June 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

in a lua script, is "session" available only after "answer"?
by Vincent Xia 12 Jun '13

12 Jun '13

i have an extension in the dialplan\default.xml <extension name="test_lua" continue="false"> <condition field="destination_number" expression="^(9\d{3})$"> <action application="answer" /> <action application="lua" data="test.lua" /> <action application="sleep" data="1000" /> </condition> </extension> the test.lua is something like: if session:ready() then freeswitch.consoleLog("info", "a leg is ready.\n" ) else freeswitch.consoleLog("info", "a leg is not ready.\n" ) end this works fine and info "a leg is ready." will be shown on the console. when im trying to swap the sequence of the dialplan as <action application="lua" data="test.lua" /> <action application="answer" /> the freeswitch gets down, with a windows message box prompting to debug or close the program. (a bug?) does this mean in a lua script, "session" is only available after "answer"?

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pickup endpoint - Break the Simultaneous Ring
by Lloyd Aloysius 12 Jun '13

12 Jun '13

Hello All The following break the Simultaneous Ring.User registered two different devices. Only one Device ring after add the *,pickup/dave(a)alcan.mydomain.ca * *bridge_data: * {sip_invite_domain=alcan.mydomain.ca ,ignore_early_media=true,force_transfer_context=alcan.mydomain.ca }[leg_timeout=20]user/dave@alcan.mydomain.ca*,pickup/dave@alcan.mydomain.ca* Thanks LLoyd * <http://www.sunteltech.ca/blog/>*

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Setting custom presence/BLF
by Kelly McDonald 12 Jun '13

12 Jun '13

François, The answer to your question (dated Apr 13 2013) is yes: I'm hoping that this matches up with your earlier post so that others can find it (I'm new to the Freeswitch mailing list, so I couldn't just reply to the message.) With the help of Allamamond here on the list, I was able to do the following: telnet localhost 8021 > Trying 127.0.0.1... > Connected to localhost. > Escape character is '^]'. > Content-Type: auth/request > auth ClueCon > Content-Type: command/reply > Reply-Text: +OK accepted > sendevent PRESENCE_IN proto: sip from: 6001(a)192.168.1.99 login: 6001(a)192.168.1.99 event_type: presence alt_event_type: dialog event_count: 1 unique-id: kpmtest1 Presence-Call-Direction: outbound Answer-State: early > Content-Type: command/reply > Reply-Text: +OK a4e12f2a-d2c3-11e2-9598-1170e5c1ad37 Unique-id should most likely be . . . um - unique (although it worked fine for me when it wasn't) And I tried a virtual id for the from and login. For example"kpmtest(a)192.168.1.99" worked fine. On the phone (yealink) I just used "6001" or "kpmtest" in the BLF without the trailing domain. Answer-State could also be "confirmed" or "terminated" Finally, this was working with 1.2 Freeswitch. HTH Kelly

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Re: [Freeswitch-users] Sending NOTIFY with perl ESL
by Kelly McDonald 12 Jun '13

12 Jun '13

Just to follow up on an email from Jun 24 2012: Vallamamod was trying to set BLF fields from ESL, and hadn't quite got it - I was having the same problems, so I PM'd him to see if he had figured it out and he had. Based on the code that he sent me, I was successfully able to alter the BLF on a Yealink phone. here is my raw text from a ESL session using telnet (well- almost raw, - I prefixed all responses from freeswitch or telnet with a >) telnet localhost 8021 >Trying 127.0.0.1... >Connected to localhost. >Escape character is '^]'. >Content-Type: auth/request > auth ClueCon >Content-Type: command/reply >Reply-Text: +OK accepted > sendevent PRESENCE_IN proto: sip from: 6001(a)192.168.1.99 login: 6001(a)192.168.1.99 event_type: presence alt_event_type: dialog event_count: 1 unique-id: kpmtest1 Presence-Call-Direction: outbound Answer-State: early >Content-Type: command/reply >Reply-Text: +OK a4e12f2a-d2c3-11e2-9598-1170e5c1ad37 ----------------------------------------------------- This changed the light - For Answer-State you can also use confirmed and terminated Thanks to Vallimamod for his help. Kelly

