January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Error in finding OpenZAP span id
by Magesh R 19 Jan '10

19 Jan '10

Dear All, I have installed Sangoma PRI card and installed wanpipe drivers. The wanrouter process started sucessfully. I had the following configurations, openzap.conf: ========== [span wanpipe PRI_1] name => OpenZAP number => 1 trunk_type => e1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 [span wanpipe PRI_2] name => OpenZAP number => 2 trunk_type => e1 b-channel => 2:1-15 d-channel => 2:16 b-channel => 2:17-31 openzap.conf.xml: =========== <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="1"/> </settings> <libpri_spans> <span name="PRI_1"> <param name="node" value="cpe"/> <param name="switch" value="ni2"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </span> </libpri_spans> </configuration> When I started the freeswitch I have received the following error, 2010-01-19 12:41:22.693212 [ERR] mod_openzap.c:2039 Error finding OpenZAP span id: name:PRI_1 Any one please tell me a way to solve this problem... Thanks, Mag.

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IVR returning info via event socket
by Earl Pinkerton 19 Jan '10

19 Jan '10

Hi All, I am trying to call an IVR remotely over the event socket port. I am testing using telnet. I call originate (similar to the following): api originate {origination_caller_id_number=8885551111,ignore_early_media=true}sofia/gateway/teliax/18885552222 &ivr(demo_ivr) This works fine (places the call, runs the IVR, accepts DTMF tones to move between menus). My problem is that I would like the remote app to send a PIN as part of the originate call, then have the callie enter the PIN via DTMF and indicate whether there was a match or not back to the calling app (which is a remote web program). We are trying to keep it simple and have the remote app just call the IVR and get the result, without having to control the IVR through asynchronous events. Is there any way for the IVR to check for a match, then somehow send info back over the port to the calling app? By the way, I tried using menu-exec-api inside ivr.conf.xml (to see if I might be able to use the echo api call or something similar), but I got the following error: 2010-01-19 01:22:55.796658 [WARNING] switch_ivr_menu.c:704 Invalid Action [menu-exec-api] Thanks.

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Bria softphone registration issue (NAT/rport issue?)
by Adam Ford 19 Jan '10

19 Jan '10

I don't know if any of you would be able to help with this, but I figured I would ask. After a few hours of inactivity, it seems my Bria softphones lose connection with FreeSWITCH. They are able to call out, but when calling in I just get silence until it goes to voicemail. It sounds like a classic NAT issue, except the NAT is completely handled by a Soincwall that has VoIP features to support SIP transformations (SonicOS Enhanced 5.x). It seems as though after a period of time the port in which the Sonicwall is assigning for the SIP UA and the port that FreeSWITCH has registered get out of sync, despite the Bria re-registering every 5 minutes. Could this be caused by 'Use rport' being enabled by default in the Bria softphone? Setup - FreeSWITCH 1.0.4 Bria 2.5.4 Sonicwall E5500 with SIP transformations and Consistent NAT enabled Thanks in advance for any help, -Adam

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Call for help - content ideas for freeswitch.org
by Michael Collins 18 Jan '10

18 Jan '10

Hello all, I'd like to ask everyone to think about things that we can put up on freeswitch.org. We are interested in anything related to FreeSWITCH, certainly, but also any VoIP or telecom news/articles on other sites that are of interest to FreeSWITCH users. (See http://www.freeswitch.org/node/229as an example.) Please email me off list if you have ideas about content that would be appropriate for our main page. Likewise, if you would like to be the author of stories and/or blog posts on the main page then definitely let me know. We would love to have some fresh perspectives represented. Thanks! -Michael

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Dial Plan bridge did not return a correct variable from reqular expression
by DJB INTERNATIONAL 18 Jan '10

18 Jan '10

Does anyone experience this problem? SVN: 16395 Dialplan: <extension name="tollfree"> <condition field="${toll_allow}" expression="tollfree"/> <condition field="destination_number" expression="^(\+1|1)?(8(00|88|77|66)[2-9]\d{6})$"> <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/external/1$ 2(a)tollfreetollfree.com"/> </condition> </extension> Possible Error: Dialplan: sofia/internal/2132345567(a)204.110.15.190 Regex (PASS) [tollfree] destination_number(18005551212) =~ /^(\+1|1)?(8(00|88|77|66)[2-9]\d{6})$/ break=on-false Dialplan: sofia/internal/2132345567(a)204.110.15.190 Action set(bypass_media=true) Dialplan: sofia/internal/2132345567(a)204.110.15.190 Action bridge(sofia/external/1(a)tollfreetollfree.com) ****** (It should return $2 as 8005551212) Thank you, Dorn B. (djbinter)

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Takes 5 minutes for FS to start (weird Test-Average-Step messages)
by Nicolas Brenner 18 Jan '10

