May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Aastra and SCA
by Chris Chen 12 May '10

12 May '10

I just tested the 2.6 firmware on my Aastra 55i, looks like they have made SCA working with my Polycom phone IP 650 now I will see if there any other problems with Aastra phone. Chris Chen On Tue, May 11, 2010 at 4:12 PM, Mark Maly <mark.maly(a)molcs.org> wrote: > FYI- > > > > Got a ticket w/Aastra and they said, > > > > “We have just released a new GA firmware that I would like for you to load > onto one of your phones to see if it resolves the issue. You can download > this from our website at www.aastratelecom.com/support and click on > Download Area and select the 6731i. The firmware version you are looking for > is 2.6 and it is listed under Current Software Release. > > > > Please load that on a test phone and let us know the results.” > > > > I’ll let you know, > > Mark > > > > > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Monday, May 03, 2010 8:00 PM > > *To:* freeswitch-users(a)lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Aastra and SCA > > > > Nope but we have exchanged some emails about it back and forth and some > beta firmware where they did half way fix it... but seems you need the > call-info header on ALL packets associated with the dialog of that call. > They are still missing a few. > > > > /b > > > > On May 3, 2010, at 7:55 PM, Mark Maly wrote: > > > > Brian, > > > > After my first reply I did receive a msg from Aastra support and whether FS > had reported the problem. The contact wondered whether they [FS] had been > given any “reference number” related to the problem. > > > > Would love to help and would pass ref num if you had one. > > > > Mark > > > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Sunday, May 02, 2010 11:14 PM > *To:* freeswitch-users(a)lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Aastra and SCA > > > > Give up their exists a bug that prevents it from working on any aastra. > Polycom and Cisco SPA work flawless. > > > > /b > > Sent from my iPad > > > On Apr 30, 2010, at 9:07 PM, "Mark Maly" <mark.maly(a)molcs.org> wrote: > > Hi, > > > > I’ve tried to patiently figure this out by reading the wiki and this list. > Unfortunately, I’ve been unable to get it right. > > > > I have 2 Aastra 6731is and a 51i and trying to get SCA working. I’m > experiencing problems similar to the Cisco thread from last month – outgoing > calls implement SCA well. Incoming calls ring all lines and appearances > work, but when one phone is answered, the line appearances are removed from > the remaining phones. I am not attempting to use any DNS. > > > > My configuration has all three phones plus FS on a local LAN. Nothing too > fancy. Each line is configured for Broadsoft SCA and SCA bridging is > enabled globally for the phones. > > > > Trying to update/replace an old phone at my church. > > > > Any help would be greatly appreciated. > > > > Thanks, Mark > > Mark.maly(a)molcs.org > > > > _______________________________________________ > 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 > > _______________________________________________ > 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 > > > > _______________________________________________ > 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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Mod_lcr question...
by Louis Ntasin 12 May '10

12 May '10

Gents- I am new to this list and would like to say hi to all. I have a question about mod_lcr. When I load mod_lcr I see errors about a missing table: npa_nxx_company_ocn. I did some searching and saw that it is meant for intra lata/state calling. I don't need this feature. Is there a flag that I can turn on to tell mod_lcr not to worry about the table? Thanks. Louis

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short ringback tone when answer
by Tony Tin 12 May '10

12 May '10

Hi All, I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is connected to a 4ess T1). Every thing works fine so far except one little problem that annoys me. Every time I dial in the ivrs, there is a very short (0.5 second) ringback tone exactly while the "<action application="answer"/>" line is executed in the dial plan. I want to disable this ringback tone because the ivrs will answer the call immediately, the ringback tone is really unnecessary in this case, but I fail to do that. I try to set the variable "ringback" to change the ringback tone, it's also not working. Could anyone please help. I'm using the native pri stack. The default.xml containts only below lines, and attached is the log file. <extension name="testing"> <condition field="destination_number" expression="^(6.*)$"> <action application="sleep" data="3000"/> <action application="answer"/> <action application="sleep" data="3000"/> </condition> </extension> Regards, Tony

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Voice to text / speech recognition
by Wellie Chao 12 May '10

12 May '10

I'm looking to provide a feature to transcribe voicemails into text for my users. I've looked at SpinVox (which was acquired by Nuance). Their licensing method doesn't seem particularly friendly or flexible. They charge per user and lock each license to a particular user's actual identity. If I have a user named John Smith, I have to buy a license from Nuance _for John Smith_. If John Smith stops using the voicemail to text feature, but I have another user named Jane Doe who wants to start using the voicemail to text feature, I have to buy a new license for Jane Doe. Since Nuance makes you pay for a license one year at a time, it gets expensive if you have any significant churn. Apart from that, it is just annoying to have to register users with another company. I'd like to buy 100 or 1000 or 10000 licenses and use them how I see fit and not have to bother with registering individual users with a provider of speech recognition services. I am wondering if other FreeSWITCH users have recommendations for good voice to text services or software. I'd be happy to consider either a service or software. If software, ideally it would be free of course, but a reasonable cost would also be acceptable. I know about PocketSphinx, but is it really sufficiently high quality as to be useful for transcribing voicemails? I don't need 100% accuracy -- probably 80% or 90% would be good enough since I will also attach the WAV file and direct my users to use the WAV file when in doubt. Any pointers or tips would be appreciated.

