Re: [Asterisk-Users] mini-ITX suggestions
http://www.voip-info.org/tiki-index.php?page=Asterisk+hardware+mini-itx New Wiki page on mini-itx with Leo Anne's comment added to it. Please help us collect experiences on Mini-itx configurations there. Seems to be a lot of interest in this hardware. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
Michael Graves wrote: Please forgive me if this is a silly question. I've been following this thread in the hope that I could put my * server and snom 200 into full-time service very soon. I need to find out how to have the lines Olle Wrote: I've opened a bug http://bugs.digium.com/bug_view_page.php?bug_id=732 Let's continue adding information there. The best way to follow up is to check the bug report, usually when we open a bug, debate goes on in the bug tracker system to make sure we have all information on testing, patches and possible new errors in the same place. There's no solution yet, but some discussion and some coding experiments. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI, transfering thru a 2nd PBX - keep original CallerID
Good day,
I want to have Asterisk as my gateway to the outside world and use
another PBX to connect my existing phones.
How do I specify the dial string to forward the original Caller ID to
over the ISDN to the second PBX?
Right now, my extensions.conf looks like this:
exten = ${OUTSIDEMSN},1,Dial,CAPI/${MSN2NDPBX}:${EXTEN}
How do I transfer the caller Id information initially coming in?
Any and all help is greatly appreciated.
rgds
pos
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[Asterisk-Users] TDM400P X101P cards, echo issues?
Let me start by saying no, this is not the normal stupid echo question :) I currently have a SIP device and an X101P, and have had the usual echo issues and have played around with the various solutions, none of which are quite perfect (understandably).. however, my girlfriend is a little more picky than I am when she uses the phone, so I'm considering a Digium TDMXXB card for the house. My questions to everyone here are: 1) with an x101p and TDM card in the same box, have you noticed any echo issues 2) if yes, how bad does it seem? Do the echo cancellers in zaptel/asterisk do a good job of killing it, without causing issues? I see the default setting for echocancelwhenbridged is no, which gives me some faith in going the TDM route instead of ATA route, but I'd like to hear some sugguestions. Also, I haven't been able to find any real documentation on the various echo cancellers in zaptel. Can anyone elaborate on the differences between the various mechanisms? (ECHO_CAN_) Finally, I tried recompiling zaptel with -DAGGRESSIVE_SUPPRESSOR, but that seems to cause more harm than good in my particular setup (if both parties are talking at the same time, results are indeterminate). I know echo is caused by a wide range of issues, a wide range of problems, and everyone's experience has been different. I'd like to get the most feedback possible, and it would probably be better to address replies directly to me ([EMAIL PROTECTED]) to avoid cluttering the list unless your answer will directly benefit us all :) Thanks! Pat ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware to build an Enterprise Asterisk Universal Gateway
Does anyone know what the hardware requirements would be to build an Enterprise Asterisk Universal Gateway ? I am thinking of something comprable to the Cisco AS5xxx Series of gateways. Thanks Expand your wine savvy and get some great new recipes at MSN Wine. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modem Communications thru *
Happy New Year, I have a project to pass modem calls through * convert them from IP to X.25 and then allow the modems at each end to talk thru the rtp stream to each other before calling modem terminates the call. Datamodem --- FXS(non *) ADSL --- * -Software Application X.25 modem While the above occurs I also wish to capture the data stream and also parse selective parts of these transactions into a database. My questions are: Should I start trying with G.711 (the modem speed is 1200Baud)? I want to use monitor app. to capture the voice stream and later convert it using text to speech? I don't control the modems and part of the stream is encrypted the rest is just text - strings and numbers Is there an easier way to do this? TIA Fred __ Do you Yahoo!? Find out what made the Top Yahoo! Searches of 2003 http://search.yahoo.com/top2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk.
I agree in stopping the thread, but I do have one question... What would Qwest think of her posting to the list under a yahoo mail account representing her company, badmouthing this community, who, in the long run, could be VERY much worth their interest? Hmm Just my $.02 - no flames please. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Balaji NJL Sent: Saturday, January 03, 2004 8:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk. Hi All, Can we stop this thread pl. This lady has no intentions to learn asterisk. She is just a troll and wasting our time. With her corporate attitude, what she expects is support that available with paid commercial products. Her company has enough money to buy commercial products, let she go there. Hey lady, whoever u are, dont waste our time. this is not for u. Lets move on to something useful pl. -B - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 03, 2004 5:36 PM Subject: RE: [Asterisk-Users] New to asterisk? RUN... don't walk. On Sat, 2004-01-03 at 14:12, Me wrote: Mr. West, Sorry to burst your bubble, but that is not me. My name is Barbara Simpson. Either you are lying or someone is trying to remove any credibility from my original post. I now stand by my original post with more conviction than ever. You had little to no credibility when you show up acting like a troll from what most people would consider a throw away account. There were a lot of insightful replies. However, none of them were able to address the real problems of the asterisk community and come up with solutions. If you can't see your own faults, you are in for a bumpy ride. This is due to the problem residing in the general population, not the community. The problem resides in users who can't be bothered to either expend energy, or patience for the software to develop. Remember you came here, we didn't go recruiting you. So if you are disappointed in your experience, blame yourself for your expectations. As far as I can tell here, you haven't paid a single person for anything, so any help you have received has been at a cost to the other people of this community. So the solution is for you to grow up. You need to learn that the comment you have made in this thread are worthless as they don't advance anything here. If you want credibility in a technical forum, you will have to show some technical skills. Otherwise you will be cast aside and hopefully ignored. Barbara Simpson Qwest Voice Over Packet Services --- Brian West [EMAIL PROTECTED] wrote: You said it good Look what this person posted to my blog... Now thats what I call grown up. Date: Thu, 1 Jan 2004 10:10:24 -0600 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] IP Address: 24.10.200.168 Name: Jeff Sowery Email Address: [EMAIL PROTECTED] URL: Comments: You're a complete idiot. Grow a brain or at least some balls. -Jeff NEXT!!! bkw On Thu, 1 Jan 2004, JR Richardson wrote: Piping in 2 cents, This is a great example of the Internet, Fast Food generation, showing their appreciation for all the magic that happens in the labs, hearts and minds of the courageous, hard working, dedicated and motivated group of people truly interested and guided to accomplish greatness. This platform for learning is one of the best tools in existence to come to a finite understanding of VoIP and legacy telephony with the versatility to expand beyond and develop originality in the field of telecommunications excellence, product development. Learn it, understand it, appreciate it, then take it past where you found it and if you're capable contribute, if not, enjoy it. But always, always maintain respect for those who created it and continue to refine it. Learning is intrinsically human, and in this world of Industry (There is no substitution for knowledge. [Edward Deming]). Find your inner child, re-capture and embrace what God has given you, the ability to learn. It will require you to put down the remote control, get off the couch and decrease your apparently frequent visits to McDonalds. Search and find the knowledge which you seek to ultimately fulfill your destiny; build an Asterisk Server that works. Hell, we all did. JR Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL
[Asterisk-Users] POTS interfacing recommendation
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello there, I'm drawing up a scheme to manage our company calls and would like to implement it with Asterisk. In order to get moving quickly I'd like some recommendations on what hardware to buy so I can start tinkering. Initially we'd like to be able to support one line to accept incoming calls, and another one for forwarding such calls to mobiles or other external phone numbers. My initial thought was that two voice-supporting modems will do this but I get the impression that the modem drivers in Asterisk will not behave as flexibly as I'd like, and/or that I'll need to get hold of very hard-to-get or overpriced modems. Are either of these impressions true, or do people run such phone systems practically with a pair of modems? If so, do I need specific modems and where do I get them? If voice modems will be hasslesome, I understand that Zaptel-based cards will give me the least problems but I'm not sure which would be the most suitable to buy for our needs, how much to expect to pay and so on. Also bonus points for pointing me at a friendly/knowledgeable UK supplier of such cards. Any advice would be greatly appreciated: once I have some known-working hardware in place, I'm cocky enough to believe I can set the software up with enough head banging :) cheers, - -- Matthew Bloch Bytemark Hosting tel. +44 (0) 8707 455026 http://www.bytemark-hosting.co.uk/ Dedicated Linux hosts from 15ukp ($26) per month -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQE/9/0zT2rVDg8aLXQRAhKhAKCC0JpZxjaRbUi/YUkmZNfgBUjaAQCbBGFN TCdY0qP6wupYOFK1eteahdg= =/Mr7 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk.
I agree in stopping the thread, but I do have one question... What would Qwest think of her posting to the list under a yahoo mail account representing her company, badmouthing this community, who, in the long run, could be VERY much worth their interest? Hmm Just my $.02 - no flames please. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Balaji NJL Sent: Saturday, January 03, 2004 8:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk. Hi All, Can we stop this thread pl. This lady has no intentions to learn asterisk. She is just a troll and wasting our time. With her corporate attitude, what she expects is support that available with paid commercial products. Her company has enough money to buy commercial products, let she go there. Hey lady, whoever u are, dont waste our time. this is not for u. Lets move on to something useful pl. -B - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 03, 2004 5:36 PM Subject: RE: [Asterisk-Users] New to asterisk? RUN... don't walk. On Sat, 2004-01-03 at 14:12, Me wrote: Mr. West, Sorry to burst your bubble, but that is not me. My name is Barbara Simpson. Either you are lying or someone is trying to remove any credibility from my original post. I now stand by my original post with more conviction than ever. You had little to no credibility when you show up acting like a troll from what most people would consider a throw away account. There were a lot of insightful replies. However, none of them were able to address the real problems of the asterisk community and come up with solutions. If you can't see your own faults, you are in for a bumpy ride. This is due to the problem residing in the general population, not the community. The problem resides in users who can't be bothered to either expend energy, or patience for the software to develop. Remember you came here, we didn't go recruiting you. So if you are disappointed in your experience, blame yourself for your expectations. As far as I can tell here, you haven't paid a single person for anything, so any help you have received has been at a cost to the other people of this community. So the solution is for you to grow up. You need to learn that the comment you have made in this thread are worthless as they don't advance anything here. If you want credibility in a technical forum, you will have to show some technical skills. Otherwise you will be cast aside and hopefully ignored. Barbara Simpson Qwest Voice Over Packet Services --- Brian West [EMAIL PROTECTED] wrote: You said it good Look what this person posted to my blog... Now thats what I call grown up. Date: Thu, 1 Jan 2004 10:10:24 -0600 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] IP Address: 24.10.200.168 Name: Jeff Sowery Email Address: [EMAIL PROTECTED] URL: Comments: You're a complete idiot. Grow a brain or at least some balls. -Jeff NEXT!!! bkw On Thu, 1 Jan 2004, JR Richardson wrote: Piping in 2 cents, This is a great example of the Internet, Fast Food generation, showing their appreciation for all the magic that happens in the labs, hearts and minds of the courageous, hard working, dedicated and motivated group of people truly interested and guided to accomplish greatness. This platform for learning is one of the best tools in existence to come to a finite understanding of VoIP and legacy telephony with the versatility to expand beyond and develop originality in the field of telecommunications excellence, product development. Learn it, understand it, appreciate it, then take it past where you found it and if you're capable contribute, if not, enjoy it. But always, always maintain respect for those who created it and continue to refine it. Learning is intrinsically human, and in this world of Industry (There is no substitution for knowledge. [Edward Deming]). Find your inner child, re-capture and embrace what God has given you, the ability to learn. It will require you to put down the remote control, get off the couch and decrease your apparently frequent visits to McDonalds. Search and find the knowledge which you seek to ultimately fulfill your destiny; build an Asterisk Server that works. Hell, we all did. JR Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL
Re: [Asterisk-Users] POTS interfacing recommendation
Check http://www.telappliant.com for their VoIP Starter kits or Telephony Cards sections. Robert -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello there, . for pointing me at a friendly/knowledgeable UK supplier of such cards. Any advice would be greatly appreciated: once I have some known-working hardware in place, I'm cocky enough to believe I can set the software up with enough head banging :) cheers, - -- Matthew Bloch Check http://www.telappliant.com for their VoIP Starter kits or Telephony Cards sections. I don't have any experience with the X100 or voice cards since my implementation is VoIP only (so far). Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] POTS interfacing recommendation
Matthew Bloch wrote: Hello there, I'm drawing up a scheme to manage our company calls and would like to implement it with Asterisk. In order to get moving quickly I'd like some recommendations on what hardware to buy so I can start tinkering. Initially http://www.voip-info.org/wiki-Asterisk+hardware+recommendations is a good place to start. You'll not find all the answers there, but some recommendations. I would really like to see other members of the community add your configurations to the Wiki. It's helpful for all newbies to see a number of different setups for various solutions. If you don't want to add it yourself, feel free to mail me a note describing your setup. Regards, /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
I seriously doubt this is actually a Quest employee. Probably just someone trying to mess with a boss or something. - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 03, 2004 8:36 PM Subject: Re: [Asterisk-Users] New to asterisk? RUN... don't walk. Justin Sinclair wrote: Your complaints about the Asterisk Community remind me very much of complaints often made about the Linux Community. Judging an entire community (and even quality of the software) based on the actions of a few people is a big mistake. It was trolling, plain and simple, and unfortunately for Qwest, a lot of VoIP-savvy types who *do* pull their own weight in the world now have a data point about the moral quality of at least one person who at least professes to be part of the Qwest VoIP effort. Barbara, who is your boss, and has s/he been watching what you're doing for Qwest's reputation out in ListLand? It is pretty hard not to see through the motive behind your post--you should have identified yourself as a Qwest employee upfront. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
Mike, The v2.03f code (alpha/beta?) did correct the multi-line problems very nicely, however I think the snom folks might have another tweak or two to make to this code. If your snom 200 is running in a business production environment, you might want to wait a little. If you're using it in a test/home/soho environment, it would appear the code is good enough to play nicely in the multi-extn environment. Hint: after upgrading to v2.03f, on the web interface goto Settings/Base and turn off the option for Challenge Response on Phone. I've not tested this extensively, but v2.03f appears to be very close to what I'd consider a production release for basic sip/asterisk use then any release that I've tested in the last several months. (Your milage may vary!) In my limited testing, the code handled two extns well, nice Call Waiting indication, call hold, alternating between extn 1 and 2 nicely, etc. The only somewhat unusual operation that I noticed was: If the phone as two calls going on (extn 1 and extn 2, one of which is on hold), hanging up the phone causes the two extns to be bridged together without any indication that happened on the phone itself (all LED's are off). That could get to be embarrasing as it would suggest two inbound calls from customers/friends/etc are now tied together without you knowing it, and no way to recover from that accident. Rich Please forgive me if this is a silly question. I've been following this thread in the hope that I could put my * server and snom 200 into full-time service very soon. I need to find out how to have the lines configured so that it does not return a busy reply when only one call instances is engaged. Am I supposed to create multiple extensions on my asterisk dialplan to reflect the 5 call instances? That is, would the snom 200 be extension 2000 or 2000-2004? Also, did the 2.03f firmware resolve the matter? Thanks, Michael On Sat, 3 Jan 2004 16:05:39 +0100, Christian Stredicke wrote: Hi Olle! I put something into trouble ticket (I guess you get this as email). BTW 2.03f is available at http://snom.com/download/share. Christian -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Olle E. Johansson Gesendet: Donnerstag, 1. Januar 2004 11:57 An: [EMAIL PROTECTED] Betreff: Re: AW: [Asterisk-Users] Snom 200 with two extns defined anyone? Christian Stredicke wrote: We at snom have problems with Asterisk when we receive calls without the line indication. When we register we place a contact like this: REGISTER sip:asterisk SIP/2.0 Contact: sip:[EMAIL PROTECTED];line=h35h345 When we receive the 200 Ok, we search for the h35h345. If we dont find it, we try to guess which line is affected. This is relatively easy on a REGISTER response, but on an incoming INVITE we have serious problems. I think some of the challenging and line-assignment problems are related to this problem. Strictly speaking, we register the contact sip:[EMAIL PROTECTED];line=h35h345, NOT sip:[EMAIL PROTECTED]! Parameters are an essential part of a URI which must not be discarded. Ok, Christian, let's fix this. First, I'm curious, is the line= parameter specified somewhere? (Always looking for documentation :-) Secondly, in many places in the sip channel, everything after the ; is discarded. I would really appreciate if Snom could help us fixing this, so Snom phones work correctly and fully with Asterisk. (Have a new Snom 200 on my desk :-) I'm not an experienced C programmer, so I can't fix this myself. However, there are experienced C programmers in the community that will fix this, but they need proper and detailed input on what to fix. I've opened a bug http://bugs.digium.com/bug_view_page.php?bug_id=732 Let's continue adding information there. BTW, there's some other Snom problems where we need input from SNOM. Search on Snom in the bug tracker. Thank you for participating in the Asterisk community! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 I am easily satisfied with the very best - Winston Churchill The questions arisen, is this a prison? Some say it is, but I say it isn't. - Ian Hunter ** Tag(s) inserted by Bandit Tagger98 -
Re: [Asterisk-Users] POTS interfacing recommendation
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Sunday 04 January 2004 12:46, rnc Info Lists wrote: Check http://www.telappliant.com for their VoIP Starter kits or Telephony Cards sections. Thanks for the pointer Robert (and from Olle too). The X100P sounds like a good deal for £60 and should let us get started with Asterisk right away. One further question: can these cards distinguish (and communicate to Asterisk) the difference between the two rings we receive on our one phone line through BT's Callsign service? I assume not but it would be very useful if so. - -- Matthew Bloch Bytemark Hosting tel. +44 (0) 8707 455026 http://www.bytemark-hosting.co.uk/ Dedicated Linux hosts from 15ukp ($26) per month -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQE/+CWpT2rVDg8aLXQRAgpGAJ4l/jHn7OgSLBebTFUxwpEybPel1ACdGlj6 i8GckXbFCW9JWtrh0kIt308= =1jHn -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk.
