Re: [asterisk-users] voice mail system
Welcome to the right mailinglist :) Asterisk can definately do that. All you need to do is make a extension, with an ivr and some sound files. Have a look at these links: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail http://www.voip-info.org/wiki/view/Asterisk+cmd+Background Trying to do that from the start might be a bit fast, i'd suggest looking for a beginner tutorial first. Greetings, Zoa On 9/7/2010 4:16 PM, Frenette, Rob wrote: Hi, Does Asterisk, provide the option within its voice mail system to be able to do the following: We want to have an option to be able to have two mail boxes on one ext. For example: we want to be able to have a French Mailbox and a English one... So I can say the following... You have reached ext 000 to continue this message in French press example 3 -- and it would then go to my French greeting... would anyone know if this is possible? Kind Regards, Rob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1 second Audio Lag
Colin, I'm working for Zoiper, you can contact us directly on supp...@zoiper.com Zoa Nick Brown wrote: Do you see the issue when calling between two softphones? Do you see the issue if you call from your mobile into an echo test? Setting TOS flags on packets will make no difference unless the gear in between is configured to treat them differently. Not that I envision this is the issue at all. Nick. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of colin mcdermott Sent: Tuesday, 27 July 2010 5:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 1 second Audio Lag Hi All (reposting after 24 hours). I will do a test call from a soft phone to my mobile. I can speak into my headset and the audio is heard instantly. But if I speak into my mobile there is a 1-2 second delay in the Audio. I am using SIP. I am only finding it in the Zoiper Softphones that we are using. We are able to make a call without lag on the X-lite softphone no problem. Sadly the customer is Quite attached to the Zoiper. I have set QOS = CS5 for both SIP and RTP packets. Altering these settings has no effect to the lag issue. We have three 24 port Gigabit switches, with the top switch connecting in the Asterisk Box. Even the stations plugged into the TOP switch have this delay and to the same extent as the other switches. No routers on the loop I have tried switching the stations to IAX. No effect. I have tried using GSM instead of G711 (alaw). No effect. I have about 30 stations. No change under heavy or light load. I have done a Wireshark trace on the stations and no issues detected when I go analyse on the RTP packets. All sequencing is correct. Is Zoiper any good? Anyone else had these problems? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality
I think that the clock resets would cause no audio or garbled audio every 20 minutes, not constant interference. Could you tell us how many simultaneous calls were in the trunk and what the size is of 1 voice packet ? Can you try putting maximum 30 calls per trunk (use multiple trunks if needed) and see if the problem goes away. Greetings, zOa Vieri wrote: --- On Thu, 5/13/10, Zoa zoach...@securax.org wrote: Can you try trunk = no ? Lifesaver... trunk=no made the interference go away. I have clean audio now. Quote: IAX Trunking needs support of a hardware timer. I'm supposing my system is using the DAHDI-driven Digium cards on my motherboard. I don't know how hardware timers work and if Digium hardware rely on the motherboard (my system clock is going too fast and my ntpd is constantly adjusting the clock by -2.6 seconds every 20 minutes). In any case, since I'm on a dedicated LAN I guess I can safely set trunk=no. Thanks! Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality
Hello, Can you try trunk = no ? How much jitter do you see on the link ? Zoa Gareth Blades wrote: There should be no noticeable difference between slin, ulaw and alaw so what you have is fine. The problem must be elsewhere. Vieri wrote: --- On Thu, 5/13/10, Gareth Blades list-aster...@skycomuk.com wrote: Show the details on the active channels when using both methods and check what codecs are being used. The audio codecs are different: Type: SIP State: Up (6) Rings: 0 NativeFormats: 0x4 (ulaw) WriteFormat: 0x40 (slin) ReadFormat: 0x40 (slin) WriteTranscode: Yes ReadTranscode: Yes Type: IAX2 State: Up (6) Rings: 0 NativeFormats: 0x8 (alaw) WriteFormat: 0x8 (alaw) ReadFormat: 0x8 (alaw) WriteTranscode: No ReadTranscode: No By the way, I have this in iax.conf: [interboxIAX2] deny=all allow=ulaw allow=gsm type=friend host=192.168.250.111 secret=mysecret auth=plaintext requirecalltoken=no qualify=yes context=mycontext trunk=yes username=interbox Shouldn't the channel details report ulaw instead of alaw? Also, if I change [interboxIAX2] and replace ulaw with alaw, the result is the same (I still experience bad audio quality). Maybe I should try slin but how do I force it? Vieri wrote: Hi, I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall). One trunk is SIP and the other IAX2. Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a third noise overlapping with a scratchy sound as if it were some kind of interference). So lately I setup calls to go through the SIP trunk and audio quality is OK (no third overlapping noise). This is happening between Asterisk 1.4.31 and a 1.2.40. I'm wondering if there's something I can tweak in IAX2 to eliminate this artifact. Could the IAX2 jitter buffer between 1.2 and 1.4 be an issue (I believe it's enabled by default)? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Re: SpiderMux?
I have played with one before, it worked quite well. (Until somebody fried it by accident). Joachim Peter wrote: Hi, I have one in stock - got it from a client who wanted to get rid of all his old IT equipment. Looks strange, did not have enough time to play with it Tried it once, looked hard to configure. It stays unused in the storage room. Peter On 29.4.2010 10:20, Tim Nelson wrote: Greetings all- I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux. It looks rather interesting. Has anyone used one? Where did you purchase it? Pricing? Operational issues? http://spidermux.com/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license
JR Richardson wrote: Zoa wrote: On friday we finally released Attrafax under a GPL2 license. It comes with its own set of modems and built in transparent gatewaying. The solution should be quite stable as long as the line quality is ok. (Some tools for measuring the line quality are included in the release, as well as some fax2mail scripts). There is an example implementation included for Asterisk 1.4, if someone wants to porting it to the new fax backend or more recent asterisk versions and needs some help, let us know. I tested Attrafax this afternoon and was very pleased to see that it worked first time right out of the box. I tested the gateway function with the Asterisk source in the tarbal, Zaptel 1.4.10.1 and a Digium TE405P ver2 4-port T1 card. I really like the console output while processing faxes. Very impressive. Would anyone mind sharing any performance statistics based on real word usage or even high volume lab testing? I'm wondering how many concurrent T38 to PRI faxes could be handled with high end server hardware. Where are the bottlenecks for the software stack, RAM, PCI Bus, Proc Speed, Disc I/O? Would there be a problem running 3 to 4 PRI's full of T38 to SIP Faxes on one server? Could the Attrafax software handle that volume? Thanks in advanced for any feedback. JR CPU is the limiting factor, but on recent hardware you should be able to do 120 channels or more simultaneously, without it having to be really high end hardware. We have a test tool in the lab, similar to the show codec translations on asterisk, to estimate how many channels you could do, i will have it added to the attrafax archive in the coming week so that you can estimate things easier. To already give you some idea in the mean time, when we ran the test a long time ago on our very old single core p4 based xeon 3.06 ghz ( http://www.cpubenchmark.net/cpu_lookup.php?cpu=Intel+Xeon+3.06GHz ) we could do about 40 simultaneous channels. Ratings on recent cpu's such as the core i7 have a rating on the same website that is about 15 times higher, so i would presume that 120 channels would be handled easily. Greetings, Zoa -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.
Thanks, I have uploaded the patch to the website and will let you know the feedback we receive. Greetings, Joachim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.
I'm sorry, we do not have anything more recent ourselves. We will make some when we have a little bit more time, we are quite busy at the moment. Greetings, Joachim Matt Watson wrote: I just downloaded a copy of this, by any chances does Zoiper by any chance have diff files available for a more recent 1.4.x release? (I know 1.6 is probably out of the question) Thanks, -- Matt On Mon, Mar 8, 2010 at 12:11 PM, Matt Watson m...@mattgwatson.ca mailto:m...@mattgwatson.ca wrote: Awesome! I was an Attrafax customer and was very disappointed when it vanished and couldn;t get new modules for newer versions Asterisk with our paid license. If anybody is working on t38 gatewaying code for 1.6, it would be worth a look at this, as I can attest that Attrafax worked quite well at t38 gatewaying. -- Matt On Sun, Mar 7, 2010 at 4:52 AM, Zoa zoach...@securax.org mailto:zoach...@securax.org wrote: On friday we finally released Attrafax under a GPL2 license. It comes with its own set of modems and built in transparent gatewaying. The solution should be quite stable as long as the line quality is ok. (Some tools for measuring the line quality are included in the release, as well as some fax2mail scripts). There is an example implementation included for Asterisk 1.4, if someone wants to porting it to the new fax backend or more recent asterisk versions and needs some help, let us know. The full press release can be found here: http://www.zoiper.com/downloads/opensource_fax_stack_PR.pdf the project homepage can be found at www.zoiper.com/foip/ http://www.zoiper.com/foip/ Cheers, Zoa -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mail-2-Fax and Fax-2-Mail solution for Asterisk with T38
We released the attrafax sources under GPL 2 last week. (It's an alternative to the digium fax and the spanDSP fax) There are some fax2mail scripts included in the archive on www.zoiper.com/foip/ Zoiper biz (the softphone) also has a printer driver, for win2k and above (64 bit versions as well). Greetings, Zoa On 3/7/2010 1:50 AM, Thorolf Godawa wrote: Hi, I am looking for an Mail-2-Fax and in a second step Fax-2-Mail-solution that works via T38 with Asterisk, currently still version 1.4 but it also should work with 1.6. For Mail-2-Fax I am thinking that you either have to install a special printer-driver on your Windows-PC (Mac and Linux would be good too), where you can print your fax too and where you have to enter the destination number. Second possibility would be, that the user sends an e-mail with the attached fax to the server, the server has to open the attachment, converts it and sends it via T38 to the number found in the e-mail-body. I would prefer if the attachment would be an image or pdf only, but it also might be nessecary to support office-formats too. The Fax-2-Mail-solution might be more simple, the system receives the T38-fax, converts it in TIF, JPG or PDF and send it via E-Mail to the person that is assigned wit the fax-number. I would prefer a Linux-based opensource-solution, but if there are other good solution I might look at them too! So every suggestion would be nice, thanks a lot, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.
On friday we finally released Attrafax under a GPL2 license. It comes with its own set of modems and built in transparent gatewaying. The solution should be quite stable as long as the line quality is ok. (Some tools for measuring the line quality are included in the release, as well as some fax2mail scripts). There is an example implementation included for Asterisk 1.4, if someone wants to porting it to the new fax backend or more recent asterisk versions and needs some help, let us know. The full press release can be found here: http://www.zoiper.com/downloads/opensource_fax_stack_PR.pdf the project homepage can be found at www.zoiper.com/foip/ Cheers, Zoa -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Web operator/softphone with integration features
Hello, Give our zoiper softphones a try, you could achieve this functionality by sending a url over IAX (Sendurl) or by using the open website on incoming call. (In which you pass the callerid as a paramter to the website to open the ticket that matches that one. (You could also ask for the customer to type their ticket number with DTMF and then send that info to Zoiper). Greetings, Joachim www.zoiper.com Carlo Dimaggio wrote: Hi All, I would like to know if there is a good web operator/softphone for a little help desk environment (5-10 people). Apart from the classic features (call, transfer, conference,...), I need a small integration with the internal trouble ticket system / crm. For example when a call arrives to a number (ex: a number in sip to: header), the software should open the right TT web page or any web link. Can you help me? (names, useful links,...) Thanks and Regards, Carlo Dimaggio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT: Has SILK codec gotten anywhere?
Zoiper supports it in our wideband beta: http://www.zoiper.com/downloads/beta/Zoiper%20Communicator_Free_1.12wb2_Installer.exe (the beta is a bit old though). Cheers, Zoa On 2/20/2010 8:02 PM, Kyle Kienapfel wrote: Hi, I stumbled upon mentions of a SILK codec last night on skypes skype for sip information page. I tried looking into it further and found some blog and mailing list posts from 2009 but I can't find any mentions of anything other than skype using the codec. Has the codec not gotten anywhere so far? http://en.wikipedia.org/wiki/SILK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Random DTMF tones generated from speech in conversations
I have seen this years ago, i received complaints about women voices triggering dtmf. With some help from Mr. Underwood, it was able to confirm lots of false positives on the dtmf detection. My issues went away when we upgraded all cards to the ones with the octasic DSP chip on them. Zoa On 12/12/2009 7:49 PM, hbk wrote: Hi, My Asterisk systems runs like a dream with mISDN, SIP and even and old Digium board. But have almost in every conversation some irritating DTMF being generated. The seems to be just as often from all trunks but are worse if noise load speaker in other end. Any good advices? Where to look for forgotten DTMF detection settings? Thank you! HB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Magnetic door locks
I think you can buy some kind of ATA's to do such a job. I do not however remember any brand names Google returned these links: http://www.voip-info.org/wiki/view/Asterisk+phone+doorview_comment_id=15775 http://www.abptech.com/products/its.html Mobotix c james wrote: I have an opportunity to interface asterisk with a security system to open their magnetic door locks. The security system needs a dry contact close upon activation to signal the door. Has anyone done this before? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium announcement: new community manager - John Todd
congratz!
