary Intermediate) -----END CERTIFICATE----- -----BEGIN CERTIFICATE----- (Thawte Secondary Intermediate -----END CERTIFICATE----- Any suggestions? Thanks in advance, Adam

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Wanpipe - Wanrouter help - freetdm and freeswitch
by Ravi 16 Aug '13

16 Aug '13

Hello Everyone ! I am not sure if this error is relevant for this mailing list. My apologies if this is not the forum. I am trying to configure the Sangoma card, with the freetdm installation for Freeswitch. When I complete the wanpipe installation, and try wanrouter status, and wanrouter start I get the following: [root@bfree-server log]# wanrouter start Starting up device: wanpipe1 wanconfig: WAN device wanpipe1 driver load failed !! : ioctl(wanpipe1,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Configuring interfaces: w1g1 w1g1: unknown interface: No such device done. ___________________________________________________________________ Can anyone help me understand why this is happening ? And how to resolve it. Thanks. Ravi

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error after fax receiving completes
by Babak Yakhchali 16 Aug '13

16 Aug '13

Hi I tested fax receiving using Freeswitch stable versions 1.2.10 11 and 12 and I get same results. at the end of fax receiving some assertion fails in heap validation. I build Freeswitch on windows 7 64 bit with visual studio 2010 in 64 bit debug mode. I attached the error screen to this email. if any more information is needed I can provide it. thanks

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BLF
by Kevin Wormington 16 Aug '13

16 Aug '13

Hi, I'm doing some testing of BLF with Freeswitch and Yealink T28 phones. It works fine as far as flashing the indicator when a monitored extension is ringing, solid red when in use, etc. However, the indicator is green even when an extension is not registered. I haven't found much documentation on this, but it seems the indicator should be unlit if the extension is not registered. If I manually clear the registration with sip tracing on I see that FS sends the below content in the NOTIFY and gets a 200 OK back from the T28: Subscription-State: active;expires=1175 Content-Type: application/dialog-info+xml Content-Length: 155 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="13" state="partial" entity="sip:1102@x.x.x.x"> </dialog-info> I assume this is not working because the phone is expecting something else in order to turn off the light. Is this something that is configurable or would it need code changes? Also, is it possible to use an OPTIONS ping or otherwise to determine if a registered endpoint is available rather than wait for the expiration? I know this can be done on un-registered gateways. Thanks, Kevin

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Public and default context question
by Grant Bagdasarian 16 Aug '13

16 Aug '13

Hello, I just started to play around with FreeSwitch, and I have a couple of questions. Is the default context only used for devices which are registered with FreeSwitch? If I have a Carrier which communicates with FS through a public IP and a Kamailio SIP Proxy which communicates with FS through a private IP address, both will use the public context right? My goal is to configure FS as a SBC, and I see there is a sbc directory in the ../conf folder, but when I change the pre-process value to point to sbc/freeswitch.xml, I'm not able to start freeswitch anymore. <X-PRE-PROCESS cmd="include" data="sbc/freeswitch.xml"/> There are no devices(IP Phones for example) which will register with FreeSwitch, so basically I won't be needing the default context, correct? Regards, Grant

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Recording calls with Freeswitch - log says recorded, but no file found
by Ravi 16 Aug '13