I agree in stopping the thread, but I do have one question... What would Qwest think of her posting to the list under a yahoo mail account representing her company, badmouthing this community, who, in the long run, could be VERY much worth their interest? Come on guys, drop it!!! There isn't anyone on this list that can actually validate it was her that started the comments in the first place. Absolutely no one! So drop it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise Asterisk Universal Gateway
On Sun, 2004-01-04 at 04:35, EDWARD WILSON wrote: Does anyone know what the hardware requirements would be to build an Enterprise Asterisk Universal Gateway ? I am thinking of something comprable to the Cisco AS5xxx Series of gateways. Just to prepare you, if you ask the above question, you are not ready to ask the above question. Basically it falls down to the problem of what is needed to be done, and more so what is considered enterprise level hardware to be run upon. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modem Communications thru *
On Sun, 2004-01-04 at 05:33, fred alexander wrote: Happy New Year, I have a project to pass modem calls through * convert them from IP to X.25 and then allow the modems at each end to talk thru the rtp stream to each other before calling modem terminates the call. Datamodem --- FXS(non *) ADSL --- * -Software Application X.25 modem While the above occurs I also wish to capture the data stream and also parse selective parts of these transactions into a database. My questions are: Should I start trying with G.711 (the modem speed is 1200Baud)? I want to use monitor app. to capture the voice stream and later convert it using text to speech? I don't control the modems and part of the stream is encrypted the rest is just text - strings and numbers Do you really know what you are asking? You say you have a modem connection, and then you say you want to use text to speech? Your problem will be related to the fact that a modem call over a VoIP is not necessarily stable. If you only need 1200, I'm sure you might be able to discuss with Steve Underwood about changing the tx/rx fax app to be a modem only app. At that point you could dial in to a local port with a software modem, convert to your text string, and send it to the other side. On the other side you can then push that though another software modem to the line out. Just remember that G711 uses around 80K bandwidth to send 1200. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
I would set the Enterprise Class bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. In our network, Linux is approaching Enterprise Class and I don't see why * could not achieve this in the near future. Steven Critchfield wrote: On Sun, 2004-01-04 at 04:35, EDWARD WILSON wrote: Does anyone know what the hardware requirements would be to build an Enterprise Asterisk Universal Gateway ? I am thinking of something comprable to the Cisco AS5xxx Series of gateways. Just to prepare you, if you ask the above question, you are not ready to ask the above question. Basically it falls down to the problem of what is needed to be done, and more so what is considered enterprise level hardware to be run upon. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] POTS interfacing recommendation
On Sun, 2004-01-04 at 08:39, Matthew Bloch wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Sunday 04 January 2004 12:46, rnc Info Lists wrote: Check http://www.telappliant.com for their VoIP Starter kits or Telephony Cards sections. Thanks for the pointer Robert (and from Olle too). The X100P sounds like a good deal for £60 and should let us get started with Asterisk right away. One further question: can these cards distinguish (and communicate to Asterisk) the difference between the two rings we receive on our one phone line through BT's Callsign service? I assume not but it would be very useful if so. The two separate rings sounds like you are asking for what we call distinctive ring. Multiple phone numbers attached to a single line and that line will signal the difference via a different ring cadence. This is supposed to work, but I haven't tried it. Something to think about on all system deployments is what are the chances for expansion. Some hardware limits your expansion without scraping some of your hardware investment. For instance if you go the X100P route, and you later need 4 physical lines you may not be able to get this working with X100P cards. But if you go with a T/E100P and a nice channel bank then you should be able to build up to 24 channels in a nice mix of in and out lines. Granted it is more investment up front, but you don't scrap it later when you grow. I don't know if there is a VoIP provider in your area, but you may wish to think about the costs of a VoIP provider. You mention that your calls come in and get forwarded out. In this case a VoIP provider that allows you to have more than 1 line active or more than one account means you just do the forward at your office and then the VoIP provider does all the phone hookups. This also means you won't need to worry about the ring problem since the dialed number will be transmitted in the VoIP protocol. Also if your calls are mostly forwarded off, this is a easy way to later grow to more lines without having any hardware at all to upgrade. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
I would set the Enterprise Class bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. My Norstar Meridian system has nowhere near this. We get about 5 minutes downtime every month (usually trunk card issues). Not arguing against anything you've said, just making a datapoint. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
Doug Shubert wrote: I would set the Enterprise Class bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. In our network, Linux is approaching Enterprise Class and I don't see why * could not achieve this in the near future. Asterisk would need some kind of clustering/load balancing ability (Single IP system image for the IP phones across multiple servers) to be truely Enterprise Class in terms of both reliability and scaleability.. Obviously that would not be as relevent for the analog hard wired phones unless the channelbanks and T1/E1 lines could be automatically switched to another server.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
Rich, What form of config is required in * to get 1 extensions available. Single login/registry or multiple? Do I have to specify lines per Christiasn's earlier mail? Thanks, Michael On Sun, 4 Jan 2004 07:58:21 -0600, Rich Adamson wrote: Mike, The v2.03f code (alpha/beta?) did correct the multi-line problems very nicely, however I think the snom folks might have another tweak or two to make to this code. If your snom 200 is running in a business production environment, you might want to wait a little. If you're using it in a test/home/soho environment, it would appear the code is good enough to play nicely in the multi-extn environment. Hint: after upgrading to v2.03f, on the web interface goto Settings/Base and turn off the option for Challenge Response on Phone. I've not tested this extensively, but v2.03f appears to be very close to what I'd consider a production release for basic sip/asterisk use then any release that I've tested in the last several months. (Your milage may vary!) In my limited testing, the code handled two extns well, nice Call Waiting indication, call hold, alternating between extn 1 and 2 nicely, etc. The only somewhat unusual operation that I noticed was: If the phone as two calls going on (extn 1 and extn 2, one of which is on hold), hanging up the phone causes the two extns to be bridged together without any indication that happened on the phone itself (all LED's are off). That could get to be embarrasing as it would suggest two inbound calls from customers/friends/etc are now tied together without you knowing it, and no way to recover from that accident. Rich Please forgive me if this is a silly question. I've been following this thread in the hope that I could put my * server and snom 200 into full-time service very soon. I need to find out how to have the lines configured so that it does not return a busy reply when only one call instances is engaged. Am I supposed to create multiple extensions on my asterisk dialplan to reflect the 5 call instances? That is, would the snom 200 be extension 2000 or 2000-2004? Also, did the 2.03f firmware resolve the matter? Thanks, Michael On Sat, 3 Jan 2004 16:05:39 +0100, Christian Stredicke wrote: Hi Olle! I put something into trouble ticket (I guess you get this as email). BTW 2.03f is available at http://snom.com/download/share. Christian -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Olle E. Johansson Gesendet: Donnerstag, 1. Januar 2004 11:57 An: [EMAIL PROTECTED] Betreff: Re: AW: [Asterisk-Users] Snom 200 with two extns defined anyone? Christian Stredicke wrote: We at snom have problems with Asterisk when we receive calls without the line indication. When we register we place a contact like this: REGISTER sip:asterisk SIP/2.0 Contact: sip:[EMAIL PROTECTED];line=h35h345 When we receive the 200 Ok, we search for the h35h345. If we dont find it, we try to guess which line is affected. This is relatively easy on a REGISTER response, but on an incoming INVITE we have serious problems. I think some of the challenging and line-assignment problems are related to this problem. Strictly speaking, we register the contact sip:[EMAIL PROTECTED];line=h35h345, NOT sip:[EMAIL PROTECTED]! Parameters are an essential part of a URI which must not be discarded. Ok, Christian, let's fix this. First, I'm curious, is the line= parameter specified somewhere? (Always looking for documentation :-) Secondly, in many places in the sip channel, everything after the ; is discarded. I would really appreciate if Snom could help us fixing this, so Snom phones work correctly and fully with Asterisk. (Have a new Snom 200 on my desk :-) I'm not an experienced C programmer, so I can't fix this myself. However, there are experienced C programmers in the community that will fix this, but they need proper and detailed input on what to fix. I've opened a bug http://bugs.digium.com/bug_view_page.php?bug_id=732 Let's continue adding information there. BTW, there's some other Snom problems where we need input from SNOM. Search on Snom in the bug tracker. Thank you for participating in the Asterisk community! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 I
Re: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)
Mike Jagdis wrote: John Coll wrote: Dave You were right disallow=all allow=ulaw allow=alaw gave me two-way voice! Whew! Thanks a million. I wonder if I really should have found that for myself ... I've added it to the voip-info.org wiki OK lets see if the next step is a bit easier :) thanks again all john Note that if you don't have canreinvite=no you probably also want to disable gsm on the GS phones themselves (just change the 723 entry in the list on the admin page to a repeat of a 711). Initially * negotiates each leg and relays packets. So the disallow and allow in *'s config works. If reinvite is enabled * then about 10 seconds later the two end points will bounce SIP INVITES between each other and start sending packets direct. Since * isn't in on this negotiation the fact that it is configured to filter gsm out of the codec list is immaterial... I don't know if gsm actually works between GS phones or not, but it definitely doesn't to other stuff. They negotiate gsm fine but send gsm data to the rtp port and the GS phone replies with icmp errors. Non-gsm data is fine... Added to http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+grandstream+budgetone Thank you! Guess most of this also applies to the Handytone. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Java?
Hi! Apache + Mysql + PHP ( Ming + Actionscript ) + Asterisk is good. Dynamic effective,Easy coding and Fast response :-) That's an excellent suggestion, I agree with Ray. Masakazu, do you think you could provide a working sample either here on the list or in the Wiki? yeah. surely ok. but please just a moment to disclose my code. because that is very evalution code at now. bit buggy ;-) Ok, wounderful - I am *very* curious about this! No need to present things in a perfect shape though, we all know perfection is a function of time... By the way: I installed ming two days ago and got the php module (so didn't compile ming support into PHP), but apparently I am not able to use Flash 6 code that way, only 4 and 5). Did I maybe get the wrong php module? I found http://www16.brinkster.com/gazb/ming/ to have interesting samples, especially the dynbarchart one. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise Asterisk Universal Gateway
Steven Critchfield wrote: Just to prepare you, if you ask the above question, you are not ready to ask the above question. Quote added to http://www.voip-info.org/tiki-index.php?page=Asterisk+quotes /O ;-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail Out call
There was a post in the wiki for an
application to provide an outcall when there is a voicemail is left on
asterisk. I am having a problem
that this application will only work if the caller presses the pound sign at
the end of recording. As most people just hang up, this application
isnt working. Can any offer
suggestions to accomplish this out call?
http://voip-info.org/wiki-Asterisk+tips+callback
[macro-leave_voicemail]
;Leaveavoicemailmessage,thendopost-processing.
;oCallconfiguredphones,withanannouncementthatamessage
;iswaiting,andtheoptiontolistentothevoicemail(s)
;${ARG1}=uorbfor'unavailable'or'busy'message
;${ARG2}=mailbox
;
${ARG3}=Calluserflag
;USAGE:
; exten=s,15,Macro(leave_voicemail,u,310,1)
exten=s,1,ResponseTimeout(30)
exten=s,2,Voicemail2(${ARG1}${ARG2})
exten
= s,3,GoToIf($[${ARG3} = 0]?s|5)
exten=s,4,system(${SCRIPTS_DIR}/voicemail_callback.sh${ARG2})
exten=s,5,NoOp
exten=h,1,GoToIf($ANBSP;CLASS='WIKI'NBSP;NBSP;HREF=''${ARG3}=0?h|3)
exten=h,2,system(${SCRIPTS_DIR}/voicemail_callback.sh${ARG2})
exten=h,3,NoOp
exten=t,1,GoToIf($ANBSP;CLASS='WIKI'NBSP;NBSP;HREF=''${ARG3}=0?t|3)
exten=t,2,system(${SCRIPTS_DIR}/voicemail_callback.sh${ARG2})
exten=t,3,NoOp
Thanks,
Kevin
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
Andrew Kohlsmith wrote: I would set the Enterprise Class bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. To turn around, let's discuss what we need to focus on to get Asterisk there: Here's a few bullet points, there's certainly a lot more * Linux platform stability - how? ** Special demands when using Zaptel cards * Redundancy architecture * Development/stable release scheme Then we have some channel demands, like * Better support for SRV records in the SIP channel More? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)
WipeOut wrote: Asterisk would need some kind of clustering/load balancing ability (Single IP system image for the IP phones across multiple servers) to be truely Enterprise Class in terms of both reliability and scaleability.. Obviously that would not be as relevent for the analog hard wired phones unless the channelbanks and T1/E1 lines could be automatically switched to another server.. Anyone that have peeked into Vovidas heartbeat/cluster architecture? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Two E100P boards - could not load zaptel module - Channel 63 - no such device
Hi, I have two E100P boards connected to my PC. I wish to setup two E1-ISDN-PRI lines, the first SPAN with 15 channels and the second one with 30 channels. When I try to load zaptel modules, I get an error message: Loading zaptel framework: Loading zaptel hardware modules: wct1xxp wcusb Running ztcfg: ZT_CHANCONFIG failed on channel 63: No such device Follows my zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 span=2,0,0,ccs,hdb3 bchan=33-47,49-63 dchan=48,64 Any clue? Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
I would set the Enterprise Class bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. In our network, Linux is approaching Enterprise Class and I don't see why * could not achieve this in the near future. Linux might approach that, but * as an application won't in its present design for lots of reasons that have already been discussed. I'd be reasonable certain (you're right) it will head that direction, it just happens to not be there today. On the surface, I've not heard of anyone that is actually addressing it either. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI, transfering thru a 2nd PBX - keep original CallerID
Hi!
I want to have Asterisk as my gateway to the outside world and use
another PBX to connect my existing phones.
exten = ${OUTSIDEMSN},1,Dial,CAPI/${MSN2NDPBX}:${EXTEN}
How do I transfer the caller Id information initially coming in?
I have strong doubts that this can be done at all. One way would be to
set your ${MSN2ndPBX} to ${CALLERIDNUM}, but that would require that
capi.conf has that CALLERIDNUM listed as one of the valid outgoing MSNs.
Since you won't know in advance who'll call that'll be a problem - also I
don't think you can reconfigure capi.conf in the midst of processing a
call...
Besides: I suppose your ISDN PBX (which brand exactly?) supports CLIP (or
comes with an internal S0 bus) and you have an analog CLIP phone (or ISDN
phone) connected?
Workaround: See my last posting and other very recent discussions
concerning a simple tool that shows the current caller ID and name on
your PC using either Flash, HTML or Java. Or use astman/ gastman.
As of now I am storing the caller data through AGI in mySQL and display
that on a web page that the user needs to re-load manually when desired.
Cheers, Philipp
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Re: [Asterisk-Users] Voicemail Out call
It is a problem - but the call recording is saved by * when you hang up.
So you need to look for new files in whichever directory the call
recordings are saved and pick them up eg with a script.