Zoa
John Todd wrote:
On Tue, May 20, 2008 at 7:41 PM, John Todd [EMAIL PROTECTED] wrote:
I'd like to take a few moments to introduce myself and the new role
Hi John,
Like Jared, you need no introduction to most of us, you are a pillar
of the asterisk community. When I first heard of asterisk, the first
information I found on it was your publicly posted sip and
extensions.conf files. Since those heady days, we've had the pleasure
of talking to you about Freenum.org on the VoIP Users Conference and I
note in passing that you are able to do several text chats while
speaking coherently, a talent that may come in handy in your new
position. WELCOME! (raises glass)
Randy
Thanks! I've got my work cut out for me. :-)
JT
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Subject: [Rails-spinoffs] Class instantiation problems
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Hi Guys,
I'm converting some Mootools code to Prototype and am having problems.
I'd like to instantiate an itemScroller Class that takes an html
element as its first argument and a list of options as it's second.
The class has default options which can be overwritten.
var itemScroller = Class.create({
options: {
containerSelector: 'ul',
itemSelector: 'li',
viewPortSelector: '.wrap',
scrollerSelector: '.item_scroller',
nextSelector: '.next',
prevSelector: '.prev',
itemsPerScroll: 5,
wrapEnds: false,
mode: 'vertical'
},
initialize: function(el,options
Re: [asterisk-users] Outbound international calls over BT ISDN30
You might need to set the dialplan to international or so in the config files. Zoa Stuart Ford wrote: Hello all As always I'm trying the mailing list as a last resort as I'm out of options. I am seemingly unable to dial international numbers over our BT ISDN30 line. I've checked with BT and the number format they're expecting is: 00CCnumber (where CC is the country code). But this doesn't work. Looking at the PRI debug, the most notable error seems to be: Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] I've also tried different number formats, including: +CCnumber +0CCnumber But to no avail. Anybody know what I'm doing wrong? Here's a complete debug dump of a failed international call to the US: Many thanks Stu -- Executing [EMAIL PROTECTED]:2] Dial(SIP/sbf-b7c104e0, Zap/g1/0012127551200) in new stack -- Making new call for cr 33090 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=47 Call Ref: len= 2 (reference 322/0x142) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 83] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 3 ] [6c 0d 21 80 30 31 36 31 34 38 36 37 37 38 30] Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '01614867780' ] [70 0e a1 30 30 31 32 31 32 37 35 35 31 32 30 30] Called Number (len=16) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0012127551200' ] [a1]on*CLI Sending Complete (len= 1) q931.c:2881 q931_setup: call 33090 on channel 3 enters state 1 (Call Initiated) -- Called g1/0012127551200 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 322/0x142) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 83] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 3 ] -- Processing IE 24 (cs0, Channel Identification) q931.c:3428 q931_receive: call 33090 on channel 3 enters state 3 (Outgoing call Proceeding) -- Zap/3-1 is proceeding passing it to SIP/sbf-b7c104e0 Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 322/0x142) (Terminator) Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 8 (cs0, Cause) -- Processing IE 30 (cs0, Progress Indicator) q931.c:3563 q931_receive: call 33090 on channel 3 enters state 12 (Disconnect Indication) -- Channel 0/3, span 1 got hangup request, cause 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2716 q931_release: call 33090 on channel 3 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 322/0x142) (Originator) Message type: RELEASE (77) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] -- Hungup 'Zap/3-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Hangup(SIP/sbf-b7c104e0, ) in new stack == Spawn extension (macro-outgoing, s, 3) exited non-zero on 'SIP/sbf-b7c104e0' in macro 'outgoing' == Spawn extension (macro-outgoing, s, 3) exited non-zero on 'SIP/sbf-b7c104e0' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 322/0x142) (Terminator) Message type
Re: [asterisk-users] G729 license count...
Afaik its per encode / decoder pair. In this case you will need 32 simultaneous encoders / decoders between g729 and slin, so you would need 32 licenses. Contact digium sales/support directly and you will know for sure :) Zoa Carlos Chavez wrote: I need a refresher course on how many licenses I need to buy. I have an Asterisk server that receives calls by SIP (G729) and then sends them to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if the license is per channel or per call so I do not know if I need 32 or 64 licenses for this application. Could anyone please remind me? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_swift v1.6.1 released for Asterisk 1.6
What is app_swift ? Zoa Darren Sessions wrote: Thought I'd let everyone know I've released app_swift v1.6.1 which is entirely based off of Will Orton's work he's placed in the public domain. Works great with Asterisk v1.6.0-beta7.1. In any case, can be downloaded from my site at: http://www.darrensessions.com Go easy on me, this is my first release of anything. Thanks, - Darren ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cdr_custom outout to serial port
How about a tail -f on Master.csv ? Then you will have everything realtime and you will not need a cronjob. Zoa Col Ferguson wrote: Hello again, I can copy the file out the serial port by doing this: rename Master.csv out1.csv cat out1.csv /dev/ttyS0 If I build a script to do this every 10 or 20 seconds via cron I think it will work fine, unless someone has a better way. Cheers, Col - Original Message - From: Col Ferguson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 12, 2008 5:12 PM Subject: [asterisk-users] cdr_custom outout to serial port Hello, I have a system in a motel that needs call billing data output through its serial port so the existing motel management software can collect the call billing info. Is there any easy way to redirect the data that goes into the cdr_custom/Master.csv file to go out the serial port ? The system is Asterisk 1.4.18.1 on Centos 5.1 Thanks, Col - Original Message - From: Peder @ NetworkOblivion [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 12, 2008 1:37 PM Subject: Re: [asterisk-users] NAT issue with Fortinet Firewall FYI, I have probably 10 Fortinet units with multiple SIP phones behind each and all of the phones work flawlessly. As long as the Fortinet is ver 3.0 or newer, it does NAT so that you don't need to have nat=yes on *. No pinholes or static nat or anything, it just works. As a side note, I probably have 20+ Cisco PIX's with the same setup and they work flawlessly too. I've seen a lot of people saying fixup sip breaks phones, but not that I have seen. I just let the PIX do nat and it works fine. Carlos Chavez wrote: I have a customer with a Fortinet Firewall that is having stability issues with Asterisk and SIP endpoints (PAP2T) outside his network. The first issue I see is that Asterisk sees all phones as the IP address of the Fortinet. Since the parameter localnet defines the local network and that address falls in that range, how will Asterisk treat the endpoints? I have nat=yes for all phones and canreinvite=no as well. The externip parameter is set to the outside public IP address. Still we have calls with one way audio. This is the first setup with a firewall that rewrites the IP address of the endpoint so I do not know how that is affecting the packet flow. On my other servers I can always see the public IP of the endpoint. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.22.12/1374 - Release Date: 4/11/2008 4:59 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.22.12/1374 - Release Date: 4/11/2008 4:59 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [VOIP-Users-Conference] Potential subject for Friday - Does the Asterisk community need a 3rd party commercial software ecosystem?
You need to talk about this with digium sales, i suggest you give Jim Webster a call. Without going into details, other license agreements are possible. (Several companies including mine can distribute software as a module similar to how g729 is being sold - we do that for our t.30/t.38 fax modules, lumenvox does it for voice recognition,... ). OT, where can i find the best info on this salesforce API ? Do you see any possibilities to integrate our zoiper softphone with salesforce ? (contact me off list for that) Cheers, Zoa Dean Collins wrote: Hi BJ, Further explanation about the 3rd party ecosystem question this morning Cory Andrews from VoIP supply was on the Voip-Users conference call last week. I asked the question - how much of VoIP Supply revenue is product versus applications - he said we don't sell any services such as ITSP hosted Asterisk so I replied that wasn't what I was thinking of and gave the example of Snap Dialer which is a low cost (I think I paid $20 for it) application which allows me to dial names from outlook. I then talked about some of the consulting I did for Salesforce.com and how they have built an entire ecosystem of third party applications all built by other people but utilizing the documented API's and application security etc. My comments were that although Asterisk should always remain a free open source application that developers need to eat and pay rent as well. If there was some common marketplace that developers could sell small - low cost third party applications to the Asterisk community that Digium had some type of overview/management control over who listed etc that this would deliver a stream of revenue that would encourage further application development. The question I then posed to the group was if anyone knew how Digium managed the sale and licensing of the G729 codes. And if this was an open published standard that it could be used as the basis for the Asterisk ecosystem license model. Now I know it's not perfect and can be hacked but everything can be hacked. The idea is to build apps cheap enough that it's not worth the effort of hacking. I know there were discussions in Mexuar about how we could sell (read license) a single channel of the Mexuar Corraleta application rather than the entire server license for $2000. Earlier this week I sent an original email to Digium and told Kevin was responsible for the G729 licenses so I was hoping that this Friday we could get Kevin and possibly the developers of Snap Dialer to talk about their current license models and how they implemented payment systems and also maybe the developer of FOP to discuss if this was available to him and he was able to sell a 100 licenses or something like that a month would this provide an income stream to support further development etc. Does this make sense? Does anyone have any comments or would you like to be involved with Fridays call? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: Dean Collins Sent: Wednesday, 9 April 2008 7:53 AM To: '[EMAIL PROTECTED]' Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'; Ast-biz (asterisk- [EMAIL PROTECTED]) Subject: RE: [VOIP-Users-Conference] Subject or Guest for Friday? We can talk about the third party application idea Cory and I discussed last week - sorry still haven't posted to the list yet as I've had a client in town for the last few day but should be able to do this today. If Fop, Snap Dialer, Mexuar and or any other third party developer who has written commercial asterisk applications (preferably small low value eg $20 etc) is interested in jumping on a call with me this Friday I can put together a series of questions to run the call with. Maybe we can get Kevin from Digium to explain on the call how the g729 license registration process works and turn the call into a working discussion. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:VOIP-Users- [EMAIL PROTECTED] On Behalf Of randulo Sent: Wednesday, 9 April 2008 7:27 AM To: VOIP Users Conference Subject: [VOIP-Users-Conference] Subject or Guest for Friday? Hi, Does anyone have a guest or subject up their sleeve for this week? I have neither and I'm getting ready to move house so I don't have a lot of time to pursue. Any chance of getting a phone mfr or service provider on? Anyone out there want to take a crack at this? Present your product or service or open source contributions. Incidentally, TringMe apparently has an API available and are making it very easy to build AIR and Flash (or Flex) apps. This sounds very interesting
Re: [asterisk-users] tokbox - voice and video in the browser
Looks like a standard chatbox with flash media server in between. You can't use this with asterisk unless you write a flash media server channel or a convertor of some kind. Zoa Dean Collins wrote: Interesting to note that Tokbox now has ‘clientless’ voice and video conferencing in the browser. Does anyone know how they do this? Any thoughts on how we can leverage off this for the asterisk community. http://tokbox.com/ Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 ( Sydney in-dial). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Analog modem as phone
If you cant power off the machine, look for a sip ata or channel bank. USB/ TDMoE Channel banks: xorcom.com spidermux.com/ And for ata's or sip gateways, there are zillions of brands, Zoa Ronny Forberger wrote: Thanks for that. What channel module do I have to use then ? And can you recommend a card? Are there external ones, as I can't really power off the machine. ;) Thanks, Ronny -- Message from: Zoa [EMAIL PROTECTED] Date: Mi 02 Apr 2008 21:54:31 CEST Subject: Re: [asterisk-users] Analog modem as phone I'd say, save yourself the time and the frustration, drop the idea and buy a real voice card. Zoa Ronny Forberger wrote: Hi, maybe this has been asked before but I couldnt find a proper answer on the web or list. I want to use a analog V.92 modem to make outgoing (and possibly) incoming phone call through a standard analog phone line. I found on web it's easy been done via chan_modem.so module. But this seems removed from asterisk or buggy. So my questions are can I enable chan_modem.so to be built or what other way to connect a modem to asterisk ? Thanks in advance, Ronny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Analog modem as phone
I'd say, save yourself the time and the frustration, drop the idea and buy a real voice card. Zoa Ronny Forberger wrote: Hi, maybe this has been asked before but I couldnt find a proper answer on the web or list. I want to use a analog V.92 modem to make outgoing (and possibly) incoming phone call through a standard analog phone line. I found on web it's easy been done via chan_modem.so module. But this seems removed from asterisk or buggy. So my questions are can I enable chan_modem.so to be built or what other way to connect a modem to asterisk ? Thanks in advance, Ronny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many maximum SIP Registrations can Asterisk Handle
I'd say several thousands i n normal circumstances, but i would also like to recommend not to use asterisk for large scale registrations, its better to do those on SER for example, and route the calls through asterisk for termination, voicemail, conferencing etc. It all depends on how fast the reregistration time is set on those end devices and how much the registrations will collide in the same small interval. SER doesn't handle audio so even if the registration gets a little delayed because a flood arrives, the audio won't suffer. Zoa Abid Saleem Choudhary wrote: Hi All, I am new to this community and just subscribed. We have Asterisk running in production but I could not find out in documentation as well as web that how many maximum number of registrations an Asterisk Server can support. We have it on a 1.4 GHz Processor, 2 GB RAM and 40 GB HDD IBM Server. Please suggest urgently. Thanks. Best Regards, - Abid Saleem Choudhary Team Lead VoIP Networks Comcerto Bahrain W.L.L. Direct: (973) 13301504 Mobile: (973) 36080504 Tel: (973) 13301100, Fax: (973) 13301101 MSN: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] WebSite: http//Comcerto.net P.O. Box: 311100 Manama, Kingdom of Bahrain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] recommendable softphones / X-Lite / Zoiper for amd64?
