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monday build
by Kendall Stauffer 14 Dec '09

14 Dec '09

HI I tried to build the svn last Friday and it didn't make the sphinx dll, so I thought I would wait for the Monday build (web site says update every Monday). Is there going to be a windows build today, and or is the sphinx dll build problem fixed if I build it myself? Thanks!!

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FreeSWITCH 1.0.5pre9 is now available
by Michael Collins 14 Dec '09

14 Dec '09

FYI, The latest pre-release is now available. Usual information is available here: http://www.freeswitch.org/node/222 Please update as soon as you can. (SVN trunk users do the "make current" thing please.) We need your testing and feedback please! Many thanks for continuing to support FreeSWITCH. -Michael

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pfSense with Freeswitch - so far so good, but no calls going through
by mailinglist 14 Dec '09

14 Dec '09

Hi All Ok, so next episode in the saga of getting this monster of the ground :-) I've gotten the FS up and running pretty much I guess, but I'm missing something. It has been set up as per the 'multi-homed' document (http://wiki.freeswitch.org/wiki/Multi_home_tutorial) I want to use the webinterface in pfSense, as it is the easiest for me to manage, and gives me a better overview. If I do a reloadxml it gives me this output on the console: freeswitch(a)firewall.fribert.dk> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) 2009-12-07 21:21:40.719721 [INFO] mod_enum.c:808 ENUM Reloaded 2009-12-07 21:21:40.719721 [INFO] switch_time.c:661 Timezone reloaded 530 definitions API CALL [reloadxml()] output: +OK [Success] I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Looking at the status I see: sofia status profile internal API CALL [sofia(status profile internal)] output: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.11.12.25 Ext-RTP-IP 10.11.12.25 SIP-IP 10.11.12.25 Ext-SIP-IP 10.11.12.25 URL sip:mod_sofia@10.11.12.25:5060 BIND-URL sip:mod_sofia@10.11.12.25:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 10 FAILED-CALLS-IN 5 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: MThhODdkOWFkMGM4YTk5OWU1MTMzMjg5NmFjOGFhNWU. User: 1001(a)10.11.12.25 Contact: "1001" <sip:1001@10.11.12.145:59650;rinstance=7c2be2513804dcea;fs_nat=yes;fs_path=sip%3A1001%4010.11.12.145%3A59650%3Brinstance%3D7c2be2513804dcea> Agent: X-Lite release 1103k stamp 53621 Status: Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:03:40) Host: firewall.fribert.dk IP: 10.11.12.145 Port: 59650 Auth-User: 1001 Auth-Realm: 10.11.12.25 Call-ID: OTc2NTJkMmU3MGQ0MDNkN2NiZDgzZDFjYzQ1MzYxMDY. User: 1002(a)10.11.12.25 Contact: "1002" <sip:1002@10.11.12.195:4117;rinstance=8c18309a2c957d6a;fs_nat=yes;fs_path=sip%3A1002%4010.11.12.195%3A4117%3Brinstance%3D8c18309a2c957d6a> Agent: 3CXVoipPhone 3.1.6288.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:25:09) Host: firewall.fribert.dk IP: 10.11.12.195 Port: 4117 Auth-User: 1002 Auth-Realm: 10.11.12.25 ================================================================================================= And... sofia status profile external API CALL [sofia(status profile external)] output: ================================================================================================= Name external Domain Name N/A DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 87.61.18.196 Ext-RTP-IP 87.61.18.196 SIP-IP 87.61.18.196 Ext-SIP-IP 87.61.18.196 URL sip:mod_sofia@87.61.18.196:5080 BIND-URL sip:mod_sofia@87.61.18.196:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 2 Registrations: ================================================================================================= ================================================================================================= In my Dialplan I've created these two entries: ----- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 ----- and ----- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$1 I'm not sure if the latter is correct or needed to make local calls? But anyways, it doesn't seem to react as per my intentions. If I try and make a local call from 1001 to 1002 it says 2009-12-07 21:40:02.776210 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001(a)10.11.12.25 [b115e2b1-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:40:02.776210 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-07 21:40:02.796449 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [aa22e3b1-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:40:02.874492 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-07 21:40:02.894599 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-07 21:40:02.894599 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001(a)10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1085 Session 15 (sofia/external/$1) Ended 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1085 Session 14 (sofia/internal/1001(a)10.11.12.25) Ended 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001(a)10.11.12.25 [CS_DESTROY] As I read it it goes into context default, and then tries to create an external channel, which I don't understand why? And then it fails of course. Then if I try to do an external call (with the leading 0) it gives me: 2009-12-07 21:41:33.655915 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001(a)10.11.1.25 [25260ce8-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:41:33.655915 [INFO] mod_dialplan_xml.c:252 Processing 1001->012345678 in context dfault 2009-12-07 21:41:33.655915 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [36e10ce870e3-de11-af59-000c29b7b4cb] 2009-12-07 21:41:33.755921 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NOROUTE_DESTINATION] 2009-12-07 21:41:33.755921 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATON 2009-12-07 21:41:33.755921 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001(a)10.11.12.25 [C_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1085 Session 17 (sofia/external/$1) Ened 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [S_DESTROY] 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1085 Session 16 (sofia/internal/1001(a)1.11.12.25) Ended 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/100110.11.12.25 [CS_DESTROY] So for me, it looks like it never comes to the dialplan I've entered into the pfsense interface??? I've used the gateway value instead of the profile value in my bridge. So the question is, do I go and enter the 'default.xml' for the dialplan, or what do I do? What have I missed here???