Iain
--On Sunday, January 04, 2004 12:07:35 -0500 Kevin [EMAIL PROTECTED]
wrote:
There was a post in the 'wiki' for an application to provide an outcall
when there is a voicemail is left on asterisk. I am having a problem
that this application will only work if the caller presses the pound sign
at the end of recording. As most people just hang up, this application
isn't working. Can any offer suggestions to accomplish this out call?
http://voip-info.org/wiki-Asterisk+tips+callback
[macro-leave_voicemail]
; Leave a voicemail message, then do post-processing.
; o Call configured phones, with an announcement that a message
; is waiting, and the option to listen to the voicemail(s)
;${ARG1} = u or b for 'unavailable' or 'busy' message
;${ARG2} = mailbox
; ${ARG3} = Call user flag
; USAGE:
; exten = s,15,Macro(leave_voicemail,u,310,1)
exten = s,1,ResponseTimeout(30)
exten = s,2,Voicemail2(${ARG1}${ARG2})
exten = s,3,GoToIf($[${ARG3} = 0]?s|5)
exten = s,4,system(${SCRIPTS_DIR}/voicemail_callback.sh ${ARG2})
exten = s,5,NoOp
exten =
h,1,GoToIf($ANBSP;CLASS='WIKI'NBSP;NBSP;HREF='${ARG3}NBSP;=NBSP;0'
${ARG3} = 0?h|3) exten =
h,2,system(${SCRIPTS_DIR}/voicemail_callback.sh ${ARG2}) exten =
h,3,NoOp
exten =
t,1,GoToIf($ANBSP;CLASS='WIKI'NBSP;NBSP;HREF='${ARG3}NBSP;=NBSP;0'
${ARG3} = 0?t|3) exten =
t,2,system(${SCRIPTS_DIR}/voicemail_callback.sh ${ARG2}) exten =
t,3,NoOp
Thanks,
Kevin
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RE: [Asterisk-Users] Two E100P boards - could not load zaptel module - Channel 63 - no such device
Yes, I think it should be: span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 span=2,0,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 Cheers, Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Bichara Sent: Sunday, January 04, 2004 5:11 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Two E100P boards - could not load zaptel module - Channel 63 - no such device Hi, I have two E100P boards connected to my PC. I wish to setup two E1-ISDN-PRI lines, the first SPAN with 15 channels and the second one with 30 channels. When I try to load zaptel modules, I get an error message: Loading zaptel framework: Loading zaptel hardware modules: wct1xxp wcusb Running ztcfg: ZT_CHANCONFIG failed on channel 63: No such device Follows my zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 span=2,0,0,ccs,hdb3 bchan=33-47,49-63 dchan=48,64 Any clue? Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help - recording both sides of a conversation
Does some kind Asterisk soul have an example from extensions.conf that shows
how to record both sides of a conversation?
Thanks!
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Sent: Sunday, January 04, 2004 9:23 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] CAPI, transfering thru a 2nd PBX - keep
original CallerID
Hi!
I want to have Asterisk as my gateway to the outside world and use
another PBX to connect my existing phones.
exten = ${OUTSIDEMSN},1,Dial,CAPI/${MSN2NDPBX}:${EXTEN}
How do I transfer the caller Id information initially coming in?
I have strong doubts that this can be done at all. One way would be to
set your ${MSN2ndPBX} to ${CALLERIDNUM}, but that would require that
capi.conf has that CALLERIDNUM listed as one of the valid outgoing MSNs.
Since you won't know in advance who'll call that'll be a problem - also I
don't think you can reconfigure capi.conf in the midst of processing a
call...
Besides: I suppose your ISDN PBX (which brand exactly?) supports CLIP (or
comes with an internal S0 bus) and you have an analog CLIP phone (or ISDN
phone) connected?
Workaround: See my last posting and other very recent discussions
concerning a simple tool that shows the current caller ID and name on
your PC using either Flash, HTML or Java. Or use astman/ gastman.
As of now I am storing the caller data through AGI in mySQL and display
that on a web page that the user needs to re-load manually when desired.
Cheers, Philipp
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[Asterisk-Users] OT: Anyone going to Open Source Telephony Summit in Geilenkirchen from North Germany?
Hello, anyone from northern germany planning to go to http://www.guug.de/veranstaltungen/telephony-summit-2004/ If yes, could you contact me off list. Maybe we can save some money by car-pooling?! -- Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise Asterisk Universal Gateway
On Sun, 2004-01-04 at 11:07, Olle E. Johansson wrote: Steven Critchfield wrote: Just to prepare you, if you ask the above question, you are not ready to ask the above question. Quote added to http://www.voip-info.org/tiki-index.php?page=Asterisk+quotes While funny, it makes at least some sense in context. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI, transfering thru a 2nd PBX - keep original CallerID
Philipp von Klitzing worte:
I want to have Asterisk as my gateway to the outside world and use
another PBX to connect my existing phones.
exten = ${OUTSIDEMSN},1,Dial,CAPI/${MSN2NDPBX}:${EXTEN}
How do I transfer the caller Id information initially coming in?
I have strong doubts that this can be done at all. One way would be to
[..]
Besides: I suppose your ISDN PBX (which brand exactly?) supports CLIP (or
comes with an internal S0 bus) and you have an analog CLIP phone (or ISDN
phone) connected?
Correct. It would save me from buying FXS cards. (The PBX is a Gesko
2108 Office).
Thanks for pointing out your workaround. It is a feasible solution for
times when the computer is near the phone, most of the time, the phone
is away.
--
Best regards
Peer Oliver Schmidt
the internet company
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Re: [Asterisk-Users] help - recording both sides of a conversation
Paul, you broke the thread! Please create your own top posting - or better, search the list archive! Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?
Mike, What form of config is required in * to get 1 extensions available. Single login/registry or multiple? Do I have to specify lines per Christiasn's earlier mail? In my implementation, the two extns are treated as though they are separate phones with separate (independent) logins, like: [3006] type=friend host=dynamic username=3006 secret=password context=from-sip mailbox=3006 [3007] type=friend host=dynamic username=3007 secret=password context=from-sip mailbox=3007 And on the phone, complete the a) two line entries under Settings/SIP/Lines (w/Action=proxy) b) two authentications under Settings/SIP/Authentication (Line 1 2) c) two Function Keys under Settings/SIP/Key Mapping (P1 P2) I've not figured out for sure when a phone reboot is required and when it is not. Seems some changes are accepted dynamically while others require a reboot. To keep from getting caught with unknowns, I rebooted the phone. After the reboot, the Settings/SIP/Lines show Registered for both lines. In an off-list discussion, it turns out that if you have two lines in use (one active and one on hold) and then hang the phone up, the two extns are bridged together. Apparently that's an expectation in European ISDN, but certainly a surprise in the US. That apparently only happens if two lines are actually in use (if four lines are on hold and one line active, it doesn't bridge all five extns, but I've not tried it either). That's the only surprise I've seen thus far in v2.03f code, other then turning off the Challenge Response on Phone option (under Settings/Base). MOH (from *), MWI, nice Call Waiting tone with extn LED blinking, etc, all work very nicely. (I don't use the phonebook, xml or anything like that, so didn't bother testing those things.) If you select button 2 to initiate a call, the proper CallerID is used, etc. Ringer is very loud for an office environment and there does not appear to be any way to adjust the volume. Maybe that'll come later. I have noticed that what is displayed in Settings/Base under Version is not accurate until after the second reboot (following a firmware upgrade). However, the phone's panel does display the correct version after a single reboot. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
On Sun, 2004-01-04 at 10:14, Doug Shubert wrote: I would set the Enterprise Class bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. In our network, Linux is approaching Enterprise Class and I don't see why * could not achieve this in the near future. I may be wrong, but I think the 5 9's relates to full system not to individual pieces especially when talking about a class4/5 switch. On a small scale deployment, that will be a problem as you won't implement full redundancy. Redundancy adds quite a bit to the cost of your deployment. As far as linux goes, it is at that level if you put forth the effort to make it's environment decent. I have multiple machines approaching 2 years of uptime, and many over a year of uptime. I have not had a machine in my colo space go down since we removed the one machine with a buggy NIC. So next step, is asterisk. Outside of a couple of deadlocks from kernel problems when I was compiling new modules, I haven't had asterisk knock over while doing normal calls. The downtime could have been dealt with by having some redundancy in the physical lines. I would have lost the calls on the line, but the calls could be reconnected immediately. I can say up front that I have asterisk installs running multiple months without problems. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail Out call
There was a post in the wiki for an application to provide an outcall when there is a voicemail is left on asterisk. I am having a problem that this application will only work if the caller presses the pound sign at the end of recording. As most people just hang up, this application isnt working. Can any offer suggestions to accomplish this out call? http://voip-info.org/wiki-Asterisk+tips+callback I'm not running the app, but pure 100% guess, in voicemail.conf try: [general] maxsilence=10 ; number of secs of silence before vm drops the connection ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)
WipeOut wrote: Asterisk would need some kind of clustering/load balancing ability (Single IP system image for the IP phones across multiple servers) to be truely Enterprise Class in terms of both reliability and scaleability.. Obviously that would not be as relevent for the analog hard wired phones unless the channelbanks and T1/E1 lines could be automatically switched to another server.. Switching a T1 automagically seems like it would be an easy hack, but it wouldn't be needed for customers who had more than one T1 (like, say, most Enterprises :-). The exception to this is people who are muxing their internal phones, of course. Anyone that have peeked into Vovidas heartbeat/cluster architecture? Yes, I've played with it a bit. It's pretty simplistic... the clustering just keeps several servers in sync with each other. I suppose that would be easy to do with Asterisk, especially if configuration data was stored in a RDBMS that could do replication. Even now, setting up a copy/reload routine isn't difficult. It also seems that if you had a load balancer set up in front of your * servers to balance the call requests, you'd have enough clustering to keep one failure from taking down the whole system. Since the load balancer keeps an affinity table (and monitors to make sure the servers aren't going down) all VoIP connections could end up at the same * box once they had been allocated, unless a server goes down, in which case the call probably gets dropped. Any planned downtime could be made without any disruptions, since you could stop the load balancer from allocating any more connections to the * box and use 'stop when convenient' to wait for all current calls to end. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Anyone going to Open Source Telephony Summit in Geilenkirchen from North Germany?
Thanks for the info. I would like to go. Is it in German or English? I only speak English. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver schmidt Sent: 04 January 2004 18:10 To: Asterisk User List Subject: [Asterisk-Users] OT: Anyone going to Open Source Telephony Summit in Geilenkirchen from North Germany? Hello, anyone from northern germany planning to go to http://www.guug.de/veranstaltungen/telephony-summit-2004/ If yes, could you contact me off list. Maybe we can save some money by car-pooling?! -- Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)
On Sun, 2004-01-04 at 12:53, Nick Bachmann wrote: WipeOut wrote: Asterisk would need some kind of clustering/load balancing ability (Single IP system image for the IP phones across multiple servers) to be truely Enterprise Class in terms of both reliability and scaleability.. Obviously that would not be as relevent for the analog hard wired phones unless the channelbanks and T1/E1 lines could be automatically switched to another server.. Switching a T1 automagically seems like it would be an easy hack, but it wouldn't be needed for customers who had more than one T1 (like, say, most Enterprises :-). The exception to this is people who are muxing their internal phones, of course. Switching a T1 has been discussed, it needs a special adapter. Anyone that have peeked into Vovidas heartbeat/cluster architecture? Yes, I've played with it a bit. It's pretty simplistic... the clustering just keeps several servers in sync with each other. I suppose that would be easy to do with Asterisk, especially if configuration data was stored in a RDBMS that could do replication. Even now, setting up a copy/reload routine isn't difficult. A database doesn't make this easier. I would suggest you look into a revision control system and the ability to register applications/scripts to be run on check in of changes. Your benefit here is a quick roll out on hardware failure, plus roll back. You probably have seen people doing mailings based on CVS check ins, you could have those trigger a script on the clients that pulled fresh copies and did a reload. Fairly simple over all. It also seems that if you had a load balancer set up in front of your * servers to balance the call requests, you'd have enough clustering to keep one failure from taking down the whole system. Since the load balancer keeps an affinity table (and monitors to make sure the servers aren't going down) all VoIP connections could end up at the same * box once they had been allocated, unless a server goes down, in which case the call probably gets dropped. Any planned downtime could be made without any disruptions, since you could stop the load balancer from allocating any more connections to the * box and use 'stop when convenient' to wait for all current calls to end. The problem here is that you do have a single point of failure, the load balancer. It would be better to have multiple machines that you selectively placed as primary and backup in your VoIP phones. It isn't true load balancing, but it does allow you only loose a specific amount of calls in progress at any time if a machine fails. Calls could then be picked up and restarted via the other machine. This would give fault tolerance, and would give the impression of having 5 9's as long as the failures are sufficiently spaced out. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
Andrew Kohlsmith wrote: I would set the Enterprise Class bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. To turn around, let's discuss what we need to focus on to get Asterisk there: Here's a few bullet points, there's certainly a lot more * Linux platform stability - how? ** Special demands when using Zaptel cards * Redundancy architecture * Development/stable release scheme Then we have some channel demands, like * Better support for SRV records in the SIP channel More? Better sip phone support for primary/secondary proxy (and failover) (note: some phones don't support a second proxy at all; some say they do, but fail at it.) Maybe some sort of HSRP (hot spare standby protocol, or whatever) Some form of dynamic config sharing between pri/sec systems Won't mention external pstn line failover as that's sort of a separate topic, or loss of calls in flight, etc. I'd guess part of the five-9's discussion centers around how automated must one be to be able to actually get close? If one assumes the loss of a SIMM the answer/effort certainly is different then assuming the loss of a single interface card (when multiples exist), etc. I would doubt that anyone reading this list actually have a justifiable business requirement for five-9's given the expontential cost/effort involved to get there. But, setting some sort of reasonable goal that would focus towards failover within xx number of seconds (and maybe some other conditions) seems very practical. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)
Nick Bachmann wrote: Yes, I've played with it a bit. It's pretty simplistic... the clustering just keeps several servers in sync with each other. I suppose that would be easy to do with Asterisk, especially if configuration data was stored in a RDBMS that could do replication. Even now, setting up a copy/reload routine isn't difficult. It also seems that if you had a load balancer set up in front of your * servers to balance the call requests, you'd have enough clustering to keep one failure from taking down the whole system. Since the load balancer keeps an affinity table (and monitors to make sure the servers aren't going down) all VoIP connections could end up at the same * box once they had been allocated, unless a server goes down, in which case the call probably gets dropped. Any planned downtime could be made without any disruptions, since you could stop the load balancer from allocating any more connections to the * box and use 'stop when convenient' to wait for all current calls to end. Nick As long as what ever system is used only presents a single IP address on the network, the reason being that if a SIP UA is behind NAT the NAT router will have opened a path for the response from the server it contacted, if the request was offloaded to another IP address then the response would not get through.. Also the servers in the cluster would have to share SIP registration information so that all servers would know all availible UA's and all servers would have to communicate to that UA on the same IP address.. These things could have major issues when it came to the RTP streams.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
Steven Critchfield wrote: On Sun, 2004-01-04 at 10:14, Doug Shubert wrote: I would set the Enterprise Class bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. In our network, Linux is approaching Enterprise Class and I don't see why * could not achieve this in the near future. I may be wrong, but I think the 5 9's relates to full system not to individual pieces especially when talking about a class4/5 switch. On a small scale deployment, that will be a problem as you won't implement full redundancy. Redundancy adds quite a bit to the cost of your deployment. As far as linux goes, it is at that level if you put forth the effort to make it's environment decent. I have multiple machines approaching 2 years of uptime, and many over a year of uptime. I have not had a machine in my colo space go down since we removed the one machine with a buggy NIC. So next step, is asterisk. Outside of a couple of deadlocks from kernel problems when I was compiling new modules, I haven't had asterisk knock over while doing normal calls. The downtime could have been dealt with by having some redundancy in the physical lines. I would have lost the calls on the line, but the calls could be reconnected immediately. I can say up front that I have asterisk installs running multiple months without problems. Steven, You often mention your servers uptime, I am assuming you don't count reboots since you must have had to patch your kernel at least a few times in the last year and the reboot would have reset your uptime.. If that is the case then I have a server that is also around the 2 year uptime mark.. The longest single runtime between reboots for updated kernels is only 127 days.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
Rich Adamson wrote: Andrew Kohlsmith wrote: I would set the Enterprise Class bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. To turn around, let's discuss what we need to focus on to get Asterisk there: Here's a few bullet points, there's certainly a lot more * Linux platform stability - how? ** Special demands when using Zaptel cards * Redundancy architecture * Development/stable release scheme Then we have some channel demands, like * Better support for SRV records in the SIP channel More? Better sip phone support for primary/secondary proxy (and failover) (note: some phones don't support a second proxy at all; some say they do, but fail at it.) Maybe some sort of HSRP (hot spare standby protocol, or whatever) Some form of dynamic config sharing between pri/sec systems Won't mention external pstn line failover as that's sort of a separate topic, or loss of calls in flight, etc. I'd guess part of the five-9's discussion centers around how automated must one be to be able to actually get close? If one assumes the loss of a SIMM the answer/effort certainly is different then assuming the loss of a single interface card (when multiples exist), etc. I would doubt that anyone reading this list actually have a justifiable business requirement for five-9's given the expontential cost/effort involved to get there. But, setting some sort of reasonable goal that would focus towards failover within xx number of seconds (and maybe some other conditions) seems very practical. A failover system does not solve the scalability issue.. which means that you have a full server sitting there doing nothing most of the time when if the load were being balanced across the servers in a cluster senario you would also have the scalability.. Also a failover system would typically only be 2 servers, if there were a cluster system there could be 10 servers in which case five 9's should be easy.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)
On Sun, 2004-01-04 at 12:53, Nick Bachmann wrote: Yes, I've played with it a bit. It's pretty simplistic... the clustering just keeps several servers in sync with each other. I suppose that would be easy to do with Asterisk, especially if configuration data was stored in a RDBMS that could do replication. Even now, setting up a copy/reload routine isn't difficult. A database doesn't make this easier. I would suggest you look into a revision control system and the ability to register applications/scripts to be run on check in of changes. Your benefit here is a quick roll out on hardware failure, plus roll back. You probably have seen people doing mailings based on CVS check ins, you could have those trigger a script on the clients that pulled fresh copies and did a reload. Fairly simple over all. Yes, agree that CVS works well for this (that's how I manage my stuff). I like a RDBMSs for this kind of work, though, because the replication works well and they are much faster than text files when you've got lots of data. Rolling back transactions is also pretty simple with most databases, but I agree CVS is easier in this regard. It also seems that if you had a load balancer set up in front of your * servers to balance the call requests, you'd have enough clustering to keep one failure from taking down the whole system. Since the load balancer keeps an affinity table (and monitors to make sure the servers aren't going down) all VoIP connections could end up at the same * box once they had been allocated, unless a server goes down, in which case the call probably gets dropped. Any planned downtime could be made without any disruptions, since you could stop the load balancer from allocating any more connections to the * box and use 'stop when convenient' to wait for all current calls to end. The problem here is that you do have a single point of failure, the load balancer. It would be better to have multiple machines that you That's why you buy a load balancer with its own redundancy :-). The Allied Telesyn SB series, for example, have two system controllers. Cisco stuff is probably similar. The ATI stuff says about 30 seconds of downtime when one SC fails, I would guess Cisco delivers less. selectively placed as primary and backup in your VoIP phones. It isn't true load balancing, but it does allow you only loose a specific amount of calls in progress at any time if a machine fails. Calls could then be picked up and restarted via the other machine. This would give fault tolerance, and would give the impression of having 5 9's as long as the failures are sufficiently spaced out. I remind you that close only counts in horseshoes, hand grenades, and nuclear weapons, and not with my users' uptime :-). Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Anyone going to Open Source Telephony Summit in Geilenkirchen from North Germany?