Contact me at [EMAIL PROTECTED] and ask for a beta for the 64 bit build of zoiper Cheers, Zoa martin f krafft wrote: Hi, I am on amd64 Linux and not really too happy with twinkle, linphone and ekiga. Unfortunately, X-Lite and Zoiper, even though they provide Linux versions (w00t!) have only x86 versions for download. Do you guys know of amd64 versions of those, or can you recommend other softphones that will run on amd64, or which come with source code? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Want to know Frequency and lenght of Frame
Mojo with Horan Company, LLC wrote: [EMAIL PROTECTED] wrote: I am planning to write a module to find if a Special Information was detected or not. Can anyone please help me to figure out the below fields? 1. The Frequency of a frame 2. Length of frame in milliseconds Aren't all the frames in asterisk 20ms long, no exceptions? Isn't ilbc the exception ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Sip phones for call centers
Are you looking for a softphone, a hardphone ? In the case of a softphone - give our zoiper a try (www.zoiper.com) you can easily integrate it with your contact center software through the api available at http://www.zoiper.com/downloads/Zoiper_API_Documentation.pdf and it will also enable the agents to send an receive faxes. Headset : go for plantronics or GN netcom, (stick with the usb models) In the case of a hardphone, don't go for the ultra cheap, go for aastra, polycom, snom, cisco, linksys. Pick a model based on - do you need power over ethernet ? - do you need them to have a built in switch ? - How many lines will your agents handle ? - do you need busy lamp fields - do they need to be provisioned through tftp ?: Zoa Mail list wrote: Hello Can anyone suggest sip phones with headset for use in call centers . They should be fully inter operable with Asterisk over sip . Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Fwd: OT - CEBIT next week!] - updated list
So far these people let me know there are going to be there, who else is going and wants to do some networking Joachim Vanheuverzwijn (zoachien AT securax.org) - Attractel.com - wednesday / thursday. Tan Aksoy - Telappliant - wednesday / thursday Antoine Megalla - SAND - wednesday / thursday loic didelot - wednesday / thursday Olle E. Johansson - edvina - wednesday / thursday Marius Savelberg - onecentral BV - thursday skyler??? - Digium ---BeginMessage--- Hi, Just added my name to the bottom of the list :) Joachim Vanheuverzwijn (zoachien AT securax.org) - Attractel.com - wednesday / thursday. Tan Aksoy - Telappliant - wednesday / thursday skyler??? - Digium Antoine Megalla - SAND - wednesday / thursday Antoine Megalla Products Manager SAND S.A.E. http://www.sandcti.com Tel: +20 (2) 26393117 Mob: +20 (12) 2139129---End Message--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - CEBIT next week!
Any Asterisk people going to Cebit ? Let's meet! If you go and would like to go for a drink and meet some others from the voip business, please add your name to the list below Joachim Vanheuverzwijn (zoachien AT securax.org) - Attractel.com - wednesday / thursday. Tan Aksoy - Telappliant - wednesday / thursday skyler??? - Digium [add name here:P] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for business-grade SIP Softphone
Have a look at our Zoiper (http://www.zoiper.com/oem.php) - it does all 4 items you are looking for. Zoa Dovid B wrote: Try EyeBeam. It is the paid version of X-Lite. - Original Message - *From:* Mike mailto:[EMAIL PROTECTED] *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' mailto:asterisk-users@lists.digium.com *Sent:* Friday, January 18, 2008 10:57 PM *Subject:* [asterisk-users] Looking for business-grade SIP Softphone Hi, I am looking for a good (not necessarily free) business-grade SIP Softphone that supports: 1) G729 2) Outlook contact integration (click on number to dial) 3) Remote provisioning (not a must, but a very nice to have) 4) Customizable skin (again, not a must but a nice to have) I've seen X-Lite (which has only 2 lines, not enough). The commercial version of X-Lite looks nice, but doesn't support provisioning. At the moment, it's my fallback plan. Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
T.38 will not work with the fxo card. Zoa Fernando Berretta wrote: Dear All, Are you telling me Asterisk 1.6.0b2/4 has support for t38 and rxfax etc. and will be able to receive faxes and negotiate with voip CPE's like ATA's to transmit faxes which comes from FXO cards to VoIP Devices using T38 ? it is possible to compile this version of app_fax to work with Asterisk 1.4x ? Someone has tried it ? Best Regards, Fernando Thomas Kenyon wrote: Steve Underwood wrote: I thought * was still not capable for T.38 gateway operation. Doesn't beta 4 just added T.38 termination? And, I believe it misses out some key elements of doing that properly. Note that T.38 termination is an addon, so it can't be used with, say, G.729. The only real option available at the moment is to keep one PSTN line on an ATA with an FXO port and T.38 support available and direct calls from the fax machines through to it. However, I should point out that while I believe this should be possible, I haven't actually tried it myself. The new asterisk T.38 functionality is from the Asterisk addons 1.6.0b2 version of app_fax (and a few small changes in 1.6.0b4), which I thought someone would have mentioned to you, since it does use spandsp. (Or at least the configure script checks for spandsp, I haven't actually looked at the code). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
Fernando Berretta wrote: Tzafir, I'm sorry, my question wasn't clear. Apparently Asterisk 1.6.0b2 and b4 has support for t38 because of some modifications on app_fax so the questions are: 1 - If I use Asterisk 1.6.0b2 o b4 and a fax is received from a FXO Card and this FXO port is forwarded to other ATA/Gateway is asterisk going to transmit this fax using t38 ? PSTN FAX MACHINEASTERISK(1.6.0b2) FXO CARD---t38?ATA/Gateway-FAX MACHINE No this is not going to work with the code you find in add-ons (Steve Underwood was right, my last email was a bit vague). The FAX - ASTERISK - t.38 part will not work. 2 - If the first answer is yes, if we compile app_fax with asterisk 1.4x same behavior could be achieved ? Regards, Fernando Tzafrir Cohen wrote: On Mon, Feb 25, 2008 at 05:32:24PM -0300, Fernando Berretta wrote: Dear All, Are you telling me Asterisk 1.6.0b2/4 has support for t38 and rxfax etc. and will be able to receive faxes and negotiate with voip CPE's like ATA's to transmit faxes which comes from FXO cards to VoIP Devices using T38 ? it is possible to compile this version of app_fax to work with Asterisk 1.4x ? Someone has tried it ? You have rx_fax for 1.4 . You also have fax detection in chan_zap, and thus can send faxes from the PSTN to rx_fax. Not exactly the same, but maybe this is actually what you're looking for. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
I think you are missing something. Steve means that since its in add-ons its probably a GPL addition and not compatible with the g729 licensing. A t.38 gateway involves more than origination and termination, those 2 are pretty easy and do not involve any modems, the gatewaying is the harder part. Zoa. Rob Hillis wrote: T.38 is a codec in exactly the same way that GSM or G.729 is a codec, so yes it /can/ be used at the same time as any other codec - just that only /one/ codec will be used at a time. What often happens is that the call will initially be established with a codec such as G.729 or G.711a, but once fax tones are detected the call will change codecs to T.38. According to the release notes for 1.6.0-b4... - 11873, Added core API changes to handle T.38 origination and termination (The version of app_fax in Asterisk-addons now supports this.) This should be all that is necessary to run a T.38 gateway. Steve Underwood wrote: Rob Hillis wrote: Not unless you're running CallWeaver or Asterisk 1.6.0-beta4. Asterisk has had passthrough support for T.38 for a while (somewhere in 1.4 it became available IIRC) but is currently completely incapable of terminating or encoding a fax call to T.38. I thought * was still not capable for T.38 gateway operation. Doesn't beta 4 just added T.38 termination? And, I believe it misses out some key elements of doing that properly. Note that T.38 termination is an addon, so it can't be used with, say, G.729. The only real option available at the moment is to keep one PSTN line on an ATA with an FXO port and T.38 support available and direct calls from the fax machines through to it. However, I should point out that while I believe this should be possible, I haven't actually tried it myself. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk G722
Asterisk does not support that yet. Zoa rachid wrote: Hello, I have some problems to use G722, when my client sent an invite request to asterisk using G722/16000 codec asterisk respond with G722/8000 codec. I dont know exactly if Asterisk supports G722/16000 codec?? If yes how can I activate It?? Thanks. Rachid. Below wireshak trace: My Invite: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.10.12:5060;branch=z9hG4bK2600322761 From: Manager sip:[EMAIL PROTECTED];tag=3871604470 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 21 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Authorization: Digest username=Manager, realm=asterisk, nonce=1c8c3fd9, uri=sip:[EMAIL PROTECTED], response=5d32f87fa423cd2f1bf9aefb8cf920b6, algorithm=MD5 Max-Forwards: 70 User-Agent: wengo/v1/wengophoneng/wengo/rev54/trunk/ Expires: 120 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 365 v=0 o=userX 2001 2001 IN IP4 192.168.10.12 s=A call c=IN IP4 192.168.10.12 t=1202402970 1202406570 m=audio 10600 RTP/AVP 0 8 109 3 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:109 G722/16000/1 a=rtpmap:3 GSM/8000/1 a=rtpmap:101 telephone-event/8000/1 m=video 10702 RTP/AVP 34 31 a=rtpmap:34 H263/9/1 a=rtpmap:31 H261/9/1 Asterisk response: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.12:5060;branch=z9hG4bK2600322761;received=77.203.231.140 From: Manager sip:[EMAIL PROTECTED];tag=3871604470 To: sip:[EMAIL PROTECTED];tag=as5c1447b6 Call-ID: [EMAIL PROTECTED] CSeq: 21 INVITE User-Agent: Asterisk PBX SVN-trunk-r102777 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: sip:[EMAIL PROTECTED]:5060 Content-Type: application/sdp Content-Length: 397 v=0 o=root 1999706631 1999706631 IN IP4 91.121.31.80 s=Asterisk PBX SVN-trunk-r102777 c=IN IP4 91.121.31.80 b=CT:384 t=0 0 m=audio 18950 RTP/AVP 109 0 8 101 a=rtpmap:109 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 18692 RTP/AVP 34 a=rtpmap:34 H263/9 a=sendrecv ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Softphones and Citrix ?
I'm working for zoiper.com and i'm willing to help out with ours when needed. Zoa d4rk f1br wrote: Anyone aware of any SIP softphones that might virtualize well with Citrix presentation server? I suspect I know the answer already as I have been researching softphones that work with Cisco CallManager that can be virtualized if you will with Citrix and have come to learn that its not something that seems to be doable at this time. I have to assume that the issues affecting the virtualization of cisco softphones with Citrix will come into play with SIP softphones as well. Seems that the two biggest issues revolve around wrapping the UDP stream up with the ICA protocol, and possibly issues with the various mics and speakers and having to interface with them I think. However, I am also a firm believer that anything is possible, practical well not usually, and it may just be the time has not come yet for this. There is a good article about this over at: http://www.brianmadden.com/content/article/How-should-Citrix-integrate-VoIP-with-Presentation-Server Any thoughts, comments or insight into this and your experiences around any of this is appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maybe a little OT---USB Handset
Iirc, there used to be such an adaptor in the digium dev kit years ago. Maybe somebody here remembers what it was exactly ? Zoa John Millican wrote: Hello All, This may be a little OT for the list but it seems to be to be the place to get the best answer. I have looked at the many Skype/Yahoo phones out there and none seem to be what I am looking for. I have a need for a USB handset that I can use with an Asterisk server. This will be on the server itself and an extension on that server, most likely the only extension. The handset needs the ability to generate its own on hook/off hook and DTMF so that I would not need to load a soft phone. I will eventually be needing many of these so if the set up requires a lot of hacking to the phone it may not be feasible. Having said that any suggestions will be appreciated. I know I could use an ATA and a PSTN Phone from wally world, but this will not fit the project or the need. Thanks, JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maybe a little OT---USB Handset
Gordon Henderson wrote:
On Sun, 27 Jan 2008, John Millican wrote:
Tzafrir Cohen wrote:
On Sun, Jan 27, 2008 at 11:27:16AM -0500, John Millican wrote:
Hello All,
This may be a little OT for the list but it seems to be to be the place
to get the best answer. I have looked at the many Skype/Yahoo phones out
there and none seem to be what I am looking for.