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Re: [Freeswitch-users] Unable to get K3765 dongle signalling using mod_gsmopen
by Silviu CHINGARU 11 Jun '13

11 Jun '13

Thank you for your reply. Can you give me a small example of usage of minicom with dongles? Baud rate, etc... Should I call only from minicom via AT commands sent to interface or use freeswitch to init the call and only sniff the interface with minicom. Sorry if is noob question but I never tryed those low level commands. -- Best regards, Fiftyz -------- Original message -------- From: Dmitry Lysenko <dvl36.ripe.nick(a)gmail.com> Date: 12/06/2013 00:01 (GMT+02:00) To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Unable to get K3765 dongle signalling using mod_gsmopen Hi! > I think the serial interface is ok because I can send AT commands to init the call but I don't get any signaling. You should try other ttyUSBx. One of ttyUSBx used for commands only, but another for commands AND signalling too. You can open ttyUSBx with minicom and make call to number of SIM card inserted to modem. If you will see RING you will find "right" serial port. Best wishes, Dmitry. 2013/6/11 Silviu CHINGARU <silviuchingaru(a)gmail.com> Hi, I'm trying to make mod_gsmopen work with my testing freeswitch. I have the following setup: K3765 Huawei dongle Freeswitch 1.2.9 with mod_gsmopen I setup a gateway using gsmopen.conf.xml (see attached) and an extension in dialplan. I've set consolelog 9 and I'm able to make call via mod_gsmopen from any switch phone, but the signaling is not working. So when I answer to mobile called by gsmopen module, sip phone is still displaying "Conecting...", I can talk to mobile phone via sip phone and reverse but when I close the conversation from mobile side is still "Conecting...". In console I don't see any signaling from gsmopen like ringing, answered and so on. I think the serial interface is ok because I can send AT commands to init the call but I don't get any signaling. Also the audio interface is ok because I have audio both sides (via Early Media because the call is never signaled as answered). I've attached: - reload.log (output from freeswitch when I type reload mod_gsmopen) - call.log (output from freeswitch when I init the call via mod_gsmopen) - gsmopen.conf.xml (I've changed / masked context, ip and phone number for privacy) I've used this dongle also with chan_dongle in asterisk and was working fine but on freeswitch I'm unable to configure it. Please help, I don't even know what to read! What could be the problem, what is wrong? How can I figure out what is wrong? Thank you very much for your time! Best regards, Fiftyz _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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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USSD support for mod_gsmopen
by Yuriy Ostapchuk 11 Jun '13

11 Jun '13

Hi, I've submitted a patch that implements USSD messaging support. (useful for checking and refilling account balance and other provider-specific usage) Here is the JIRA ticket: http://jira.freeswitch.org/browse/FS-5078 If this feature is considered useful, I would appreciate if it's integrated in git. Thanks, Yuriy

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SIP-S and openssl
by mehroz 11 Jun '13

11 Jun '13

Hi all, The question is beyond the SIP SSL/TLS support, everything working perfectly fine with creating keys/certs, generating agent.pem and client with cafile.pem. I am looking for a way to use different Ciphers for encryption/decryption before keys get involved. I dont know if freeswitch could directly or indirectly help to acquire the desired results, but since FS is using openssl at TLS layer, it might have a link ! Ciphers are being traced out in wireshark packets and i can see ClientHello, sending some particular cipher, and Server respond with serverHello, deciding the Cipher from preferences... However, changing the Cipher from client creates issues at server end and normally, "Handshake failures" are returned, even though those offered ciphers are supported by Server. Please comments if any body have previously worked or dev guys could help so? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-S-and-openssl-tp7591496.h… Sent from the freeswitch-users mailing list archive at Nabble.com.