Craig Waddington wrote: anyone from northern germany planning to go to http://www.guug.de/veranstaltungen/telephony-summit-2004/ Thanks for the info. I would like to go. Is it in German or English? According to the site mostly english. rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
Andrew Kohlsmith wrote: I would set the Enterprise Class bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. To turn around, let's discuss what we need to focus on to get Asterisk there: Here's a few bullet points, there's certainly a lot more * Linux platform stability - how? Even more than Linux itself is the x86 platform... I've thought about this a bit when considering * boxes for big customers. When one actually comes along, I'll have to actually make a decision :-). From where I stand, the best thing to do for smaller customers is give them a box with RAID and redundant power supplies, if they can afford it. You can overcome most of those problems by buying good quality hardware. If you buy your * server from your local Taiwanese clone shop, you're asking for trouble. A big, beefy machine from Dell would be better. But if I were to have a big customer with deep pockets, I'd really like * on a big Sun beast with redundant-everything (i.e. you can hot swap any component and there's usually n+1 of everything). The problem is that I don't think there's any Solaris support for Digium cards, since it's kind of a chicken-and-egg problem. Nope. No Solaris support, but you might be able to get away with Linux/Solaris...but then you lose a lot of the hot-swapability. In my experience, though, the only things I've ever been able to hotswap were power supplies and hard drives...and thats not software/os dependant. One of these days, I may convince myself to buy a modern Sun box (maybe the ~$1000 Blade 100s) and see what can be done. The only problem I could conceive would be endian-ness, but I read about Digium cards in a PowerPC box, so that won't be a problem, right? Nick Endian-ness is really only a driver issue. Its when programmers who believe that the world revolves around Linux/i386 that you have problems. Personally, I'd stick my Digium cards into an Alpha of some sort. A DS-10L for 1U mounting with 1 card or a DS-20 for multiple cards where you need lots of processor zoobs. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help - recording both sides of a conversation
Iain -
First off, all of this is heavily borrowed from others. For those who see
their code embedded here, I thank you and give you full credit.
Here's how I do it. It's a bit convoluted, but I didn't want to record
everything. So, if a call comes in and I want to record it, I send it here:
[ext-surrept]
exten = _57XXX,1,Answer
exten = _57XXX,2,Macro(record-enable)
exten = _57XXX,3,BackGround(for-quality-purposes)
exten = _57XXX,4,BackGround(this-call-may-be)
exten = _57XXX,5,BackGround(recorded)
exten = _57XXX,6,Dial(SIP/${EXTEN:1},120,tm)
exten = _57XXX,7,Macro(rg-inbound,10,tr)
exten = _57XXX,8,Goto(aa-nooneavail,s,1)
By transferring a call to 5 + the extension I'm at, I enable the call
recording, let the caller know he might be recorded and then send the call
right back to myself.
Here's the Macro:
[macro-record-enable]
exten = s,1,AGI(set-timestamp.agi)
exten = s,2,SetVar(CALLFILENAME=${timestamp}-${CALLERIDNUM}-${MACRO_EXTEN})
exten = s,3,Monitor(wav,${CALLFILENAME})
It starts the recording and calls set-timestamp.agi
Here's the agi file:
#!/bin/sh
longtime=`date +%Y%m%d-%H%M%S`
echo SET VARIABLE timestamp $longtime
It sets a timestamp, which if you scour the asterisk list, you'll see that
it is necessary for mixing the in and out audio later.
I have one hangup extension set for my internal phones; it looks like this:
exten = h,1,Macro(record-cleanup)
And the record-cleanup macro looks like this:
[macro-record-cleanup]
exten = s,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor)
exten = s,2,GotoIf($[${CALLFILENAME} = ${FOO}]?6:3)
exten = s,3,System(/usr/scripts/mix_monitor_files.pl ${MONITORDIR}
${CALLFILENAME}-in.wav ${CALLFILENAME}-out.wav ${CALLFILENAME}.wav)
exten = s,6,NoOp
Don't forget to make the /var/spool/asterisk/monitor directory!
Finally, mix_monitor_files.pl does the mixing job and combines the in and
out files:
#!/usr/bin/perl
$monitordir = shift;
$infile = shift;
$outfile = shift;
$finishfile = shift;
chdir($monitordir);
$infile_output = `sox $infile -e stat 21`;
$outfile_output = `sox $outfile -e stat 21`;
$infile_output =~ /Samples read:\s+(\d+)/;
$infile_samples = $1;
$outfile_output =~ /Samples read:\s+(\d+)/;
$outfile_samples = $1;
if($outfile_samples $infile_samples)
{
$diff_samples = $outfile_samples - $infile_samples;
system(sox -v 3 $outfile temp${outfile} trim ${diff_samples}s);
system(wmix $infile temp${outfile} $finishfile);
system(rm -f $infile temp${outfile} $outfile);
}
elsif($infile_samples $outfile_samples)
{
$diff_samples = $infile_samples - $outfile_samples;
system(sox -v 3 $infile temp${infile} trim ${diff_samples}s);
system(wmix temp${infile} $outfile $finishfile);
system(rm -f temp${infile} $outfile $infile);
}
else
{
system(wmix $infile $outfile $finishfile);
system(rm -f $infile $outfile);
}
You'll need wmix from http://tph.tuwien.ac.at/~oemer/wavetools.html and
sox, which was already on my system and is pretty standard.
The only problem I've found is that my in channel is a bit low, with respect
to volume. It's probably a sox issue, but I haven't had time to mess with
the settings yet. It's only an annoyance; you can definitely hear both
sides of the conversation.
John
P.S. I record my outbound calls by prefixing my outbound calls with a 5,
which similiarly call record-enable. In that case, the other party doesn't
know they're being recorded. IANAL. Check your state laws first! In some
states both parties must know about calls being recorded. In mine, TX, only
the calling party must know, but it must be first person. For this reason,
I do not let asterisk record everything, because my employees must
themselves determine what they're going to record.
- Original Message -
From: Iain Stevenson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, January 04, 2004 12:51 PM
Subject: Re: [Asterisk-Users] help - recording both sides of a conversation
* always records both sides of the conversation - but stores them in
separate files in
/var/spool/asterisk/monitor/. You need to combine the in and out
parts
using soxmix.
Iain
--On Sunday, January 4, 2004 9:59 am -0800 Paul Mahler
[EMAIL PROTECTED] wrote:
Does some kind Asterisk soul have an example from extensions.conf that
shows how to record both sides of a conversation?
Thanks!
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Sent: Sunday, January 04, 2004 9:23 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] CAPI, transfering thru a 2nd PBX - keep
original CallerID
Hi!
I want to have Asterisk as my gateway to the outside world and use
another PBX to connect my existing phones.
exten = ${OUTSIDEMSN},1,Dial,CAPI/${MSN2NDPBX}:${EXTEN}
How do I transfer the
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
On Sun, 2004-01-04 at 13:28, WipeOut wrote: Steven Critchfield wrote: On Sun, 2004-01-04 at 10:14, Doug Shubert wrote: I would set the Enterprise Class bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. In our network, Linux is approaching Enterprise Class and I don't see why * could not achieve this in the near future. I may be wrong, but I think the 5 9's relates to full system not to individual pieces especially when talking about a class4/5 switch. On a small scale deployment, that will be a problem as you won't implement full redundancy. Redundancy adds quite a bit to the cost of your deployment. As far as linux goes, it is at that level if you put forth the effort to make it's environment decent. I have multiple machines approaching 2 years of uptime, and many over a year of uptime. I have not had a machine in my colo space go down since we removed the one machine with a buggy NIC. So next step, is asterisk. Outside of a couple of deadlocks from kernel problems when I was compiling new modules, I haven't had asterisk knock over while doing normal calls. The downtime could have been dealt with by having some redundancy in the physical lines. I would have lost the calls on the line, but the calls could be reconnected immediately. I can say up front that I have asterisk installs running multiple months without problems. Steven, You often mention your servers uptime, I am assuming you don't count reboots since you must have had to patch your kernel at least a few times in the last year and the reboot would have reset your uptime.. Why do you assume I would have to patch a kernel? Not all machines must run the most current kernels, and some kernels can be such that they are sufficiently minimal enough to present low risk. Plus all the recent problems require a local user to exploit. I subscribe to the theory to only give access to critical machines to people I can quickly level a shotgun to their head. With that knowledge, and my users acknowledgment or witness to my accuracy, they don't wish to screw with the systems. BTW, my accuracy goes up with the number of concurrent targets by about 4 percent. If that is the case then I have a server that is also around the 2 year uptime mark.. The longest single runtime between reboots for updated kernels is only 127 days.. :) I have 2 machines at this moment that are halfway to looping the uptime counter again at 497 days. Webserver is at 497 + 197 days Old almost decommissioned file server is at 497 + 194 days A VPN machine is at 414 days DB server is at 245 days A almost decommissioned distro server is at 497 + 165 days due to some upgrades, I now have fewer machines holding high uptimes. My mail server was updated just over 2 months ago and it was swapped to the distro server. So the distro server that is about to be decommissioned is really just waiting for me to go take it out of the rack. Those are real uptimes with no reboots. What makes those 4 machines with more than a year uptime interesting is that 1 is a dell, one is a supermicro, the other 2 are homebuilt systems. So I can attest to x86 being able to be stable. Maybe not always, and I would like some more swappable parts. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversalGateway
Perhaps I was being somewhat ambitious by posting five 9's for Enterprise Class using a three tiered approach, five 9's (5.25 min. per year) for Carrier Class four 9's (52.56 min. per year) for Enterprise Class three 9's (8.76 hrs. per year) for User/SOHO Class each Class having specific requirements of hardware, software, network and power redundancy. I presume this is needed to deliver an Service Level Agreement (SLA). Olle E. Johansson wrote: Andrew Kohlsmith wrote: I would set the Enterprise Class bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. To turn around, let's discuss what we need to focus on to get Asterisk there: Here's a few bullet points, there's certainly a lot more * Linux platform stability - how? ** Special demands when using Zaptel cards * Redundancy architecture * Development/stable release scheme Then we have some channel demands, like * Better support for SRV records in the SIP channel More? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
Andrew Kohlsmith wrote: I would set the Enterprise Class bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. To turn around, let's discuss what we need to focus on to get Asterisk there: Here's a few bullet points, there's certainly a lot more * Linux platform stability - how? Even more than Linux itself is the x86 platform... I've thought about this a bit when considering * boxes for big customers. When one actually comes along, I'll have to actually make a decision :-). From where I stand, the best thing to do for smaller customers is give them a box with RAID and redundant power supplies, if they can afford it. You can overcome most of those problems by buying good quality hardware. If you buy your * server from your local Taiwanese clone shop, you're asking for trouble. A big, beefy machine from Dell would be better. Yeah, but nothing like a nice, big Sun machine. A cluster of Dell machines is reliable, but a midrange Sun box puts them to shame. But if I were to have a big customer with deep pockets, I'd really like * on a big Sun beast with redundant-everything (i.e. you can hot swap any component and there's usually n+1 of everything). The problem is that I don't think there's any Solaris support for Digium cards, since it's kind of a chicken-and-egg problem. Nope. No Solaris support, but you might be able to get away with Linux/Solaris...but then you lose a lot of the hot-swapability. In my experience, though, the only things I've ever been able to hotswap were power supplies and hard drives...and thats not software/os dependant. With the big boxes like the 4800, you can hot swap CPUs and memory and such as well. You're right that all that stuff is pretty Solaris-dependent, which is why I wanted to see if I couldn't get Asterisk to run on a little Solaris machine (and then sell it to people who own the big ones). One of these days, I may convince myself to buy a modern Sun box (maybe the ~$1000 Blade 100s) and see what can be done. The only problem I could conceive would be endian-ness, but I read about Digium cards in a PowerPC box, so that won't be a problem, right? Nick Endian-ness is really only a driver issue. Its when programmers who believe that the world revolves around Linux/i386 that you have problems. But it can also be a problem if you have on-card firmware, I've heard. Personally, I'd stick my Digium cards into an Alpha of some sort. A DS-10L for 1U mounting with 1 card or a DS-20 for multiple cards where you need lots of processor zoobs. I like the Alphas too, but they're being discontinued last I heard, and being replaced with the Itanium. Even VMS is being ported (now _there's_ an OS for * :-) Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help - recording both sides of a conversation
you nolonger need set-timestamp.agi as we have ${TIMESTAMP} in that format
by default now.
bkw
On Sun, 4 Jan 2004, John Baker wrote:
Iain -
First off, all of this is heavily borrowed from others. For those who see
their code embedded here, I thank you and give you full credit.
Here's how I do it. It's a bit convoluted, but I didn't want to record
everything. So, if a call comes in and I want to record it, I send it here:
[ext-surrept]
exten = _57XXX,1,Answer
exten = _57XXX,2,Macro(record-enable)
exten = _57XXX,3,BackGround(for-quality-purposes)
exten = _57XXX,4,BackGround(this-call-may-be)
exten = _57XXX,5,BackGround(recorded)
exten = _57XXX,6,Dial(SIP/${EXTEN:1},120,tm)
exten = _57XXX,7,Macro(rg-inbound,10,tr)
exten = _57XXX,8,Goto(aa-nooneavail,s,1)
By transferring a call to 5 + the extension I'm at, I enable the call
recording, let the caller know he might be recorded and then send the call
right back to myself.