I have a need for a USB handset that I can use with an Asterisk server.
A USB handset is basically a sound device (and not a great one, usually)
along with a small keyboard. Linux will usually easily identify the
sound device and you can use the phone as chan_{oss,alsa,console}.
Using the keyboard in it may be trickier.
Do any of the above support cancelling acustic echo? Is it actually
needed in this case?
Tzafrir,
Thanks for the reply. Acusitic echo cancel may not be needed as this
will not be used in a noisy work place, only in possibly quieter home
environments. There will also be no need for speaker phone operation.
Enabling the keypad is definitely the tricky part. I am trying to avoid
loading a soft phone since I don't want to have to instruct the users on
how to use one (mostly NON-technical types). If the set looks and feels
like a phone they will be OK on their own. I guess I may have to go
with a decent, hopefully inexpensive, basic IP desk phone.
I had a little success with a cheap USB 'phone' (From Tesco in the UK)
which was a Yealink device. Linux has a driver for the keypad on it which
makes it work just like a regular keyboard (limited number of keys,
obviously!), but the issue is still that you'd need a program of some
sorts to take the keypad input and translate it to an asterisk console
command dial, if using it as a console phone.
I did use it successfully some time back with idefisk, although idefisk
didn't have a keyboard equivalent of 'hang up' at the time (zoiper might
have now though). The down-side was that you needed to put the mouse over
the idefisk application so it had keyboard input focus )-:
In the mean time, idefisk (now called zoiper) has support for the
yealink chipset phones on windows, (no need to focus), but he is looking
for a solution without softphone.
I'm unsure if they have an SDK for linux, chances are bigger that the
intel HID standard based chipsets might have linux drivers that support
the hangup buttons etc too. (as those dont exist on a normal keyboard)
Oh for a command-line IAX client, but it's something I just don't have
time to put together myself.
Gordon
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Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
You can find it here: http://www.zoiper.com/downloads/free/linux/zoiper201-linux.tar.gz Note that the linux version does not support TLS and SRTP yet. * Instructions: * 1) Download zoiper201-linux.tar.gz 2) Extract Zoiper. If you don't use a GUI application for archive processing, here is the command line: tar zxf zoiper201-linux.tar.gz ./zoiper 3) Start Zoiper. *ZoIPer depends on ALSA library, so it* **must** *be installed! * Zoa Robert Moskowitz wrote: zoa wrote: Have you tried our Zoiper softphone yet (www.zoiper.com) - new version scheduled for in a couple of days ? If so, can you send me any remarks of list so that we can keep those things in mind for future versions ? Do you know where I can get it as an rpm to install on Centos 5 with Gnome? I do not have the time resources to do compiles. I am really a security protocol researcher and would be very interested in seeing what you have done for SIP TLS and SRTP. But for the later, I am all Linux. The one XP system is a corp box that I cannot add any software too. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
Thank you very much for the feedback, i definately like the suggestions and i will do my best to get this on the roadmap. (which should be pretty easy as i actually kind of make the roadmap :p), so expect in done in one of the following releases. The things to turn it into a callcenter application are already there, not with a TCP port, but you could use it with command line options (even if the phone is already running) or through a com object. Documentation can be found here: http://www.zoiper.com/downloads/Zoiper_API_Documentation.pdf Examples can be found here : http://www.zoiper.com/biz3.php I have an example for jscript somewhere tool, contact me offlist if you want it. Let me know offlist if you need any biz licenses to try it out, i;d be happy to provide you with them. Zoa. Christian Ejlertsen wrote: Ok good piece software easy on the eyes as they say and I have to say this before I start listing a lot of things that I would love to see, for it to be usable as a good high performance phone. Working with industrial pc switchboards and soft phones of various vendors for some years now, and it all boils down to. How much functionality you can boil into the keyboard. No mouse action should be needed to search a number add an F-key for it. No mouse action should be needed to dial or transfer a number. No mouse action should be needed unless absolutely unavoidable. A_PARTY = caller B_PARTY = operator / called person C_PARTY = number to transferred to STATES: Example to keep it within the numeric key-pad when you receive a call and transfer it. STEP 1 A call is presented. LINE_STATE: Ringing TRANSFER_STATE: inactive TALKING_TO_STATE: inactive STEP 2 Press numeric enter to pick up call. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: inactive TALKING_TO_STATE: A_PARTY STEP 3 Transfer the call Scenario 1: Search out the number in the phonenbook by pressing ex: F10, while talking to the caller, the phone book appears search by name, number or whatever is available and mark the number with arrow keys and dial with NUM-enter. Scenario 2 Press enter a new dial box appears. Type in the number to call. Press enter. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CALLING_C_PARTY TALKING_TO_STATE: DIALBACKTONE STEP 4 The person transferring the call can now make a choice either to do a attended transfer or a blind transfer. Scenario Blind transfer: Simply pressing NUM-enter should do a blind transfer, and the call handling is done and all states are reset, C_PARTY becomes the B_PARTY and so on. The phone is ready for a new call. LINE_STATE: inactive TRANSFER_STATE: inactive TALKING_TO_STATE: inactive Scenario: Attended transfer: The person transferring the call can talk to the C_PARTY LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY Should the operator wish for switching back do the previous call that currently placed on hold it could be done by pressing the NUM+ key placing the C_PARTY on hold and reconnecting the A_PARTY LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: A_PARTY Switch back by NUM+ LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY Connect the call by NUM-enter at any point talking to either the A_PARTY or C_PARTY. The call handling is done and all states are reset, C_PARTY becomes the B_PARTY and so on. The phone is ready for a new call. LINE_STATE: inactive TRANSFER_STATE: inactive TALKING_TO_STATE: inactive Scenario: disconnect the party you are talking to Press NUM- If the states are as follows. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY The C_PARTY would be disconnected and the states would go to. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: inactive TALKING_TO_STATE: A_PARTY And the here we go again with a new transfer or a goodbye and hang up with NUM-. Some side notes: The calling transfer functions are already in the phone alle that needs to be done is associate the functions to the states and numeric keys. The features could be activated by putting the phone in operator mode, if this was the case you could turn of the DTMF and just start typing the new number and hit NUM-enter twice to transfer the call fast. 1 enter to dial number the other to transfer. DTMF could be turned of since the operator rarely calls any ivr, that needs a DTMF response, if so you could leave dtmf open on the QWERTY number keys HEX 30 31 33 34 so on. A tcp port on the phone that allowed for picking up calls and hanging up calls, and perhaps being able to read the number status would make is possible for people write some very nice callcenter agent software for this phone, without having to worry about
Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
Hello, Have you tried our Zoiper softphone yet (www.zoiper.com) - new version scheduled for in a couple of days ? If so, can you send me any remarks of list so that we can keep those things in mind for future versions ? Greetings, Joachim Philipp Kempgen wrote: Andre Herrlich wrote: any one advise a good, strong and free softphone that can work with SIP or/and IAX lines and supports attended transfer ? IMHO there are no good softphones - at least not for Mac OS X and I think that is true for Linux as well. They're either not stable or their interface is unusable. Regards, Philipp Kempgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS damaged at TDM22B
I'd say check with Digium, maybe it's supposed to not break (i personally don't think it would break it, i'd have noticed it already :) if you plug it to the wrong thing and you will get a replacement for free. Zoa bilal ghayyad wrote: Hi All; If one of my FXS port damaged at TDM22B because we connected the Telephone Line cable to the FXS port while it should be connected to the FXO port, then can I order S110M FXS Module and fix it instead of the damaged FXS? (This if we assume my problem that really the FXS port damaged). Rregards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replacing Skype with Asterisk Peering Servers - and Security
Philipp Kempgen wrote: Siju George wrote: What are the security ramifications of peering two Asterisk servers on remote locations and sending the VOIP traffice through the internet using IAX2 ? Can this traffic be sniffed and the Voice be captured and heard by any third party? Yes. If so is ther a way to prevent it? IPSec. Or the built in encryption in iax2 Is there a way in asterisk to do that No. Yes :) or should i be using some VPN technique like IPSEC between the two end points to encrypt VOIP traffic? Yes. Regards, Philipp Kempgen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
There are many, (i'm one of the people working for zoiper): Look at the iaxclient homepage, There are iaxcomm, loudhush, kiax, mediax , diax and many more, (you could also easily make your own). Cheers, Zoa Vincent wrote: On Fri, 30 Nov 2007 09:52:59 +0100, randulo [EMAIL PROTECTED] wrote: I have used SIP and IAX for about three years now. We don't do a lot of traffic, but I haven't really seen a difference in quality or dropped calls. Sorry for jumping in, but besides ZoIPer/Idefisk, are there IAX-capable softphones for Windows? Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
IAX had some stability issues in the past, the recent releases have a lot of iax2 fixes and should no longer have those issues. Zoa Steve Totaro wrote: randulo wrote: Hi, We all know what the principal advantage of IAX is, doing it all on a single port, right? But now and again I hear complaints about it. What specific griefs have you had with IAX and has it stopped you from using it entirely? Under what conditions have you had problems? I have used SIP and IAX for about three years now. We don't do a lot of traffic, but I haven't really seen a difference in quality or dropped calls. What have others on the list experienced? tia randy I am not sure why, what versions, under what conditions, but audio cutting out has been seen many times. Simply switching to SIP has solved these issues. I think trunking (one of the main selling points of IAX due to less overhead) may be a common denominator. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
The jitter buffer is actually the same. Zoa Dr. Michael J. Chudobiak wrote: randulo wrote: On Nov 30, 2007 1:40 PM, Steve Totaro [EMAIL PROTECTED] wrote: solved these issues. I think trunking (one of the main selling points of IAX due to less overhead) may be a common denominator. That does tend to explain why I've never experienced (or at least noticed) problems. I never trunk which is, as you state, another important advantage of IAX. I find the audio quality to be better on IAX - better jitter buffer! I don't trunk. - Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] trunk working under windows!