18 Jan '10

I got a server on Rackspace Cloud (www.rackspacecloud.com) I compiled FreeSWITCH from svn trunk and once it starts it works fine, no problem at all. But, it takes FreeSWITCH about 5 minutes to start. It prints, literally, hundreds of messages like the ones below, until the value of "Test:" reaches 0. I looked for info about this on the wiki and the mailing list but couldn't find anything. Is there a way to suppress this? Why does it happen? Is it because the server is a virtual machine? Thanks! 2010-01-18 19:12:00.373937 [CONSOLE] switch_time.c:188 Test: 611 Average: 10000 Step: 13 2010-01-18 19:12:01.373937 [CONSOLE] switch_time.c:188 Test: 598 Average: 10000 Step: 12 2010-01-18 19:12:02.373937 [CONSOLE] switch_time.c:188 Test: 586 Average: 10000 Step: 11 2010-01-18 19:12:03.373937 [CONSOLE] switch_time.c:188 Test: 575 Average: 10000 Step: 10 2010-01-18 19:12:04.373937 [CONSOLE] switch_time.c:188 Test: 565 Average: 10000 Step: 9 2010-01-18 19:12:05.373937 [CONSOLE] switch_time.c:188 Test: 556 Average: 10000 Step: 8 2010-01-18 19:12:06.373937 [CONSOLE] switch_time.c:188 Test: 548 Average: 10000 Step: 7 2010-01-18 19:12:07.373937 [CONSOLE] switch_time.c:188 Test: 541 Average: 10000 Step: 6 2010-01-18 19:12:08.373937 [CONSOLE] switch_time.c:188 Test: 535 Average: 10000 Step: 5 2010-01-18 19:12:09.373937 [CONSOLE] switch_time.c:188 Test: 530 Average: 10000 Step: 4 2010-01-18 19:12:10.373937 [CONSOLE] switch_time.c:188 Test: 526 Average: 10000 Step: 3 2010-01-18 19:12:11.373937 [CONSOLE] switch_time.c:188 Test: 523 Average: 10000 Step: 2 2010-01-18 19:12:12.373937 [CONSOLE] switch_time.c:188 Test: 521 Average: 10000 Step: 1 2010-01-18 19:12:13.373937 [CONSOLE] switch_time.c:188 Test: 520 Average: 10000 Step: 1 2010-01-18 19:12:14.373937 [CONSOLE] switch_time.c:188 Test: 519 Average: 10000 Step: 1

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Re: [Freeswitch-users] More playing with sessions in lua
by Pete Mueller 18 Jan '10

18 Jan '10

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How to register from FS to Huawei SoftX3000
by Dome Charoenyost 18 Jan '10

18 Jan '10

Dear sir, I found some provider use Huawei SoftX3000 and can limit use call from they softphone only. (use eyeball SDK). They can limit some account can register and call by sip server like an FS and Asterisk. but some account can't. (register and call by softphone). and i don't know how they can do that. So i try to use wireshark to debug sip headeder when use softphone with both account type. it's nothing diferent. I want to use both account work by FS register to Huawei SoftX3000. Can someone help me. i can give you softphone and both account type for test. Best Regards. Dome C.

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Call Manager
by John 18 Jan '10

18 Jan '10

Is there an active project for a client call control software for Freeswitch, such as ShoreTel's or Cisco's Call Manager? Looking for the ability for a Windows or Mac user to be able to transfer calls, See presence, conference calls etc. Thanks

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More playing with sessions in lua
by Troy Anderson 18 Jan '10

18 Jan '10

I think there may be other ways to do this, but if I could understand how to do it from lua, I think it would help me understand more about what's going on and make the mod_lua even more valuable to me and anyone else struggling with this kind of issue. I am trying to answer a call in lua, exchange a bit of information with the caller, then I want to originate a call to another endpoint, and depending on what their response to some audio questions is, connect the original caller to them, or send the original caller away. The problem is that I want the original caller to her ring tones or music on hold while they're waiting. In addition to the way presented here, I've tried parking the caller using uuid_park, but still can't figure out how to play music/ringback for them. session:answer(); session:sleep(1000); session:execute("playback","pleasehold.wav") local targetEndpoint = "1100@default" -- or wherever -- ringback works only AFTER the bridge line, below. How do I get it to start immediately? local destSession = freeswitch.Session("{ringback='myringback.wav'}".. targetEndpoint) -- this "originates" a call to targetEndpoint local digit destSession:playAndGetDigits(1,1,3,3000,"#","instructions.wav","[1-3]") if (digit == "1") then freeswitch.bridge(session,destSession) else session:execute("playback","goodbye.wav") session:hangup() destSession:hangup() end I appreciate any help on this. -Troy

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