Here's the Macro:
[macro-record-enable]
exten = s,1,AGI(set-timestamp.agi)
exten = s,2,SetVar(CALLFILENAME=${timestamp}-${CALLERIDNUM}-${MACRO_EXTEN})
exten = s,3,Monitor(wav,${CALLFILENAME})
It starts the recording and calls set-timestamp.agi
Here's the agi file:
#!/bin/sh
longtime=`date +%Y%m%d-%H%M%S`
echo SET VARIABLE timestamp $longtime
It sets a timestamp, which if you scour the asterisk list, you'll see that
it is necessary for mixing the in and out audio later.
I have one hangup extension set for my internal phones; it looks like this:
exten = h,1,Macro(record-cleanup)
And the record-cleanup macro looks like this:
[macro-record-cleanup]
exten = s,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor)
exten = s,2,GotoIf($[${CALLFILENAME} = ${FOO}]?6:3)
exten = s,3,System(/usr/scripts/mix_monitor_files.pl ${MONITORDIR}
${CALLFILENAME}-in.wav ${CALLFILENAME}-out.wav ${CALLFILENAME}.wav)
exten = s,6,NoOp
Don't forget to make the /var/spool/asterisk/monitor directory!
Finally, mix_monitor_files.pl does the mixing job and combines the in and
out files:
#!/usr/bin/perl
$monitordir = shift;
$infile = shift;
$outfile = shift;
$finishfile = shift;
chdir($monitordir);
$infile_output = `sox $infile -e stat 21`;
$outfile_output = `sox $outfile -e stat 21`;
$infile_output =~ /Samples read:\s+(\d+)/;
$infile_samples = $1;
$outfile_output =~ /Samples read:\s+(\d+)/;
$outfile_samples = $1;
if($outfile_samples $infile_samples)
{
$diff_samples = $outfile_samples - $infile_samples;
system(sox -v 3 $outfile temp${outfile} trim ${diff_samples}s);
system(wmix $infile temp${outfile} $finishfile);
system(rm -f $infile temp${outfile} $outfile);
}
elsif($infile_samples $outfile_samples)
{
$diff_samples = $infile_samples - $outfile_samples;
system(sox -v 3 $infile temp${infile} trim ${diff_samples}s);
system(wmix temp${infile} $outfile $finishfile);
system(rm -f temp${infile} $outfile $infile);
}
else
{
system(wmix $infile $outfile $finishfile);
system(rm -f $infile $outfile);
}
You'll need wmix from http://tph.tuwien.ac.at/~oemer/wavetools.html and
sox, which was already on my system and is pretty standard.
The only problem I've found is that my in channel is a bit low, with respect
to volume. It's probably a sox issue, but I haven't had time to mess with
the settings yet. It's only an annoyance; you can definitely hear both
sides of the conversation.
John
P.S. I record my outbound calls by prefixing my outbound calls with a 5,
which similiarly call record-enable. In that case, the other party doesn't
know they're being recorded. IANAL. Check your state laws first! In some
states both parties must know about calls being recorded. In mine, TX, only
the calling party must know, but it must be first person. For this reason,
I do not let asterisk record everything, because my employees must
themselves determine what they're going to record.
- Original Message -
From: Iain Stevenson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, January 04, 2004 12:51 PM
Subject: Re: [Asterisk-Users] help - recording both sides of a conversation
* always records both sides of the conversation - but stores them in
separate files in
/var/spool/asterisk/monitor/. You need to combine the in and out
parts
using soxmix.
Iain
--On Sunday, January 4, 2004 9:59 am -0800 Paul Mahler
[EMAIL PROTECTED] wrote:
Does some kind Asterisk soul have an example from extensions.conf that
shows how to record both sides of a conversation?
Thanks!
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Sent: Sunday, January 04, 2004 9:23 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] CAPI,
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
This is my favourite response to this post RUN... don't walk.. WELL SAID! John Haigh - Original Message - From: asterisk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 4:24 PM Subject: Re: [Asterisk-Users] New to asterisk? RUN... don't walk. Here's the deal: Asterisk is free. If we go with * we will save $50k. It does almost anything. I can make it open my garage door. My installation records all conversations and then archives them as timestamped stereo MP3s. Our VB windows application can dial out with a click. All for free. It's not done. We are not at v1.0. Mr Spencer is a busy guy. It might not solve 'your' problem. We contracted the AgentCallbackLogin Queue stuff. That part works great. If you want it modified or fixed, pay for it or do it yourself. If you change your own oil, do your own plumbing, have more that 3 computers at home, or have [EMAIL PROTECTED] running, you are either a do-it-yourselfer or a geek. Asterisk might be for you. On the other hand, if you can't change a lightbulb or don't know what a dipstick is and have lots of money, then pay someone for a phone system. But please stop whining. I have 3 kids. Gettin' tired of it. Good day. - Original Message - From: Me [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 2:37 PM Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. __ Do you Yahoo!? Find out what made the Top Yahoo! Searches of 2003 http://search.yahoo.com/top2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail was scanned for viruses using BitDefender Sent by 602Pro LAN SUITE - http://www.software602.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)
Dave I note your suggestion you probably also want to disable gsm on the GS phones themselves (just change the 723 entry in the list on the admin page to a repeat of a 711 My GS phone has the following codec options: PCMU, PCMA, G.723.1, G729A/B, G726-32, G728. Half an hour's research and reading tells me that PCMU and PCMA are G.711. Can you confirm Dave, that I should ONLY have PCMU and PCMA in all the six options that GS provide for selecting codecs - or is it OK to have G.729A/B, G.726-32 and G.728 but not to have G.723.1? thanks john -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: 04 January 2004 17:03 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :) Mike Jagdis wrote: John Coll wrote: Dave You were right disallow=all allow=ulaw allow=alaw gave me two-way voice! Whew! Thanks a million. I wonder if I really should have found that for myself ... I've added it to the voip-info.org wiki OK lets see if the next step is a bit easier :) thanks again all john Note that if you don't have canreinvite=no you probably also want to disable gsm on the GS phones themselves (just change the 723 entry in the list on the admin page to a repeat of a 711). Initially * negotiates each leg and relays packets. So the disallow and allow in *'s config works. If reinvite is enabled * then about 10 seconds later the two end points will bounce SIP INVITES between each other and start sending packets direct. Since * isn't in on this negotiation the fact that it is configured to filter gsm out of the codec list is immaterial... I don't know if gsm actually works between GS phones or not, but it definitely doesn't to other stuff. They negotiate gsm fine but send gsm data to the rtp port and the GS phone replies with icmp errors. Non-gsm data is fine... Added to http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+grandstream+budg etone Thank you! Guess most of this also applies to the Handytone. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)
Thanks to Dave I now have two Grandstream phones with a voice path. Yippee! Wanting to learn from the experience I compared the sip debugs from before and after adding the disallow=all, allow=ulaw, allow=alaw lines to sip.conf to see what I should have noticed in the debug that would have pointed me to the problem. I see that during negotiation I got the following Found audio format UNKN Found audio format ALAW Found audio format ULAW Found audio format UNKN Found audio format GSM Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found description format G728 Capabilities: us - 524302, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Now why didn't asterisk and grandstream negotiate a common codec? Why does only allowing ulaw and alaw work better than allowing everything? That sounds to me as if one or other end is failing to negotiate correctly. Interestingly if I remove both of the allow lines for both phones I get *CLI WARNING[81926]: File chan_sip.c, Line 1954 (process_sdp): No compatible codecs! but the voice path works fine - perhaps using no compression??? In a FAQ I read: Q What Codec should I use for my Granstream phone? (http://www.grandstream.com/FAQ.htm#Q15) A: By default, PCMU(G711u) will be used. Both PCMU and PCMA will give you toll quality but their bandwidth consumption is also the highest (64kbps). If your network bandwidth is low, you can choose other lower-bit-rate codec such as G723 or G729 which will give you near toll quality at much smaller bandwidth consumption (G723 consumes 5.3/6.3kbps and G729 consumes 8kbps). If bandwidth is not a concern and you want good voice quality, try using PCMU or PCMA, or even the new wide band codec G722 (64kbps) which will provide hi-fidelity voice that is better than toll quality. The phrase by default seems to imply that negotiation will sort out the codecs. Eduardo Goncalves seems to raise essentially the same issue on December 16, 2003, Re: [Asterisk-Users] codec negotiation but no answer seems to emerge. I've pretty much had enough of this problem so don't spend long on a detailed response but I am curious to know why * and the GS phones failed to negotiate the right codec. Is there a bug / incompatibility issue? thanks john -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dave Cotton Sent: 03 January 2004 18:11 To: Asterisk List Subject: RE: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :) On Sat, 2004-01-03 at 18:59, John Coll wrote: Dave You were right! In the words of that welsh comedian I know because I was there. As others have said there's a steep learning curve for *, but as one who's climbed just some of it, I can say it's worth it. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie - MWI
With help I got two phones communicating - PCMA/PCMU was the problem. Next stpe is to try voicemail. VM works fine, I can leave a mesage and then retrieve it - but no MWI on the phone and no stutter dialtone. I promise I've spent the requisite 4 hours + on google etc. but have really no further ideas. The setup is 2 Grandstream phones (latest firmware) and an asterisk on a LAN. The cofig files I am using are shown below. Any suggestions would be appreciated. john - ; ; liza:/etc/asterisk/sip.conf ; [general] port = 5060 bindaddr = 0.0.0.0 externip = 10.0.1.198 [5702] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #1 5702 mailbox=5702 disallow=all allow=ulaw allow=alaw ; dtmfmode=rfc2834 dtmfmode=info username=5702 ; not convinced this is needed nat=yes [5703] same as above in effect ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie - MWI
Sorry for the partial post a moment ago
With help I got two phones communicating - PCMA/PCMU was the problem.
Next stpe is to try voicemail. VM works fine, I can leave a mesage and then
retrieve it - but no MWI on the phone and no stutter dialtone.
I promise I've spent the requisite 4 hours + on google etc. but have really
no further ideas.
The setup is 2 Grandstream phones (latest firmware) and an asterisk on a
LAN. The cofig files I am using are shown below. Any suggestions would be
appreciated.
john
-
;
; liza:/etc/asterisk/sip.conf
;
[general]
port = 5060
bindaddr = 0.0.0.0
externip = 10.0.1.198
[5702]
type=friend
host=dynamic
context=johnhome
reinvite=no
canreinvite=no
qualify=300
callerid=John workroom #1 5702
mailbox=5702
disallow=all
allow=ulaw
allow=alaw
; dtmfmode=rfc2834
dtmfmode=info
username=5702 ; not convinced this is needed
nat=yes
[5703]
same as above in effect
-
;
; liza:/etc/asterisk/extensions.conf
;
[general]
static=yes
writeprotect=no
;
[globals]
CONSOLE=Console/dsp
[johnhome]
exten = 5702,1,Dial(SIP/5702,20,Ttr)
exten = 5702,2,Voicemail(u5702)
exten = 5702,102,Voicemail(b5702)
exten = 5702,103,Hangup
exten = 5703,1,Dial(SIP/5703,20,Ttr)
exten = 5703,2,Voicemail(u5703)
exten = 5703,102,Voicemail(b5703)
exten = 5703,103,Hangup
exten = 88,1,VoicemailMain(${CALLERIDNUM})
-
;
; /etc/asterisk/voicemail.conf
;
[general]
format=wav49|gsm|wav
[johnhome]
5702 = 5702,John Coll,john
5703 = 5703,John Coll,john
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Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)
Hi! The affinity table makes the RTP stuff OK, but I agree that sharing SIP registrations is a concern. These are stored in the Asterisk DB. Type this at your CLI: database show SIP/Registry Consequently it shouldn't be a a problem to sync the registry data. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie - MWI
John Coll wrote: With help I got two phones communicating - PCMA/PCMU was the problem. Next stpe is to try voicemail. VM works fine, I can leave a mesage and then retrieve it - but no MWI on the phone and no stutter dialtone. I promise I've spent the requisite 4 hours + on google etc. but have really no further ideas. The setup is 2 Grandstream phones (latest firmware) and an asterisk on a LAN. The cofig files I am using are shown below. Any suggestions would be appreciated. ...and voicemail.conf + extensions.conf ? Only sip.conf can't help us debugging for you. Try again! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
I'd guess part of the five-9's discussion centers around how automated must one be to be able to actually get close? If one assumes the loss of a SIMM the answer/effort certainly is different then assuming the loss of a single interface card (when multiples exist), etc. I would doubt that anyone reading this list actually have a justifiable business requirement for five-9's given the expontential cost/effort involved to get there. But, setting some sort of reasonable goal that would focus towards failover within xx number of seconds (and maybe some other conditions) seems very practical. A failover system does not solve the scalability issue.. which means that you have a full server sitting there doing nothing most of the time when if the load were being balanced across the servers in a cluster senario you would also have the scalability.. Also a failover system would typically only be 2 servers, if there were a cluster system there could be 10 servers in which case five 9's should be easy.. Everyone's response to Olle's proposition are of value including yours. For those that have been involved with analyzing the requirments to achive five-9's (for anything), there are tons of approaches, and each approach comes with some sort of cost/benefit trade off. Once the approaches have been documented and costs associated with them, it's common for the original requirements to be redefined in terms of something that is more realistic in business terms. Whether that is clustering, hot standby, or another approach is largely irrelavent at the beginning of the process. If you're a sponsor of clustering and your forced to use canreinvite=no, lots of people would be unhappy when their RTP system died. I'm not suggesting clustering is a bad choice, only suggesting there are lots of cost/benefit trade-offs that are made on an individual basis and there might be more then one answer to reliability/uptime question. In an earlier post, you mentioned a single IP address issue. That's really not an issue in some cases as a virtual IP (within a cluster) may be perfectly fine (canreinvite=yes), etc. Pure guess is that use of a virtual IP forces some other design choices like the need for a layer-3 box (since virtual IP's won't fix layer-2 problems), and probably revisiting RTP standards. (And, if we only have one layer-3 box, guess we need to get another for uptime, etc, etc.) Since hardware has become increasingly more reliable, infrastructure items less expensive, uptimes moving towards larger numbers, software more reliable (in very general terms over years), using a hot spare approach could be just as effective as a two-box cluster. In both cases, part of the problem boils down to assumptions about external interfaces and how to move those interfaces between two or more boxes; and, what design requirements one states regardling calls in progress. (Olle, are you watching?) 1. Moving a physical interface (whether a T1, ethernet or 2-wire pstn) is mostly trevial, however what signal is needed to detect a system failure and move the physical connection to a second machine/interface? (If there are three systems in a cluster, what signal is needed? If a three-way switch is reqquired, does someone want to design, build, and sell it to users? Any need to discuss a four-way switch? Should there be a single switch that flip-flops all three at the same time (T1, Ethernet, pstn)?) Since protecting calls in progress (under all circumstances and configurations) is likely the most expensive and most difficult to achive, we can probably all agree that handling this should be left to some future long-range plan. Is that acceptable to everyone? 2. In a hot-spare arrangement (single primary, single running secondary), what static and/or dynamic information needs to be shared across the two systems to maintain the best chance of switching to the secondary system in the shortest period of time, and while minimizing the loss of business data? (Should this same data be shared across all systems in a cluster if the cluster consists of two or more machines?) 3. If a clustered environment, is clustering based on IP address or MAC address? a. If based on an IP address, is a layer-3 box required between * and sip phones? (If so, how many?) b. If based on MAC address, what process moves an active * MAC address to a another * machine (to maintain connectivity to sip phones)? c. Should sessions that rely on a failed machine in a cluster simply be dropped? d. Are there any realistic ways to recover RTP sessions in a clustered environment when a single machine within the cluster fails, and RTP sessions were flowing through it (canreinvite=no)? e. Should a sip phone's arp cache timeout be configurable? f. Which system(s) control the physical switch in #1 above? g. Is sharing static/dynamic operational data across some sort of high-availability
Re: [Asterisk-Users] Newbie - MWI
- Original Message -
From: John Coll [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 05, 2004 9:07 AM
Subject: [Asterisk-Users] Newbie - MWI
Sorry for the partial post a moment ago
With help I got two phones communicating - PCMA/PCMU was the problem.
Next stpe is to try voicemail. VM works fine, I can leave a mesage and
then
retrieve it - but no MWI on the phone and no stutter dialtone.