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mod_callcenter doesn't play very nicely with mod_xml_curl
by Cal Leeming [Simplicity Media Ltd] 11 Jun '13

11 Jun '13

Hello, Having a few issues with mod_callcenter and mod_xml_curl. If anyone would like to take a look, please feel free; http://jira.freeswitch.org/browse/FS-5504 Any thoughts/comments would be very much appreciated. Thanks Cal

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Debian package dependency:
by Julian Pawlowski 11 Jun '13

11 Jun '13

Hi, since a few days the package dependencies for freeswitch-meta-codecs seems to be broken in the 1.2 stable tree. I get the following message during install: The following packages have unmet dependencies: freeswitch-meta-codecs : Depends: freeswitch-mod-b64 (= 1.2.11~n20130608T103852Z-1~wheezy+1) but it is not installable I had a look to our Debian repository and yes, the named package freeswitch-mod-b64 indeed is missing. However it exists in the 1.3 unstable tree so I guess it's new and the reference to this package was mistakenly transported to the stable tree. Travis, would you mind to correct this so our Gemeinschaft5 build will be running again? ;-) That would be awesome! Cheers, Julian

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Re: [Freeswitch-users] NAT problem: Use local IPv4 for specific LAN IP address range
by Ben 11 Jun '13

11 Jun '13

Steve, Sorry for the late reply (4 months!!!), the immediate need for a fix went away, until now. The problem with setting up an additional profile would be that the end user would have to change port numbers when moving from the internal network to an external network. Unfortunately the end user does not have access to these sorts of settings with the app we're using, and more importantly, I need this to be seamless between networks. Does anyone else have any ideas? I'm starting to think this problem is unsolvable with the current functionality. Implementing an equivalent of the Asterisk "nat.conf" with a "localnet=" line, which will force the Freeswitch local IPv4 address as the SDP for any requests originating from one of these "localnets" would hopefully resolve the issue. Ben From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Steven Ayre Sent: 07 February 2013 13:07 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Use local IPv4 for specific LAN IP address range One simple option would be to have a 2nd profile listening on another port (eg 5080) that does not use ext-*-ip and connect to that internally. -Steve On 7 February 2013 11:44, Ben <ben122uk(a)gmail.com> wrote: Greetings all, A question on NATing in Freeswitch - I have my internal SIP profile to use a static public IP for the ext-sip-ip and ext-rtp-ip. The Freeswitch server is on a local LAN IP, for example 192.168.0.10. External SIP clients can access the server via the external IP, as there's a static NAT on the router to map the public IP to the private IP. I have another SIP client on a different LAN, but behind the same firewall, for example, 10.0.0.10. This client is allowed to talk to the server on its private ip of 192.168.0.10, by allowing inter-vlan traffic to pass through. My challenge is, how do I get Freeswitch to recognise this internal SIP client as being allowed to use the internal IP for SIP/RTP communications, and therefore present the client with internal addresses in the SIP header for SIP/RTP? Due to statically setting ext-sip-ip and ext-rtp-ip, Freeswitch tells the client to go via the public IP. The firewall has a policy to not accept traffic to the public IP if sourced from an internal device. I have explored the NATing options in Freeswitch, but can't find anything that fits my scenario. It seems to be more towards how to enable NAT in certain situations, not disable it. I have also looked at re-writing the SDP, but this re-writes the incoming SDP from the clients. My intention was to re-write the outgoing SDP from the Freeswitch to include the internal IP address as the RTP connect in address. I don't have much Asterisk experience, but I seem to remember a nat.conf file where you could specify subnets that were treated differently. These could be classed as internal subnets, and would be given the private IP in the SDP from asterisk, everything got the public IP. I'm also aware that I could create another SIP profile easily with different NATing/IP options, but I want to find out if there's anything I can do with the above first. Many thanks for the help! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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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