November 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_lcr and 302 (Moved temporarily)
by Armando Kumbara 19 Nov '13

19 Nov '13

I have configured the diaplan to make a database query using the lcr module and respond with a 302 (moved temporarily) the destination number with a new prefix to the gateway that is sending the call. The process of manipulatinfg the digits works fine and the 302 message is generated bur the call fails because of a problem in the 302 message below: * SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 192.168.8.23:5060;rport=5060;branch=z9hG4bK-528b3362-c488a182-4cc5 Max-Forwards: 50 From: "Armando Kumbara"<sip:1234567@192.168.8.23 <sip%3A1234567(a)192.168.8.23>>;tag=45826b5f-13c4-528b3362-c488a182-473c To: <sip:9903645879714@95.107.130.71:5080 <http://sip:9903645879714@95.107.130.71:5080>>;tag=vSU04ySty4yvQ Call-ID: 12b6c96c-45826b5f-13c4-528b3362-c488a182-670e(a)192.168.8.23 <12b6c96c-45826b5f-13c4-528b3362-c488a182-670e(a)192.168.8.23> CSeq: 1 INVITE Contact: "unknown" <sip:9902645879714@192.168.8.26:5060 <http://sip:9902645879714@192.168.8.26:5060>> User-Agent: FreeSWITCH-mod_sofia/1.5.6b+git~20131104T192948Z~384ddf1517~32bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=31;text="NORMAL_UNSPECIFIED" Content-Length: 0 Remote-Party-ID: "9903645879714" <sip:9903645879714@192.168.8.25 <sip%3A9903645879714(a)192.168.8.25>>;party=calling;privacy=off;screen=no* Contact: "unknown" <sip:9902645879714@192.168.8.26:5060> should be Contact: "9902645879714" <sip:9902645879714@192.168.8.26:5060>. the dialplan entry for theDB query is the following: < *condition field="destination_number" expression="9903645879714"> <action application="lcr" data="$1 test"/> <action application="capture" data="mynum|${lcr_auto_route}|([^/]*)$"/> <action application="redirect" data="${mynum[1]}@192.168.8.26:5060 <http://192.168.8.26:5060>>"/> </condition>* Basically it should receive the call and query the db, capture the last digits after the "/" from the returned string and return it to the gaetway via the 302 message. Can you please advice me how i can fix this issue? Is there any way i can fix the contact information from the dialplan?

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read_codec and write_codec xml_cdr
by FERNANDO VILLARROEL 19 Nov '13

19 Nov '13

Dear All. I am trying to use xml_cdr for save the cdr in backend (postgresql). My problem is only the answered calls have the read_codec and write_codec codec variables. Any idea why the cancel calls have not this variables on xml_cdr. I appreciate some help.

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xml_curl: initiator type
by vad fabi 19 Nov '13

19 Nov '13

Hi all, How to indentify call initiator type (gateway or locally registered user) then freeswitch sends xml_curl dialplan request ? As i see all post variables almost same for both types.

2 4

Ignore busy signal when dialling out on an FXO port
by Oren Hecht 19 Nov '13

19 Nov '13

Hi, I'm new to freeswitch so please excuse me if I'm not using the right terms. I have freeswitch connected to a POTS line using an FXO port, and I am trying to use it to dial out without success. The POTS line requires me to dial '9' first to get a line for dialing out, and as long as I don't press 9 it gives a 'busy' tone. I am trying to bridge an extension to freetdm port with adding a '9' prefix, but I get "Failed to open channel" and the reason is "NORMAL_CIRCUIT_CONGESTION". When I look at 'ftdm dump' I see: "last_state: BUSY". I guess the reason for failure is that freeswitch hears the busy tone and thinks the line is not free and aborts. I also thought this is related to the early media signaling so I tried adding {ignore_early_media=true} thinking it will ignore the busy tone, but it dodn't help. How can I tell freeswitch to ignore the busy signal and just dial the destaination number, (the '9' prefix will give it a free line)? Thank you very much for the help, Oren Hecht.

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No ringback tone on bridge
by mbo 19 Nov '13

19 Nov '13

Hello, we receive calls from gateway A and bridge them using gateway B to the destination number . <extension name="test"> <action application="bridge" data="sofia/gateway/B/+49123456789" /> </extension> The problem is, that we do not hear a ringtone at all. If we put a "ring_ready" before the bridge, we can hear a ringtone immediately, but then the callers hear a ringtone even if the bridged destination is not ringing yet. <extension name="test"> <action application=ring_ready /> <action application="bridge" data="sofia/gateway/B/+49123456789" /> </extension> Is there a solution that the callers hears a ring tone as soon as the destination is ringing, but not before ? Thanks Markus

4 3

Anyone really familiar with Confluence that whats to lend me a hand with a project?
by Ken Rice 19 Nov '13

19 Nov '13

Contact me off list or find me on IRC SwK on #FreeSWITCH via irc.freenode.net Thanks K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

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Channel variable to detect WebRTC is being offered
by Joegen Baclor 19 Nov '13

19 Nov '13

I am playing with FreeSWITCH 1.4 beta branch to create a WebRTC to SIP Bridge via ESL. Reading the code, it seems media_webrtc variable is only being set for outbound channel. What is the best way to detect that the inbound channel offer is a WebRTC stream? - joegen

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module install
by Joli Martinez 19 Nov '13

19 Nov '13

Hello, I have installed Freeswitch via yum on a CentOS box. Not sure how to add a module on Freeswitch. Could someone help me since this is installed via yum and not compiled. Thanks,

2 1

Transferring calls - can Freeswitch instruct the up-stream provider to do the transfer?
by Clive Lansink 18 Nov '13

18 Nov '13

Hi everyone. I have an IVR application written in Python that utilises Freeswitch to handle all the SIP and VOIP stuff. However in this case, Freeswitch is really only the end point and is not the client's PBX system. The client would like an option that will cause the incoming call to be transferred to a specified destination. The destination is an outside number and not a Freeswitch extension. I am playing with this now. I believe I can use the API originate command to originate a call to the specified destination and use the bridge application to bridge them. But even if that works, it would mean the incoming call is still being routed right through Freeswitch and out again to the destination. Now is there a way I can tell Freeswitch to issue like a call transfer command to the up-stream provider so Freeswitch can then hang up and the call can be handled elsewhere? In the old days, and in fact I have done this in a much earlier version of this same IVR application, I would have sent a flash to the outside world, wait for dialtone then dial the destination number, wait for ring and then hang up. Can anyone advise how would I do something like that now? Cheers. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150

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tuning/measuring - hostile networking
by Jon Hancock 18 Nov '13

18 Nov '13

Hi list, I’m working with FreeSwitch 1.5.x in BigBlueButton 0.81 We have a tough networking problem: Calls between China and New Zealand. We’re running FreeSwitch on a Hong Kong VPS. This is presumed to be the best middle ground to have a server handle the calls. The China domestic internet is messy and there’s a lot of filtering of traffic in/out of China. To compound the problem, traffic from New Zealand up to Hong Kong isn’t the greatest. For example, ping tests between a China user and the Hong Kong server are around 400ms and ping between New Zealand and the Hong Kong server are between 250-400ms. What I’d like to do is find methods to measure/log the latency, packet loss, etc and associate this with subjective call quality. This is our first step. As a second, I have an option of having the China side of the calls coming in over a VPN from China which could improve matters. I would need to understand how to have FreeSwitch use this VPN connect for one side of the call. Also, if someone knows some good settings for codecs, etc for such a hostile environment, that might be a good starting point. Any ideas appreciated! thanks, Jon -- Jon Hancock

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