Re: [asterisk-users] Decent Voip Phones for enterprise
On Oct 28, 2008, at 5:13 PM, Kev Szaszvari wrote: Hi there Can anyone reccomend any voip phones( Cisco, Polycom, SNOM ) that have * Central Management for all the phones (We dont mind if we have to buy the software to manage them) * Programable shortcut buttons, So i can program in on certian phones quick dials to queues. * Optional but bonus, The ability to have a shared address book accross the phones. We just rolled out Snom phones and it was the easiest thing in the world. 1) Yes, you can centrally manage your phones. If you realtime SIP with a database then you can do a complete plugplay setup. We use a few scripts to do this here. a) Script to respond to the Snom plugplay request (SIP broadcast message), redirects to PHP script (b). b) A few PHP scripts that update the firmware, provision the phones (via the database), define all the standard buttons, and allow overrides based upon extension number. 2) The Snom's let you program every single button. If you want to re- program the conference button to be a hold button, *shrug* go for it. You can program a button to function as a BLF, speed-dial and call- pickup button all at the same time. (Current 7.3.7 has a bug that only lets you speed-dial and call-pickup when the phone is on-hook, latest beta fixes that). 3) Nearly perfect support for LDAP directory. I say nearly because if you enable the number lookup feature (in addition to the name lookup) then anytime you dial it will immediately match by name and not let you see the number you are dialing. It basically forces you to dial-by- directory, kind of annoying. I got a bug report in on that. In regards to 1b, the PHP script gets the MAC address from the phone (via the URL requested), queries the database, sends back an XML file with all the registration information. With SIP realtime, what this means is that you get a new phone, put in the registration information in your database along with the MAC address of the phone, then plug the phone into the network. Come back 10 minutes later and it has updated itself to the latest firmware and is ready to make receive calls. If you want more specific information on this I would be happy to give you the scripts. As others have noted, the Linksys may be able to do what you want. But if you do end up switching I recommend the Snom if you want the best bang for your buck. Cisco Polycom are good phones, but getting a big enough phone that has programmable buttons etc. gets really pricy. Grandstream is okay, but after comparing them with the rest of the phones they audio quality just isn't there. And for us, the Snom is the only phone I could successfully program to do single-button call parking, which was a major requirement. Daniel Thanks in advance Regards, Kev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)
2008/10/28 Robert Boardman [EMAIL PROTECTED] Olivier wrote: 2008/10/5 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Kevin P. Fleming wrote: Olivier wrote: 2. R Hook-flash key is now available to transfer calls. In s450IP web management server, its defaults settings are : Application-type: dtmf-relay Application-signal: 16 Is there anything to configure in features.conf, extensionsconf or elsewhere to trigger transfers when R key is pressed ? I don't believe there is any support for hook-flash style transfers over SIP in Asterisk; that key should be programmed to use standard SIP transfer methods, not DTMF emulation methods. do you have a suggestion, there is only two fields that can be filled in that to refer to the R key, Application-type: I think this is content type Application-signal: what it sends? Hello, Reading this thread, I think I should have opened in the first place, 2 different threads as a common title is misleading to this R Hook-Flash key topic. Now, Gigaset S450IP base configuration web offers 2 fields to set R key : Application-type: Application-signal: When those 2 fields are respectively valued to Application-type: dtmf-relay Application-signal: 16 ... anytime this R-key is pressed, the base station would send a SIP INFO message to Asterisk. This SIP info is ended with : ... User-Agent: S450 IP02123000 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Type: application/dtmf-relay Content-Length: 22 Signal=16 Duration=86 This 16 signal is interpreted as : Receiving INFO! * DTMF-relay event received: FLASH In my testing, I changed values like this Application-type: foo Application-signal: 16 2 and got a (single) SIP INFO message like this: User-Agent: S450 IP02123000 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Type: application/foo Content-Length: 22 Signal=16 2 As Kevin told previously, Hook Flash transfer is not supported by Asterisk SIP stack. At the same time, it is written