Re: [asterisk-users] Fax Machine Options
On Tue, May 20, 2008 at 12:23 AM, Lee Howard [EMAIL PROTECTED] wrote: Andreas van dem Helge wrote: Cisco gateway with T.38 support. That's the only real way to do faxing through asterisk. Although this statement has marginally more truth to it given the SIP-only context that the original poster provided, it is still substantially inaccurate. So, mr accurate, please tell us what is the best way to fax via Asterisk. Keep in mind you can't always control the latency between the endpoint and the asterisk machine/media gateway (yes, I know SIP to asterisk on the same VLAN + send the call via PRI works pretty well). And how do we know he is using SIP only? He asked about the IAXy! There are several ways to do T.38 other than with a Cisco gateway. Now, if you meant that T.38 is the only way in the SIP-only context (and not specifically Cisco-branded T.38) then that has significantly more accuracy to it. Doesn't have to be Cisco. But there is no Asterisk T38Gateway so either way you need to buy a 3rd party gateway (or use a service that supports it... check out www.gafachi.com) Audiocodes we can take off the list because the violate the GPL so next down my list is Cisco. But any SIP-compliant T.38 gateway will work. However, if by *real* you also mean *reliable* then be aware that T.38 over SIP/UDP has an inherent weakness due to the medium that make it, in my experience, significantly less-reliable than simply having a fax machine hooked up to a traditional analog line. Yes but significantly the most reliable way to fax at the moment via VoiP... well you have that magic answer but I haven't heard it yet. Honestly if the OP is using the phones voip-only and can make calls and understand the remote party and be understood there should be no issues. The problem is even in an environment where voice quality is perceived very well faxing could still be an issue. If you have an issue where your t38 calls can't complete due to packet loss, latency or other such issues I doubt you can make an intelligible phone call. When my clients come to me with the same issue I generally do not push them into a corner with T.38. In almost all cases they find that it is worth the $20-50 monthly for the analog fax line... and if that expense is too much then the on-line fax service provider is an easy recommendation. You're not my client and I'm not trying to pressure anyone. What if they have 500 fax machine spit over 23 locations is keeping phone lines still cost effective? He's asking on the Asterisk mailing list regarding faxes so I assume 'get a phone line' isn't an option. Note that if you have a fax machine that performs some variant of T.37 (fax-over-email) and you have an on-line service provider that is willing to work with you... then you can rather easily get your fax machine faxing through their service. (Which is yet another option.) And how many (low cost... not $5000 copiers) do that? and to the OP about the hardware, even a cheap grandsteam ATA will work just fine... that's what I use on my personal fax machine and it has no issues. I can't recall a time this year a fax has failed. This is going over the public internet and then also back out to a voip provider. We do use decent networks but that's about it. No QoS anywhere down the line. ___ -- 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] Googles 411 services
Yea... tf.voipmich.com went down ~ 1 week ago when you call through it the calls just ring forever. Some issue seems to happen with tf.voipmich.com at least once a year and it always takes a long time to fix. tollfreegateway.com seems to be working.. On Sat, May 17, 2008 at 1:38 PM, Adrian Marsh [EMAIL PROTECTED] wrote: All, Does anyone know of a SIP URI direct to googles 800-GOOG-411 service? When I put calls via sipbroker, half the time the calls fail. An enum lookup shows 3 URIs listed, none of them seem to be google directly, and I think 1 of them fails 100%, and the remaining one fails at other random times. Thanks, Adrian ___ -- 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] Recording problems, reinvites
So why don't you just disable reinvite? Using 1.4.15 here with no issues with MixMonitor. Then again I've *ALWAYS* disabled reinvite because it never works for me. On Mon, May 19, 2008 at 4:33 PM, Trevor Peirce [EMAIL PROTECTED] wrote: Hello, I'm wondering if anyone else has been observing problems lately with 1.4.18 and higher releases of asterisk not properly recording calls. When using MixMonitor, the resulting file is only a few bytes long. I think this is because asterisk is doing Native bridging even though MixMonitor should block that. Did something change around the release of 1.4.18 that would have changed the behaviour? I thought that when ChanSpy, MixMonitor, and the like are enabled on a channel it would be prevented from reinviting the audio to bypass asterisk. Thanks, Trevor Peirce ___ -- 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] Fax Machine Options
Cisco gateway with T.38 support. That's the only real way to do faxing through asterisk. I think a VG200 with newer firmware will support SIP + T.38 but don't buy on my suggestion because I've never used that device outside call manager configuration. Or see if your VoIP provider supports T.38 fax but you must use SIP in that case. It will work very well once you get it working hint: check udptl.conf On Mon, May 19, 2008 at 11:27 PM, Joseph L. Casale [EMAIL PROTECTED] wrote: Is my only solution to add a fax machine to our VOIP only setup by using an IAXy? I should specify the office people want a traditional fax machine in the sense that fax's be sent and received from a physical unit, they don't want an email to fax setup. They have a dedicated sip did provisioned just for the fax. What are others using? Thanks! jlc ___ -- 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] playing .gsm sounds through a web browser
A posting to the correct mailing list? Or at least a post with the details of the issue? What OS? Can you play these same .gsm files in any media player your OS might have? On Thu, May 15, 2008 at 7:26 PM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: I have a lot of recordings from asterisk in a .gsm format. I would like to play these files from a web browser (IE, firefox and opera) What do I need to do in order to achieve this goal ? Thanks Julian ___ -- 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] Shared line appearance phones?
The docs as far as I can tell are not correct. E.g. Zaptel is required (because it seems that it uses MeetMe) but none of that is documented. So yes please do see if you can make the feature work and please post a working example config for a Polycom phone. On Fri, Nov 30, 2007 at 8:10 PM, Russell Bryant [EMAIL PROTECTED] wrote: Mark Wiater wrote: I fought with this in 1.4.5 with polycom phones. I was hoping to share a DID from a PRI on several Polycom IP430's. Might you be willing to share some specific configurations for such a situation? There are some basic examples in doc/sla.pdf in the 1.4 tree. However, I have on my to-do list to spend a week with an SLA test environment and coming up with an extensive set of examples of the different ways it can be used. I will post something to this list when that is available. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --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] Setting CallerID UNKNOWN on an outgoing call
On PRI SetCallingPres works fine it should work with ISDN because its the same signaling. -= Info about application 'SetCallerPres' =- [Synopsis] Set CallerID Presentation [Description] SetCallerPres(presentation): Set Caller*ID presentation on a call. Valid presentations are: allowed_not_screened: Presentation Allowed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen: Presentation Prohibited, Passed Screen prohib_failed_screen: Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable On Wed, May 14, 2008 at 2:08 AM, Stefan Guenther [EMAIL PROTECTED] wrote: Hello, on my ISDN phone I can configure that on the next outgoing call, my telephone number should not be transmitted, instead it should be UNKNOWN. How can I configure Asterisk to do the same? Is this a feature/parameter of the driver (chan_capi) that I'm using? BTW: I'm using ISDN and Deutsche Telekom, if the provider makes any difference. Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- 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] No-mobo PC for USB Drives Enclosure?