I promise I've spent the requisite 4 hours + on google etc. but have
really
no further ideas.
The setup is 2 Grandstream phones (latest firmware) and an asterisk on a
LAN. The cofig files I am using are shown below. Any suggestions would be
appreciated.
john
-
;
; liza:/etc/asterisk/sip.conf
;
[general]
port = 5060
bindaddr = 0.0.0.0
externip = 10.0.1.198
[5702]
type=friend
host=dynamic
context=johnhome
reinvite=no
canreinvite=no
qualify=300
callerid=John workroom #1 5702
mailbox=5702
disallow=all
allow=ulaw
allow=alaw
; dtmfmode=rfc2834
dtmfmode=info
username=5702 ; not convinced this is needed
nat=yes
[5703]
same as above in effect
-
;
; liza:/etc/asterisk/extensions.conf
;
[general]
static=yes
writeprotect=no
;
[globals]
CONSOLE=Console/dsp
[johnhome]
exten = 5702,1,Dial(SIP/5702,20,Ttr)
exten = 5702,2,Voicemail(u5702)
exten = 5702,102,Voicemail(b5702)
exten = 5702,103,Hangup
exten = 5703,1,Dial(SIP/5703,20,Ttr)
exten = 5703,2,Voicemail(u5703)
exten = 5703,102,Voicemail(b5703)
exten = 5703,103,Hangup
exten = 88,1,VoicemailMain(${CALLERIDNUM})
-
;
; /etc/asterisk/voicemail.conf
;
[general]
format=wav49|gsm|wav
[johnhome]
5702 = 5702,John Coll,john
5703 = 5703,John Coll,john
John,
You have your voicemail within the johnhome context, so for your sip
config, your phone entry for voicemail should be [EMAIL PROTECTED]
Paul
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Re: [Asterisk-Users] Re: Grandstream Early Dial
Where / how do I set DTMF payload type to 101? - Original Message - From: Josh Roberson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 01, 2004 3:17 PM Subject: RE: [Asterisk-Users] Re: Grandstream Early Dial I've never had early dial working, however, I resolved my multiple digit issue by simply putting both the GS phones and asterisk in INFO mode. This worked on both 10.0.3.81 firmware on the budgetone and the ATA286, as well as 10.0.4.30 firmware. I'm not saying I don't believe you, but doubelcheck your lines in asterisk to be dtmfmode=info and the gs devices are on SIP INFO method, and your DTMF Payload type is 101. Just my $.02 -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen R. Besch Sent: Wednesday, December 31, 2003 12:59 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Grandstream Early Dial I've just checked my voicemail with 1.0.4.30 and get the same multiple digits problem. sip.conf and GS config are both at info, for me this is a new problem voicemail has always worked perfectly with the GS. This has come up many times in this list, with no consensus for a solution. According to Grandstream, the multiple digit problem arises from a difference in the interpretation of the SIP standard. I'm not sure I really understand this, so no flames please, but, paraphrasing a conversation I had with GS, apparently they retransmit the digit as long as the key is pressed and expect asterisk to know that it is a re-transmission by examining other data in the packet. Asterisk does not handle the SIP packet in the way GS expects, resulting in multiple digit transmission. This flaw (?) is avoided by setting DTMF to INBAND. Why this behaviour is not repeatable on everyones installations escapes me. However, I have noticed one thing that may be a clue. I have one phone that is older hardware (redial button instead of send and an unused battery compartment on the bottom). This phone behaves differently than all the other, later, models. For example, it is the only phone on which the flash button actually works to answer the alternate line (eg when an incoming call waiting call arrives). All phones are using 3.81 firmware. Early dial has never worked for me, and I just upgraded to the 1.0.4.30 load yesterday. Now, I am having DTMF recognition issues, making it impossible to check my voice mail. This is an acknowleged bug on the GS. They have connected to my * server and acknowleged the problem. A fix has been promised but not yet delivered. Until then, the only solution is to turn early dial off and let the phone send the entire dial string in one packet. Since this does not affect later single digit transmission for IVR's, etc, the only consequence is the irritating delay between the last entered digit and the actual placing of the call. But, you can always hit the send key. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.554 / Virus Database: 346 - Release Date: 12/20/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.554 / Virus Database: 346 - Release Date: 12/20/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
1. Moving a physical interface (whether a T1, ethernet or 2-wire pstn) is mostly trevial, however what signal is needed to detect a system failure and move the physical connection to a second machine/interface? (If there are three systems in a cluster, what signal is needed? If a three-way switch is reqquired, does someone want to design, build, and sell it to users? Any need to discuss a four-way switch? Should there be a single switch that flip-flops all three at the same time (T1, Ethernet, pstn)?) Simple idea: Have a process on each machine pulse a lead-state (something a s simple as DTR out a serial port or a single data line on a parallel port) out to an external box. This box is strictly discrete hardware and built with timeout that is retriggered by the pulse. When the pulse fails to arrive, the box switches the T1 over to the backup system. Since protecting calls in progress (under all circumstances and configurations) is likely the most expensive and most difficult to achive, we can probably all agree that handling this should be left to some future long-range plan. Is that acceptable to everyone? Its going to be almost impossible to preserve calls in progress. If you switch a T1 from one machine to the other, there's going to either going to be a lack of sync (ISDN D-channels need to come up, RBS channels need to wink) that's going to result in the loss of the call. 2. In a hot-spare arrangement (single primary, single running secondary), what static and/or dynamic information needs to be shared across the two systems to maintain the best chance of switching to the secondary system in the shortest period of time, and while minimizing the loss of business data? (Should this same data be shared across all systems in a cluster if the cluster consists of two or more machines?) 3. If a clustered environment, is clustering based on IP address or MAC address? a. If based on an IP address, is a layer-3 box required between * and sip phones? (If so, how many?) Yes. You'll need something like Linux Virtual Server or an F5 load balancing box to make this happen. You can play silly games with round robin DNS, but it doesn't handle failure well. b. If based on MAC address, what process moves an active * MAC address to a another * machine (to maintain connectivity to sip phones)? Something like Ultra Monkey (http://www.ultramonkey.org) c. Should sessions that rely on a failed machine in a cluster simply be dropped? d. Are there any realistic ways to recover RTP sessions in a clustered environment when a single machine within the cluster fails, and RTP sessions were flowing through it (canreinvite=no)? e. Should a sip phone's arp cache timeout be configurable? Shouldn't need to worry about that unless the phone is on the same physical network segment. f. Which system(s) control the physical switch in #1 above? A voting system...all systems control it. It is up to the switch to decide who isn't working right. g. Is sharing static/dynamic operational data across some sort of high-availability hsrp channel acceptable, or, should two or more database servers be deployed? DB Server clustering is a fairly solid technology these days. Deploy a DB cluster if you want. 4. If a firewall/nat box is involved, what are the requirements to detect and handle a failed * machine? a. Are the requirements different for hot-spare vs clustering? b. What if the firewall is an inexpensive device (eg, Linksys) with minimal configuration options? c. Are the nat requirements within * different for clustering? 5. Should sip phones be configurable with a primary and secondary proxy? a. If the primary proxy fails, what determines when a sip phone fails over to the secondary proxy? Usually a simple timeout works for this..but if your clustering/hot-spare switch works right...the client should never need to change. b. After fail over to the secondary, what determines when the sip phone should switch back to the primary proxy? (Is the primary ready to handle production calls, or is it back ready for a system admin to diagnose the original problem in a non-production manner?) Auto switch-back is never a good thing. Once a system is taken out of service by an automated monitoring system, it should be up to human intervention to say that it is ready to go back into service. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 12sp+ program update
does anyone have a running cisco12sp+ or 30 vip phone on their network? and if so could you also tell me what tftp files you actually use and if there are any special settings in skinny.conf that i need? (I ran several searches for setup, nothing has come up so far so i'll ask for advice now.) Thanks Rohde
[Asterisk-Users] Voicepulse DID fast busy
I just signed up for Voicepulse with a DID. I can register with Voicepulse and dialout just fine. Only problem is that when I dial my DID from my POTS line I just get a fast busy and nothing in the console. Any ideas?
RE: [Asterisk-Users] Newbie - MWI
Thanks Paul very much!
john
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Paul Liew
Sent: 04 January 2004 22:32
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie - MWI
- Original Message -
From: John Coll [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 05, 2004 9:07 AM
Subject: [Asterisk-Users] Newbie - MWI
Sorry for the partial post a moment ago
With help I got two phones communicating - PCMA/PCMU was the problem.
Next stpe is to try voicemail. VM works fine, I can leave a mesage and
then
retrieve it - but no MWI on the phone and no stutter dialtone.
I promise I've spent the requisite 4 hours + on google etc. but have
really
no further ideas.
The setup is 2 Grandstream phones (latest firmware) and an asterisk on a
LAN. The cofig files I am using are shown below. Any suggestions would be
appreciated.
john
-
;
; liza:/etc/asterisk/sip.conf
;
[general]
port = 5060
bindaddr = 0.0.0.0
externip = 10.0.1.198
[5702]
type=friend
host=dynamic
context=johnhome
reinvite=no
canreinvite=no
qualify=300
callerid=John workroom #1 5702
mailbox=5702
disallow=all
allow=ulaw
allow=alaw
; dtmfmode=rfc2834
dtmfmode=info
username=5702 ; not convinced this is needed
nat=yes
[5703]
same as above in effect
-
;
; liza:/etc/asterisk/extensions.conf
;
[general]
static=yes
writeprotect=no
;
[globals]
CONSOLE=Console/dsp
[johnhome]
exten = 5702,1,Dial(SIP/5702,20,Ttr)
exten = 5702,2,Voicemail(u5702)
exten = 5702,102,Voicemail(b5702)
exten = 5702,103,Hangup
exten = 5703,1,Dial(SIP/5703,20,Ttr)
exten = 5703,2,Voicemail(u5703)
exten = 5703,102,Voicemail(b5703)
exten = 5703,103,Hangup
exten = 88,1,VoicemailMain(${CALLERIDNUM})
-
;
; /etc/asterisk/voicemail.conf
;
[general]
format=wav49|gsm|wav
[johnhome]
5702 = 5702,John Coll,john
5703 = 5703,John Coll,john
John,
You have your voicemail within the johnhome context, so for your sip
config, your phone entry for voicemail should be [EMAIL PROTECTED]
Paul
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Re: [Asterisk-Users] Newbie - MWI
lots of snips - ; ; liza:/etc/asterisk/sip.conf ; [general] port = 5060 bindaddr = 0.0.0.0 externip = 10.0.1.198 [5702] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #1 5702 mailbox=5702 disallow=all allow=ulaw allow=alaw ; dtmfmode=rfc2834 dtmfmode=info username=5702 ; not convinced this is needed nat=yes ; ; /etc/asterisk/voicemail.conf ; [general] format=wav49|gsm|wav [johnhome] 5702 = 5702,John Coll,john 5703 = 5703,John Coll,john John, You have your voicemail within the johnhome context, so for your sip config, your phone entry for voicemail should be [EMAIL PROTECTED] Paul Why shouldn't the mailbox definition inherit the context defined on the SIP entry? Why should we have to create each SIP/IAX/(etc) entry, define it's context, and then also define the context it's voicemail is in? [default] has no rights privelidges that should put it above any other context, does it? Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multi-line help
I am looking for common practice ideas on how to handle multiple line phones. Is it common with asterisk to have the lines appear as programmable buttons? Or to just have itcm like buttons and use the dial 9 approach? What I am specifically interested in, is to have my line one appear on the first button (sip polycom phones) line two appear on the second button, and use the third as an intercom (internal extension) button. I have managed to get the line 1 to ring on the line 1 button and the same for line two. I have even managed to get extension transfers to happen on the itcm button. The trouble I have is that I don't know if someone else is on the particular line, and when I dial, it picks up the first available button (line) so even if I dial an extension, it looks like I am dialing from line 1 to the extension. How do I make it pick the third button, etc... Confusing? I have read the handbook and countless searches through wiki and Google, but cannot find practical examples of multi-line use with asterisk. Thanks a ton. I have been testing asterisk and on the mailing list for about a month now... I would be happy to send all my config files for perusal. Sean Garland - Siskiyou Technology Consultants ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - MWI
There are no guarantees that the voicemail will be in the same context as the extension. By giving you the ability and flexibility of defining everything independently, there's not much you can't do! Remember, the context call in the sip.conf refers to the context in extensions.conf. the johnhome at the end of the [EMAIL PROTECTED] refers to the context in voicemail.conf. Maybe I'm missing your point, and I apologize if I am. Sean -Original Message- From: Andrew Thompson [mailto:[EMAIL PROTECTED] Sent: Sunday, January 04, 2004 7:32 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie - MWI lots of snips - ; ; liza:/etc/asterisk/sip.conf ; [general] port = 5060 bindaddr = 0.0.0.0 externip = 10.0.1.198 [5702] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #1 5702 mailbox=5702 disallow=all allow=ulaw allow=alaw ; dtmfmode=rfc2834 dtmfmode=info username=5702 ; not convinced this is needed nat=yes ; ; /etc/asterisk/voicemail.conf ; [general] format=wav49|gsm|wav [johnhome] 5702 = 5702,John Coll,john 5703 = 5703,John Coll,john John, You have your voicemail within the johnhome context, so for your sip config, your phone entry for voicemail should be [EMAIL PROTECTED] Paul Why shouldn't the mailbox definition inherit the context defined on the SIP entry? Why should we have to create each SIP/IAX/(etc) entry, define it's context, and then also define the context it's voicemail is in? [default] has no rights privelidges that should put it above any other context, does it? Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Earpiece Connections
Does anyone know of a piece of hardware that can allow multiple earpices to be connected directly to a server running Asterisk. I hope I am not being to vague but basically I am looking to allow a call center to user the server to do all of the Pickup and Hangup functions. The operators will merely have to have th earpiece in their ear. I have seen serial pieces of hardware that do this (41D switch matrix?) But I need to find one that asterisk can use. I will then build some custom scripts to handle the Pickup and Hangup parts of it. Anyway any ideas or websites I could research for this type of thing would be most helpful. Thanks Micahel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco to Cisco - poor quality
I am just starting to deploy asterisk in our office to use as our primary phone system - we plan to use a Voicetronix OpenLine4 card as our PSTN gateway - but one thing at a time... haven't got that far yet. Currently, i'm trying simple IP to IP calls within the office using our Cisco 7960's phones running SIP. When I make a call between these two phones, the conversation is of a quality so bad that it is barely audible (5% makes sense). I recall having this same problem when I tested asterisk briefly one year ago. However, I did also try on this occasion to make a call between the cisco phone and MSN - that worked fine. So it would seem that the cisco phone is to blame? - but why? Does anyone know why two phones of the same type should have so much problem talking to each other? Thanks! Terence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dutch/DTMF Caller ID
hi, since development of dtmf caller id under * is prolly going to only be done if someone stumps up the cash I've been looking for alternatives... Hoving found a number of projects which turn out to be mad prototypes or unavailable details i nearly gave up.. then I found this: http://www.artech.com.tw/html/english/ex200/ex200.htm http://www.artech.com.tw/html/english/ex200/ex200me.PDF The units are pretty cheap if i recall my conversations correctly...if anyone else in .nl interested in one of these... perhaps we could get together to reduce shipping costs... any takers? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to Cisco - poor quality
what firmware are you using? is it SIP? to check, push settings then status and firmware you should have a load ID like this 'POS3-04-4-00' also check the preferred CODEC we use g711ulaw as the default Terence Parker wrote: I am just starting to deploy asterisk in our office to use as our primary phone system - we plan to use a Voicetronix OpenLine4 card as our PSTN gateway - but one thing at a time... haven't got that far yet. Currently, i'm trying simple IP to IP calls within the office using our Cisco 7960's phones running SIP. When I make a call between these two phones, the conversation is of a quality so bad that it is barely audible (5% makes sense). I recall having this same problem when I tested asterisk briefly one year ago. However, I did also try on this occasion to make a call between the cisco phone and MSN - that worked fine. So it would seem that the cisco phone is to blame? - but why? Does anyone know why two phones of the same type should have so much problem talking to each other? Thanks! Terence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to Cisco - poor quality
On Sun, 2004-01-04 at 17:45, Terence Parker wrote: When I make a call between these two phones, the conversation is of a quality so bad that it is barely audible (5% makes sense). You must be doing something wrong (maybe codec problems), because I've had absolutely no problems with Cisco to Cisco calls, and I've got almost 50 deployed across the company. (For what it's worth, I'm using the ulaw codec.) Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to Cisco - poor quality
Thanks for the replies. My cisco firmware is only POS3-04-2-00, though it is SIP. It used to work fine under vocal though - which was strange. Is this definitely nothing to do with asterisk? I do note however that my firmware is fairly old... except cisco aren't exactly generous with firmware upgrades. I have tried both g729a (default on my phone) and g711ulaw with no success. But i'll have another fiddle and try to get it to work. Thanks again. Terence what firmware are you using? is it SIP? to check, push settings then status and firmware you should have a load ID like this 'POS3-04-4-00' also check the preferred CODEC we use g711ulaw as the default -- snip -- You must be doing something wrong (maybe codec problems), because I've had absolutely no problems with Cisco to Cisco calls, and I've got almost 50 deployed across the company. (For what it's worth, I'm using the ulaw codec.) Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 12sp+ program update