here (http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP) that : * Enable the R-button in SIP mode /fixed 14/09/2007/ So, what does this exactly mean ? Which values are to be typed in Application type and Application signal to make this R key be of any use ? Is it possible to pass several DTMF signals in a single SIP INFO so that Asterisk would receive a *2 anytime the R-key is pressed ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'll reply to the correct thread [featuremap] blindxfer = ## ; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = A ; Attended transfer so set atxfer to 'A' and DTMF relay Application signal on the Gigaset to 'A' (without quotes) and transfer works as expected Robb Thanks for replying ! I'll give it a try and report to the list ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI with Siemens Gigaset S450IP
2008/10/28 Robert Boardman [EMAIL PROTECTED] Olivier wrote: 2008/10/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi, 1. Here http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP it is mentioned MWI is now working. In my testings with lastest 02123 firmware, MWI is blinking when missed calls but not when a message in present in voicemail. With SIP debug I can see 481 Call Leg/Transaction Does Not Exist replies to NOTIFY announcing new messages. With previous firmware, I had 415 Unsupported Media if my memory is correct. Has anyone been any further ? Regards Replying to myself, for an unknown reason, MWI is weirdly working : - Phone icon inconsistently shows awaiting voicemails, - NOTIFY message from Asterisk are still replied with 481 Call Leg/Transaction Does Not Exist When base station is restarted, it will SUBSCRIBE its endpoints to Voicemail Notifications : - you can see SUBSCRIBE message - you can see NOTIFY answer - you can't see any 481 Call Leg/Transaction Does Not Exist reply to this NOTIFY message From then on, further NOTIFY messages are replied with 481 Call Leg/Transaction Does Not Exist and obviously not taken into account as endpoint GUI remains unchanged. Looking deeper into this here are : NOTIFY message accepted by S450IP NOTIFY sip:[EMAIL PROTECTED]:5060 http://sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport From: asterisk sip:[EMAIL PROTECTED];tag=as4ea953db To: sip:sip:[EMAIL PROTECTED]:5060 http://sip:sip:[EMAIL PROTECTED]:5060;tag=2580238520 Contact: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 2/0 (0/0) NOTIFY message rejected by S450IP (rejected means 481 reply) NOTIFY sip:[EMAIL PROTECTED]:5060 http://sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK3d83e7f6;rport From: asterisk sip:[EMAIL PROTECTED][EMAIL PROTECTED] mailto:[EMAIL PROTECTED][EMAIL PROTECTED] ;tag=as5e574490 To: sip:[EMAIL PROTECTED]:5060 http://sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 96 Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED][EMAIL PROTECTED] mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Voice-Message: 3/0 (0/0) The only difference I see between both is that new NOTIFY don't include : Subscription-State: active Do you see something else ? Is it possible to easily add this Subscription-State field without patching Asterisk source (I'm unable to do that) ? Your thoughts ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Just worked out a good way of getting transfer working Using features .conf [featuremap] blindxfer = ## ; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = A ; Attended transfer DTMF A-D are valid DTMF signals but are not usually shown on standard phones so set atxfer to 'A' and DTMF relay Application signal on the Gigaset to 'A' (without quotes) and transfer works as expected Robb Hi, What about MWI and Subscription-State: active ? I can see that Asterisk sends NOTIFY messages with and without this Subscription-State: active statement in header. I can see that NOTIFY messages without Subscription-State: active are rejected by Gigaset base station. Is it possible to either configure : 1. Gigaset to accept NOTIFY messages without Subscription-State: active 2. Asterisk to send NOTIFY messages with Subscription-State: active Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any dialplan action on received jabber msgs?
2008/10/29 Brian J. Murrell [EMAIL PROTECTED] So I have (and have had) jabber configured for some time, specifically for GTalk, but something has occurred to me. If somebody happens to send an IM (text) to that account, nobody is going to be receiving it. I'd like to send a canned message back to any sender of an IM. Possible? b. I've been told a JabberReceive application belongs to one of Asterisk branches. As it needs an active channel, I can't tell how it could help when no channel is active. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone using an Intel Atom ?