This will work: http://www.newegg.com/Product/Product.aspx?Item=N82E16899705001 I assume you have devised a way to power the USB to serial adapters from the PC power supply. FWIW I think your system is inefficient but maybe you do need 750gb per each installation. Each to his own. On Tue, May 13, 2008 at 10:22 PM, Matthew Rubenstein [EMAIL PROTECTED] wrote: I have over a half-dozen different SATA hard drives, each with different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each one's different user groups and applications. Each one's load on the Asterisk server is small enough that one server can host them all, accessed easily over USB. But right now, each one is in its own external USB enclosure on a powered USB hub. I want to combine them all into a single large enclosure. I tried to use a single PC chassis, leaving the USB hub inside with the drives screwed into it, and powered from the PC power supply as internal drives on the proper drive power output plugs. But without a PC motherboard plugged into the power supply, too, the power supply won't start up to power the drives. I don't want to add a motherboard: that costs money, and sucks power, and is totally unnecessary. I just want to make this gutted PC chassis power my drives only, and have them connect to the complete PC sitting next to it via the single USB cable coming out of the drive chassis. How do I do that? Is it possible to use the extra, unused floppy power plugs to power more hard drives, with an adapter? Is it possible to split the existing hard drive power plugs to each power multiple drives? How many drives can I split each power plug into? The power supply is a cheap 300W unit, and the drives draw max under 9W each: http://www.wdc.com/en/products/products.asp?driveid=311 . So can I power 25-30 of these drives, or at least 10? -- (C) Matthew Rubenstein ___ -- 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] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect?
srv04*CLI show application Dial srv04*CLI -= Info about application 'Dial' =- [Synopsis] Place a call and connect to the current channel *SNIP* p- This option enables screening mode. This is basically Privacy mode without memory. P([x]) - Enable privacy mode. Use 'x' as the family/key in the database if it is provided. The current extension is used if a database family/key is not specified. n- This option is a modifier for the screen/privacy mode. It specifies that no introductions are to be saved in the priv-callerintros directory. N- This option is a modifier for the screen/privacy mode. It specifies that if callerID is present, do not screen the call. On Sun, May 11, 2008 at 12:24 PM, Robert DeVries [EMAIL PROTECTED] wrote: GrandCentral has a feature where when you call the GrandCentral number it can ring multiple phones. However, it's not the first phone to answer that gets connected, but the first phone to answer AND play a touch-tone after hearing a recording. The advantage of this is that if one of the called phones has voicemail, it won't get connected to the calling party because the VM won't send a touch tone in response to the recording, unlike a live person. I have always resisted implementing a multiple ring scenario with Asterisk that included a cellphone because of the voicemail answering problem, but this seems to be a solution. Anyone know how to implement it with Asterisk? ___ -- 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] 3U server chassis Digium TE405P?
A quality 3U chassis will mount the cards parallel to the mainboard with the use of a riser card, just as a 1U chassis does. If you are intent on sourcing the components yourself may I suggest a Tyan or Supermicro barebones server? I think that is the best solution for integration in these sort of specialized systems. I know they've saved me many headaches. On Mon, May 12, 2008 at 11:29 AM, Sherwood McGowan [EMAIL PROTECTED] wrote: Gentlemen, First let me say it's great to be back on the Asterisk mailing lists. Those of you who have been around for a while will remember me as Rushowr. I look forward to answering questions and whatnot in the future, but for the moment I have a minor question that I cannot find a definitive answer for online. I am in possession of a Digium TE405P card which I _know_ will fit in a 4U chassis, but we are building a new server and cannot get a 4U from the supplier that my current client wants to use. However, we can get a 3U chassis. My question is, will this card fit? Does anyone out there have a 405 out there that they have installed in a 3U? Thanks in advance for any help that can be offered, Sherwood McGowan VoIP / Telecom Solutions Consultant ___ -- 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] Out-Going Callerid
I've had the same issues with T-Mobile. Thanks for pointing out the exact cause. I did speak to someone in their back-end offices and they did resolve it for a short time, then it reverted back for a bit and started working again. Now it seems to work 90% of the time. On Sun, May 11, 2008 at 9:40 PM, Alexander Lopez [EMAIL PROTECTED] wrote: This happened to me here in the US. T-Mobile was the carrier that I had a hard time with, land lines, and all other carriers worked fine. It seams that T-Mobile, was not accepting calls that it could not confirm the ANI on. The solution was on the Telco side. I had enabled a feature that allowed me to change the ANI of outgoing calls to those within my DID block. Any calls outside that block the telco would send out with a ANI flag of 'presented, but not-confirmed' message. Try setting your CallerID to your BTN (Billing Telephone number, or ANI) and see what happens. Hope this helps, don't know much about the UK NTL lines but this had me stumped for a while.. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tim Guy Sent: Thursday, May 08, 2008 5:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Out-Going Callerid Well.. Now I'm confused. Recap.. Couldn't appear to get out going callerid to work on a UK NTL PRI connection. Id been testing it with my Orange Mobile phone.. Dial the 07973xx and it displays private. Called my girlfriend tonight on our land line (all be it NTL again but this time analogue), got her to do 1471 and feck me, it read back the callerid Id been putting through. Only been able to try it on Orange and NTL residential at the moment. Ill try it to a BT line tomorrow morning. I'm really stumped now.. Why does it work on one and not on the other? Tim This message is sent in confidence for the addressee only. Unless specifically stated, the contents are not to be disclosed to anyone other than the addressee. Unauthorised recipients must preserve this confidentiality and should please advise the sender immediately of any error in transmission. The views an opinions expressed in this e-mail message are the sender's own and do not necessarily represent the views and opinions of NS Optimum Ltd. Although this e-mail and attachments are believed to be free of any virus or other defects which may affect any computer or IT systems into which they are received, no responsibility is accepted by NS Optimum Ltd for any loss or damage arising in any way from the receipt or use thereof. Place of registration: England, Registered Office: Jenton Road, Sydenham Ind Est, Leamington Spa, Warwickshire, CV31 1XS, Registered No 3018839 ___ -- 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
[asterisk-users] Fwd: Polycom Advanced Features
Anyone have shared lines (sla.conf) working with Polycom phones? Also, has anyone figured out if its possible to do 1 button call park with the softkeys? ___ -- 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] T38 Passthrough Verification
The call is still going to show up as the codec with which the voice segment was established. Have you viewed the SIP debug messages and confirmed that T.38 is not being used? FWIW the device that is receiving the T.38 fax (generally callee) should be issuing the T.38 re-invite, so you might want to start at that end. Make sure t38pt_udptl = yes is defined. On Thu, May 8, 2008 at 8:55 AM, JR Richardson [EMAIL PROTECTED] wrote: JR Richardson wrote: I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet ! sip show channels shows the call setup with ulaw. Try setting canreinvite=no for the peer doing T.38. It looks like the code in Asterisk 1.4 will not allow re-invites for an established T.38 passthrough call. I saw a post about the re-invites, so I tried it both ways, canreinvite=yes/no with the same results. Thanks. JR --- JR Richardson Engineering for the Masses ___ -- 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] MOH and Licensed G729 codec
In menuconfig did you select the g729 music on hold? Or you can try this one: http://app5.netjdn.com/~joako/sounds/SampleAudioSource.g729.wav (remove the .wav extension) On Thu, May 8, 2008 at 5:46 PM, Nitesh Divecha [EMAIL PROTECTED] wrote: Hello All, Recently, I build three Asterisk 1.4 box and installed licensed copy of G729 codec. Before installing the G729 codec I tested the MOH on all three Asterisks box and it was working fine. So I install G729 codec and retested MOH and it was all wavy... Meaning the music was going up and down and missing bits and pieces and choppy... Any idea what did I do wrong? The MOH files are the default ones which comes with Asterisk. Cheers, Neel ___ -- 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] Best Linux distribution to use in Asterisk server
