[asterisk-users] MixMonitor and g729 licenses
Hi, I recently bought a handful of g729 licenses and moved all my equipment over to use it. We terminate most of our calls with a provider that supports g729, so it's g729 all the way through from the phone on the desk to the provider. Asterisk works very well in passthrough mode, simply moving the bits from the phone to the provider. Good work. The problem happens when I record a call using MixMonitor. Even though it's recording natively in g729, a single call uses 2 decoders and one encoder! The only explanation I can think of for that is that MixMonitor is transcoding the g729 streams to something else, muxing them, then encoding the muxed stream out to g729. This seems ridiculous - why go through all that work and licenses? Does anyone know for sure what's going on here? I could go back to using Monitor, I suppose, but MixMonitor is somewhat less hacky. Thanks jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Melbourne Asterisk Group meeting Thursday
Hi all, We're having a meeting this coming Thursday, June 1, 7:00 at Pint on Punt (same as last time). Peter Fern has graciously volunteered to lead a discussion on ENUM and DUNDi: What they are, how they work, and how they work with Asterisk. Of course there will also be time to talk about phones and what's new in the Asterisk community. Date: Thursday June 1 Time: 7:00pm Place: Pint on Punt, 42 Punt Road Windsor (Near St Kilda Road trams and Windsor train station) http://pintonpunt.com.au/ Come one, come all. Bring some cool toys if you like. Last time, we ended up building an Asterisk server with a bunch of phones, and connected it wirelessly to the phone network. Pretty good for a bunch of people in a pub. :-) We also have our own community mailing list: http://melbn.com/mailman/listinfo/voip See you there, jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone BT-101 audio problems
I didn't think SIP had a jitter buffer (yet). There's nothing in the wiki about it, as it pertains to sip.conf. In the meantime, I'll try a different NTP server, to see if that changes anything. On 01/02/06, Cristian Draghici <[EMAIL PROTECTED]> wrote: > No. If the same NTP server is used and both phones are in the same > switch, I don't see how disabling NTP would help. > > I'd rather disable jitterbuffer if it was on when the problem is > manifesting itself. > > -- > c > > > On 2/1/06, jurgen <[EMAIL PROTECTED]> wrote: > > NTP? Really? I would never have thought of that one. All the phones > > are configured to use pool.ntp.org, and they keep accurate time. Do > > you think disabling NTP altogether would be a good idea? > > > > Thanks! > > > > > > On 31/01/06, Cristian Draghici <[EMAIL PROTECTED]> wrote: > > > Are you using the same NTP server for both phones? > > > Are you using NTP at all? > > > > > > Is jitterbuffer enabled on the asterisk server? > > > > > > Not sure about SIP, but on IAX if the timestamps go haywire, you can > > > loose audio from one side. > > > > > > hth, > > > c > > > > > > > > > On 1/31/06, jurgen <[EMAIL PROTECTED]> wrote: > > > > Hi all, > > > > > > > > I'm having a really frustrating time with a bunch of BT-101 phones. > > > > They've been trouble-free and working very well for the past several > > > > months. A couple of days ago, some of the phones (but not all of them, > > > > yet) have started acting very strangely. All phones are running > > > > firmware 1.0.6.7, and are identically configured (except for the > > > > user/authenticate/password things) on both the phone side and in > > > > sip.conf. > > > > > > > > After a few hours of relatively heavy use, the phone stops sending the > > > > remote party's voice to the BT-101 person. Someone else (me, heh heh) > > > > listening in to the call via ZapScan can hear both sides just fine, so > > > > it doesn't seem to be Asterisk's problem, at least directly. > > > > > > > > I've tried simply power cycling the phones. Doing that buys me a bit > > > > of time, sometimes a minute, sometimes ten. But the problem always > > > > comes back relatively quickly. Moving the phones to another physical > > > > Ethernet connection does nothing either. > > > > > > > > The way to make the problem go away for about 24 hours is to swap them > > > > around. I move a spare from my desk to the person with the bad phone, > > > > simply by changing the user/auth/pass strings. I set the broken phone > > > > up with my testing user/auth/pass stuff, and they both start working > > > > again. Now get this: Simply changing the user/auth/pass strings on the > > > > bad phone to something else, then setting them back to what they were > > > > before *doesn't work*. The actual phones have to be swapped around. > > > > The phones are plugged into the same switch as the Asterisk server, no > > > > funny stuff there. > > > > > > > > I've tried resetting the phones back to the factory settings, and > > > > reconfiguring them from scratch. That doesn't eliminate the problem > > > > either, just delays it another 24 hours. > > > > > > > > So. That pretty much covers it. Only three of the phones are doing > > > > this, and they only started a few days ago. They were working *fine* > > > > for months! Does anyone have any ideas here? I'm about ready to throw > > > > these phones into a tree shredder. > > > > > > > > Thanks! > > > > > > > > ..jurgen > > > > > > > > -- > > > > [EMAIL PROTECTED] is jurgen's gmail address. > > > > Visit http://jurgen.ca/ for more yummy goodness. > > > > ___ > > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > > > Asterisk-Users mailing list > > > > To UNSUBSCRIBE or update options visit: > > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > Asterisk-Users