On Wed, 29 Oct 2008, Peter Evans wrote: Gordon Henderson wrote: I just wish there was a fanless version - one feature which I like in the VIA boards I use. MSI Wind Board. No idea about outside Japan, but its fanless, almost certainly needs convection. That's because it's called a Wind Board... Great name!!! Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)
Olivier wrote: snip / I'll reply to the correct thread [featuremap] blindxfer = ## ; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = A ; Attended transfer so set atxfer to 'A' and DTMF relay Application signal on the Gigaset to 'A' (without quotes) and transfer works as expected Robb Thanks for replying ! I'll give it a try and report to the list I just tested this and it seems to work with the Siemens S685IPs. This thread was such a coincidence. We were trying to get attended transfer to work last night but setting the atxfer to normal things like *2 just didn't work. I just set my S685IP base station to A for the Application Signal and set A in the features.conf and behold, when I now press the R key, Attended Transfer :-) Thanks Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Decent Voip Phones for enterprise
On Wed, 29 Oct 2008, Kev Szaszvari wrote: Hi there Our company is using the Linksys SPA-942 Phones, and they are pretty useless. They dont have any central management or provisioning, as well as a pretty bad interface. Can anyone reccomend any voip phones( Cisco, Polycom, SNOM ) that have * Central Management for all the phones (We dont mind if we have to buy the software to manage them) I always wondered about this - my target is the SME - say 4-150 seats - people don't move desks, change office that often, staff churn is typically low, so I program the phones once then leave them there. If you move desk you take your phone with you. If you leave then the phone can be renamed via it's web interface relatively easily. Maybe I'm just dealing with simple (dumb?) offices, but I'm curious as to what people do with the phones that require this sort of central management. (And regular phone updating) * Programable shortcut buttons, So i can program in on certian phones quick dials to queues. How about implementing this in the PBX. * Optional but bonus, The ability to have a shared address book accross the phones. Same here. So some phones do have nice programmable buttons - and that's good, but in my PBX I have the space for about 600 speed-dials (3-digit extensions) which are web managed by the admin, and 30 personal ones settable on the phone *00 through *29 ... (I know this sometimes might clash with a phones own 'star' codes though) But maybe this is just me ... When I started playing with asterisk I bought a small number of different phones to get a feel for them and was frustrated by a lack of common functions across them, so put all features back into the PBX - things like diverts, follow-me, voicemail and so on are all handled by my asterisk system rather than relying on a particular SIP phone to handle it... However if you want to know what phones I use, it's mostly Grandstream for now. I provision them using gsutil, and when customers want something a bit more posh, it's Snoms. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Decent Voip Phones for enterprise
hi O use around 500 atcom530, they are work perfect www.atcom.com.cn Gordon Henderson wrote: On Wed, 29 Oct 2008, Kev Szaszvari wrote: Hi there Our company is using the Linksys SPA-942 Phones, and they are pretty useless. They dont have any central management or provisioning, as well as a pretty bad interface. Can anyone reccomend any voip phones( Cisco, Polycom, SNOM ) that have * Central Management for all the phones (We dont mind if we have to buy the software to manage them) I always wondered about this - my target is the SME - say 4-150 seats - people don't move desks, change office that often, staff churn is typically low, so I program the phones once then leave them there. If you move desk you take your phone with you. If you leave then the phone can be renamed via it's web interface relatively easily. Maybe I'm just dealing with simple (dumb?) offices, but I'm curious as to what people do with the phones that require this sort of central management. (And regular phone updating) * Programable shortcut buttons, So i can program in on certian phones quick dials to queues. How about implementing this in the PBX. * Optional but bonus, The ability to have a shared address book accross the phones. Same here. So some phones do have nice programmable buttons - and that's good, but in my PBX I have the space for about 600 speed-dials (3-digit extensions) which are web managed by the admin, and 30 personal ones settable on the phone *00 through *29 ... (I know this sometimes might clash with a phones own 'star' codes though) But maybe this is just me ... When I started playing with asterisk I bought a small number of different phones to get a feel for them and was frustrated by a lack of common functions across them, so put all features back into the PBX - things like diverts, follow-me, voicemail and so on are all handled by my asterisk system rather than relying on a particular SIP phone to handle it... However if you want to know what phones I use, it's mostly Grandstream for now. I provision them using gsutil, and when customers want something a bit more posh, it's Snoms. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial() - any way to limit waiting for a RINGING state?
Hello! Just trying to find out how to limit waiting for a RINGING state for an initiated call by Dial() - This is necessary since I want to inform the CALLER that destination is not available if RINGING state was not received within, say 20 seconds. This applies for mostly SIP and IAX2 calls - Is that possible without hacking the app_dial? Regards, Anton. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone using an Intel Atom ?