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bidirectional SRTP
by Helmut Kuper 14 Dec '09

14 Dec '09

Hello, today I found that my SRTP calls are only SRTP in one way (from snom to FS). The other direction is still RTP (at least if I believe wireshark). How can I get both directions using SRTP? I do a simple call to an announcement in FS. My SIP-profile has: <param name="inbound-late-negotiation" value="true"/> I followed the document of FS Wiki: http://wiki.freeswitch.org/wiki/Secure_RTP But it seems there is no ch_var like sip_has_crypto. Is there a way to have both directions/streams per leg using SRTP? Please find attached the pcap trace of my snom phone. There you can see that snom offers crypto to FS. FS sends an OK with SRTP and RTP. When you look into the RTP data you see SRTP from snom to FS and RTP from FS to snom. Any hints? regards Helmut

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Link between User context and dialplan
by Otis 14 Dec '09

14 Dec '09

Hi folks I am so sorry if this is such a basic thing. well, when a user/extension eg 8888 is created in with say a user context - SWAHILI-SPEAKERS Please hear are my questions: 1. What dialplan will that user/extn use. 2. I guess I have to create a dialplan Should the dial-plan also be called WAHILI-SPEAKERS (is the case relevant ) ? Or could it be any name ? 3. And how does FS know to load that dialplan for that user. 4. Where should that xml file be stored ? 5. Is there a means of determinig which dialplan was used for a call ? Thanks I think I have demonstrated enough thickness for now

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Which ATAs to chose for modem connections?
by Peter P GMX 14 Dec '09

14 Dec '09

We currently use Patton gateways SN4116 for attaching fax and modem equipment to our Freeswitch system. Freeswitch is in bypass-media-mode, so media flow goes the following way: Modem/Fax => Patton_SN4116 => Patton_SN46XX =>PSTN/ISDN However modem connections are not very reliable. We exchanged the SN4118 against a "Fritzbox" ATA and the situation improves. However Fritzboxes do not deliver the number of ports we need. What is your experience? Which ATA is the best choice for modem connections? Best regards Peter

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embedded freeswitch compatable hardware
by Brian May 14 Dec '09

14 Dec '09

Hello, I asked this question on my local linux user group mailing list, and got the recommendation to ask here. Anyway, at the moment I am running Asterisk on an IP04 embedded system. http://www.rowetel.com/ucasterisk/ip04.html It works well most of the time, however there are some bugs that do, under circumstances lead to less then desirable behaviour (such as on some occasions which I don't fully understand sometimes the remote system fails to generate any audio packets when there is no audio - almost like silence suppression was supported by the remote system - and asterisk fails to generate any audio packets in return; on another slower computer running the same SIP software and on the same network everything works fine; as far as I can tell the software - twinkle - doesn't even support silence suppression). I suspect at least some - if not all - of the issues I have encountered may be resolved with Freeswitch, however I don't really want to replace my small, energy efficient, embedded system, with a large, power hungry computer system. Overkill. An added complication is I need at least 1 analogue port to connect to the Australian based telephone line (2 ports exchange ports and 1 extension port would be ideal but not essiential). Unfortunately, I have been told that the IP04 hardware isn't compatable with the requirements of Freeswitch. Such as not having a MMU. So there doesn't appear to be much effort porting Freeswitch to IP04 as a result. I do have a spare TDM400p card, although as it is full height, suspect this isn't going to help. Are there any other good alternatives? Thanks. -- Brian May <brian(a)microcomaustralia.com.au>

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Java ESL
by Niall Crosby 13 Dec '09

13 Dec '09

Hi, I am about to start writing a Java Event Socket Library as I can't find one already written thats available. 1 - Is there one already out there? 2 - If not, any pointers as to what design I should follow? Which of the current ESL's is the best modal to follow? Thanks, Niall. -- -- The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the sender. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer.

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CDR question.
by William King 13 Dec '09

13 Dec '09

Anyone know a good way to determine which extension picked up a call that was bridged to 10+ extensions? -William

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Link between Use-context and dialplan
by Otis 13 Dec '09

13 Dec '09

Sorry I posted this earlier but did not do the due diligence and sent it with so much typo them meaning does not come out: In a nutshell I would like to know : 1. How FS would know which dialplan to use for an extension with user context other than default. 2. If a file file has to be created does the name matter 3. Where should that file be located. Thanks.

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