Re: [Asterisk-Users] Grandstream Budgetone BT-101 audio problems
NTP? Really? I would never have thought of that one. All the phones are configured to use pool.ntp.org, and they keep accurate time. Do you think disabling NTP altogether would be a good idea? Thanks! On 31/01/06, Cristian Draghici <[EMAIL PROTECTED]> wrote: > Are you using the same NTP server for both phones? > Are you using NTP at all? > > Is jitterbuffer enabled on the asterisk server? > > Not sure about SIP, but on IAX if the timestamps go haywire, you can > loose audio from one side. > > hth, > c > > > On 1/31/06, jurgen <[EMAIL PROTECTED]> wrote: > > Hi all, > > > > I'm having a really frustrating time with a bunch of BT-101 phones. > > They've been trouble-free and working very well for the past several > > months. A couple of days ago, some of the phones (but not all of them, > > yet) have started acting very strangely. All phones are running > > firmware 1.0.6.7, and are identically configured (except for the > > user/authenticate/password things) on both the phone side and in > > sip.conf. > > > > After a few hours of relatively heavy use, the phone stops sending the > > remote party's voice to the BT-101 person. Someone else (me, heh heh) > > listening in to the call via ZapScan can hear both sides just fine, so > > it doesn't seem to be Asterisk's problem, at least directly. > > > > I've tried simply power cycling the phones. Doing that buys me a bit > > of time, sometimes a minute, sometimes ten. But the problem always > > comes back relatively quickly. Moving the phones to another physical > > Ethernet connection does nothing either. > > > > The way to make the problem go away for about 24 hours is to swap them > > around. I move a spare from my desk to the person with the bad phone, > > simply by changing the user/auth/pass strings. I set the broken phone > > up with my testing user/auth/pass stuff, and they both start working > > again. Now get this: Simply changing the user/auth/pass strings on the > > bad phone to something else, then setting them back to what they were > > before *doesn't work*. The actual phones have to be swapped around. > > The phones are plugged into the same switch as the Asterisk server, no > > funny stuff there. > > > > I've tried resetting the phones back to the factory settings, and > > reconfiguring them from scratch. That doesn't eliminate the problem > > either, just delays it another 24 hours. > > > > So. That pretty much covers it. Only three of the phones are doing > > this, and they only started a few days ago. They were working *fine* > > for months! Does anyone have any ideas here? I'm about ready to throw > > these phones into a tree shredder. > > > > Thanks! > > > > ..jurgen > > > > -- > > [EMAIL PROTECTED] is jurgen's gmail address. > > Visit http://jurgen.ca/ for more yummy goodness. > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream Budgetone BT-101 audio problems
Hi all, I'm having a really frustrating time with a bunch of BT-101 phones. They've been trouble-free and working very well for the past several months. A couple of days ago, some of the phones (but not all of them, yet) have started acting very strangely. All phones are running firmware 1.0.6.7, and are identically configured (except for the user/authenticate/password things) on both the phone side and in sip.conf. After a few hours of relatively heavy use, the phone stops sending the remote party's voice to the BT-101 person. Someone else (me, heh heh) listening in to the call via ZapScan can hear both sides just fine, so it doesn't seem to be Asterisk's problem, at least directly. I've tried simply power cycling the phones. Doing that buys me a bit of time, sometimes a minute, sometimes ten. But the problem always comes back relatively quickly. Moving the phones to another physical Ethernet connection does nothing either. The way to make the problem go away for about 24 hours is to swap them around. I move a spare from my desk to the person with the bad phone, simply by changing the user/auth/pass strings. I set the broken phone up with my testing user/auth/pass stuff, and they both start working again. Now get this: Simply changing the user/auth/pass strings on the bad phone to something else, then setting them back to what they were before *doesn't work*. The actual phones have to be swapped around. The phones are plugged into the same switch as the Asterisk server, no funny stuff there. I've tried resetting the phones back to the factory settings, and reconfiguring them from scratch. That doesn't eliminate the problem either, just delays it another 24 hours. So. That pretty much covers it. Only three of the phones are doing this, and they only started a few days ago. They were working *fine* for months! Does anyone have any ideas here? I'm about ready to throw these phones into a tree shredder. Thanks! ..jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Evening in Melbourne: Feb 2!