Cool, i'll help out a bit with the windows port, i will start right away with a new project on asteriskguru making nightly executable builds and installers - will post the links in -users when i'm done. Well done luigi, this will make it a lot easier for a lot of non linux guys to make their first steps in the asterisk world Crossposted to -users. Zoa Luigi Rizzo wrote: As a result of the commit below, now trunk can be built and run under Windows/cygwin, including the building of modules. Haven't checked yet the functionality - some modules surely cause ill side effects or deadlocks on exit, so you need to play a bit with modules.conf . If you want to play with a very minimal version the following does something: ; -- modules.conf [modules] autoload=no load = res_monitor.so load = res_features.so load = chan_sip.so Unfortunately, loading other modules is a bit critical and depending on the order or the timing you get crashes etc. To build trunk under windows/cygwin you need at least the following pieces: bash binutils curl gcc libiconv minires (resolver library) libdb4.3(probably db4.2 too) and a bit of patience because the build takes around 15min or more. cheers luigi On Tue, Nov 20, 2007 at 04:12:11PM -, SVN commits to the Asterisk project wrote: Author: rizzo Date: Tue Nov 20 10:12:10 2007 New Revision: 89454 URL: http://svn.digium.com/view/asterisk?view=revrev=89454 Log: Fix building of modules under cygwin. After this commit we can actually load modules under windows, and we can start debugging more interesting problems related to the load order and functionality of modules. Modified: trunk/Makefile.moddir_rules trunk/apps/Makefile trunk/channels/Makefile trunk/pbx/Makefile trunk/res/Makefile Modified: trunk/Makefile.moddir_rules URL: http://svn.digium.com/view/asterisk/trunk/Makefile.moddir_rules?view=diffrev=89454r1=89453r2=89454 == --- trunk/Makefile.moddir_rules (original) +++ trunk/Makefile.moddir_rules Tue Nov 20 10:12:10 2007 @@ -66,9 +66,8 @@ ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) # linker options and extra libraries for cygwin SOLINK=-Wl,[EMAIL PROTECTED] -shared - LIBS+=-L../main -lasterisk -L../res + LIBS+=-L../main -lasterisk -L../res $([EMAIL PROTECTED]) # additional libraries in res/ - LIBS_RES:= -lres_monitor -lres_adsi -lres_features endif endif Modified: trunk/apps/Makefile URL: http://svn.digium.com/view/asterisk/trunk/apps/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/apps/Makefile (original) +++ trunk/apps/Makefile Tue Nov 20 10:12:10 2007 @@ -39,3 +39,9 @@ all: _all include $(ASTTOPDIR)/Makefile.moddir_rules + +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + LIBS+= -lres_features.so -lres_ael_share.so -lres_monitor.so -lres_speech.so + LIBS+= -lres_smdi.so +endif + Modified: trunk/channels/Makefile URL: http://svn.digium.com/view/asterisk/trunk/channels/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/channels/Makefile (original) +++ trunk/channels/Makefile Tue Nov 20 10:12:10 2007 @@ -64,6 +64,10 @@ include $(ASTTOPDIR)/Makefile.moddir_rules +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + LIBS+= -lres_monitor.so -lres_features.so +endif + clean:: rm -f gentone $(MAKE) -C misdn clean Modified: trunk/pbx/Makefile URL: http://svn.digium.com/view/asterisk/trunk/pbx/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/pbx/Makefile (original) +++ trunk/pbx/Makefile Tue Nov 20 10:12:10 2007 @@ -25,6 +25,10 @@ include $(ASTTOPDIR)/Makefile.moddir_rules +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + LIBS+= -lres_ael_share.so -lres_monitor.so +endif + clean:: rm -f ael/*.o Modified: trunk/res/Makefile URL: http://svn.digium.com/view/asterisk/trunk/res/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/res/Makefile (original) +++ trunk/res/Makefile Tue Nov 20 10:12:10 2007 @@ -25,6 +25,13 @@ include $(ASTTOPDIR)/Makefile.moddir_rules +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + # order-only dependency: build res_monitor before res_features + res_features.so: | res_monitor.so + # res_features uses some functions from res_monitor + res_features.so_LIBS:= -lres_monitor.so +endif + ael/ael_lex.o: ael/ael_lex.c ../include/asterisk/ael_structs.h ael/ael.tab.h ael/ael_lex.o: ASTCFLAGS+=-I. -Iael
Re: [asterisk-users] Kirk IP600/3 Wireless Server SIP config
I would stay with DECT, the battery in WIFI devices only lasts a couple of hours. (Unless you want to take the phone with you and use it on public hotspots etc) Zoa Luis Antonio Prata Barbosa wrote: Some days ago, I was looking for some mobility solutions... My conclusion is Wi-Fi phones are growing up fast and I think it's only a time question they became a standart for mobility in pbx, as well as pure IP telephony. Even manufactures of DECT systems are preparing their products line to Wi-fi. Of course, DECT is a mature technology but If you could spend some time in tests and adjusts I suggest you to think in wi-fi as an option. Any opinions ??? Thanks. Luis A P Barbosa 2007/10/25, Remco Barendse [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi list! Is anyone using the Kirk IP600/3 with SIP firmware connected to Asterisk? Any experiences / caveats? If anyone would be willing to share the dump of their IP600 config file, i would really appreciate it. Is there anything special i should put in my asterisk config? Thanks !!! Remco ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum manager connections
Use the astmanproxy and move the load elsewhere. (If you just want to passively listen to messages, your box is about 100 times faster than you need :) Zoa Roberto wrote: Have anyone maided like 200 simultaneous connections to asterisk AMI (manager). ?? How many connections can I made without problems ? I’m using a Quad core DELL poweredge machine. *Roberto Fernandes Lopes* *Diretor Presidente*** *Dialtech Telecom. e Sistemas Ltda.*** *(11) 6986-8886*** No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.14.8/1063 - Release Date: 11/10/2007 09:11 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Having problems posting to the list
Same here lenz wrote: Mee too, a lot of the messages I'm sending seem to disappear. l. In data Tue, 02 Oct 2007 22:38:26 +0200, robert boardman [EMAIL PROTECTED] ha scritto: Hi All I'm having problems posting to this list, no bounces the mails just dont show any advice how to get the postings through is there filtering? robb _ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Java Softphone?
Mexuar is the best known one i think, they showed me a demo on astridevcon, seemed to work ok. Zoa Matthew Rubenstein wrote: Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center SoftPhone with Auto Answer
Zoiper can do it when you use the provisioning, contact me offlist on [EMAIL PROTECTED] Zoa Joao Pereira wrote: I don't think so, because in paging/intercom, the phones must support Auto Answer. The link you sent says: SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. I'm using X-Lite, and in X-Lite I can't force the users to answer the call. The users can put Auto Answer = Off. Also, the response from Counterpath was weird, as they said they're engineering team cannot remove the Auto Answer option: To have the auto-answer permanently on in the context that you wish to have is a feature that our engineering team cannot hard code into the phone. It can be turned on and off in the menu So, if someone knows a nice softphone for an Asterisk Call Center, please advice me. Thanks Regards Joao Pereira Ed Pastore wrote: On Sep 17, 2007, at 11:11 AM, Joao Pereira wrote: But still, the user can choose not to answer the phone. I want to force the users to accept the calls. Wouldn't that be the same as paging/intercom, then? http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
Gordon Henderson wrote: On Sun, 3 Jun 2007, Andrew Kohlsmith wrote: On Sunday 03 June 2007 4:30 pm, Alex Crow wrote: No frills, specs look good, price seems excellent! http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519 That's a terrible phone. I've tried them. the screen is pretty much useless, the buttons are *TINY*, the battery life horrible, and the ringtones gimmicky. I have to disagree on at least one point here - Battery life. I don't think 3 or 4 days standby and several hours of talk time makes for a horrible battery life. The F1000G has other faillings, but battery life isn't one of them! If you compare this battery life to a decent DECT phone, it's still miserable. I'm used to these dect phones : http://www.bang-olufsen.com/UserFiles/File/Products/Technical%20Specifications/BeoCom6000_en.pdf I haven't tried WEP or WPA on these things, but the phones I've gotten rid of long ago due to their problems. Mine works with both WEP and WPA. It just occasionally won't re-attach to an access point (iwthout rebooting) and won't roam between access points at all. I think for the cost (and you can get them cheaper than from Scan!) they're not bad, but still in the experimental/toy category than something I'd deploy to clients. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
We have it (in belgium) http://www.voipsolutions.be/phones/dect-sip-phones/siemens-gigaset-sl75-wlan-voip-phone.html I still think DECT is better though :) Zoa Alex Crow wrote: Alban, Thanks! Where on earth did you source this? I can't seen to find hide nor hair of it here in the UK :( Alex On Mon, 2007-06-04 at 16:01 +0200, Alban wrote: Hi, I've tested several wifi phone (UtStarcom, Hitachi 3000 and 5000, and one Siemens). The Siemens is the best one, for a really cheaper price than hitachi. And was the only one which roams well between AP (same SSID, same channel) with WPA. Battery is still a problem, especially if the coverture is not very good everywhere. But that was the best one I could test... The reference is : Gigaset SL75 WLAN. Hope it helps Alban Le Lundi 4 Juin 2007 14:53, Bryan Laird a écrit : On Jun 4, 2007, at 8:31 AM, Andrew Kohlsmith wrote: On Monday 04 June 2007 8:24 am, Bryan Laird wrote: - Physically the phone feels very light and cheap, that if you were to drop it that it might not survive very many of them. The buttons feel more like a toy than anything else but once you get beyond that it works. How are they for big hands? I'll have to do some checking around to see if I can find a rubber case for it or something, it's all concrete floors here. Considering I too have the sausage finger problem... the buttons are incredibly similar to what you find on the Nokia candy bar style phones. - Address book storage is ok the interface from the phone is fairly standard for what you would see in a cell phone and adding entries isn't really all that horrid of a task. You can also add entries via the web interface which does make for an easier way to add several entries but the lack of anything resembling a 'sync' function could be considered bothersome. Bugger. Last thing, one neat thing about the wip300 if you are adventurous is the fact that the firmware is under GPL... so if you really felt like it you could probably change the behavior of the phone. This I was not aware of. I will certainly evaluate this phone and it's bigger brother. Anyway sorry for the long message but I felt like chiming in on this. All in all I don't think it's a horrible phone I do however think it's over priced for what it is but not enough demand on this type of device is always going to keep the price up in the air. Your message is *exactly* the kind of reply I was hoping to get. Thank you so much for taking the time to write such a long response. I truly appreciate it. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk High-Capacity Stability
Several people do use it for handling 50k minutes a day. (I'm one of them). Yes, you need to know what you are doing, and have a nice design, but it is possible.Our code is only slightly altered. (mainly for billing purposes). Zoa Daryl Jurbala wrote: On May 12, 2007, at 4:11 PM, Atlanticnynex wrote: Thanks Alex, some great ideas. I think, however, I'm leaning towards Asterisk at this point- since I have quite a bit of experience there, and very little with SER. At this point, I'm wondering from a dimensioning standpoint, what kind of capacity my machine will have (Dual Core Xeon 2.4GHz 4GB RAM). As I said, I don't plan to do any transcoding. I read the voip-info page on dimensioning and it seems theres some mixed feelings about Asterisk in high-capacity environments. I guess I'm looking for input as to whether Asterisk could handle roughly one DS3's worth of calls (672 calls) just doing the LCR (I've seen some pre-built LCR apps, looks like they all do on-the-fly MySQL queries- I think I'd write my own AGI that would use a cache). With my hardware, could Asterisk run stable for this amount of traffic? What stability issues does Asterisk have at this scale? Simply put, NO. I am on a project now where a client had an OpenSER box acting as an SBC and registrar passing traffic to several asterisk boxes which are doing LCR lookups on the fly as well as writing custom CDRs all through PHP AGI scripts to a Postgres DB. The Asterisk boxes do not scale, and randomly start swallowing calls or, more often, restart the process (safe_asterisk is handling this). There is some light IVR type usage for reporting account balances and the like. With anything more than 80 or 90 calls on the box, the IVR prompts start to break up. Ben through replacing hardware, more memory, different Asterisk builds, etc. I've had an open issue with Digium support on this for at least a couple of weeks, and the best advice so far was try using the SVN build. That makes things better, but it's still not anywhere close to fixed.. It's absolutely incredible that Asterisk works at all for some of the situations its been put in - major kudos to the developers. But I don't think using it for what you're talking about is a long-term business strategy. When the highlight of the 1.6 release is bridging channels, you know high volume sip to sip usage in a carrier class call routing environment is NOT what development is focused on. And that's fine. If you use a wrench to do the job of a screwdriver, you shouldn't complain when you bust your knuckles That being said, I don't meant to trash Asterisk at all. It's a fantastic feature server, and a great PBX, both of which things I use it for very successfully. I just don't think it's ready to handle 50k plus minutes a day SIP to SIP with LCR and billing data, no matter what you do with it. I'm 100% positive there are people out there doing it successfully, but those are the exception, not the rule. And I doubt they are running unmodified code. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone that supports central provisioning?
James, How about we provide you with a free copy to see if it suits your needs ? :) It's not exactly rocket science (we already store everything in an XML file for ages anyway) so i don't expect too much glitches with it. Email [EMAIL PROTECTED] mentioning this email if you want to give it a try. Cheers, Zoa James FitzGibbon wrote: Has anyone found a softphone that supports pulling it's configuration from a central server via TFTP/FTP/HTTP, much like hard desk phones use? I'm looking for something for a call center that I can provision from a central location by generating config files. If the phone has soft keys (yes, I know they're all soft - but you know what I mean; programmable buttons whose function comes from the provisioning system), even better. I know idefisk Biz says they'll do this, but it's not in the release candidate and will make it's debut in the final version, which is a little too much early adoption for my liking. Other than that, I'm back at X-Lite/eyeBeam, which stores it's configs in binary files, preventing me from I'm open to SIP/IAX, so long as I don't have to jump through hoops to get it talking to *. Thanks for any experience you can share. -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone that supports central provisioning?