I think it's all personal preference I'd never recommend anyone use ubuntu for anything, honestly. SLES is my #1 pick with CentOS / PNAELV being a close second... problem with Cent is there's not central administration like there is in SuSE (YaST2... it's so simple! gotta setup a network no ifconfig up this route add that just point and click or use the ncurses interface.. same for just about every service)... not even an interactive package manager thats usable from the CLI. You might be able to get by with openSuSE but remember the lifecycles are short, like Fedora. On Fri, May 9, 2008 at 11:19 AM, equis software [EMAIL PROTECTED] wrote: Hi, I allways use Gentoo y my Asterisk servers and work well, but what do you think about to use Ubuntu or another distibution?? 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] Best Linux distribution to use in Asterisk server
Oh, and FWIW a Cisco uses PNAELV as the basis for one of it's most popular voice products. http://www.bouncethem.com/5455 On Fri, May 9, 2008 at 11:19 AM, equis software [EMAIL PROTECTED] wrote: Hi, I allways use Gentoo y my Asterisk servers and work well, but what do you think about to use Ubuntu or another distibution?? 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
[asterisk-users] Polycom Advanced Features
Anyone have shared lines (sla.conf) working with Polycom phones? Also, has anyone figured out if its possible to do 1 button call park with the softkeys? ___ -- 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] Newbie alert: VoIP hardware
I think there are quite a few aspects to the issue. I agree I've used the X101p cards which really are a Windmodem with a resistor removed and I had nothing but echo problems but then again I could have tried harder. 1) It was an early digium product. I think the Sangoma cards and the newer Digium cards are better designed. Hardware echo cancelation does not hurt, especially if you have a history of echo problems. 2) It's a modem, probably the people who designed the card and also the chip did not expect such scrutiny to be placed on its voice abilities they probably figured primary use would be for data or fax and maybe some voicemail use. 3) The process of converting the signal to digital brings out qualities that you might not normally notice physical wiring issues, reversed tip and ring pairs etc. Before giving up I would use certified Cat 5 or at least Cat 3 cabling, make sure all connectors and junctions are at minimum Cat 3, none of the pairs are reversed at any point. And heck just for troubleshooting remove any unneeded devices and wiring, i.e. everything else, including DSL modems and filters (does not hurt to use better quality ones). Even then there are thousands of feet of wiring beyond control, but that last few hundered tend to be the messiest. With that said many people will use these and not encounter issues. And then the majority that do encounter issues will be with echo. Many times this can be resolved by correcting any major wiring issues and tweaking the software echo cancellation. On Wed, May 7, 2008 at 4:15 AM, Marco [EMAIL PROTECTED] wrote: Alan Lord wrote: If you only have one analogue line why not just get a simple x100p card? When you use OSLEC with them they work great here in the UK. I bought my card from a USA based eBay seller. Total cost for card and shipping was about £17.00 Respectfully, I don't agree. I've purchased an original clone :-P of the X100P card, on the long period they almost always have some drawbacks... Faxing have been troubling for me. Don't know if it was for the line or else, but with a Digium card I had no problem at all. No sponsoring in here, ok, but certified hardware works better, therefore it's a better investment, I think. ___ -- 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] VOICEMAIL OPTIONS help needed
see voicemail.conf.sample all the options you need are documented there. maxmsg delete On Wed, May 7, 2008 at 12:49 PM, Steve Johnson [EMAIL PROTECTED] wrote: Hi everyone, We have a particular user on our Asterisk 1.4.x system who always listens to his voicemail messages via email. - Is there some way to send the voicemail ONLY to email and not retain them on the phone? - Alternatively, can the voicemail system only keep, say, just the last 10 messages (as backup in case of email delivery failure or a message getting deleted in email accidentally before it is heard), purging out the oldest when a new one is received? (If we set the option maxmsg=10 on his mailbox in voicemail.conf, I think it will stop accepting voicemails after 10 messages, not turf the oldest one and accept a new one in its place). Everyone else uses the normal voicemail options on their phones, so the solution should be just for this single user. Thanks for any suggestions. S. ___ -- 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] Running Asterisk as root
I totally agree. Someone filed a bugreport for this? Also asterisk init script should be installed by default too. I am going to give Cesar's instructions a try (sans removing /bin/sh) and hope it works! On Tue, May 6, 2008 at 3:24 AM, Stelios Koroneos [EMAIL PROTECTED] wrote: In general, if your asterisk is accesible from the internet its much better to have it run as a non-root process. (My opinion is that this should be the default out-of-the-makefile ;) asterisk behaviour) This is the norm for more of the servers/services running on a linux system, and can act as a safety-net when things go bad Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Sent: Tuesday, May 06, 2008 3:00 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Running Asterisk as root Hi all, I have seen discussions on this earlier on, but just want to hear some quick thoughts. I am running v1.6 of Asterisk on my Ubuntu installation, I did make config to make it run at boot. Since I've got a firewall and don't have any other servers running I am not worried. I have been htinking about running Asterisk as a seperat user, but haven't done that yet. Everything is working fine. What do you think? Thanks, Christian ___ -- 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] [OT] wireless headphone that can answer a call?
Some of the polycom phones support this with a specific firmware and Plantronics headset. Read the polycom SIP release notes/changelog for details On Mon, May 5, 2008 at 5:29 AM, Louis-David Mitterrand [EMAIL PROTECTED] wrote: Hello and sorry for the OT, Is it possible for a wireless headset of which the base is connected to a Polycom IP601 to remotely answer a call? In the same way as a bluetooth headset. 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] Zaptel Compatibility
.build file is missing in the kernel-source package. Solutions is: Once you have the appropriate kernel sources installed you will need to configure them. Execute the following commands: cd /lib/modules/`uname -r`/build make mrproper Execute one of the following commands based on your hardware configuration (again, the exact file names may vary): cp -f configs/kernel-2.4.2-i586.config arch/i386/defconfig cp -f configs/kernel-2.4.2-i586-smp.config arch/i386/defconfig cp -f configs/kernel-2.4.2-i686-enterprise.config arch/i386/defconfig Verify that the kernel Makefile EXTRAVERSION information matches the version that you are running with respect to smp support. make oldconfig make dep Similar to 'make cloneconfig' in SuSE Linux. On Thu, May 1, 2008 at 3:06 AM, Alan Lord [EMAIL PROTECTED] wrote: Mik Cheez wrote: Hmph...and it appears no kernel-smp-source exists. You should be able to compile going to a non-SMP kernel, but there must be a better solution. I can't believe this hasn't come up before. Sorry. You only need the kernel headers in reality I believe. Why not just mail RH and ask them for the headers? Al -- The way out is open! http://www.theopensourcerer.com ___ -- 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
[asterisk-users] Shared Line Appearance
Could someone please add to the documentation that Zaptel is required for SLA to work? It becomes sort of frustrating when you read the documentation a few times, keep on trying to get the thing to work for a few hours only to discover there is a minor omission in the documentation. ___ -- 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] Zaptel Compatibility
Is Zaptel 1.4.10 compatible with RHEL 3 (2.4.21-53.ELsmp)? Because I can compile 1.2.20.1 just fine but 1.4 says: echo You do not appear to have the sources for the 2.4.21-53.ELsmp kernel installed. You do not appear to have the sources for the 2.4.21-53.ELsmp kernel installed. exit 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/zaptel-1.4.10' make: *** [all] Error 2 Yes kernel-source is installed, there is no kernel-devel. I read one account where if I use non-SMP kernel it might work. But there's no fun it that. 1.2 works why not 1.4? Failing getting 1.4 to work can I use Zaptel 1.2 with Asterisk 1.4? I think not but just wanted to make sure. ___ -- 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] Shared Line Appearance
System with Zaptel: *CLI sla show stations = === Configured SLA Stations = = === System without Zaptel: unisoft*CLI sla show stations No such command 'sla show' (type 'help' for help) On Wed, Apr 30, 2008 at 12:53 PM, Patrick [EMAIL PROTECTED] wrote: On Wed, 2008-04-30 at 08:56 -0700, Andreas van dem Helge wrote: Could someone please add to the documentation that Zaptel is required for SLA to work? It becomes sort of frustrating when you read the documentation a few times, keep on trying to get the thing to work for a few hours only to discover there is a minor omission in the documentation. Can you please explain what you mean with zaptel is required for SLA to work? Thanks, Patrick ___ -- 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] Zaptel Compatibility