On 31/01/06, Skeeve Stevens <[EMAIL PROTECTED]> wrote: > Any idea if there will be a Sydney one? There was some talk about Sydney doing something, but as far as I know, no one has done anything about it yet. There's a page in the wiki about Sydney, with some contact information of the person who's starting things: http://www.voip-info.org/wiki/index.php?page=VoIP+User+Groups+Australia jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Evening in Melbourne: Feb 2!
Hi all, Come one come all! We're having the next Asterisk evening at the Fujitsu Centre for Excellence! This is Fuji's state of the art show-off centre - they're promising lots of interesting toys to play with. As usual, we'll be discussing developments in Asterisk land over the past couple of months. If you've got some interesting toys yourself, please bring them along! Date: Thursday February 2nd (the day after tomorrow!) Time: 7pm - late Place: Fujitsu Centre for Excellence, 1 Southbank Boulevard, Southbank (Inside the Pacific Internet building) After Fujitsu makes us leave, we'll be heading to a local cafe or pub to continue our collective phone geekiness. If you have trouble finding us, please give me a call (my number is in the .sig below). Many thanks to Joseph Sirucka for securing the venue for this evening, and to Fujitsu for letting us use it. More about Fujitu's playground here: http://www.fujitsu.com/au/news/pr/archives/2005/20051109-01.html Best, ...jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk Evening in Melbourne Australia!
Just a quick reminder - this is happening *TONIGHT*. Hope to see all local Asteriskers come out (except PaulH, who went to great lengths to avoid us this time). jurgen On 14/10/05, jurgen <[EMAIL PROTECTED]> wrote: > Hi all, > > Come out come out! If you're involved in Asterisk and live around the > Melbourne area, please come out and join us for an evening of geeking > out with Asterisk, socialising and generally having fun. > > Please note, people who have before, the venue has changed from last > time because it was invaded by an annoying DJ. > > Date and time: Thursday October 20th at 7pm. > Location: Mitre Tavern: http://www.melbournepubs.com/v/487/ > > If it's a warm evening, we'll be outside in the courtyard, but if it's > not so warm, look for us inside. I'll bring along an old skool Telecom > 9600 PABX phone and put it on the table. If anyone else has some > classic technology, bring it along for a laugh. We've been thinking > about doing a more geeky, less social evening as well, so we'll be > talking about that - plus whatever else everyone has been up lately. > > Questions? Send them to [EMAIL PROTECTED], or give me a ring on 0415 276 127. > > See you there! > > ...jurgen > > > -- > [EMAIL PROTECTED] is jurgen's gmail address. > Visit http://jurgen.ca/ for more yummy goodness. > -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Evening in Melbourne (again!) next Thursday
Hi all, If you're in Melbourne Australia and interested in Asterisk, you're invited to join us for the second in an irregularly scheduled casual evening to talk about Asterisk, VOIP, networks, and just generally get geeky about IP phone stuff. About a dozen of us got together a couple of months ago, and had a good time chatting about all things Asterisk. Beverages were also consumed. Anyone with an interest is welcome; from Asterisk Gods to newbies who have recently downloaded it, from people administering several hundred seats to people playing with it at home and annoying their families. When: Next Thursday evening, the 16th, at 7pm. Where: Niagara Hotel, 383 Lonsdale Street (between Queen and Elizabeth) in the city. The Niagara's a relaxed, comfortable place, people seemed to like it last time. Also, like last time, I'll get an old phone and put it on the table, so those of us who haven't met will be able to recognise each other. Any questions, you can reach me on 0415 276 127, or email [EMAIL PROTECTED]. Hope to see you there! ...jurgen-- [EMAIL PROTECTED] is jurgen's gmail address.Visit http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do you name yours