For things running inside the browser, i think java is a reasonable choice. Yes you could do it with active-x too, but it won't work on all OS'es. I hate java, probably for the same reasons you do, but in same cases its the best option. Zoa Dean Collins wrote: Lol, yep you missed something but do you really want to be taught something you already think you know? Regards, Dean Collins [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph Call Button http://click.mexuar.com/webuser/click/7/userurl/Cognation ** www.Mexuar.com ** http://www.mexuar.com/ Want to voice enable your website? Use Corraleta to reach your customers in 10 seconds or less. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Saturday, 21 April 2007 8:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphone that supports central provisioning? What's your objection to a softphone in java ? Java is slow and the interface is always ugly and doesn't fit into the window manager etc. you are used to. :-P I never understood why I would use Java to write software when I could use C(++) or when a script language would do. The simple fact that people have 2 or 3 GHz doesn't mean that I have to burn them for nothing. The only point may be portability. Do I miss something? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Astertest - Asterisk stressing tool
Hello, As i was involved in the development i can say that it is for the moment abandoned by the developers, we might get back to it later but are first focussing on some other projects. (Idefisk being the main one) I don't think it will work with any recent version of asterisk and even if it did it would be a pita to set up). I think you would be better of writing a script that generates call files. Zoa Sebastien Cruaux wrote: Hi, Did someone ever managed to make Astertest (http://www.asteriskguru.com/tutorials/astertest.html) work ? I followed all the instructions of this tutorial and corrected the mistakes pointed by the users but it still doesn't work. I can compile it and load app_securax_cpuinfo.so. When trying to load app_securax_serverload.so I have this error : WARNING[31477] : loader.c: 325 __load_resource: /usr/lib/asterisk/modules/app_securax_serverload.so: undefined symbol: scx_load_global_config_value I tried with Asterisk 1.4.2 and 1.2.17 but it's more or less the same result. However, even people who managed to load both modules report that they can't make the origination server make any call. Looks like this software could be outdated and that it is no longer followed by its developers :( Please help me I'm quite stuck and I really need to stress test my Asterisk server. Thanks for your help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stun
Joe Acquisto wrote: . . . http://sourceforge.net/projects/stun/ Which is linked from: http://www.vovida.org/applications/downloads/stun/ That's what I'm running. Gordon Thanks. Looking there, why would I need a stun client if the device/softdevice already has STUN support? All I should need is the linux daemon thing-let, correct? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The linux daemon is also downloadable there i think ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off Topic: Open Source USB Softphone
You could use an rebranded (OEM) idefisk - does sip and IAX and uses XML for the config files, not the registry - making it possible to use it on a usb stick. More info : http://www.asteriskguru.com/idefisk/oem/ (But its not open source, nor free). Joachim Mike Lynchfield wrote: sip would be the required one as iax..well.. also openwengo wont work.. to much overhead .. broswrer needed.. ie component + flash + css+js etc.. not viable.. so im also asking anyone have one ? since ihave a supply of around 2000 of the vonage usb stick OEM.. On 3/30/07, *Michael Van Donselaar* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Which USB Phone? I have written custom versions of iaxcomm for various people, and have a version that works with the Yealink phone. On Thu, 29 Mar 2007 11:33:07 -0300, Luis Claudio Santos [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX best practices
Some tricks: If you have a high latency link, watch out i've experienced problems with it in the past. (high latency = 300ms) Stay away from trunking unless you have a lot of time to spend, if you do use trunks, do not use a jitter buffer, make sure it works in both directions and don't make it too big (although that might be fixed in more recent asterisk versions, there was a patch included for it). If you use a VPN, make sure it is UDP based, and check it for low latency. Make sure both asterisk's are on a public ip. (Although you should be able to connect to an asterisk B registered to Asterisk A, it - at least used to - not work very well in production). If you do a lot of simultaneous calls, make sure your vpn servers can handle the load. Zoa www.asteriskguru.com Michelle Dupuis wrote: You will likely have latency issues - causing choppiness. Start with a traceroute to validate latency. Michelle Dupuis Technical Support Specialist Generation Software - Linux and Asterisk solutions and support Visit us at www.generationd.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Friday, March 02, 2007 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] IAX best practices Thanks Steve! What are usually the best approaches in troubleshooting the audio quality issues and QoS related stuff when putting two Asterisk boxes together via IAX? Have you ever tried connecting Asterisk boxes in the same VPN (but still in different countries)? Regards, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, March 01, 2007 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX best practices Asterisk wrote: Hi guys, I am planning to connect two Asterisk boxes that are currently running in two different countries, using IAX. I was wondering if anyone could provide me with some links or suggestion regarding best practices in connecting two Asterisk in such way. I guess many of you have already tried this, and already have some know-how (what I should be careful about, what to avoid, etc...)? Regards, Alex Bandwidth and latency. IAX2 is remarakably good at traversing NAT and even double NATs. It should just work. The issues that I ran into are low bandwidth and latency. Not much you can do about latency besides getting a better route and putting QoS on your equipment and hoping that your provider either observes your tagging or is not very latent to begin with. The other is bandwidth which I found SPEEX works wonders (but adds to latency). In my experience, bandwidth issues result in choppy audio and latency results in delays which cause people to talk on top of each other and can be extremely annoying. Try pinging a router or device at the remote side to get an idea of how latent your connection will be. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX using T38
So does asterisk (Albeit with a commercial package) http://www.attractel.com/t38.html Lee Howard wrote: Matt Riddell [NZ] wrote: Does OpenPBX do a T.38 gateway then? Yes, it does. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple simultaneous calls
I wouldn't do that with softphones, unless the softphones are designed to do this. The delay will vary depending on the audio card, OS, and drivers. (And the phones might not all answer at the same time, but if you use music on hold or so to play that should not be a problem). [EMAIL PROTECTED] wrote: Hi Guys, I am a novice of Asterisk and I need some experts help to understand what I can get out of it. I need to make multiple calls (let say 50) at once to autoanswering softphones on a LAN and send all of them the same message that they will repeat with loudspeakers in the same environment. I am a little concerned about synchronization of the phones and moreover it is not much clear to me if I have to open 50 connections and send 50 times the same packets or if can use in some way the multicast. Is there anybody that may give me some idea. Thanks in advance, Stefano :. CONFIDENZIALE: Questo messaggio e gli eventuali allegati sono confidenziali e riservati. Se vi è stato recapitato per errore e non siete fra i destinatari elencati, siete pregati di darne immediatamente avviso al mittente. Le informazioni contenute non devono essere mostrate ad altri, né utilizzate, memorizzate o copiate in qualsiasi forma. CONFIDENTIALITY : This e-mail and any attachments are confidential and may be privileged. If you are not a named recipient, please notify the sender immediately and do not disclose the contents to another person, use it for any purpose or store or copy the information in any medium. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX call limit
Hello, Send an email to [EMAIL PROTECTED] i think we the upcoming version has some fix for this iirc Zoa Nir Simionovich wrote: Hi Philipp, Thanks for the tip, but that is not what I initially meant. I'm using IDEfisk, and I would like it when a call comes Into IDEfisk to generate a BUSY signal, if there is already a call in the client. Any ideas ? Nir S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Thursday, January 18, 2007 12:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX call limit Nir Simionovich wrote: Stupid and silly question - is there a way to limit the number of concurrent calls an IAX client can make? something in the similar sense of incominglimit and outgoing limit on SIP? It can be done in the dial plan: http://www.voip-info.org/wiki/view/Asterisk+Queue+with+limited+calls+per+IAX+agent Best regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ebay Unwired Buyer, Using Asterisk?
Allison is not exclusively working for asterisk, she also does other recordings. Zao Steve Totaro wrote: Just got a call from Ebay's unwired buyer and The Voice is Allison Smith. Adoption is wide but who is willing to give away their competitive edge (although ebay doesn't really have any real competition). Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Problems with Digium TE410
Return the card and ask for a new one. (i have seen this problem before with a broken 411, a new card fixed it). Zoa. Tony Mountifield wrote: In article [EMAIL PROTECTED], Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello List Just want to check if anybody else is having this problem. Every time the PRI connections are disconnected, the card freezes, and I have to reload the driver, to make it work again. We are very seriously considering switching to Sangoma at this moment, due to this and other problems, but I want to know if there is a solution, and to make sure it isn't asterisk that's freezing the cards, and that the problem would re-appear on the sangoma cards. Could be a faulty card. I've used several TE405 and TE410 cards and never observed this problem. Cheers Tony ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk timing
You need a timing device on both ends. Zoa Vicky wrote: If the other server doesnt have any hardware device that can act as timer. then just compile zaptel and modprobe ztdummy .. This kernel module should act as timing source i think . ( it works with meetme ) . On 16/01/07, *Andy Hester* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have read that an IAX trunk requires a timing device. What wasn't clear to me was whether it is like TDM ie 1 timing device for the trunk, or if each end requires a timing device. I have a zaptel card in one server; do I have to have one in the second server in order to do an IAX trunk? I set up a trunk and so far calls can be made one way, but not the other. It is probably just not configured correctly, but I just wanted to make sure as I can't seem to find any reason at the moment. Thanks, Andy ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
Does somebody know a similar device that does the same for GSM networks ? Zoa Dovid B wrote: There has been talk about it before and I think people have done it. Paging Sam Tam - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Tuesday, January 02, 2007 4:56 PM Subject: Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source Do you know If its possible to do the same with Dock and Talk and an ATA GrandStream HandyTone 386? Thanks Joao Pereira Jonathan Attwood wrote: I use a Dock-n-Talk in conjuction with a Sipura SPA3000 Asterisk. Because I'm using Asterisk, I cannot use voice dialling, however inbound outbound calls work extremely well. I have Asterisk outbound routes set up to make a calls to cell phones go through the Dock-n-Talk. On 1/1/06, *Brian McEntire* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Is anyone familiar with cell phone switches that allow routing cell phone calls through in-home wiring? One example of these devices is the Phone Labs Dock-N-Talk. It says it keeps your cell charged when you are home and connects your cell (for incoming and outgoing calls) to your home wiring or cordless phones. But it also has features such as allowing speed dialing and voice dialing from extensions if your cell phone has those features. So I'm not sure if the device offers a fully compatible FXO signalling. I'm currently running Asterisk with 1 POTS and 1 VOIP (via Sipura 3000) lines coming into Zaptel FXS modules, and then I have two FXO modules for two extensions. I'm thinking of doing away with the land line. Should something like the Dock-N-Talk allow substituting a cell phone line for the POTS line? ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 license counting
Yes Zoa Michel wrote: Hello, How many licenses to buy?? : From what we understood from digium website, we must buy as many licenses as the number of maximum simultaneous calls using G729 Codec we wish to make. For example, If we want to be able to make a maximum of 10 simultaneous calls using G729 Codec, we must buy 10 licenses. Is it right? Thanks you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some queries on g729 license.
I did some tests a long time ago and the speed was roughly the same. ( I think digium's was slightly faster). I think the IPP version also doesn't work on AMD out of the box. It's just 10$ a channel, that's not even worth the hassle of trying something else. Joachim Al Bochter wrote: Matthew I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 So tell me what you think. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have unhappy employees that might be the way they hear about the violation in the first place. Important consideration: Bankruptcy law generally excludes debts created by things like malicious or criminal acts. Matthew Rubenstein wrote: As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: Date: Mon, 08 Jan 2007 13:47:39 +0800 From: Leo Ann Boon [EMAIL PROTECTED] Subject: Re: [asterisk-users] Some queries on g729 license. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd
Re: [asterisk-users] How big a pipe can IAX2 go?
It used to be a problem to have very big iax2 trunks (e.g. 100 channels). This should be resolved in asterisk 1.4, in older versions you can just work around it by making several smaller trunks. Zoa Noah Miller wrote: Hi Adrian - (Happy new year!) How big can an IAX channel grow to in size? (Realistically) Eg, if I have a 2Mb pipe between two A*k servers, can IAX grow to use the whole 2Mb with no issues, or do I need to create separate IAX channels (and if so, how do you do that in the config). It will go as big as your connection. I have one * install on a 100mbps fiber connection. You don't need to do anything to configure it to use all available bandwidth. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect many fax lines?