No such thing. [EMAIL PROTECTED] [/usr/src/zaptel-1.4.10]# up2date --showall | grep kernel kernel-2.4.21-53.EL.athlon kernel-2.4.21-53.EL.i686 kernel-BOOT-2.4.21-53.EL.i386 kernel-doc-2.4.21-53.EL.i386 kernel-hugemem-2.4.21-53.EL.i686 kernel-hugemem-unsupported-2.4.21-53.EL.i686 kernel-pcmcia-cs-3.1.31-19.i386 kernel-smp-2.4.21-53.EL.athlon kernel-smp-2.4.21-53.EL.i686 kernel-smp-unsupported-2.4.21-53.EL.athlon kernel-smp-unsupported-2.4.21-53.EL.i686 kernel-source-2.4.21-53.EL.i386 kernel-unsupported-2.4.21-53.EL.athlon kernel-unsupported-2.4.21-53.EL.i686 kernel-utils-2.4-8.37.15.i386 On Wed, Apr 30, 2008 at 2:47 PM, Mik Cheez [EMAIL PROTECTED] wrote: Have you tried kernel-smp-devel? Andreas van dem Helge wrote: Is Zaptel 1.4.10 compatible with RHEL 3 (2.4.21-53.ELsmp)? Because I can compile 1.2.20.1 just fine but 1.4 says: echo You do not appear to have the sources for the 2.4.21-53.ELsmp kernel installed. You do not appear to have the sources for the 2.4.21-53.ELsmp kernel installed. exit 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/zaptel-1.4.10' make: *** [all] Error 2 Yes kernel-source is installed, there is no kernel-devel. I read one account where if I use non-SMP kernel it might work. But there's no fun it that. 1.2 works why not 1.4? Failing getting 1.4 to work can I use Zaptel 1.2 with Asterisk 1.4? I think not but just wanted to make sure. ___ -- 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] Discover connected Zap lines
in the CLI you can issue the command zap show status e.g.: pbxserver-doral*CLI zap show status Description Alarms IRQbpviol CRC4 Wildcard X101P Board 1 RED0 0 0 In this case the phone line is unplugged and the hardware is just a cheap winmodem with the resistor removed. If the phone line were plugged in there would not be a red alarm. Certainly any digital line will provide the same sort of status notifications. On Wed, Apr 30, 2008 at 3:07 PM, Vinz486 [EMAIL PROTECTED] wrote: Hi, i have 2 FXO ports on my host (Asterisk 1.4 on Debian).. In the production env i will not know if will be analog cable plugged in port 1, 2 or both. How can i discover this programmatically? I see NOTICE messages on CLI upon plug and unplug lines: ho get these info? Thanks. -- PicoStreamer - the real WEB live streaming software vinz486.com ___ -- 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] Zaptel Compatibility
/lib/modules/2.4.21-53.ELsmp/build [EMAIL PROTECTED] [/]# ll /lib/modules/2.4.21-53.ELsmp/build lrwxrwxrwx1 root root 35 Apr 30 04:48 /lib/modules/2.4.21-53.ELsmp/build - ../../../usr/src/linux-2.4.21-53.EL/ There's something wrong with this system usr/src/linux-2.4.21-53.EL/.build is missing and I get errors trying to do 'make cloneconfig' On Wed, Apr 30, 2008 at 4:57 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Apr 30, 2008 at 09:21:37PM +0300, Tzafrir Cohen wrote: On Wed, Apr 30, 2008 at 02:00:57PM -0400, Andreas van dem Helge wrote: Is Zaptel 1.4.10 compatible with RHEL 3 (2.4.21-53.ELsmp)? Because I can compile 1.2.20.1 just fine but 1.4 says: echo You do not appear to have the sources for the 2.4.21-53.ELsmp kernel installed. You do not appear to have the sources for the 2.4.21-53.ELsmp kernel installed. exit 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/zaptel-1.4.10' make: *** [all] Error 2 Yes kernel-source is installed, there is no kernel-devel. I read one account where if I use non-SMP kernel it might work. But there's no fun it that. 1.2 works why not 1.4? Failing getting 1.4 to work can I use Zaptel 1.2 with Asterisk 1.4? I think not but just wanted to make sure. Zaptel will look as the kernel source for (in this specific order) 1. Whatever you explicitly set in KSRC (if you did) 2. /lib/modules/$KVERS/build (if you set KVERS explicitly) 3. /lib/modules/`uname -r`/build 4. /usr/src/linux-2.4 5. /usr/src/linux 'build' in (2) and (3) is normally a symlink to the path of the kernel. I forgot to mention that there's an additional test done: the source directory found (KSRC) has to have a file called .config in it . Which is the first of those directories that you actually have? To better debug this, edit the Makefile. Find the line with that error message and add the word '$(KSRC)' (without quotes) to it. This should help you see what the makefile thought is the kernel source tree. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] OT: Polycom 3.0
How do they get away with that? On Mon, Apr 28, 2008 at 7:23 PM, Jonathan C. Bailey [EMAIL PROTECTED] wrote: Try the RPM from Trixbox. If you need something to open the file on Windows, 7zip works fine.. http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html -Jon - Original Message - From: Darrick Hartman (lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 28, 2008 5:18:26 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] OT: Polycom 3.0 Andreas van dem Helge wrote: Anyone have a download link for 3.0 SIP firmware? If you are going to say ask polycom or ask your vendor don't even waste your time posting. I've asked the Nazis and they'll probably take 1 week. Suggest you get a different vendor then. I got a response from mine within a few hours. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- 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] OT: Polycom 3.0
The thing is that's still the case. If they really wanted change they'd post the newest version of the firmware at www.polycom.com which they dont (well technically yes but you need to be a member of their reseller crap) The byproduct of the corporate bureaucracy,. Isn't it great? On Tue, Apr 29, 2008 at 8:54 AM, Eric Wieling [EMAIL PROTECTED] wrote: An amazing change from the old days when you could only get firmware from a Polycom authorized distributer. Jonathan C. Bailey wrote: Polycom is affiliated with the project in some way.. They also have an official Polycom moderated vendor forum. -Jon - Original Message - From: Andreas van dem Helge [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 29, 2008 3:21:30 AM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] OT: Polycom 3.0 How do they get away with that? On Mon, Apr 28, 2008 at 7:23 PM, Jonathan C. Bailey [EMAIL PROTECTED] wrote: Try the RPM from Trixbox. If you need something to open the file on Windows, 7zip works fine.. http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html ___ -- 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
[asterisk-users] OT: Polycom 3.0
Anyone have a download link for 3.0 SIP firmware? If you are going to say ask polycom or ask your vendor don't even waste your time posting. I've asked the Nazis and they'll probably take 1 week. Thanks, Andy ___ -- 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] Siemens Gigaset S685IP Review
AFAIK Siemens ceased distribution of their Gigaset line in North America a few years ago either you find a wholesaler that is importing grey market items or you buy it from a distributor overseas. On Sun, Apr 27, 2008 at 11:16 AM, Michael Graves [EMAIL PROTECTED] wrote: On Sun, 27 Apr 2008 13:17:11 +0100, Alan Lord wrote: Hi there, in case anyone is interested, I've just taken ownership of a small home network (3 handsets) of the brand new Siemens DECT PSTN/VOIP phone. It works great with Asterisk. Here's my overview and review so far... http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ Most excellent review! I've been waiting for the C6xxIP series to be available in the US. I hope that it happens soon. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- 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] Manual Wardialer
So I can't dial my own number blocks for auditing? I do this manually right now dial 1 number, dial another on and on it gets very tedious and sometimes you loose your place. Approx every 2 months per number. The companies using these numbers have very specific reasons for requiring these audits, but franky I don't think its needed. AFAIK in my state doing that is legal because: 1) Its not telemarketing 2) its with the intent to communicate (if someone answers an 3) its for a legit business purpose, so its not harassment 4) The owner of the numbers (my company) and the users of the number (the clients) have expressly authorized this, although the law does not mention authorization I think this would be justification enough. I am not familar with any FTC / federal regulations since we don't telemarket I didn't think they were relevant but you do remind me when anything crosses a state line it can usually be considered interstate commerce... any resource you might have for interstate phone calling laws? I was thinking VCDial too... let me give that a try I've always wanted to mess with it anyways. I think I could load all the number ranges at one time also instead of doing one range at a time like I was thinking. And yes this is not war dialing because I looked up the definition and it seems war dialing is just scanning for modems, which is not the case here. On Sun, Apr 27, 2008 at 9:23 AM, Matt Florell [EMAIL PROTECTED] wrote: Hello, Sequential auto-dialing like this is pretty much illegal in the USA. The FTC has specific regulations against this as well as several states. Obligatory Simpsons reference: http://www.internerd.com/frink.retired/frinkv.3/inventions/at5000-2.gif http://www.snpp.com/episodes/4F01.html My servers generally don't have built in legs or otherwise any way to automatically relocate itself :) ___ -- 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] Manual Wardialer
Does anyone have a script for manual wardialer for asterisk? not sure if wardialer is the correct term but basically I want to call X number say 555- through 555-0050 and be able to listen to each call and when I hang up or press a key it will dial the next number for me. I guess sort of like scanning an exchange but I want to be on the line and if possible complete / talk on certain calls. ___ -- 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] G729 license count...