In my varied career, I've named servers after: Composers (until I realised people couldn't spell "Tchaikovsky") Film directors (ditto for "Kieslowski") Streets of Vancouver (cuz that's where I was) Rivers of Australia (cuz that's where I am) On 11/05/05, David John Walsh <[EMAIL PROTECTED]> wrote: Hello listwe are installing 2 new servers (to run asterisk) shortly, for a"stand alone" service. Ignoring our current naming convention, we'dlike to name them something.. but we are not sure what. a consideration is that on the screens of the phones it shows[EMAIL PROTECTED] (eg [EMAIL PROTECTED]) (all extensions are numeric) sothe users will see it everydayi'm not creative in this way, it doesn't need to be a silly reference (like jarjar and anikin etc) per sebut im curiouswhat would you name them?___Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- [EMAIL PROTECTED] is jurgen's gmail address.Visit http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Minimal hardware requirements
(Thanks Paul) And that same box now has a TDM-400 card in it, all 4 ports used. Two ATs are registered with the server as well. Most of the time, it doesn't even break a sweat. I would not want to use it for anything close to production though. On Tue, 22 Feb 2005 14:25:00 +1100, Paul Hales <[EMAIL PROTECTED]> wrote: > And I forgot - once we were finished with the 600 we gave it to Jurgen. > > Caring and sharing, > > PaulH > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales > Sent: Tuesday, 22 February 2005 2:03 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] Minimal hardware requirements > > We did our proof on concept on a celeron 600 - which was fine to run 2 > software and 2 hardware phones off. > > Original testing was with sjphone and xlite. (software phones) > > We didn't fit a TDM card until we set up our first box, which was a dual > p3-1000. > > Later, > > PaulH > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf > Ladyzhenskii > Sent: Tuesday, 22 February 2005 11:35 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Minimal hardware requirements > > Hi, all > > I am doing "prrof of concept" system. I will have two IP phones connected to > Asterisk box. Box itself will have 1 PSTN conenction and one analog phone > conenction. A basic minimal configuration. > > At the moment I am planning to use an old PII-350 with 128M of RAM I have > lying around. I can not test anything yet, as I am waiting for phones to > arrive, so question is will that be enough to demonstrate? > > Thanks, > Rudolf > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > CAUTION: This email message and accompanying data may contain information > that is confidential. If you are not the intended recipient, you are notified > that any use, dissemination, distribution or copying of this message or data > is prohibited. If you have received this email message in error, please > notify us immediately and erase all copies of this message and attachments. > Thank you. > CAUTION: This email message and accompanying data may contain information > that is confidential. If you are not the intended recipient, you are notified > that any use, dissemination, distribution or copying of this message or data > is prohibited. If you have received this email message in error, please > notify us immediately and erase all copies of this message and attachments. > Thank you. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > CAUTION: This email message and accompanying data may contain information > that is confidential. If you are not the intended recipient, you are notified > that any use, dissemination, distribution or copying of this message or data > is prohibited. If you have received this email message in error, please > notify us immediately and erase all copies of this message and attachments. > Thank you. > CAUTION: This email message and accompanying data may contain information > that is confidential. If you are not the intended recipient, you are notified > that any use, dissemination, distribution or copying of this message or data > is prohibited. If you have received this email message in error, please > notify us immediately and erase all copies of this message and attachments. > Thank you. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: * > Mobile Phone > Mobile Network