Have a look at www.spidermux.org Zoa Allen Casteran wrote: We have an application for Asterisk that will require connecting 144 fax ports into the system. Faxes will route externally over a PRI. The 144 ports are for local fax machines within the building. Not all will be faxing simultaneously. We just need to be able to provide ports in the building to plug in lots of fax machines. The plan is to run an Asterisk server for about 100 phones and these fax ports. The big question is what's the best way to connect these fax ports to *? 1) We could use an 8-port T1 card and link 6 Rhino FXS channel banks. 2) We could put 6 MultiTech MultiVOIP FX-24 boxes on the * LAN. 3) or ??? Anyone do something like this before? Any suggestions? I personally like the simplcity of the MultiVOIP boxes, plus the fact that they don't require T1 ports. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Tones A-B-C-D
Idefisk will do that - www.asteriskguru.com . (And asterisk will accept it). Zoa Al Bochter wrote: Ok does anyone know of any softphones that will dial DTMF tone keys A B C D And do you know if Asterisk will take the DTMF Tones for A B C D ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Tones A-B-C-D
Hmm, if the latest free version does not have all 16 keys, email [EMAIL PROTECTED], there should not be a difference in the amount of DTMF keys between biz and free version. Zoa Bob Chiodini wrote: The free version 1.31 has all 16 keys on the keypad. Bob... Al Bochter wrote: Are you sure IdsFisk will do all 16 DTMF tones? I have the free ver of IdeFisk and that one does only the base 12 DTMF tones Base 12 DTMF are 1 2 3 4 5 6 7 8 9 0 # * The 16 DTMF are 1 2 3 4 5 6 7 8 9 0 # * A B C D If the paid ver of IdeFisk has that may have to pay the money but first I must know for sure. :-) I want to use A B C D for control IVR's Not Everyone knows about the 16 tones like the Hams do 8-) Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 -- For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email -- Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email -- Need Voice Mail? http://www.bochterservices.com/?t=VMSt=email --For new and used security items http://www.bochterservices.com/?j=storet=email --BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Zoa wrote: Idefisk will do that - www.asteriskguru.com . (And asterisk will accept it). Zoa Al Bochter wrote: Ok does anyone know of any softphones that will dial DTMF tone keys A B C D And do you know if Asterisk will take the DTMF Tones for A B C D ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0660-0, 12/19/2006 - 12/19/2006 11:16:05 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax2 softphone attended transfers
Idefisk 2.0 will have it. Zoa Mail list wrote: Is there any good iax2 softphone capable of attended transfer ( like sjphone for sip ) . ? I tried iaxcomm and idefisk both seems unable to handle attended transfers. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax2 softphone attended transfers
It is scheduled for 9 januari. (If you ask nicely on [EMAIL PROTECTED] and promise to give good feedback, you might be able to get a beta version earlier ;) Zoa Vicky wrote: I have configure it by using the *2 atxfer feature of asterisk but its not as good as other attended transfer which sipphones give ( like sjphone where you can switch between two anytime ) . Also tried zoiper but it do not have even blind transfer yet . Any idea when idefisk 2.0 is going to be released :( or any other iax phone . On 16/12/06, *Zoa* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Idefisk 2.0 will have it. Zoa Mail list wrote: Is there any good iax2 softphone capable of attended transfer ( like sjphone for sip ) . ? I tried iaxcomm and idefisk both seems unable to handle attended transfers. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 200+ analog phones connected to FXS modules
You could go for 2 quad pri cards + channel banks or for TDMoE or usb channel banks. The last option would be the cheaper and more scalable one imho www.spidermux.org www.xorcom.com Joachim John covici wrote: You could put at least two Rhino quad t1 cards and that would give you 8 times 24 ports and I heard of one with those cards plus a dual t1 card which is 240 extensions on one server. this would take up 3 pci slots. on Thursday 11/30/2006 Jon Schøpzinsky([EMAIL PROTECTED]) wrote I think It would be cheaper to use dedicated VoIP PSTN Gateways, such as audiocodes or similar (audiocodes is actually a bad example, as their not that cheap). But dedicated ATA hardware with 24 or more ports. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vieri Sent: 30. november 2006 10:15 To: asterisk-users@lists.digium.com Subject: [asterisk-users] 200+ analog phones connected to FXS modules I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating Asterisk servers (cluster) mainly due to PCI slot availability. I am aware of the TDM2400P card. One could put 6 FXS uqad-modules and would serve 24 analog phones. However, I would need at least 9 of these PCI cards which could be placed in 2 or 3 servers. Is there another way of doing this (hopefully cheaper and more convenient)? Thank you for your suggestions. Vieri Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 200+ analog phones connected to FXS modules
Interesting product, I didn't know about this one until just now. I've heard that TDMoE is more trouble than it's worth, though, and may eventually be phased out of Asterisk. Can anyone from Digium give some more information or suggestions? -A. I'm not from digium but am the proud owner of a preproduction sample of the spidermux, i also took it to Astricon Dallas. (they are already being produced but are not being sold yet). The TDMoE implementation in asterisk works, but is not used by a lot of people or hardware yet, so it needs some work (Especially to make it work with recent kernels). I know the spidermux people already have a bunch of patches ready to be released to fix the issues that exist now. I've never heard something about tdmoe being phased out of asterisk. Zoa. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Siemens Gigaset SL75
http://www.voipsolutions.be/phones/dect-phones/gigaset-sl75-wlan.html Zoa Andrew Joakimsen wrote: Where can it be purchase? On 11/21/06, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, yes I tested this one week ago and it worked without problems. It is a nice wlan-phone with some (in my opinion) unnecessary features. Regards, Jens -Original Message- *From:* Olivier [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] *Sent:* Friday, November 17, 2006 10:20 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Siemens Gigaset SL75 Hi, Has anyone tested Siemens Gigaset SL75 with Asterisk ? How would you rate its performances ? Cheers ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable clock with 2.6 and without Digium hardware.
Can you tell us how you do the testing ? Zoa. Anton Tinchev wrote: Anybody sucessfully got stable 1000Hz clock without Digium harware and kernel 2.6? We need to consult some peoples how to clock asterisk stable with exactly 1000 Hz without much kernel/drives patching/tweaking. Some test results we made so far: 2.6 with digium card - stable 1000 Hz. 2.6 with ztdummy - uses RTC and the clock is 1024, not 1000. 2.6 with some Realtime kernel patch - provides stable 1000 Hz for some time, but in moments stops/misses interrupts/goes away from 1000 Hz 2.6 with ztdummy USB_UHCI - don't works, needs some tweaking. Somebody knows good patch for it? 2.6 with ztdynamic as primary clock sources - some issues with 2.6 (ztdynamic not ported well to 2.6?) with the mainstream versions, somehow patches solves it. 2.6 with kernel clock - needs kernel recompiling and work stable with switched off kernel Preemption. Long time tests in progress now. 2.4 with digium card - stable 1000 Hz 2.4 with ztdummy UHCI - stable 1000 Hz 2.4 with ztdynamic clock source - stable 1000 Hz/Depends on network conditions. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Port Range
Whatever suits you. (and whatever suits your internet provider that might have some ports filtered without telling you). 10.000 to 20.000 is the default range afaik. zoa Matt wrote: Ok, So what exactly is the RTP port range support to be? Lots of people are claiming 1,000 - 2,000, but then there are others claiming 16,384 - 32,768. What is it suppose to be? Then someone else told me it should be 10,000 - 20,000. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] say Asterisk to answer
It's not possible. The idefisk however has a button to auto answer. Zoa Gregory Duchatelet wrote: Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on “Accept” button. Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] considering purchasing a t1 card, any recommendations?
I think the recent Digium and Sangoma cards are quite similar. (and about the same price) I didn't try sangoma so far, never had any issues with the digium cards, I have no clue how the digium helpdesk is, i never needed to call them. (well not really correct i did call them once, years ago for a firmware problem with their first te410p revision, causing a crash once every few months they had the distributor send me replacement cards right away, before i returned the old ones, so that i could swap them without having to shut down the server for a week). Configuration and installation for the cards is pretty straightforward, all you need to do is compile the kernel modules for your kernel. I personally installed at least 20 digium pri cards, all on different hardware without problems related to the digium hardware. (sometimes i did have bad cables, bad pri's, oh and my embedded pc didn't provide enough power for FXO ports). You will probably find more people on the list with problems with digium than people with problems with sangoma. This might be because a lot more people seem to use the digium cards with asterisk than sangoma cards with asterisk. (Based on the people i speak to, i'd guess 1 to 5% use sangoma?). The biggest choice you need to make is if you want onboard echo cancellation or not, you might not need it and if you want it its going to cost you a lot more than without. (both for sangoma and digium hardware). - They both seem to use exactly the same Octasic echo cancellation module. If you need on board echo cancellation but don't need 4 ports, digium is the only choice with their 2 port card with Octasic echo cancellation module. (Afaik sangoma doesn't have such a 2 port board with on board E.C. but i could be wrong.) Btw, there are more options, dialogic has compatible cards and so does eicon. (you will need deeper pockets though, the eicon retails at +/- 12000 euro for a quad span i think - people who buy these for asterisk usually do so for hardware faxing or interconnection to different carriers at the same time.) Some people prefer digium over sangoma because they sponsor the asterisk development that way. I'm not one of them, i buy digium cards (or tell my customers to buy them) because i'm happy with their product. Dislaimer: I know some of the people within Digium quite well, so maybe i get exceptional support or they ship me handpicked gold plated, overclocked versions of their cards (not really since i just buy them from a reseller). Cheers, Zoa. Dovid B wrote: Can I now 5th it ? All this makes me wonder why Digium dosent work harder. I have mainly only seen others praise Sangoma over Digium. - Original Message - *From:* Tom Vile mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Wednesday, October 18, 2006 4:22 PM *Subject:* Re: [asterisk-users] considering purchasing a t1 card,any recommendations? I 4th it. On 10/18/06, *Matthew Thompson* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On 17 Oct 2006, at 22:09, Richard wrote: I would have to second the Sangoma buy. Their tech support is second to none and more then helpful. I've never had any problems with their products that wasn't my own fault. Thirded - I've just done another install with a Sangoma A102 - the setup guides you through all the way and takes no more than 30 minutes (Including recompiling zaptel, which it does for you) [EMAIL PROTECTED] :o) -- Matthew Thompson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman
Re: [asterisk-users] IAX Terminal
Lets change the question to : does somebody know good iax phones, that are ROHS compliant and without enormous delivery problems ? Neil Tancock wrote: Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm looking for a NAT-friendly solution and my SIP phones are good but not dependable. Neil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio on chan_gtalk
From our experience, chan_jabber doesnt work behind nat. We tried to patch it (in a similar way as nat=yes in chan_sip) but quickly bumped into other problems. (problems explained on mantis). Zoa. Gustavo Hernandez Baratta wrote: Hi! I´m trying with 1.4b2, chan_jabber and chan_gtalk. Jabber client register fine on talk.google.com, and when I start a call from gtalk to asterisk, I can see the incoming call and I see that asterisk play prompts (ie: demo and thank-you), but i can´t hear audio. If I redirect incoming call to a sip client, at sip I can hear but I can't in google talk. Asterisk is at public no firewalled network. Google Talk are behind a nat. Could anybody help me? Thanks in advance, Gustavo Hernandez ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what hardware and is it possible
I have such a setup here myself, although not for 100 people. Any recent server will do, but make sure you don't call 100 people the same second, spread them a little over time. Google for .call files Zoa. Ady Wicaksono wrote: Imagine i want to create application like SMS Alert, however it's a call alert when something happened, for example server is crashed, i want to call 100 of my staff (administrator, manager, and others) using asterix, when they pick up their phone, my asterix will play an audio file Is it possible? what is the correct hardware for this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio on chan_gtalk
Yes, its the same as what we tried. Gustavo Hernandez Baratta wrote: Hi Zoa: Thanks for your answer. Let me explain: Asterisk are not behind a NAT, google talk user are. Do you think that is the same problem? Thanks a lot! gus At 10:28 a.m. 17/10/2006, you wrote: From our experience, chan_jabber doesnt work behind nat. We tried to patch it (in a similar way as nat=yes in chan_sip) but quickly bumped into other problems. (problems explained on mantis). Zoa. Gustavo Hernandez Baratta wrote: Hi! I´m trying with 1.4b2, chan_jabber and chan_gtalk. Jabber client register fine on talk.google.com, and when I start a call from gtalk to asterisk, I can see the incoming call and I see that asterisk play prompts (ie: demo and thank-you), but i can´t hear audio. If I redirect incoming call to a sip client, at sip I can hear but I can't in google talk. Asterisk is at public no firewalled network. Google Talk are behind a nat. Could anybody help me? Thanks in advance, Gustavo Hernandez ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple TE110P cards in one chassis
Digium sells cables to interconnect them for timing. (dunno if thats only for the 412 cards). zoa Don wrote: As long as you have no interrupt conflicts...don't see why not... We have 3 TE410P cards in a Dell 2850...had to disable hyperthreading in the bios...and then make sure we had no shared interrupts on them... Work fine though...See no reason why you should have any problem with more than 1 TE110P. - Original Message - *From:* Thermal Wetland mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Thursday, October 12, 2006 2:01 AM *Subject:* [asterisk-users] Multiple TE110P cards in one chassis Does anyone know if you can have multiple TE110P cards in one chassis? -Thermal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.2/471 - Release Date: 10/10/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GPL Softphones
Xlite is not GPL! Joe Dennick wrote: The X-Ten is probably the most know free soft-phone availible. You can find it at http://www.xten.com/index.php?menu=Productssmenu=xlite Gregory Duchatelet wrote: Hi, I’m searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Kind of OT : Europeans going to Astricon
I will be also on a flight from frankfurt (lufthansa), but a few days early. Zoa. Stelios Koroneos wrote: Greetings ! Its kind of OT, but if there are any Europeans going to Astricon in Dallas, please send a message of-list. It's possible we will be on the same flight,(i am flying from Frankfurt) ;) so it will be a good way to know it's other and spend some of the 10 hours + flight time . Regards Stelios ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on 64bit xeon
Yes Akpome Akpoguma wrote: Hi All, Would asterisk and zaptel compile on 64bit dual xeon hardware?? Rgds From: Martin Joseph [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Asterisk 1.2.12 - Can NOT make call out /Asterisk terminate Date: Tue, 10 Oct 2006 00:49:30 -0700 On 2006-10-09 05:31:30 -0700, Benny Amorsen [EMAIL PROTECTED] said: PB == Peter Bowyer [EMAIL PROTECTED] writes: PB Fair enough - that's a bit different to 'Asterisk 1.2 is not ready PB for PRIME TIME' though, isn't it? There are plenty of stable 1.2 PB releases, all of which have many fewer bugs than your 1.0.x PB version. Unfortunately they also have security issues. It would be nice if someone made a 1.2.7.2 with the security issues fixed. Either way it is rather unfortunate that the latest version of 1.2 is unstable. That depends on your configuration and usage. Works fine for me on a couple of systems so far... (hope I am not spoiling my luck). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] isdn cross-over ...