On Fri, Apr 18, 2008 at 3:53 PM, Godwin Stewart Horwich IT Services [EMAIL PROTECTED] wrote: On Fri, 18 Apr 2008 08:37:32 -0800, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: If you care to use ping pong balls and the atlantic ocean as your medium, you should be able to interface with the g729 codec if you still needed to :D I've heard that RFC1149-compliant devices work well with g729 as well :) And how did you manage to get the latency down to a tolerable level? I mean... it's a great idea an all but when I'm talking to someone on the phone I like to hear their response that same hour and using RFC1149-type connections just makes the VoIP latency unbearable for me. Have you run this setup in production with any clients? What have been their thoughts on it? ___ -- 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] Need some input for Quad T1 and channel banks.
I want to 3rd this. They admitted some of their hardware runs GPL code (Linux, IPTables, wget and more) yet refuse to provide the source code or evidence of an alternate license agreement with the authors of the software (which I doubt they did I just like to give people that benefit of the doubt). But I do think their engineering is excellent. What a waste. On Thu, Apr 3, 2008 at 1:26 AM, Alex Balashov [EMAIL PROTECTED] wrote: Al lists wrote: Bad memories from AudioCodec :) Por que? I'm curious. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Zaptel data mode not supported?
On Sun, Apr 6, 2008 at 1:52 PM, Alex Kauffmann [EMAIL PROTECTED] wrote: Thank you for the replies. It was my understanding that rebuilding the kernel was necessary in 2.4 but everything needed was already included in 2.6 series. *HEAVILY* dependent on the distribution! ___ -- 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] interrupting MOH
I think that's still a better idea than using a dump the caller into meetme hack and is actually what I was going to suggest. If you want something simpler than a queue then inject the sounds into the moh already. On Tue, Apr 1, 2008 at 3:09 PM, Rob Hillis [EMAIL PROTECTED] wrote: You may be able to achieve the desired result using queues rather than Dial statements. Overkill perhaps, but it's the only way I can think to implement it at the moment. John Millican wrote: Tilghman Lesher wrote: On Tuesday 01 April 2008 05:14:25 Pete Kay wrote: I am hoping someone can help me out on this. I want to be able to interrupt MOH every X seconds after the DIAL command is executed. The interrupt greeting is something like please wait while we transfer your call. How can I do that? Within the DIAL options, I can't see any announce frequency or options that can help. Could anyone please tell me how that function can be accomplished? The only way to do that currently is to implement the prompt within the MOH stream itself. Just off the top-o-my head(YMMV), couldn't you create a meetme and play hold music into the meetme and then also play the prompt into the meetme at the same time without interrupting the hold music? This would obviously not work for high load but... 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 ___ -- 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] Newbie Polycom: DND answered as on the phone instead of not available
What is your extensions.conf setup? that has alot to do with it (I strongly suggest you use macros.) What SIP NNN code does the phone return when DND? On Mon, Mar 17, 2008 at 2:00 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: I am using Polycom IP600 phone. If I call a phone which has DND (do not disturb) enabled, the message to the caller will be The person on extension ... is on the phone, please leave a message Is there a way to pick the person ... not available message instead? ___ -- 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] Call deflection on ISDN PRI in Sweden
*CLI show application Transfer
-= Info about application 'Transfer' =-
[Synopsis]
Transfer caller to remote extension
[Description]
Transfer([Tech/]dest[|options]): Requests the remote caller be transferred
to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only
an incoming call with the same channel technology will be transfered.
Note that for SIP, if you transfer before call is setup, a 302 redirect
SIP message will be returned to the caller.
The result of the application will be reported in the TRANSFERSTATUS
channel variable:
SUCCESS Transfer succeeded
FAILURE Transfer failed
*** UNSUPPORTED Transfer unsupported by channel driver ***
So what you need to do is use app_dial instead of app_transfer.
Everything else should be able to remain the same.
On Fri, Mar 28, 2008 at 4:25 AM, Hanna Wallin
[EMAIL PROTECTED] wrote:
Hello List!
We're having trouble making call deflection on ISDN PRI. We would like to
transfer a call to an external extension but keeping the callerid of the
caller so it can be presented to the receiver of the transferred call.
At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware
TE420B. We've ordered the service (CD) from the phone company.
The zapata.conf file inlcludes:
Transfer= yes
Facilityenable=yes
Callerid=asreceived
In extensions.conf we try to transfer a call to an external extension as:
Transfer(ZAP/g0/ ) but that fails with the ${TRANSFERSTATUS} =
UNSUPPORTED.
Ideas anyone? We would really appreciate it!
Kind regards,
Hanna
Hanna Wallin
System Development
Direct: +46 (0)8 736 77 29
Mobile: +46 (0)73 414 13 38
Fax: +46 (0)8 736 77 91
E-mail: [EMAIL PROTECTED]
PocketMobile Communications AB
Wenner-Gren Center
Sveavägen 168, 3 tr
113 46 Stockholm
Nordic web page: www.pocketmobile.se
International web page: www.pocketmobileworld.com
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Re: [asterisk-users] IAXy device
It's not bad in the sense of stability (well the original ones are claimed to have overheating issues..). But its that it lacks ANY features. The IAXy has no features at all. Also no security, it MUST be placed behind a firewall, as the configuration doesn't have any sort of security whatsoever. Did I mention it has no features besides DHCP? Not even DNS. Also it's very expensive. I could understand if it was a full-featured device with a webinterface, DNS support 2 Ethernet phone ports I wouldn't complain of the price. But it was released at approx USD $100 at a time when most full-featured adapters sold for a little less, and still sells for $90 today. If they sold them for $40 I wouldn't bash them either.. because honestly thats what they really should be worth. I'd rather use a Grandstream HT than an IAXY honestly. On Thu, Mar 27, 2008 at 3:08 PM, Steve Totaro [EMAIL PROTECTED] wrote: I had a customer using an IAXY (old gen) for an FXO fax machine and it worked almost all the time so it cannot be that bad. Maybe because the fax was very old and did not have high transmit rates. Thanks, Steve Totaro On Thu, Mar 27, 2008 at 2:11 PM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: I guess I've never run asterisk without ANY echo cans :) It's just that the echo was minor enough that MG2 et. al did a fine job. Thanks! Moj Eric Wieling wrote: You will never get latency on a network low enough for echo to be perceived as sidetone (like on analog). If you want to get rid of echo you must cancel echo. Mojo with Horan Company, LLC wrote: Sean Dennis wrote: bilal ghayyad wrote: Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal The device has no echo cancellation and sounds horrible (lots of echo) on about half of the analog phones I tried it on. I wouldn't recommend it unless you absolutely need IAX. It's also very expensive for a 1 port ATA. ___ -- 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 Echo may be the result of latency on the network. I've not had any echo problems that I remember with my IAXy and I make ten calls a day, five days a week, for the last few years, to all sorts of numbers/areas. I know that this isn't representative of typical business use, but residential use, but I've been using in my business and have never been disappointed :) I will agree that's is fairly expensive, but I WOULD recommend it to people who are on the go often. After setup, it really is plug-n-play IMO. Moj ___ -- 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 ___ -- 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] Anyone know of a pass through ATA