On Wed, Oct 29, 2008 at 5:46 AM, Peter Evans [EMAIL PROTECTED] wrote: On Wed, Oct 29, 2008 at 08:45:39AM +, Gordon Henderson wrote: I wrote: MSI Wind Board. No idea about outside Japan, but its fanless, almost certainly needs convection. That's because it's called a Wind Board... Great name!!! I didn't see any today, but I was looking for something else. I think that particular model uses the atom 230 as opposed to the new sexy dual core 330 or whatever. I wonder why they use the antique 945 chipset which actually consumes more power than the cpu and not something shinier like g31. That's about the only thing that stops me from getting one to test with. (I do digital sign stuff.) http://global.msi.com.tw/html/popup/bb/windpc_en/ultra_quiet.html (230 based) http://global.msi.com.tw/index.php?func=proddescmaincat_no=1cat2_no=170prod_no=1495 (wind bored, note, the big heatsink covers cpu and north bridge) Given the spec, it should be trivial to get your favourite flavour of linux running on this, and with a cf/ide adaptor, a 2 or 4 gb flash and youre well on your way to no-moving-parts bliss. P The power supply probably consumes about as much as the processor! My question and getting more off topic, but what would one need as far as battery and solar panels to keep one of these running sans moving parts? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Decent Voip Phones for enterprise
Gordon, My guess is that you're a contractor so I can understand why you'd want to keep yourself in high demand by steering clear of the methods that simplify deployment and redeployment. As an employee on the other hand, I want to make things as easy and integrated as I can in order to simplify my own work and keep my employer happy. This mandates the central management features and integration with an existing active directory. Dave --snip-- I always wondered about this - my target is the SME - say 4-150 seats - people don't move desks, change office that often, staff churn is typically low, so I program the phones once then leave them there. If you move desk you take your phone with you. If you leave then the phone can be renamed via it's web interface relatively easily. --snip-- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP ACCOUNT CODE not included in CDR when SIP Status is Unknown
Please help with this strange issue. When sip show peers returns status Unknown the CDR does not include the accountcode even though the call is correctly processed. I'm using A2 Billing and it uses the accountcode to determine the authentication. Asterisk version 1.4.21.2 I'm calling from a Quintum device. I'm very puzzeled. Name/username HostDyn Nat ACL Port Status 1532497439/1532497439 (Unspecified)D 0UNKNOWN The SIP settings are: [1532497439] type=friend host=dynamic username=1532497439 secret=wspiov8729 accountcode=1532497439 callerid=90002 regexten=90002 amaflags=billing context=OutboundWS disallow=all allow=g729 trunk=yes qualify=6000 qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 directrtpsetup=no Thanks Shaun___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial() - any way to limit waiting for a RINGING state?
Sure it is: exten = blah,1,Dial(SIP/blah,30) Where 30 is the time in seconds the application will wait before quitting and setting the DIALSTATUS variable to NOANSWER. Atenciosamente, Vinícius Fontes Núcleo de Tecnologias Convergentes Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Convergent Technologies Core Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Anton [EMAIL PROTECTED] escreveu: Hello! Just trying to find out how to limit waiting for a RINGING state for an initiated call by Dial() - This is necessary since I want to inform the CALLER that destination is not available if RINGING state was not received within, say 20 seconds. This applies for mostly SIP and IAX2 calls - Is that possible without hacking the app_dial? Regards, Anton. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Complete OS/Asterisk disk
What options are available for installing an asterisk system onto a bare-metal system ? Ones that I have seen: pbx-in-a-flash trixbox astlinux What I am trying to achieve is to be able to shove a cd / usb into a machine and have it install asterisk, complete with my .conf files. I also need Cepstral installing. Ideally, I would like to be able to mount an .iso file, chroot into it, and update / compile / build whatever I need before burning the iso file. This would allow me (for example) to update the asterisk 1.4 source as and when i desire. Does anyone know of anything that comes close to this ? I have tried astlinux, but cannot seem to get curl and jabber working properly, and haven't even tried to get Cepstral working :) Alternatively, is there any software that can turn an existing system into a package / .iso that can then be installed on another machine ? Thanks Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] codec not in channel variables
Hi,
I'm trying to access audionativeformat / other codec variables in the hangup
handler of a call (with ${CHANNEL(audioreadformat)}), but I get no response.
Also 'core show channel ...' doesn't list those variables. Are they always set
by asterisk, or only in some scenarios? It's a simple SIP-SIP call with audio
passing through asterisk, same codecs on both sides.