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OT: Disabling 3pcc in CallManager 7.1
by Kristian Kielhofner 11 May '10

11 May '10

Hello everyone, I've got a SIP trunk up to a Cisco CallManager 7.1 system. I've been able to get it to work by enabling 3pcc in FreeSWITCH but I'd like to disable 3pcc and be able to have a standard INVITE w/SDP, 183, 200, conversation. Does anyone have any clue how to configure CallManager appropriately? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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Re: [Freeswitch-users] Aastra and SCA
by Brian West 11 May '10

11 May '10

Nope but we have exchanged some emails about it back and forth and some beta firmware where they did half way fix it... but seems you need the call-info header on ALL packets associated with the dialog of that call. They are still missing a few. /b On May 3, 2010, at 7:55 PM, Mark Maly wrote: > Brian, > > After my first reply I did receive a msg from Aastra support and whether FS had reported the problem. The contact wondered whether they [FS] had been given any “reference number” related to the problem. > > Would love to help and would pass ref num if you had one. > > Mark > > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West > Sent: Sunday, May 02, 2010 11:14 PM > To: freeswitch-users(a)lists.freeswitch.org > Subject: Re: [Freeswitch-users] Aastra and SCA > > Give up their exists a bug that prevents it from working on any aastra. Polycom and Cisco SPA work flawless. > > /b > > Sent from my iPad > > On Apr 30, 2010, at 9:07 PM, "Mark Maly" <mark.maly(a)molcs.org> wrote: > > Hi, > > I’ve tried to patiently figure this out by reading the wiki and this list. Unfortunately, I’ve been unable to get it right. > > I have 2 Aastra 6731is and a 51i and trying to get SCA working. I’m experiencing problems similar to the Cisco thread from last month – outgoing calls implement SCA well. Incoming calls ring all lines and appearances work, but when one phone is answered, the line appearances are removed from the remaining phones. I am not attempting to use any DNS. > > My configuration has all three phones plus FS on a local LAN. Nothing too fancy. Each line is configured for Broadsoft SCA and SCA bridging is enabled globally for the phones. > > Trying to update/replace an old phone at my church. > > Any help would be greatly appreciated. > > Thanks, Mark > Mark.maly(a)molcs.org > > _______________________________________________ > 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 > _______________________________________________ > 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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In-band vs. Out-of-band DTMF
by VM Knott 11 May '10

11 May '10

I'm trying to test DTMF signalling using in-band DTMF. Currently my application monitors the signals via the ESL socket, which works great for out-of-band DTMF. My problem is that when I run my current test using in-band, I'm using a phone line that I know supports out-of band DTMF, so I'm not sure that the data I'm seeing in the ESL socket is out-of-band or in-band. Does anyone know: a) if in-band signalling appears the same across the ESL socket? b) if I should be looking somewhere specific for the in-band signals. c) is there a way to disable out-of-band signalling so I can test only the in-band? Any assistance would be greatly appreciated. Victor

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Socket problem
by Jor3l Boa 11 May '10

11 May '10

Hello there, I'm trying to setup FreeSwitch but I'm getting this error everytime I run it.. *mod_event_socket.c -> Socket Error Could not listen on 127.0.0.1:8021* Also, an error here: *mod_xml_curl.c:230 xml_url_fetch() Received HTTP error 0 trying to > fetch ...* The question is, is something needed to avoid the 8021 error? I'm running Ubuntu 10.04 with ufw enabled (port 8021/tcp allowed). Note: Googled but nothing helps Thanks in advance

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one way audio (sometimes)
by lands＠freenet.de 11 May '10

11 May '10

Hi, I have a strange problem with my conference setup on freeswitch. my setup is as followed: - freeswitch quick and dirty installation according to http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install - freeswitch release: svn-17188 - http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR, - made my DNS SRV records - account from iptel.org - "phoner" softphone - UDP ports are opened inbound from 5060 till 64000 on the firewall, which is in front of the FS box - all udp ports outbound are open on the firewall - the FS box uses a public IP Well, I can dial the SIP URI. The connection is working. I can hear the IVR. Well, sometimes at least. I make a call => I can hear everything. Seconds later, I make a call with the same client on the same PC => audio is coming through. I cancel the connection and make the call again => there´s no audio coming through, and so on... When I do 10 calls within 20 seconds with 4 ones there´s no audio. According to wireshark the inbound RTP stream is sometimes missing on some calls. According to freeswitch debug, the used RTP ports are alway between 5062 and 64000 It sounds like a firewall problem with closed port. But it can´t as the firewall ports are opened. Any ideas what else could be wrong? Chris

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about pastebin
by budi wibowo 11 May '10

11 May '10

dear all just wonder about pastebin, anytime i click always ask username and password. is it the same as freeswitch username and password? i already create account in fs but cant access pastebin also TIA budi

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