SPA1001 is 1FXS only. SPA3102 is 1FXS + 1FXO... if you don't need FXO why not get the SPA2102? It should be cheaper and you have the extra port for future use. On Sat, Mar 15, 2008 at 12:08 AM, Thermal Wetland [EMAIL PROTECTED] wrote: That is good to know. I will be using the the device to connect a bunch of analog phones back to a centralized server. Since each office has only one ethernet connection do I want to: 1. Install a small switch to let a SPA1001 PC use the one drop 2. Install a SPA3102 to let the PC share the one drop Even with an extra patch cord and probably the occasional power strip number one will be cheaper, but #2 seems like a better way to go. -Thermal On Fri, Mar 14, 2008 at 6:41 PM, [EMAIL PROTECTED] wrote: its not a bad device - I have 2 problems with it. It doesn't do echo cancellation very well is particularly badly matched to the PSTN here in Oz. Hint: keep it well cooled - echo goes up badly when its hot it runs very hot if there is no ventilation. I use the 3102 to bridge a mythtv box instead of putting in an extra switch - works except for the occaisional failure to get a dhcp address. I use a linux gateway for dhcp, most devices (3102+mythtv box, lynksys PAP2, bt100 ip phones, wireless and hosts) are all dhcp -Original Message- From: Thermal Wetland [EMAIL PROTECTED] Subj: Re: [asterisk-users] Anyone know of a pass through ATA Date: Sat 15 Mar 2008 10:23 Size: 2K To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com That is awesome. I dont know why the manual doesnt mention that. I want to have the device use a static IP the computer use DCHP from a central DHCP server...sounds like it wont be a problem. Thanks. On Thu, Mar 13, 2008 at 8:14 PM, W.Kenworthy [EMAIL PROTECTED] wrote: sipura 3102 set to bridge. Works but I find that when rebooting a PC bridged it sometimes (randomly) doesnt get a dhcp lease, necessitating a powercycle of the 3102. I think the PC drops the ethernet as it reb oots and the sipura doesnt recognise it coming back. BillK On Thu, 2008-03-13 at 19:59 -1000, Thermal Wetland wrote: Anyone know of a company that makes a pass through ATA? By pass through I mean have an Ethernet switch built into the ATA, like most desktop phones have. All of the dual ethernet ATAs I have seen have WAN/LAN ports, not two LAN ports. I fooled around with DMZ etc...but it just doesnt work as well. Thermal ___ -- 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 ___ -- 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] Mail Server
tail /var/log/mail or /var/log/maillog On Thu, Mar 13, 2008 at 5:04 PM, Mike Hammett [EMAIL PROTECTED] wrote: I need to setup a small mail server on a local network. It only needs SMTP ability as it's just so Asterisk can send out emails. The machine has sendmail installed. My primary mail server seems to be rejecting the messages. Some research says something isn't configured properly. What do I have to do so the outside world accepts emails from my Asterisk box? It is behind a NAT. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- 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] Help: DTMF problem
1) Use RFC2833 2) Make sure all devices are properly configured 3) Try another provider. On Thu, Mar 13, 2008 at 4:54 PM, Jarga Jallow [EMAIL PROTECTED] wrote: Hi, I have polycom 301 IP phones most of them especially when I call a direct line with extensions, I cannot dial an extension. It does not recognize my key inputs. If the number is an 800 number it seems to work fine. I have used dtmfmode=inband with my sip trunks and my extensions as rfc2833. Any suggestions will really be appreciated. Thanks in advance Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 www.2mcctv.com ___ -- 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] Anyone know of a pass through ATA
Linksys SPA2102 does. It even has the option to auto-detect so if it is assigned an RFC1819 address it will act as a switch and otherwise just as a NAT router. So does Grandstream HT496 (and I'm sure others) but it must be manually configured. On Fri, Mar 14, 2008 at 1:59 AM, Thermal Wetland [EMAIL PROTECTED] wrote: Anyone know of a company that makes a pass through ATA? By pass through I mean have an Ethernet switch built into the ATA, like most desktop phones have. All of the dual ethernet ATA's I have seen have WAN/LAN ports, not two LAN ports. I fooled around with DMZ etc...but it just doesn't work as well. Thermal ___ -- 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] Call tracing - Asterisk 1.4
On Tue, Mar 11, 2008 at 9:41 AM, Louwrens Benadé [EMAIL PROTECTED] wrote: Better reporting through a new call event logging capability in Asterisk 1.6 will allow complete tracking of events that take place during a call. The goal, according to Fleming, is to provide more detail than traditional CDR (Call Detail Recording) features offer and to allow for more granular tracking and auditing. Sounds semi-CALEA-compliant... ___ -- 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] T.38 SIP Issues
Asterisk receives T.38 RTP packet from one SIP peer and sends it to the other SIP peer, it does not process the packets. By your argument I can't do T.38 @ 1440bps unless the manufactures of the Ethernet cable, switch, router, keystone jacks, network cards, CPU, RAM, etc all paid for the royalties for the T.38 patent. It's like G729 pass-thru Just the endpoints need to have the codec. On Fri, Mar 14, 2008 at 7:58 AM, Mindaugas Kezys [EMAIL PROTECTED] wrote: Hello, Higher speeds then 9600kbps are not permited by patents. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing Solution for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: Friday, March 14, 2008 3:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T.38 SIP Issues Has someone submitted a bugreport regarding enabling 9600kbps fax? I always wonder why it would never negociate 14400kbps... when it did work a single page on fine resolution would take 4 minutes. Thank you very much for that link. I knew there had to be more possible configurations for T.38. I will give it a try... but I think I can get away without patching chan_sip.c, no? that just seems to enable higher bitrates. And Linksys SPA2102 is one of the exact devices I have in my lab. On Thu, Mar 13, 2008 at 1:52 PM, Mindaugas Kezys [EMAIL PROTECTED] wrote: Hello, This can help: http://80.86.84.71/kolmiwiki/index.php/Send_Receive_Fax-T38 Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: Thursday, March 13, 2008 5:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T.38 SIP Issues Is there any trick to getting T.38 fax to work with SIP? I had it working and one day with no changes *poof* it stopped working and hasn't worked for months. The only common factor is Asterisk 1.4.x (always try to use the latest version) and NAT. I've tried: -Linksys ATA -Grandstream ATA -Audicodes ATA All do the same thing. Call connects, hear the first 2sec of fax tone and then just silence, but the call usually stays open. I've tried two T.38-capable providers. I've tried two different routers: -Linksys WRT54GS running DD-WRT (Linux) -Dell Optiplex 170L running PFSense (BSD) Different Linux distros on the servers: -SuSE 64bit -RHEL 32bit -SuSE 32bit Is there any magic to get this to work? As far as I can tell the only possible config option is t38pt_udptl = yes which I have set under [general] the peer. ___ -- 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 ___ -- 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] Anyone know of a pass through ATA