16 Aug '13

Hello Everyone ! I am trying to record calls received in an extension (1003). The dialplan is given below extension name="003"> <condition field="destination_number" expression="^003$"> <action application="set" data="record_file_name={recordings_ dir}/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.wav" inli$ <action application="record_session" data="{record_file_name}"/> <action application="bridge" data="user/1003"/> </condition> </extension> When I call this extension from an extension 1006, this is what I get in the log. But when I go to the folder /usr/local/freeswitch/recordings/ there are no files at all. I am just wondering if this has got to do with any file permissions, or if I am missing something. Please help. Thanks. Ravi freeswitch@bfree-server> 2013-08-14 06:53:51.185709 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/1006(a)10.0.0.16 [2d3314f4-0480-11e3-9a2f-4d1cb7d8b5b4] 2013-08-14 06:53:51.245710 [INFO] mod_dialplan_xml.c:558 Processing Ravi <1006>->1003 in context default 2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *1 execute_extension::dx XML features 2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1006.2013-08-14-06-53-51.wav 2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *3 execute_extension::cf XML features 2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *4 execute_extension::att_xfer XML features 2013-08-14 06:53:51.245710 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/sip:1003@10.0.0.10:5060 [2d3d1a26-0480-11e3-9a4d-4d1cb7d8b5b4] 2013-08-14 06:53:51.445707 [NOTICE] sofia.c:5884 Ring-Ready sofia/internal/sip:1003@10.0.0.10:5060! 2013-08-14 06:53:51.445707 [INFO] switch_ivr_originate.c:1190 Sending early media 2013-08-14 06:53:51.445707 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1006(a)10.0.0.16! 2013-08-14 06:53:52.705700 [NOTICE] sofia.c:6547 Channel [sofia/internal/sip:1003@10.0.0.10:5060] has been answered 2013-08-14 06:53:52.725708 [NOTICE] switch_ivr_originate.c:3437 Channel [sofia/internal/1006(a)10.0.0.16] has been answered 2013-08-14 06:54:05.005696 [NOTICE] sofia.c:715 Hangup sofia/internal/sip:1003@10.0.0.10:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2013-08-14 06:54:05.025711 [NOTICE] switch_ivr_bridge.c:1575 Hangup sofia/internal/1006(a)10.0.0.16 [CS_EXECUTE] [NORMAL_CLEARING] 2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1567 Session 139 (sofia/internal/sip:1003@10.0.0.10:5060) Ended 2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1571 Close Channel sofia/internal/sip:1003@10.0.0.10:5060 [CS_DESTROY] 2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1567 Session 138 (sofia/internal/1006(a)10.0.0.16) Ended 2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1571 Close Channel sofia/internal/1006(a)10.0.0.16 [CS_DESTROY]

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squelch repeated log entries for Ping failed
by Jeff Pyle 16 Aug '13

16 Aug '13

Hello, When a gateway is not reachable I see messages like this in the log: [WARNING] sofia.c:4880 Ping failed gw1 with code 408 - count 1/0/1, state DOWN This message continues to repeat until the gateway becomes reachable. I do like to know a gateway is down, but I don't need to know if a gateway is *still* down. It fills up the syslog on a relatively limited embedded device in this case. Is there a way to limit logging to only when a gateway changes state? I tried the "$RepeatedMsgReduction on" option in rsyslog but it seems that's no longer available. Hmm. Version 1.2.11-n20130813T230126Z-1~wheezy+1+git~20130813T212504Z~2131cd6cf2 (-n20130813T230126Z-1~wheezy+1git 2131cd6 2013-08-13 21:25:04Z) Regards, Jeff

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Establishing SRTP from SBC to endpoint
by Peter 15 Aug '13

15 Aug '13

In my environment, I have the following (simplified) setup: FS1 ---- FS SBC --- FS2 Phones registered to FS1 (100x) use TLS/SRTP and phones registered to FS2 (200x) use SIP/RTP FS1 has inbound-bypass-media set to true to allow SRTP peer to peer and direct to the SBC. If I make an inbound call (eg: 1000 to 2000), SRTP is correctly established between the phone and SBC with RTP on the other side of the SBC to the internal phone. However, when I try it the other way, I can't get SRTP established from the SBC to the external phone. I've been using https://wiki.freeswitch.org/wiki/Secure_RTP as a guide. I've even tried explicitly setting sip_secure_media to true on the SBC and FS1. The dialplan on the SBC has: <extension name="outgoing"> <condition field="destination_number" expression="^(10[0-9][0-9])$"> <action application="set" data="sip_secure_media=true"/> <action application="bridge" data="sofia/external/${ destination_number}(a)10.1.1.204"/> </condition> </extension> And on FS1, the dialplan has: <extension name="Local-Numbers"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="sip_secure_media=true"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> </condition> </extension> Note that I've been testing this against two phones with SRTP enabled, but only one that is using TLS. I get the same result calling each phone. On a related point, what it the step required for a TLS connection from the SBC to the phone? I'm assume the phone just needs the CA cert from the SBC. Correct? Any information as to where I'm going wrong will be gratefully accepted. Thanks Peter

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