I'll second PaulH's recommendation of the Telular SX-5e units. They plug into an FXO on the Asterisk machine (or wherever). Put anyone's capped SIM into the thing, and you're communicating with the GSM network. As a little added bonus, there's a serial port for sending and receiving text messages. You can either send AT-commands to the thing (it works like a modem), or get some software like SMS Server Tools to do it for you. There are also PRI-interface based units, but they're expensive and really designed for high-end uses. Also expensive (but said to be coming down in price soon) is a SIP-based box called the VoiceBlue. jurgen On Mon, 21 Feb 2005 15:43:43 +0300, AR Tarzi <[EMAIL PROTECTED]> wrote: > I've used a Nokia 32 unattended (remote) for the past year or so. > > "David Uzzell" <[EMAIL PROTECTED]> wrote in message > news:[EMAIL PROTECTED] > | Ok I have a question. Seen it come and go around the mailling list for a > | while but never really seen an answer that seems to sort it out. > | > | What is needed is some interface from * > Mobile Phone > Mobile Network > | Service. > | > | At this point all the providers in AUS that I have found are charging a > | Premium Rate for Land Line > Mobile Network services. > | > | What I would like to do is be able to purchase a low rate Mobile SIM > | that I can chuck into a Mobile Phone and have it setup so that I route > | the Mobile calls through it. > | > | Rembering that most if not all mobile phones can be accessed via RS232 > | interface. > | > | Anyone done this or seen it done or know how to do it using * and whatever? > | > | Cheers > | David > | ___ > | Asterisk-Users mailing list > | Asterisk-Users@lists.digium.com > | http://lists.digium.com/mailman/listinfo/asterisk-users > | To UNSUBSCRIBE or update options visit: > |http://lists.digium.com/mailman/listinfo/asterisk-users > | > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Melbourne Asterisk Users meet TONIGHT
Hi all, Just a quick reminder: If you're in Melbourne and want to talk Asterisk or VOIP in general, tonight's the night. Come out come out! It ought to be a fun evening. Details below: -- Forwarded message ------ From: jurgen <[EMAIL PROTECTED]> Date: Thu, 10 Feb 2005 12:54:43 +1100 Subject: Melbourne Asterisk Users meet next Thursday To: Asterisk-Users@lists.digium.com Hi all, If you're in Melbourne Australia and interested in Asterisk, you're invited to join us for a casual evening to talk about Asterisk, VOIP, networks, and just generally get geeky about IP phone stuff. Ultimately, I think it would be interesting and useful to turn this into a monthly get-together, so I'd like to talk about that too. Anyone with an interest is welcome; from Asterisk Gods to newbies who have recently downloaded it, from people administering several hundred seats to people playing with it at home and annoying their families. When: Next Thursday evening, the 17th, at 7pm. Where: Niagara Hotel, 383 Lonsdale Street (between Queen and Elizabeth) in the city. The Niagara's a relaxed, comfortable place. I'm going to try and get us a table, and put an old analogue phone on it, so you'll know how to find us. Any questions, you can reach me on 0415 276 127, or email [EMAIL PROTECTED] Hope to see you there! ...jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wireless LANs and Asterisk
Hi Mike, This is interesting - it's something that I've been considering doing for the Asterisk rollout at my company. We don't have enough Ethernet ports and I'm not thrilled about the expense of re-wiring the place. Have you tried D-Link's dual-channel gear for even more bandwidth, or do you feel that bandwidth is not really a problem? How resilient is 802.11g against interference from other sources? Microwave ovens, gigarange phones, etc. Thanks for reporting your success here to the list, just proves I'm not alone with my funny ideas. ..jurgen On Thu, 10 Feb 2005 12:39:00 -0600, Mike Meyer <[EMAIL PROTECTED]> wrote: > Has anyone had any experience with wireless LANs and Asterisk? > > We have and here are my impressions. > > We configured an Asterisk in the office as a precaution to see how it > would work for our own retail customers. Our office is open space, about > 800 sq ft. (20x40 area). We use Snom200 and Grandstream SIP phones. > > Using the latest Linksys wireless access point (WAP54g) and 3 wireless > bridges (WET54g), I have found that it works most of the time with WPA > encryption on, but will occasionally drop voice (loosing packets). With > no encryption on the WLAN it seems to work without a hitch! Using a less > CPU intense encryption such as 64bit WEP, things also work fine. There > must be too much delay with higher rate encryption. > > Also we had one bridge that seemed to be a week puppy in the litter. It > could only muster 60-70% signal strength. It seemed to have problems > under all configurations. Finally we positioned it such that it too > works well running WEP 64b. I wonder if having 3 wireless bridges in > close proximity would have anything to do with the signal strength? I > would doubt it though. > > Anyone else with other experiences to share regarding wireless LANs and > encryption? I'd me interested to hear them. > > Thanks, > Mike Meyer > GenDesign Corporation > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Melbourne Asterisk Users meet next Thursday