I know about them... KCL.net in Miami does (did?) the same assigning 10.x address for basic home connections... What's your point? It's configurable on the 2102 as switch/nat/autodetect, ht496 as nat/switch. If you have such ISP and want to use the device as a router just manually set the NAT option. On Fri, Mar 14, 2008 at 11:47 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Fri, Mar 14, 2008 at 02:32:54AM -0400, Andreas van dem Helge wrote: Linksys SPA2102 does. It even has the option to auto-detect so if it is assigned an RFC1819 address it will act as a switch and otherwise just as a NAT router. Clearly, someone neglected to tell them about ISPs like Rose.NET in Thomasville, GA, who assign -1918 addresses to their customers over DSL and cablemodems. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] T.38 SIP Issues
Has someone submitted a bugreport regarding enabling 9600kbps fax? I always wonder why it would never negociate 14400kbps... when it did work a single page on fine resolution would take 4 minutes. Thank you very much for that link. I knew there had to be more possible configurations for T.38. I will give it a try... but I think I can get away without patching chan_sip.c, no? that just seems to enable higher bitrates. And Linksys SPA2102 is one of the exact devices I have in my lab. On Thu, Mar 13, 2008 at 1:52 PM, Mindaugas Kezys [EMAIL PROTECTED] wrote: Hello, This can help: http://80.86.84.71/kolmiwiki/index.php/Send_Receive_Fax-T38 Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: Thursday, March 13, 2008 5:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T.38 SIP Issues Is there any trick to getting T.38 fax to work with SIP? I had it working and one day with no changes *poof* it stopped working and hasn't worked for months. The only common factor is Asterisk 1.4.x (always try to use the latest version) and NAT. I've tried: -Linksys ATA -Grandstream ATA -Audicodes ATA All do the same thing. Call connects, hear the first 2sec of fax tone and then just silence, but the call usually stays open. I've tried two T.38-capable providers. I've tried two different routers: -Linksys WRT54GS running DD-WRT (Linux) -Dell Optiplex 170L running PFSense (BSD) Different Linux distros on the servers: -SuSE 64bit -RHEL 32bit -SuSE 32bit Is there any magic to get this to work? As far as I can tell the only possible config option is t38pt_udptl = yes which I have set under [general] the peer. ___ -- 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
[asterisk-users] T.38 SIP Issues
Is there any trick to getting T.38 fax to work with SIP? I had it working and one day with no changes *poof* it stopped working and hasn't worked for months. The only common factor is Asterisk 1.4.x (always try to use the latest version) and NAT. I've tried: -Linksys ATA -Grandstream ATA -Audicodes ATA All do the same thing. Call connects, hear the first 2sec of fax tone and then just silence, but the call usually stays open. I've tried two T.38-capable providers. I've tried two different routers: -Linksys WRT54GS running DD-WRT (Linux) -Dell Optiplex 170L running PFSense (BSD) Different Linux distros on the servers: -SuSE 64bit -RHEL 32bit -SuSE 32bit Is there any magic to get this to work? As far as I can tell the only possible config option is t38pt_udptl = yes which I have set under [general] the peer. ___ -- 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] Pager (beeper) Emulation Script
Does anyone have a script that will emulate a normal numeric pager but send the number to an email address? Also anyone happen to have the traditional tones used in North America? ___ -- 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] ata device but for a soundcard
What are you trying to accomplish exactly? They sell SIP overhead speakers or you can use a SIP phone with an adapter on the 2.5mm headset jack. On Wed, Feb 20, 2008 at 2:44 PM, Jerry Geis [EMAIL PROTECTED] wrote: I am looking for an ATA like device but instead of VOIP to analog phone I want VOIP to low level audio out. Something that looks like a sound card output. I know I can use cheap PC's but that then you have HD's to setup etc... HD failures etc... Anyone know of something like that? Jerry ___ -- 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] Grandstream GXP2000 Loses Connectivity
I've had the opposite problem. Press mute while the call is still ringing and it will say MUTE on the display but the microphone is not muted. It was very embarrassing to discover this bug. On Wed, Feb 13, 2008 at 2:03 AM, Thomas Kenyon [EMAIL PROTECTED] wrote: Lutgring, Sam wrote: I take it you've also not had the problem where the handset microphone stops working. (This is apparently already fixed and will be available in the next beta firmware release (1.1.6.x), when they've fixed some more problems that have been very difficult to track down.) I'd like to go back to 1.1.5.15 if nothing more than for the improved audio quality and the on-hook dialling. ___ -- 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] Hardware requirements
On 10/15/07, Doug [EMAIL PROTECTED] wrote: Case: 1 CodeGen 4U Server Case $80 http://tinyurl.com/bnobz http://tinyurl.com/95s2b http://www.newegg.com/Product/Product.aspx?Item=N82E16811182566 Or: 1 Eagle Tech ET-RMAL2025-SL Beige 2U Server Case 2 External 5.25 Drive Bays http://www.newegg.com/Product/Product.aspx?Item=N82E1687111 Power Supply: 1 Dual 450 W. power supply -- IStar https://www.ewiz.com/detail.php?name=PS-TC50R8A http://www.directron.com/tc400r8.html Or: 1 535W power supply -- Enermax https://www.mwave.com/mwave/viewspec.hmx?scriteria=BA23110 Motherboard, CPU 1GB of memory: AMD ATHLON 64 X2 5000+ (ADO5000DDBOX) ENERGY EFFICIENT RETAIL BOXED W/512KB X 2 CACHE 65NM 65W (BRISBANE) BUNDLE W/ ASUS M2NPV-VM Don't get me wrong, the M2NPV are great boards we use them all the time for home appliances type devices they run 24/7 and process alot of media. And also for frontend because they have the HD video outputs. However I'd prefer to use a server mainboard for dedicated Asterisk systems. I've had great luck with the Tyan bareones systems. Or SuperMicro is great too. Its too bad you can't find any cheap 4U barebones... 1U $500.. 4U $1200 it makes no sense. Spend a few more dollars on a server or workstation mainboard I've found in 2 years reliability is greater. ___ --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] really sorry about this - E1 vs T1
On 10/11/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: Yes, see the t1e1override module parameter in wct4xxp/base.c. IIRC, it's 0xff to hard code to E1 mode, and set it to 0 for T1 mode. -1 is to use the jumper settings. Seems like a bad design. Why not just make it a software choice?? ___ --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] Remote voicemail in two Asterisk
http://en.wikipedia.org/wiki/Network_File_System_(protocol) On 10/12/07, Pepo [EMAIL PROTECTED] wrote: Using two Asterisk connected between they, How do I can check the voicemail in a remote system but working like *97? I mean dont want ask the voicemail box, just the password and go to the voicemail of caller. If I have the same extensions in the two Asterisk it doesn't work. Thanks. -- Linux User Registered #232544 Jabber : [EMAIL PROTECTED] Ekiga : [EMAIL PROTECTED] ICQ : 337889406 GnuPG-key : www.keyserver.net --- dum loquimur, fugerit invida aetas: carpe diem, quam minimum credula postero. ___ --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] Distributed FAX - How to best complement asterisk ?