Any card using this chip: http://www.colognechip.com/isdn/controllers/frame-hfc-s-usb.htm (=usb 1 port doing TE and NT mode) This would be the cheapest way to test them, hook the 1 port up to the asterisk server and call an echo application, then try the same with the second one. I dont know any 1 port cards that do NT mode. Zoa Ejay Hire wrote: Hi. A cross-over cable won't work, the isdn network provides signalling and adressing functions. When I was studying for my CCIE, an ISDN simulator cost an arm-and-a-leg, around $1k used from ebay. -ejay -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cesc Sent: Monday, October 09, 2006 10:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] isdn cross-over ... Hi, I am sure this is a bit off topic, but maybe the people here have the knowledge. Quick question: I have two ISDN S/T phones. What is the quickest way to test them (call one from the other)? can i make an isdn cross-over cable, taking the correct pinning, of course? What i need is to avoid the need for an NT connection (via a PBX). If the above is not possible ... where can I buy a cheap, small, simple ISDN PBX with at least two NT ports, so that i can connect my two phones and call each other? Tks! Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fonality acquires trixbox ([EMAIL PROTECTED]) ?
Looks like phonality has bought trixbox. (I suppose they failed to buy digium :) http://news.asteriskguru.com/10/773/2006/10/5/Fonality_Aquires_trixbox_([EMAIL PROTECTED]) Earlier on they found venture capitalist: http://www.fonality.com/press/20060109.htm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp (foip)
linksys spa3102 or 2100 are known to work. Grandstream also should do it with recent firmware. Don't be fooled by what is written on the box, lot of ata's out there claim t.38. (while the firmware doesnt contain anything related to t.38) Zoa Christopher Corn wrote: lee, Thanks for the feedback. in most diagrams explaining t38, it shows, the sending fax machine connecting to a pots before connecting to a gateway,then the internet. but if i've read and understood correctly, the sending end can use an ATA with t38 support instead of a pots. in that case, where does the packetization of the t30 data happen? at the ATA? level i presume? http://www.answers.com/topic/t-30-protocol-figure-01-jpg also, can you recommened a good asterisk compatible ATA adapter with t38 support? i believe cisco has one. Thanks in advance. */Lee Howard [EMAIL PROTECTED]/* wrote: Christopher Corn wrote: May I ask, from your own personal experience. is it not necessaritly worth (the headaches) of investing mytime into setting up SPANDSP into my asterisk system, but rather invest it into going to a company, like packet8 that offers t38 conversion? I am not really in a position to tell you what something will be worth to you - especially when I've not even used that something myself. I know and use spandsp as a library, with IAXmodem and HylaFAX, but I do not have any experience with spandsp in txfax/rxfax applications or in its new T.38 gatewaying. I suspect that I'll eventually get into spandsp's T.38 aspects, but without that I've only had a limited amount of hands-on exposure to T.38 applications in the form of t38modem and Cisco gateways (which experience was somewhat disenchanting - mostly because of the gateway T.30 processing). If you have a T.38 fax machine or if you have a T.38-capable ATA connected to a fax machine and you do not have your own PSTN lines then I would suspect that it would be worthwhile to use T.38 pass-through on Asterisk 1.4 or OpenPBX in conjunction with a T.38-supporting FoIP provider. (Because otherwise you don't have any straight-forward, reliable means for faxing from your internal fax machines.) what does the future of faxing lean towards? before entering an era when all fax machines run the t38 protocol. will we see more t38 termination services or faxing through g711? T.38 is the end-all solution for faxing over IP networks. So I suspect that you will see the pervasiveness of T.38 implementations increase along with the pervasiveness of VoIP in general. That said, VoIP has its own fair share of problems that keeps it from being capable of replacing PSTN circuits entirely, and so as long as those problems are not generally resolvable for your average business or service provider then you'll continue to also see more of the same, traditional, modem-ing fax machines. So I strongly suspect that you'll see more of T.38, but I don't think that the PSTN (and traditional fax machines with it) is going away any time soon. from what i've read, using a service that does t38 termination, seems to be where i should go. I would say that it entirely depends upon whether or not you have PSTN lines yourself. If you do, then I would take whatever efforts you can to avoid the additional points of T.30 processing/relaying (therefore avoiding T.38 gatewaying). But if you do not have PSTN lines, then take whatever efforts you can to properly implement T.38 to your FoIP provider who will gateway for you. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Very high ping times from 7960 phones
I can confirm the same. It doesnt mean the audio will be delayed, the phone is just slow with replying to the sip messages. Zoa Michiel van Baak wrote: On 09:42, Mon 25 Sep 06, Tomislav Par?ina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times 150ms. Every single one. I've scoured the settings on the 7960's and have looked and looked for why this might be the case. Cisco ata's (186) on the same network report ~ 10 ms. An xlite softphone reports ~ 5ms regardless of what computer it's installed on. I have the same here. All between 150 and 250 ms. The phones do work perfectly, only the time in sip show peers is higher then any other phone/device. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's
I'm not sure irqbalance is a good idea. (although i'm not familiar with it, its sounds like it balances it all the time, not just spreads it and leaves it). Maybe its best to do it manually ?. have a look at something i wrote ages ago: http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html and search for _Put your networkcard and pri card on a different CPU Zoa. _ Robert Jenkins wrote: Hi, On Centos IRQBalance should already be available. You should be able to run 'setup' from a console/terminal, go to System Services enable irqbalance. It will then be enabled on boot. To start it without re-booting, use service irqbalance start If it's already marked as enabled in the services list, the problem is elsewhere. Hope this helps, Robert Jenkins. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bart Fisher Sent: 25 September 2006 02:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's Hmm, this must not be installed: # locate irqbalance # /usr/src/kernels/2.6.9-34.0.2.EL-smp-i686/include/config/irqbalance.h How do I install this? Bart Álvaro Palma wrote: It appears that CPU1 in not taking any interrupts - What steps do I need to do bring up CPU1 and share IRQ requests for a Linux noob? Run the IRQ balance daemon (/usr/sbin/irqbalance). It's part of the kernel-utils RPM. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] new in 1.4?
I was thinking the same thing when reading the press release on sineapps and writing a news article for asteriskguru. I think this covers most of it: - Generic Jitter Buffer - t.38 passthrough - Dial plan programming language (AEL v2) - Asterisk can talk to googletalk and Jabber networks - imap storage for voicemail - whisper paging - Autoconf configuration - menuselect (graphical module select tool similar to the kernel config system) - higher quality prompts (in English, French and Spanish). - watch out they are restructured a little Zoa. Roy Sigurd Karlsbakk wrote: Hi all I've read through the UPGRADE.txt file, but AFAIK it does not quite discuss all the new stuff with 1.4. Neither the jitterbuffer nor the packetization patch (#5162, if that ever made it into 1.4) are mentioned. So, is there a document somewhere describing what's new in asterisk? thanks roy --- Humans mostly aren't particularly evil. They just get carried away by new ideas, like dressing up in jackboots and shooting people, or dressing up in white sheets and lynching people, or dressing up in tie-dye jeans and playing guitars at people - Terry Pratchett --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 analog phones
quad port T1 card 3 channel banks. Zoa mike wrote: Dear list which hardware solution would you suggest for connecting 60 analog phones to asterisk ? thank you very much .mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Volume events causing talk off on Asterisk with Digium 411P
I have the same problem on on of our systems, but i always thought it to be a problem in the ATA's connected to this server. (My customer has a lot of traffic on the lines and only sometimes hears this problem). It seemed to happen especially with loud woman voices, but i was unable to reproduce it on command. I have several other te410p's on different locations (with different carriers), without those complaints. Does this also happen on pri to pri calls for you ? Maybe its a combination of carrier volume with the te410p ? Zoa Servetas, Andrew wrote: We are experiencing random talk off events when we hear a loud volume event on the PSTN side of our calls. We do not always hear the spurious DTMF, but I can see it in the console when I have the debug and verbose levels turned up. We do however always have the associated brief periods of silence that immediately follow. Sometimes they are only a matter of seconds, other times they can be as long as a minute. We hear it most often if the remote party is on a cellular phone with a lot of background noise, or if a loud noise happens during the call. Neither party can hear the other when this happens. It almost reacts like an AGC circuit is muting the call. We are using a Digium TE411P quad-span T1 card on 1.2.5. I called Digium support and we have played with the VPM_DEFAULT_DTMFTHRESHOLD in the WCT4XXP.C driver as recommended, and all the RELAXDTMF and GAIN settings in Zapata.conf are set according to their recommendations. Has anyone else experienced this, and if so, what have you done to correct it? //Andy Servetas// CTI Support Engineer Dirigosoft Corporation Portland, ME www.dirigosoft.com http://www.dirigosoft.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Volume events causing talk off on Asterisk with Digium 411P
But does it help ? Is it better than before ? Do you have a good way of debugging ? (like an audio recording that i could play ?) Does it show something on the cli when it happens ? Zoa Servetas, Andrew wrote: They recommended changing the default value of 1000 up or down incrementally until it works better. We’re currently at 2000, and we’re still not completely free of events. What were the proposed changes to VPM_DEFAULT_DTMFTHRESHOLD ? -- -- Steven http://www.glimasoutheast.org Servetas, Andrew andrew.servetas at dirigosoft.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote in message news:28289145AD231E418DB8CABE0BE392AA016F8176 at casco.stroudwater.net... http://lists.digium.com/mailman/listinfo/asterisk-users We are experiencing random talk off events when we hear a loud volume event on the PSTN side of our calls. We do not always hear the spurious DTMF, but I can see it in the console when I have the debug and verbose levels turned up. We do however always have the associated brief periods of silence that immediately follow. Sometimes they are only a matter of seconds, other times they can be as long as a minute. We hear it most often if the remote party is on a cellular phone with a lot of background noise, or if a loud noise happens during the call. Neither party can hear the other when this happens. It almost reacts like an AGC circuit is muting the call. We are using a Digium TE411P quad-span T1 card on 1.2.5. I called Digium support and we have played with the VPM_DEFAULT_DTMFTHRESHOLD in the WCT4XXP.C driver as recommended, and all the RELAXDTMF and GAIN settings in Zapata.conf are set according to their recommendations. Has anyone else experienced this, and if so, what have you done to correct it? Andy Servetas CTI Support Engineer Dirigosoft Corporation Portland, ME www.dirigosoft.com -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prompts playback changing tempo w/ SMP kernel
Check the timer frequency, it might have a different setting on the two kernels. RR wrote: Hi all, (2nd attempt) this is probably a weird question and something I'm not doing right but I got this bizarre thing going on here. When I boot the system with the SMP kernel and compile (*) with the smp kernel source (actually even if I don't compile, but as long as I boot into the SMP kernel), I get this problem where calling into the system, say to check my voicemail, the prompt playback continously changes tempo. The prompts are played in slow-motion, and then it speeds up to its normal speed, then goes back in slow-mo and so on. It happens (I think) at constant periods. Only the tempo changes, not the pitch of the prompt. Does anyone have any idea what could be happening? I have watched topconstantly but haven't noticed anything bizarre in terms of CPU or Mem usage. This is on a 100mbps LAN with nothing much else on it. And it only happens when it's booted into the smp kernel. So it's something to do with smp, thread scheduling, or some buffer BUT I don't know what exactly. All you champs out there, esp. the asterisk-dev people, any light you can shed on this? Thanks much \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