Hi all, If you're in Melbourne Australia and interested in Asterisk, you're invited to join us for a casual evening to talk about Asterisk, VOIP, networks, and just generally get geeky about IP phone stuff. Ultimately, I think it would be interesting and useful to turn this into a monthly get-together, so I'd like to talk about that too. Anyone with an interest is welcome; from Asterisk Gods to newbies who have recently downloaded it, from people administering several hundred seats to people playing with it at home and annoying their families. When: Next Thursday evening, the 17th, at 7pm. Where: Niagara Hotel, 383 Lonsdale Street (between Queen and Elizabeth) in the city. The Niagara's a relaxed, comfortable place. I'm going to try and get us a table, and put an old analogue phone on it, so you'll know how to find us. Any questions, you can reach me on 0415 276 127, or email [EMAIL PROTECTED] Hope to see you there! ...jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitor calls timeout
Hi Trevor, > That's because * is getting tired of waiting for the caller to dial an > extension. Try this > > exten => s,1,Answer > exten => s,2,Monitor(wav,testrecord,m) > exten => s,3,Wait(600) > exten => s,4,Goto(s,3) Awfully clever, Trevor. It works brilliantly! Also gives the added bonus of being able to specify a maximum timout to prevent runaway recordings. I've raised the Wait(600) to a Wait(900) (15 minutes), and eliminated the Goto. With busy detection, once both parties hang up, the call is terminated and a recording is generated. Or if the recording goes over 15 minutes, it's automatically aborted. Thanks very much! ...jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap channels in AU hanging up on STD pips
Hi Howard, Which provider are you with? We're with Primus Business here in Melbourne, and haven't had anything like what you're describing. For reference, here's a snip of my zapata.conf: [channels] language=en context=local signalling=fxs_ks usecallerid=no echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=5 Sometimes the busydetect hack hits a false positive and disconnects during a conversation, so I'm thinking of upping the busycount, but aside from that, calls through this are quite reliable. Best, ...jurgen On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes <[EMAIL PROTECTED]> wrote: > Is anyone having/had a problem with a TDM400P card hanging up on STD > outbound calls as soon as the called party answers. > > I'm guessing that * is responding to the STD pips in some way. > > -- > Howard. > LANNet Computing Associates; > Your Linux people <http://www.lannetlinux.com> > -- > "When you just want a system that works, you choose Linux; > when you want a system that just works, you choose Microsoft." > -- > "Flatter government, not fatter government; > Get rid of the Australian states." > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitor calls timeout
On Mon, 31 Jan 2005 09:26:05 +0800, el Flynn <[EMAIL PROTECTED]> wrote: > did you try setting using AbsoluteTimeout in the context? e.g. > > exten => s,1,Answer > exten => s,2,AbsoluteTimeout(0) > exten => s,3,Monitor(wav,testrecod,m) Thanks for the suggestion, but it's no good. It still times out after 10 seconds. It seems to be something in the Monitor application, rather than anywhere else. I can playback a sound (like the monkeys, or MOH) forever and ever without timing out. Monitoring kills itself though. Oh - and using Monitor the way it's "supposed" to be used works just fine, with no problems or timeouts (I have one of the Zap channels set to record everything, and that works all the time). jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitor calls timeout
Hi all,
We're in a transition between OldPhoneSystem and Asterisk. One of the
things that's needed to be done right now with OldPhoneSystem is the
ability to record calls. I thought "Asterisk can record calls", so I
set about to make it happen. And it does, sort of.