On Sun, 2004-01-04 at 17:04, Rohde wrote: does anyone have a running cisco 12sp+ or 30 vip phone on their network? and if so could you also tell me what tftp files you actually use and if there are any special settings in skinny.conf that i need? (I ran several searches for setup, nothing has come up so far so i'll ask for advice now.) When the phone boots up, look in the right side of the LCD, you should see a version of the call already installed. Mine said G2.04. Then you go to channels/chan_skinny.c and edit the version_id on line 71 to match your version. Here is my line. static char version_id[16] = P002G204; Don't forget to make clean install after the change and restart asterisk. At that point, you don't need any tftp files. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-line help
On Sun, 2004-01-04 at 18:18, Sean Garland wrote: I am looking for common practice ideas on how to handle multiple line phones. Is it common with asterisk to have the lines appear as programmable buttons? Or to just have itcm like buttons and use the dial 9 approach? What I am specifically interested in, is to have my line one appear on the first button (sip polycom phones) line two appear on the second button, and use the third as an intercom (internal extension) button. I have managed to get the line 1 to ring on the line 1 button and the same for line two. I have even managed to get extension transfers to happen on the itcm button. The trouble I have is that I don't know if someone else is on the particular line, and when I dial, it picks up the first available button (line) so even if I dial an extension, it looks like I am dialing from line 1 to the extension. How do I make it pick the third button, etc... Confusing? I have read the handbook and countless searches through wiki and Google, but cannot find practical examples of multi-line use with asterisk. The reason you didn't find anything is because the multiline approach doesn't scale beyond a small handful of lines. It shouldn't matter what line a call is on if you are supposed to answer it. If you have hunt or rollover on your lines, it doesn't matter what line you dial out on. In the long, the only thing that your phone should know is how to get you to the PBX, the pbx will take care of the rest. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco to Cisco - poor quality
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terence Parker Sent: Sunday, January 04, 2004 8:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco to Cisco - poor quality Thanks for the replies. My cisco firmware is only POS3-04-2-00, though it is SIP. It used to work fine under vocal though - which was strange. Is this definitely nothing to do with asterisk? I do note however that my firmware is fairly old... except cisco aren't exactly generous with firmware upgrades. I have tried both g729a (default on my phone) and g711ulaw with no success. But i'll have another fiddle and try to get it to work. How are the phones talking to each other? Directly, or through asterisk? (canreinvite=what? in the sip.conf for each of them?). What I'm trying to get at here is, it is a problem between the phones, or are you having a problem possibly with the asterisk box? Some other things to know: are you running voicemail yet? If so and you can dial into it from either of the phones, how does it sound? If not, how about anything from the * boxlike the demo annoucment stuff? Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to Cisco - poor quality
see if you can upgrade to firmware 4-3 or 4-4 another point to note, are you using a full duplex 10/100 switch? if so, you should have 'Port1 Full 100' for full duplex 100Mbit under the 'Network Statistics' If you like to email me your config settings, I will check them against our phones. telnet to the phone, and capture 'Phone show config' Doug Terence Parker wrote: Thanks for the replies. My cisco firmware is only POS3-04-2-00, though it is SIP. It used to work fine under vocal though - which was strange. Is this definitely nothing to do with asterisk? I do note however that my firmware is fairly old... except cisco aren't exactly generous with firmware upgrades. I have tried both g729a (default on my phone) and g711ulaw with no success. But i'll have another fiddle and try to get it to work. Thanks again. Terence what firmware are you using? is it SIP? to check, push settings then status and firmware you should have a load ID like this 'POS3-04-4-00' also check the preferred CODEC we use g711ulaw as the default -- snip -- You must be doing something wrong (maybe codec problems), because I've had absolutely no problems with Cisco to Cisco calls, and I've got almost 50 deployed across the company. (For what it's worth, I'm using the ulaw codec.) Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pager reminder script
Since the list community has done so much for me in my humble asterisk beginnings I have put together a simple little script written in php that serves as a paging reminder script. If anyone is interested in a copy of it contact me off list and I'll forward you a copy. The basics of the script are as follows: It queries an asterisk inbox of your choosing for the existence of a file that has been there longer than the time set in the script (you are free to set this time to whatever you like). If it finds the existence of the file it emails a reminder to the email/pagers/text message device of your choosing. If the time exceeds a second time set by you in the script, it will email a second pager(s)/email(s)/text messenger(s) with a different message. The logic is very simple: you install the script where you want it installed, have cron run it at whatever interval you choose and you never have to worry about it again. In my case, we have an incidence where someone is oncall 24 hours a day. This person is assigned an alpha-numeric pager, when an initial call is received to the oncall mailbox, asterisk sends a message to the pager. Our employee has 20 minutes to answer the page, we have cron set up to run the script every 5 minutes. If the script finds that a msg.gsm is in the mailbox and the timestamp on the message is equal or greater to 20 minutes, it sends out a reminder page to the pager. Since cron runs every 5 minutes it sends in this message every 5 minutes until the message equals 40 minutes old. At that point it sends the employee a message that the supervisor is being notified and at the same time it sends a page to the oncall supervisor's pager. It doesn't matter if there is more than one message in the mailbox because when the employee calls to get the first one they will see the pressence of the others. And even if the employee forgets to delete the message, asterisk automatically moves the message to the old mail folder once it has been listened to. For anybody who wants the script, just mail me. Also feel free to modify it any way you see fit. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Grandstream Early Dial
On the config webpage, its on the bottom. Kevin Original Message Subject: Re: [Asterisk-Users] Re: Grandstream Early Dial From: Aaron Martin [EMAIL PROTECTED] Date: Sun, January 04, 2004 3:49 pm To: [EMAIL PROTECTED] Where / how do I set DTMF payload type to 101? - Original Message - From: Josh Roberson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 01, 2004 3:17 PM Subject: RE: [Asterisk-Users] Re: Grandstream Early Dial I've never had early dial working, however, I resolved my multiple digit issue by simply putting both the GS phones and asterisk in INFO mode. This worked on both 10.0.3.81 firmware on the budgetone and the ATA286, as well as 10.0.4.30 firmware. I'm not saying I don't believe you, but doubelcheck your lines in asterisk to be dtmfmode=info and the gs devices are on SIP INFO method, and your DTMF Payload type is 101. Just my $.02 -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sun Servers with UltraSparc Processors
Hi, I'm just considering buying two Telecoms grade Sun Netra's to run a lab-based VoIP solution. Not my immediate thoughts as a VoIP platform, but from what I've heard, they can run Linux, and run it well. Only thing is: The Wiki and the Whitepaper just state that Asterisk is for the x86 architecture, but has been compiled to run on PPC architectures. No mention of UltraSparc. If I can get it compiled, what would I be loosing in terms of functions or what problems might I face? Would other 3rd party code (add ons and bolts on) work too, are these tied to platforms, or just to Asterisk itself? I've not got long to decide about the machines - so any feedback would be most welcome!! And directly too if possible!!! Finally, will ISDN4Linux run on an UltraSparc version of Linux? My intention is to stick a dual ISDN BRI card into each (2x ISDN lines on each card, or 4x B-Channels in total). I've had trouble trying to find decent second-hand/refurbished or new rackable servers within the research budget that can even be considered usable! These are the first machines that have a high spec, and are designed for the Telecom's industry. Best, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sun Servers with UltraSparc Processors
Forgot to add, the Sun Netra's UltraSparc is 64bit... However, various pointers indicate that it can run both 32bit and 64bit compiled code. Best, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 4 X100P Cards
Has anyone had any success using more than one or two X100P cards? I have 4 in a system, and channels 2 3 and 4 all seem to work just fine. Channel 1 however is acting up. I get random red alarms, disconnects, etc. I have checked the /proc/interrupts and everything is sitting on it's own IRQ. Also checked memory addresses and everything looks good there. Not looking for anything specific, just any pointers out there that anyone might have would be greatly appreciated. (Also the Channels are in a group and ring SIP Polycom Phones) TIA, Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hold and transfer problem
I got a Cisco 7960 phone recently, and have downloaded and set up Asterisk version 0.5.0. Very nice! I've set up the software on a test box for now and have configured the system to route calls that start with 7 to FWD. Once I'm happy with my various tests, I will set this all up on a dedicated box, will get a couple of POTS interface cards and will set up some proper routes etc. Anyway, the FWD stuff works as expected, as does the answering machine. The problem I have is when I put a call on hold, or attempt to transfer a call (same issue, I expect). Once the call is put on hold, I can't resume it and, after a couple of seconds, I get a Maximum retries exceeded error (verbose mode). If I put a call on hold and just leave it that way then I get the same message and, of course, am unable to resume the call. I suspect that the Cisco 7960 configuration is at fault here, and am wondering if anyone has any advice. I have searched around with Google but have not found any answers, although I have found a few similar (and unanswered) questions. I have tried the latest Asterisk version from CVS but that doesn't seem to like my phone at all. For instance, the voicemail application won't read the DTMF tones and so won't allow me to enter a password. I suspect that there's some silly little Cisco configuration value that I've overlooked, but I just can't seem to find it. Thanks for any help that can be offered with this. I can live with version 0.5.0 and just not use hold/transfer, but I'd prefer to find a solution. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: SIP + DTMF problem
Reposted because my original post was mangled by a nasty webmail client. I would really like some help with this one if anyone has any ideas. -- I am having a problem interacting with a remote IVR system when the outbound call is going via SIP. The only way that I have been able to get a response from the IVR is to set dtmfmode=info in sip.conf. Unfortunately that doesn't quite fix the problem because it will still only accept DTMF input once the voice response has finished on the IVR. If I try and press anything while the IVR is still talking it doesn't recognize any of the digits. I get the same result no matter what handset I use to originate the call i.e IP or POTS handset. The SIP IVR works fine if I connect directly to it using a SIP proxy + IP phone so I am pretty sure the problem is with the way that * is handling the DTMF. I found a post on the mailing list which suggested removing DTMF detection while the far end is still talking. If that has been implemented then that could explain the problem. Anyone got any suggestions? Thanks Hamish
Re: [Asterisk-Users] Sun Servers with UltraSparc Processors
On Sunday 04 January 2004 20:25, Adthrawn wrote: The Wiki and the Whitepaper just state that Asterisk is for the x86 architecture, but has been compiled to run on PPC architectures. No mention of UltraSparc. If I can get it compiled, what would I be loosing in terms of functions or what problems might I face? You shouldn't face any problems with endianness. In fact, the core code should probably work right out of the box. However, the drivers for the hardware, if you ever wanted to use them, might be a problem. I say might, because I really don't know that architecture. Would other 3rd party code (add ons and bolts on) work too, are these tied to platforms, or just to Asterisk itself? No, 3rd party code with the possible exception of any drivers should work just fine. And I think it's excellent that you're thinking about doing this in the lab, first, and not buying your production machines without testing, first. There've been a few users who jump a little too fast, then get disappointed when their first machines didn't always perform to their expectations. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-line help
On Sun, 2004-01-04 at 18:18, Sean Garland wrote: I am looking for common practice ideas on how to handle multiple line phones. Is it common with asterisk to have the lines appear as programmable buttons? Or to just have itcm like buttons and use the dial 9 approach? What I am specifically interested in, is to have my line one appear on the first button (sip polycom phones) line two appear on the second button, and use the third as an intercom (internal extension) button. I have managed to get the line 1 to ring on the line 1 button and the same for line two. I have even managed to get extension transfers to happen on the itcm button. The trouble I have is that I don't know if someone else is on the particular line, and when I dial, it picks up the first available button (line) so even if I dial an extension, it looks like I am dialing from line 1 to the extension. How do I make it pick the third button, etc... Confusing? I have read the handbook and countless searches through wiki and Google, but cannot find practical examples of multi-line use with asterisk. The reason you didn't find anything is because the multiline approach doesn't scale beyond a small handful of lines. It shouldn't matter what line a call is on if you are supposed to answer it. If you have hunt or rollover on your lines, it doesn't matter what line you dial out on. In the long, the only thing that your phone should know is how to get you to the PBX, the pbx will take care of the rest. Steve, I'll have to beg to differ with you. In some cases, which line or extn is used does make a difference (from a business perspective). Example, if x3002 is to be answered Customer Service and x3008 is to be answered as President (or whatever), you really do want to know which extn is ringing. Likewise, if you make a call from the Presidents line to certain employees, there is some value/meaning when an employee sees the call is coming from the President and not just 'another' customer service call. Other examples: share tennant services (how would you answer a phone that has six extensions, each belonging to a different business and you are the shared operator? Happens all the time, at least in the US. Departmental accounting is another (which department pays for which calls originated from the exact same phone; sales vs collections vs cust services). Or, the shared services (again) and the operator is asked to call a dozen foreign locations (who pays for that is determined by which button the very non-technical hardly-trainable operator pushes). So, his question is very valid. With Cisco phones, no problem. We define each button to be whatever extn we want and any callerid we want, and by pushing that button on the phone, we can initiate a call from that extn as well. Operates more like the older US key systems. As of yesterday, the same is true with the Snom 200 running v2.03f code. But the snom prior to that operated under what I've been told is the European ISDN approach, where apparently there is less/no sensitivity to which extn is actually used to initiate a call. I don't have any polycom phones, so no idea how that one functions nor how to program it, but since your asking, sounds like it follows the snom European approach. I could probably list several dozen valid business reasons for doing what he's asking, but probably very few (if any) valid technical reasons. More of an issue in small business then in large ones, but I also know the same kind of thing goes on in government offices as well. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sun Servers with UltraSparc Processors
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Sunday, January 04, 2004 9:58 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sun Servers with UltraSparc Processors [...] You shouldn't face any problems with endianness. In fact, the core code should probably work right out of the box. Not as of 3 months ago, the last time I tried. Test platform was Debian running on a Netra T1 120. I did little to try to go further, as I wasn't sure how I was going to get a timing source even if * did compile. And I think it's excellent that you're thinking about doing this in the lab, first, and not buying your production machines without testing, first. There've been a few users who jump a little too fast, then get disappointed when their first machines didn't always perform to their expectations. I second that. And I'd love to get * working on a Sparc. As a matter of fact, I've for a SunFire V120 doing absolutely nothing. (along with a few T1's and soon to be an E220r...I'm phasing sun out of my NOCs due to insane new hardware costs, wonkiness and expense of Solaris-based management platforms (can you say dependency hell to the 1000th degree?) and how ridiculously easy VMWare makes managing multiple low-usage installations). Maybe I'll give it a shot. But I'm not a developer by any means. Maybe if one or more of the * devs and/or hackers want to help on this, I'll consider providing test boxes hosted at on of my NOCs. Chime in if you're interested ([EMAIL PROTECTED]). Daryl G. Jurbala BMPC Network Operations Tel: +1 215 825 8401 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