On 10/10/07, Ex Vito [EMAIL PROTECTED] wrote: Hi list, I'm evaluating a private telephony scenario of about 20 locations - 300 phones, 50 FAX machines. More than 1 PRI? All other locations, small by themselves, would get SIP phones managed by asterisk, since there is good IP connectivity between all sites. Private network? How good? How saturated? Could be possible to just run ulaw if the quality is as good as your LAN 1. On the locations where asterisk is installed, the solution is trivial; either by connecting FAXes to FXS ports on channelbanks or by managing faxes with iaxmodem + Hylafax. Probably a combination of both... Why channel banks? 2. On the remaining locations we have a problem b) T.38 is the answer to FoIP c) asterisk 1.2 does not support T.38 d) asterisk 1.4 only does T.38 passthrough, not good enough Use a VoIP provider with t.38 for your faxes... easy solution. e) CallWeaver seems to support T.38 gatewaying, although I'd rather move on with asterisk so as to leverage current experience and knowledge and to keep installed base with the same software. I've been waiting for callwaver 1.2 final for a while. Tried some betas and T38 gateway didnt work even when we put a Sangoma card in the machine. Problem was on the SIP side. [PSTN] ---PRI--- [asterisk] ---PRI--- [PRI-to-T38 GW] ... ... --SIP/T.38--- [T.38 ATA] ---FXS--- FAX machine Too many PRI... Try: PSTN ---PRI AS5300 --SIP- Asterisk 1.2 PSTN ---PRI AS5300 --SIP- Asterisk 1.4 -SIP T.38 ATA PSTN ---PRI AS5300 --SIP- T.38 ATA 4. Of course, I could use CallWeaver as a PRI-to-T.38 gateway... But then again, how solid would it be ? With which ATAs ? The CallWeaver website shows a very small amount of ATAs confirmed to be 100% working in T.38. There's a reason why CallWeaver is beta. As much as I'd love to support their stuff. It's still in beta. 5. Would I need to have a SIP proxy between the PRI-to-T.38 gw and the T.38 ATAs or would they be able to talk to each other directly ? (I'd say this would depend on the specific equipment, but...) If that would be a requirement, which way would you go, asterisk 1.4 ? Would SER forward T.38 traffic ? SER is a SIP proxy. T.38 is irrelevant to it. I'd use 1.4, your setup seems pretty straightforward. You don't have a diverse population of SIP phones and locations to manage. ___ --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] Good Book to learn SIP
So I'm the only person that actually enjoys reading the RFC's? On 10/7/07, Brian West [EMAIL PROTECTED] wrote: Telling someone to read the RFC bah.. might as well give them a blanket and pillow because they will fall asleep. chan_sip is just ugly in every way. /b On Oct 7, 2007, at 9:26 PM, Baji Panchumarti wrote: http://www.faqs.org/rfcs/rfc3261.html as well as the source in asterisk (1.4.11 here) asterisk-1.4.11/channels/chan_sip.c ___ --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] Dry Copper Pair
On 5/13/07, Jon Pounder [EMAIL PROTECTED] wrote: what exactly was the charge ? - trespass - no its public land for the most part this stuff is on so that doesn't apply - vandalism/mischief - if no other customer was impacted I don't see how this charge would stick since there is no measurable damages. - theft of service ? Going rate for dry copper is under $20/month/pr so to get up into the 5-10k level that might justify a higher level theft charge with jail time that would take some time to add up. Stealing cable TV/satellite probably works out to about 3x the monthly rate of dry copper and I have never heard of anyone being told anything more than disconnect it when they get caught. Trespass -- the cross-boxes belong to the telco, not you. The telco did not grant you access to their cross-box, now did they? If you think they did, ask yourself if they would give you that in writing? I The other issue is what crime would be involved in assisting the telco to deliver a better level of service by doing work yourself ? None, but keep in mind that trespass (access to proptery thats not yours and you have not specifically been grated access to) and theft of service are crimes. Fixing something is not a justification. If someone kills your friend and your state has the death penalty its not legal to do that yourself, there are channels through which things need to be dealt with. For example I often do as much work on their side of the demarc as possible when I have an order pending, then I know its done the way I would have wanted it. I have never got anything other than a thank you when the installer shows up and I just tell them where to make the final connection. Technically the demarc is telco property but it is placed on your property. Unless its an MDU or something thats fair game. The other issue that hasn't even been touched in this thread is how easy it is to just tap someone's line when everything is so exposed like this. The tap might get found, but if it was a line powered radio transmitter, chances of tracing back to the installer are minimal unless someone saw it get installed. Well how easy is it to kill someone? With a gun all you need is a little force on the trigger. Does it make it right? ___ --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] Get sip response code
On 5/16/07, Robert Lister [EMAIL PROTECTED] wrote: I was wondering if it is possible (in 1.2.x) to get the SIP response code back after doing Dial(). Dial() seems to treat most call-setup problems as dialstatus CONGESTION, and some are NOANSWER, but I want to know the SIP response code, so I could return the right tones to the user, not just a congestion tone for every fault. Anyone know a way to find out that information, so I want the 604 out of this lot: vim chan_sip.c /Got SIP response (Hint: there are a few references, they start around line 10.000) Or where do I need to look to find a SIP response code - DIALSTATUS mapping? Are these configurable anywhere or are they hardcoded? In app_dial.c all it does is: pbx_builtin_setvar_helper(chan, DIALSTATUS, status); It shouldn't be too hard to add another variable for SIPDIALSTATUS or whatever name you fancy. If I push the response code back to the handset (Cisco 7960) then it is even more unhelpful as it uses the same error message for all SIP error type response codes: Reorder but does not tell you why the call failed to set up. If it actually put the SIP response error on the display, that would be good, but it doesn't. (at least SIP 8.6 and prior software versions) That seems like a bug you should address with the vendor. ___ --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] GXV-3000 IP Video Phone
On 5/5/07, dave cantera [EMAIL PROTECTED] wrote: nitesh, you are correct. you need 1.4.x... daveC It is supposed to have H.263, which does work with 1.2.x: [general] ... videosupport=yes .. [video-enabled-sip-phone] ... canreinvite=no disallow=all allow=ulaw allow=h263 ... ___ --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] T1/E1 Configuration
Post the output of these commands on the Asterisk CLI zap show channels zap show status pri show span 1 On 5/4/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks Everyone for the help... Got the T1 UP and insvc with Cisco AS5350, but I am failing to send the call. On the Cisco side I do not see any incoming call and on Asterisk side I get message saying Channels unavailable, while all channels are available. Can anyone post a working configuration for Asterisk T1 and Cisco conf? Please... Thank you. Regards, Nitesh Tzafrir Cohen wrote: On Fri, May 04, 2007 at 02:28:32PM -0400, Andreas van dem Helge wrote: On 5/4/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks John, How can I change my conf to NETWORK? Where can I find this information? #signalling = pri_cpe signalling = pri_net nitpicking: ;signalling = pri_cpe signalling = pri_net (The comment character is ';' . '#' is reserved for special directives of the sort of #include) ___ --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] CDR changes in 1.4.3?
On 4/27/07, Steve Murphy [EMAIL PROTECTED] wrote: I'm the guilty party. I've been trying to fix several CDR bugs, involving stuff like missing times, missing changes in state (like NO_ANSWER when the call was ANSWERED), etc. Now that we are talking about CDRs, I must ask: in 1.2.x if the CDR is forked into two the uniqueid is the same for both CDR records. Is that the intended behavior? Does that remain in 1.4.x? CDR's are complicated by the fact that they record 3 events: start, Answer, and end events. Add to that the fact that in most cases at least two channels are involved, sometimes 4 or 5, or even more, involving bridging, maquerading, parking, transfers, local channels, AGI, conferences, and more... Some cases were impossible to fix unless CDR's were attached to every channel, and merged to collect the bits and pieces that sometimes were on the wrong side of the bridge. It would be nice if the CDR engine could be configured to allow for these transactions either to be merged or not and what to do with the bits and pieces as you describe them. However it would seem logical that if various pieces are merged then ultimately they should not be logged as that would be redundant... However I'd rather see it be a configuration choice. The result is that several more cases are more accurate, but also, that rather uninteresting CDR's can be generated. In contemplating what could be done to get rid of some of these, I sometimes have to ask, is this truly something we have to get rid of?... In the meantime, uninteresting CDR's with NO_ANSWER and billsec=0, should be easy to filter out, right? I don't think CDRs with NO ANSWER disposition or billsec=0 should be discarded. Why not make it configurable? I will, in the coming days, look at some of the extraneous CDR's that are generated, and see what I can do to get rid of them. It's not always that simple. If we ring a phone, for instance, and no-one answers it, is that truly, really, something that no-one will ever be, could ever be, interested in? (just a fer-instance). From a billing standpoint no whats the point? For statistical purposes I think its useful. For VoIP serviceprovider also very useful customer probably wants full call logs. I don't think your idea is too much CALEA-compliant either. I welcome your input. Complain up a storm. I'll try my best to make you happy. Make it configurable. ___ --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] T1/E1 Configuration
On 5/4/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks John, How can I change my conf to NETWORK? Where can I find this information? #signalling = pri_cpe signalling = pri_net ___ --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] IP Phone Provisioning Tool by voip.com.sg - xml generation
Here's something similar for Linux: http://sourceforge.net/projects/vgps/ Note I do not support nor endorse Voicepulse. Actually let's get it straight, I detest Voicepulse. On 5/3/07, Stephen Bosch [EMAIL PROTECTED] wrote: Mats Karlsson wrote: Take a look here: http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html Ugh. This is a Win32 app, isn't it? -Stephen- ___ --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