I made a .call file that rings the exension that I want to have
recorded, and barges into the conversation, using a series of DTMF
codes that OldPhoneSystem understands. That bit works with no
problems. Once it's connected, the context I've placed the call into
looks like this:
exten => s,1,Answer
exten => s,2,Monitor(wav,testrecord,m)
And even that works - recording files are made called "testrecord"
that contain the conversation from the correct Zap channel.
Problem is, Asterisk times out and disconnects after 10 seconds,
stopping the recording.
If I run something else in the context, say the infamous Monkey
Sounds, everything's fine, and the call just keeps going, annoying the
people on the line with monkey sounds. For some reason, the
*monitoring* always stops after 10 seconds.
Here's what the console tells me:
-- Attempting call on Zap/4/442,55 for [EMAIL PROTECTED]:1 (Retry 1)
> Channel Zap/4-1 was answered.
-- Executing Answer("Zap/4-1", "") in new stack
-- Executing SetVar("Zap/4-1",
"RECORDFILENAME=testrecording-s-20050131-102716") in new stack
-- Executing Monitor("Zap/4-1", "wav||m") in new stack
[all good so far]
Jan 31 10:27:26 WARNING[27937712]: pbx.c:1977 ast_pbx_run: Timeout,
but no rule 't' in context 'record'
-- Hungup 'Zap/4-1'
[okay, so I don't have a 't', but it shouldn't be timing out anyway!]
monitor executing ( nice -n 19 soxmix
//var/spool/asterisk/monitor/Zap-4-1-in.wav
//var/spool/asterisk/monitor/Zap-4-1-out.wav
//var/spool/asterisk/monitor/Zap-4-1.wav && rm -f
//var/spool/asterisk/monitor/Zap-4-1-* ) &
Jan 31 10:27:26 NOTICE[27937712]: pbx_spool.c:244 attempt_thread: Call
completed to Zap/4/442,55
Does anyone have any ideas that could help here?
Thanks very much,
.jurgen
--
[EMAIL PROTECTED] is jurgen's gmail address.
Visit http://jurgen.ca/ for more yummy goodness.
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Re: [Asterisk-Users] Connecting Asterisk to GSM
Hi Martin, I've looked at a few different options, including 2N's 4-channel SIP-to-GSM gateways, and the cheapest and most reliable I've been able to find (at least for people who need fewer than 8 ports) is a combination of Digium's 4-port FXO card and four Telular PhoneCell SE5e units. When you get into larger needs than that, there are a bunch of other options available, including PRI as Matteo suggested. Check into the Wiki (http://www.voip-info.org/wiki-Asterisk+Connecting+to+the+Cellular+Network ) for lots more products and information, including an interesting option called CellSocket, which might be cheaper for you. .jurgen On Thu, 16 Dec 2004 19:27:06 +, Martin List-Petersen <[EMAIL PROTECTED]> wrote: > On Thu, 2004-12-16 at 21:47, Jean-Michel Hiver wrote: > > Hi List, > > > > I was wondering if there was any device I could use to connect * to GSM > > networks. I don't need much capacity, maybe 2-4 GSM channels. As usual, > > cheap is better :-) > > > > What you are looking for is something like the Ateus GSM to PSTN or ISDN > gateways > (http://www.mobilecomms-technology.com/contractors/gsm/2n_tele/) > > Cheaper would be some gsm to pstn adapter, that you can connect to the > cellphone. Check the archives of the asterisk-users for that, because > it's something, that commonly has been asked before. > > Another alternative would be chan_bluetooth, > (http://www.crazygreek.co.uk/content/chan_bluetooth), but that is in a > state far from working. > > Slán leat, > Martin List-Petersen > Dublin, Eire > (contact info on --> http://www.marlow.dk/) > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ateus VoiceBlue
Hi all, Simple question: Is anyone successfully using an ATEUS VoiceBlue SIP <-> GSM box with your Asterisk setup? They look pretty nifty: http://export.2n.cz/index.phtml?l1=products&object_id=138 Thanks! ...jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Check out http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as plain PABX in call centre