The comments below are certainly not intended as any form of negativism, but rather to pursue thought processes for redundant systems. 1. Moving a physical interface (whether a T1, ethernet or 2-wire pstn) is mostly trivial, however what signal is needed to detect a system failure and move the physical connection to a second machine/interface? (If there are three systems in a cluster, what signal is needed? If a three-way switch is required, does someone want to design, build, and sell it to users? Any need to discuss a four-way switch? Should there be a single switch that flip-flops all three at the same time (T1, Ethernet, pstn)?) Simple idea: Have a process on each machine pulse a lead-state (something a s simple as DTR out a serial port or a single data line on a parallel port) out to an external box. This box is strictly discrete hardware and built with timeout that is retriggered by the pulse. When the pulse fails to arrive, the box switches the T1 over to the backup system. And upon partial restoration of the failed system, should it automatically fall back to the primary? Or, might there be some element of human control that would suggest not falling back until told to do so? Since protecting calls in progress (under all circumstances and configurations) is likely the most expensive and most difficult to achieve, we can probably all agree that handling this should be left to some future long-range plan. Is that acceptable to everyone? Its going to be almost impossible to preserve calls in progress. If you switch a T1 from one machine to the other, there's going to either going to be a lack of sync (ISDN D-channels need to come up, RBS channels need to wink) that's going to result in the loss of the call. What about calls in progress between two sip phones (and cdr records)? 2. In a hot-spare arrangement (single primary, single running secondary), what static and/or dynamic information needs to be shared across the two systems to maintain the best chance of switching to the secondary system in the shortest period of time, and while minimizing the loss of business data? (Should this same data be shared across all systems in a cluster if the cluster consists of two or more machines?) 3. If a clustered environment, is clustering based on IP address or MAC address? a. If based on an IP address, is a layer-3 box required between * and sip phones? (If so, how many?) Yes. You'll need something like Linux Virtual Server or an F5 load balancing box to make this happen. You can play silly games with round robin DNS, but it doesn't handle failure well. Agreed, but then one would need two F5 boxes as it would become the new single point of failure. b. If based on MAC address, what process moves an active * MAC address to a another * machine (to maintain connectivity to sip phones)? Something like Ultra Monkey (http://www.ultramonkey.org) c. Should sessions that rely on a failed machine in a cluster simply be dropped? d. Are there any realistic ways to recover RTP sessions in a clustered environment when a single machine within the cluster fails, and RTP sessions were flowing through it (canreinvite=no)? e. Should a sip phone's arp cache timeout be configurable? Shouldn't need to worry about that unless the phone is on the same physical network segment. Which in most cases where asterisk is deployed (obviously not all) is probably the case. f. Which system(s) control the physical switch in #1 above? A voting system...all systems control it. It is up to the switch to decide who isn't working right. With probably some manual over-ride since we know that systems can appear to be ready for production, but the sys admin says its not ready due to any number of valid technical reasons. g. Is sharing static/dynamic operational data across some sort of high-availability hsrp channel acceptable, or, should two or more database servers be deployed? DB Server clustering is a fairly solid technology these days. Deploy a DB cluster if you want. Which gets to be rather expensive, adds complexity, and additional points of failure (decreasing the ability to approach five/four-9's). 4. If a firewall/nat box is involved, what are the requirements to detect and handle a failed * machine? a. Are the requirements different for hot-spare vs clustering? b. What if the firewall is an inexpensive device (eg, Linksys) with minimal configuration options? c. Are the nat requirements within * different for clustering? 5. Should sip phones be configurable with a primary and secondary proxy? a. If the primary proxy fails, what determines when a sip phone fails over to the secondary proxy? Usually a simple timeout works for this..but if your clustering/hot-spare switch works right...the client should
Re: [Asterisk-Users] Multi-line help
On Sun, 2004-01-04 at 20:42, Rich Adamson wrote: On Sun, 2004-01-04 at 18:18, Sean Garland wrote: I am looking for common practice ideas on how to handle multiple line phones. Is it common with asterisk to have the lines appear as programmable buttons? Or to just have itcm like buttons and use the dial 9 approach? What I am specifically interested in, is to have my line one appear on the first button (sip polycom phones) line two appear on the second button, and use the third as an intercom (internal extension) button. I have managed to get the line 1 to ring on the line 1 button and the same for line two. I have even managed to get extension transfers to happen on the itcm button. The trouble I have is that I don't know if someone else is on the particular line, and when I dial, it picks up the first available button (line) so even if I dial an extension, it looks like I am dialing from line 1 to the extension. How do I make it pick the third button, etc... Confusing? I have read the handbook and countless searches through wiki and Google, but cannot find practical examples of multi-line use with asterisk. The reason you didn't find anything is because the multiline approach doesn't scale beyond a small handful of lines. It shouldn't matter what line a call is on if you are supposed to answer it. If you have hunt or rollover on your lines, it doesn't matter what line you dial out on. In the long, the only thing that your phone should know is how to get you to the PBX, the pbx will take care of the rest. Steve, I'll have to beg to differ with you. In some cases, which line or extn is used does make a difference (from a business perspective). Example, if x3002 is to be answered Customer Service and x3008 is to be answered as President (or whatever), you really do want to know which extn is ringing. Likewise, if you make a call from the Presidents line to certain employees, there is some value/meaning when an employee sees the call is coming from the President and not just 'another' customer service call. Like I said, a button for each line doesn't scale well. We have a PRI, I don't want to have a phone with 23 buttons to pick a line when a simple prefix chooses one for me. Scale that to a larger corporation and you may have many PRIs in one building. Do you want to bother choosing which of a hundred or more lines to choose from you will dial out on? It doesn't scale. If you need to know whether it is coming in for Pres. or Support, use CallerID. We have a similar need. We have a timeout on our main line that will ring all phones, and our tech support line rings all phones. It is acceptable for our programmers to ignore the main line if there are others in the office, but it isn't acceptable to ignore the support line ever. So we change the CallerID display for these calls. We all can choose once we see the CallerID as to what to answer. This solution scales because it is possible to send any arbitrary string to the CallerID unit without touching the phone number. Maybe I'm a bit cheap, but I don't like choosing solutions that will junked as soon as growth is experienced. My last office was such a place. They had a switch that was configured to show line appearances for 8 lines. That was a hardware limitation of the phones since they only supported 8 buttons. It was generally known that line 8 was tech support, but what if it was already in use? Some people gave out the numbers for specific lines so they knew after hours what was a personal call or a business call. Can you see how that does not scale at all. As soon as they needed 1 more phone line, they had to go buy a new switch as the old one was at the limit. Other examples: share tennant services (how would you answer a phone that has six extensions, each belonging to a different business and you are the shared operator? Happens all the time, at least in the US. Departmental accounting is another (which department pays for which calls originated from the exact same phone; sales vs collections vs cust services). Or, the shared services (again) and the operator is asked to call a dozen foreign locations (who pays for that is determined by which button the very non-technical hardly-trainable operator pushes). I haven't experienced the shared tenant situation for more than 2 weeks once. It was important for us to not have been on someone elses phone switch. It also made the pain of moving our service to a new location a non issue. I know if you are looking at providing the service, the headache of moving isn't of interest, but the ease of move in is important. It would seem to me that utilizing the CallerID is much easier to train than specific call appearances on a phone. The benefit is again that the CallerID is almost infinitely flexible where as a line appearance mechanism can only scale to the number of buttons you can manage and implement.
[Asterisk-Users] Grandstream Handytone 286 RTP Problems
I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 0 (Response) When doing traces with ethereal, I see successful SIP and SDP handshakes, but when * sends handytone RTP packets, I see a ICMP Port Unreachable messages sent from Handytone to * regarding the UDP RTP packet. * then gives up and I see a BYE from *, which handytone acks. Handytone config is default except obvious SIP registration parameters. I also have a Sipura SPA2000 and everything works perfect for that one, same extension and everything (not at same time of course). sip.conf entry: disallow=all; Disallow all codecs allow=ilbc allow=ulaw ; Allow codecs in order of preference [131] type=friend host=dynamic reinvite=no canreinvite=no qualify=300 callerid=handytone 131 mailbox=131 nat=0 Handytone info: Software Version:Program--1.0.4.17Bootloader--1.0.0.11 HTML--1.0.0.19 Both on same subnet, no NAT. I have two Handytones, both exhibit same symptoms. Anyone else have this problem? -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-line help
Could you explain in a little more detail about what you are trying to do with the multi-lines? Maybe a more in depth example would help. In my (limited) experience, I have seen two types of multi-line uses 1. The phone has a number of lines (usually) two. If the first line is busy, the call rings on the second and so the user has the option of putting the first on hold and answering the new incoming call or letting it ring out. Normally the user has only one advertised extension number (and the second line may not even have its own unique extension #). The second line is often used for inquiry calls or if the primary line is busy. Usually the phone selects the first available line when making a new call. 2. The second type of multi-line use I have seen is where one phone has lines for multiple extensions and those extensions may be represented on multiple phones (shared line). For example, the phone of a personal assistant may have a line for them and their boss. The multi-line button in this case may often shows the status (ie: busy) of the extension as it is 'shared' among multiple phones. Depending on the configuration, if the extension is called, it may ring on one or more of the phone lines that support that extension. Even in this case, the phone often has a default line to use when the handset is picked-up to make a call. Asterisk will support the type 1. above as long as the handset support multiple lines (which in your case it does,) However, case 2, I do not believe is supported by Asterisk at the moment - you can make the line ring, but you will not be able to show the status of the line on other phones. In addition, with a SIP phone, the phone will also have to have a way of receiving the notification for the status of the line (busy, not busy.) - not all SIP phones support this, but looking through the Polycomm manual, it seems they do. - Original Message - From: Sean Garland [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, January 04, 2004 7:18 PM Subject: [Asterisk-Users] Multi-line help I am looking for common practice ideas on how to handle multiple line phones. Is it common with asterisk to have the lines appear as programmable buttons? Or to just have itcm like buttons and use the dial 9 approach? What I am specifically interested in, is to have my line one appear on the first button (sip polycom phones) line two appear on the second button, and use the third as an intercom (internal extension) button. I have managed to get the line 1 to ring on the line 1 button and the same for line two. I have even managed to get extension transfers to happen on the itcm button. The trouble I have is that I don't know if someone else is on the particular line, and when I dial, it picks up the first available button (line) so even if I dial an extension, it looks like I am dialing from line 1 to the extension. How do I make it pick the third button, etc... Confusing? I have read the handbook and countless searches through wiki and Google, but cannot find practical examples of multi-line use with asterisk. Thanks a ton. I have been testing asterisk and on the mailing list for about a month now... I would be happy to send all my config files for perusal. Sean Garland - Siskiyou Technology Consultants ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sun Servers with UltraSparc Processors
I had asterisk running on SuSe 7.3 on an Ultra 2 back in May 2003. Had to make some changes to Makefiles. At that time, SuSe had no more updates planned for Linux on Sparc. I had MGCP andSIP running very well on it. Had some trouble with H323. Timing was another issue as the Ultra 2 is an SBus machine and not PCI. Basic IP Call switching works. MeetMe may not work. Machine was very reliable. Had almost no down time for the 6 months or so that I used it at home. I have been using Sparc servers for many years now and am very impressed by their reliability and performance. I think even Sonus Softswitch runs on a Sun Netra. I had documented the Makefile modification in an email to the list. If you search for Sparc in the mailing list, you should be able to find it. If not, drop me a line and I'll see if I still have it. Good luck. Ish "Adthrawn" [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]... Hi, I'm just considering buying two Telecoms grade Sun Netra's to run a lab-based VoIP solution. Not my immediate thoughts as a VoIP platform, but from what I've heard, they can run Linux, and run it well. Only thing is: The Wiki and the Whitepaper just state that Asterisk is for the x86 architecture, but has been compiled to run on PPC architectures. No mention of UltraSparc. If I can get it compiled, what would I be loosing in terms of functions or what problems might I face? Would other 3rd party code (add ons and bolts on) work too, are these tied to platforms, or just to Asterisk itself? I've not got long to decide about the machines - so any feedback would be most welcome!! And directly too if possible!!! Finally, will ISDN4Linux run on an UltraSparc version of Linux? My intention is to stick a dual ISDN BRI card into each (2x ISDN lines on each card, or 4x "B-Channels" in total). I've had trouble trying to find decent second-hand/refurbished or new rackable servers within the research budget that can even be considered usable! These are the first machines that have a high spec, and are designed for the Telecom's industry. Best, Ad.
Re: [Asterisk-Users] Cisco to Cisco - poor quality
I seem to recall that you are only sending calls from Asterisk to the Cisco, not sending calls from the Cisco to Asterisk. Is this correct? On Sun, 2004-01-04 at 19:10, Jared Smith wrote: On Sun, 2004-01-04 at 17:45, Terence Parker wrote: When I make a call between these two phones, the conversation is of a quality so bad that it is barely audible (5% makes sense). You must be doing something wrong (maybe codec problems), because I've had absolutely no problems with Cisco to Cisco calls, and I've got almost 50 deployed across the company. (For what it's worth, I'm using the ulaw codec.) Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems
Hi Mike I know exacty same situation about BT100 that sometimes lost any packets. like a DoS attack for BT100? ;-( mack_jpn [EMAIL PROTECTED] asterisk]# ping 192.168.XX.XX PING 192.168.XX.XX (192.168.XX.XX) from 192.168.XX.X : 56(84) bytes of data. 64 bytes from 192.168.XX.XX: icmp_seq=0 ttl=250 time=2 usec Warning: time of day goes back, taking countermeasures. 64 bytes from 192.168.XX.XX: icmp_seq=1 ttl=250 time=969 usec 64 bytes from 192.168.XX.XX: icmp_seq=2 ttl=250 time=766 usec 64 bytes from 192.168.XX.XX: icmp_seq=3 ttl=250 time=746 usec 64 bytes from 192.168.XX.XX: icmp_seq=4 ttl=250 time=829 usec 64 bytes from 192.168.XX.XX: icmp_seq=5 ttl=250 time=725 usec 64 bytes from 192.168.XX.XX: icmp_seq=6 ttl=250 time=735 usec 64 bytes from 192.168.XX.XX: icmp_seq=7 ttl=250 time=703 usec 64 bytes from 192.168.XX.XX: icmp_seq=9 ttl=250 time=670 usec 64 bytes from 192.168.XX.XX: icmp_seq=10 ttl=250 time=728 usec 64 bytes from 192.168.XX.XX: icmp_seq=11 ttl=250 time=711 usec 64 bytes from 192.168.XX.XX: icmp_seq=12 ttl=250 time=701 usec 64 bytes from 192.168.XX.XX: icmp_seq=13 ttl=250 time=707 usec 64 bytes from 192.168.XX.XX: icmp_seq=14 ttl=250 time=693 usec 64 bytes from 192.168.XX.XX: icmp_seq=15 ttl=250 time=692 usec 64 bytes from 192.168.XX.XX: icmp_seq=16 ttl=250 time=678 usec 64 bytes from 192.168.XX.XX: icmp_seq=17 ttl=250 time=673 usec 64 bytes from 192.168.XX.XX: icmp_seq=18 ttl=250 time=699 usec 64 bytes from 192.168.XX.XX: icmp_seq=19 ttl=250 time=683 usec 64 bytes from 192.168.XX.XX: icmp_seq=20 ttl=250 time=696 usec 64 bytes from 192.168.XX.XX: icmp_seq=21 ttl=250 time=714 usec 64 bytes from 192.168.XX.XX: icmp_seq=22 ttl=250 time=704 usec 64 bytes from 192.168.XX.XX: icmp_seq=23 ttl=250 time=701 usec 64 bytes from 192.168.XX.XX: icmp_seq=24 ttl=250 time=691 usec 64 bytes from 192.168.XX.XX: icmp_seq=25 ttl=250 time=670 usec 64 bytes from 192.168.XX.XX: icmp_seq=26 ttl=250 time=690 usec 64 bytes from 192.168.XX.XX: icmp_seq=27 ttl=250 time=698 usec 64 bytes from 192.168.XX.XX: icmp_seq=28 ttl=250 time=713 usec 64 bytes from 192.168.XX.XX: icmp_seq=29 ttl=250 time=723 usec 64 bytes from 192.168.XX.XX: icmp_seq=30 ttl=250 time=703 usec 64 bytes from 192.168.XX.XX: icmp_seq=31 ttl=250 time=694 usec 64 bytes from 192.168.XX.XX: icmp_seq=32 ttl=250 time=685 usec 64 bytes from 192.168.XX.XX: icmp_seq=33 ttl=250 time=727 usec 64 bytes from 192.168.XX.XX: icmp_seq=34 ttl=250 time=720 usec 64 bytes from 192.168.XX.XX: icmp_seq=37 ttl=250 time=687 usec 64 bytes from 192.168.XX.XX: icmp_seq=38 ttl=250 time=704 usec 64 bytes from 192.168.XX.XX: icmp_seq=39 ttl=250 time=686 usec --- 192.168.XX.XX ping statistics --- 40 packets transmitted, 37 packets received, 7% packet loss round-trip min/avg/max/mdev = 0.002/0.695/0.969/0.126 ms On Sun, 04 Jan 2004 20:16:31 -0800 Mike Machado [EMAIL PROTECTED] wrote: I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: snip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