Hi, Thanks for the reply. > Depending on why you want to get rid of the fujitsu. What is it that > makes you want to get rid of it? Not enough handsets, not enough > incoming lines, or not enough features etc... All of the above. :-) We're getting close to maxing out our 30-channel ISDN connection, and there's literally no more room for expansion in the old Fuji. We're scraping for headsets too, and there are some features (like call queuing, IVRs and voicemail) that it's just not able to do. > > We've also got a dozen or so plain analogue lines, used for fax > > machines, fax modems (Hylafax!), answering machines (real ones!) and > > other assorted weird stuff. > > AFAICT, you are best off trying to have these sorts of lines totally > by-pass the PBX, (any sort of PBX). So, really, just ignore this stuff > for the moment. Later on, you can try out the rxfax/txfax stuff to > replace hylafax and/or other fax machines. The analogue lines are currently hanging off the Fujitsu now. It provides a dial tone so these devices feel more comfortable - plus it allows me to route numbers from the ISDN block through to analogue-line devices in the office. > Of course you can do this, but you will need a channel bank for all > those non-VoIP phones you want. You will probably want to use T1 channel > banks with the TE4xx cards. One E1 coming in plus 2 or three T1 channel > banks. I guess what I'm missing here is *how*. How does Asterisk know what to do with all the extra buttons on these phones? How does it know what to do with the LCD screen? Some of the buttons simply send DTMF into the PABX, but there are other ones that are specially programmed lines etc etc. Surely there's some kind of protocol involved right now in the way the phones talk to the PABX. I'm not even sure of what to look for in this regard. > > I need some testimonials from people using Asterisk in a call centre > > environment. There must be some people out there doing it. How did you > > pull it off? What equipment are you using? What do I need to watch out > > for along the way? How's it working out for you now? I'd like to be > > able to go to the Directors with as much backup as possible. "Here are > > a bunch of similar setups, all working great and saving money". > > Isn't this sort of stuff listed on the wiki?? I found a few testimonials, but they're more geared to the VoIP user. While this is encouraging, I haven't found any having to do with using it as a plain old ordinary PABX, or in a call centre environment. I'd love to be proved a blind fool who can't search properly though. :-) Best, .jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Check out http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as plain PABX in call centre
Hi all, I've been lurking here and reading the Wiki for a month or so now, getting information on the suitability of Asterisk for my installation. I'm responsible for the technical stuff at a mostly-inbound call centre in Melbourne, Australia. Due to our rapid expansion, it's getting close to the time to put our old beloved Fujitsu 9600 PABX out to pasture. I'm evaluating different options, both hardware-based and Asterisk, to go forward. Right now, we've got a single ISDN line, with a 100-number block attached to it, and 30 incoming lines. There is also a small handful of PSTN lines that we need to maintain for historical purposes. Our phones are all Fujitsu AT-class models - about 30 of them. I'd like to hang on to them if I can, but it's looking less and less likely, unless I keep the Fujitsu PABX and run Asterisk in parallel somehow. We've also got a dozen or so plain analogue lines, used for fax machines, fax modems (Hylafax!), answering machines (real ones!) and other assorted weird stuff. Personally, I like Asterisk, and I'd like to be able to recommend it to the Directors as The Way To Go. I like the flexibility, the programmability, the database friendliness, and the openness of the code and architecture. Terminating calls through VoIP isn't that important to us right now, but it's a great enabling technology, and I'm looking forward to implementing it later on. Essentially though, right now we're looking to use Asterisk as a pretty much ordinary PABX: voicemail, call queues, call parking, music on hold, etc etc. Nothing particularly envelope-pushing. If I can avoid using IP or SIP phones right off the bat, so much the better. This is *the* most important thing to our business. Changing anything about how our phones work makes me very nervous (and the Directors even more so). So in order to convince myself, and the Directors too, I need some testimonials from people using Asterisk in a call centre environment. There must be some people out there doing it. How did you pull it off? What equipment are you using? What do I need to watch out for along the way? How's it working out for you now? I'd like to be able to go to the Directors with as much backup as possible. "Here are a bunch of similar setups, all working great and saving money". Thanks very much, in advance. jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Check out http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