I see that with ast-1.4.11.
Thanks for ideas,
Stan
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone using an Intel Atom ?
On Wed, 29 Oct 2008, Steve Totaro wrote: The power supply probably consumes about as much as the processor! My question and getting more off topic, but what would one need as far as battery and solar panels to keep one of these running sans moving parts? When I put my Atom board into my office firewall/router/server box, it ended up sucking about 55W, so a solar panel with at least double, probably 4x that capacity, feeding lead acids, depending on where you live. My usual boxes consume much less: http://unicorn.drogon.net/power.jpg So 15W running and that's a 1GHz VIA processor, external brick PSU. Where I live - South West England (aka South Wet England) for PV panels, I'd be looking for huge, although it is encouraging to see lots of little LED based road signs now having a PV panel on top of them. One of the suppliers here has: http://www.navitron.org.uk/product.php?proID=65 they have a small charger/controller to go with these too. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] network design philosophy and practice
I'm wondering how prevalent the practice of physically segregating voice and data networks is in the Real World. What are the factors that typically lead to such a decision? DIscussions of pros and cons are most welcome by me. Experiences, anybody? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Complete OS/Asterisk disk
On Wed, Oct 29, 2008 at 01:50:05PM +, Julian Lyndon-Smith wrote: What options are available for installing an asterisk system onto a bare-metal system ? Ones that I have seen: pbx-in-a-flash Builds from osurce but hides its build scripts. Good luck with fixing bugs there. trixbox A binary distriubtion. astlinux Builds everything from scratch. If you want to control everything (e.g. the kernel and libc), it might be the ideal solution. What I am trying to achieve is to be able to shove a cd / usb into a machine and have it install asterisk, complete with my .conf files. I also need Cepstral installing. The proecss is trivial to automate on just about anywhere. For instance, you can find my version of the bristuff build tarball at http://updates.xorcom.com/astribank/bristuff/1.4/ http://updates.xorcom.com/astribank/bristuff/1.4/bristuff-0.4.0-RC4-xr5.tar.gz Check INSTALL.html / INSTALL for instructions. To fit your model, run: ./download.sh copy the resulting directory to a CD / USB key, install the system, and on the target system run from that same tarball: ./prereq.sh ./compile.sh # also installs everything With a bit of work it would work well on your favourite distribution (and make it run automatically at the end of the installer) Ideally, I would like to be able to mount an .iso file, chroot into it, and update / compile / build whatever I need before burning the iso file. This would allow me (for example) to update the asterisk 1.4 source as and when i desire. Does anyone know of anything that comes close to this ? I have tried astlinux, but cannot seem to get curl and jabber working properly, and haven't even tried to get Cepstral working :) Alternatively, is there any software that can turn an existing system into a package / .iso that can then be installed on another machine ? If you package it into packages: yes. If not: yes, but you don't really want it. You should be able to adapt your installation to different environments, and not only to the specific partition size and colletion of devices that happened to be used where you installed. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On October 29, 2008 10:19:36 am Bill Michaelson wrote: I'm wondering how prevalent the practice of physically segregating voice and data networks is in the Real World. What are the factors that typically lead to such a decision? DIscussions of pros and cons are most welcome by me. Experiences, anybody? I'm a pragmatist; most offices have one network jack at each station; I run voice and data on the same physical wire, but if at all possible I try to split things off using smarter switches and VLANs. -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Complete OS/Asterisk disk
On Wed, Oct 29, 2008 at 10:25 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Oct 29, 2008 at 01:50:05PM +, Julian Lyndon-Smith wrote: What options are available for installing an asterisk system onto a bare-metal system ? Ones that I have seen: pbx-in-a-flash Builds from osurce but hides its build scripts. Good luck with fixing bugs there. trixbox A binary distriubtion. astlinux Builds everything from scratch. If you want to control everything (e.g. the kernel and libc), it might be the ideal solution. What I am trying to achieve is to be able to shove a cd / usb into a machine and have it install asterisk, complete with my .conf files. I also need Cepstral installing. The proecss is trivial to automate on just about anywhere. For instance, you can find my version of the bristuff build tarball at http://updates.xorcom.com/astribank/bristuff/1.4/ http://updates.xorcom.com/astribank/bristuff/1.4/bristuff-0.4.0-RC4-xr5.tar.gz Check INSTALL.html / INSTALL for instructions. To fit your model, run: ./download.sh copy the resulting directory to a CD / USB key, install the system, and on the target system run from that same tarball: ./prereq.sh ./compile.sh # also installs everything With a bit of work it would work well on your favourite distribution (and make it run automatically at the end of the installer) Ideally, I would like to be able to mount an .iso file, chroot into it, and update / compile / build whatever I need before burning the iso file. This would allow me (for example) to update the asterisk 1.4 source as and when i desire. Does anyone know of anything that comes close to this ? I have tried astlinux, but cannot seem to get curl and jabber working properly, and haven't even tried to get Cepstral working :) Alternatively, is there any software that can turn an existing system into a package / .iso that can then be installed on another machine ? If you package it into packages: yes. If not: yes, but you don't really want it. You should be able to adapt your installation to different environments, and not only to the specific partition size and colletion of devices that happened to be used where you installed. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Easy Vox Box is the best CentOS 5.2/FreePBX ISO in my opinion. It installs webin, Samba, MyPHP, and other (possibly depending on your preferences) very useful packages. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On Oct 29, 2008, at 9:19 AM, Bill Michaelson wrote: I'm wondering how prevalent the practice of physically segregating voice and data networks is in the Real World. What are the factors that typically lead to such a decision? DIscussions of pros and cons are most welcome by me. Experiences, anybody? In almost all cases it is much better to have two seperate networks. This may be impractical in some smaller installs, but in any office setting we always do this. The only reason I can think of not to is to eliminate the cost of the second cable. In the overall scheme though this is really a minimal cost compared to dealing with issues that may arise over having a fully integrated network. We also only install managed switches and do have seperate vlans. The vlans may be either port based or tagged. In the last five years of doing VOIP installs, we have only had one customer the refused to add the second cable, and they were also the most unhappy. They also demanded the lowest cost phone option (IP301) and a Snom for an operator console. It all worked, just not very well, and ultimately they relaced it all. I n the real world, there usually are very inexperienced people using and managing the network. What is trivial in the data side becomes critical on the voice side and since most networks are run by the data guys, having it as seperate as possible really helps keep it all working well. One of the not so obvious issues is when the data guys are having a problem and go around rebooting things, dropping phone calls. On this list we tend to only think about the voice side, just keep in mind any data operations which are also going on. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
Bill Michaelson wrote: I'm wondering how prevalent the practice of physically segregating voice and data networks is in the Real World. What are the factors that typically lead to such a decision? DIscussions of pros and cons are most welcome by me. Experiences, anybody? We chose to go with a segregated network and certainly don't regret the choice. Voice and data are on separate ports at the desk, avoiding QoS issues completely and reducing confision amongst users who still expect separate Phone and Computer plugs on the wall. The traffic does run through the same switches and inter-switch trunks but always on distinct VLANs. My experience with connecting the desktop computer through the phone has been very poor. Audio breaks up when the computer does large data transfers. Yes, Sir. I'll just look that up in our datab...baba.bas.ss..ss..se In addition our users require gigabit to the desktop. The phones are 100Mb. Worst part is the few Cisco phones we have insist on searching for VLAN (which doesn't exist) for 5 minutes on startup. Hopefully they will be replaced through attrition but despite being over-priced, over-featured and proprietary, Cisco do build robust kit. Sigh. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On Wed, Oct 29, 2008 at 10:32 AM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On October 29, 2008 10:19:36 am Bill Michaelson wrote: I'm wondering how prevalent the practice of physically segregating voice and data networks is in the Real World. What are the factors that typically lead to such a decision? DIscussions of pros and cons are most welcome by me. Experiences, anybody? I'm a pragmatist; most offices have one network jack at each station; I run voice and data on the same physical wire, but if at all possible I try to split things off using smarter switches and VLANs. -A. For me, it has come down to the customer's decision and budget. I usually recommend two drops per station (or more). I had a large pharmaceutical company have me (my cable crew) run four drops to each work station, rip out all their old 3com switches and replace them with Cisco switches and new routers, replace all the workstations with new Dell's with three or five year gold same day onsite support, tear out their Definity G3 and replace it with a 3Com NBX system, and replace all of their servers with new IBM servers and migrate everything over. Anyways, two of the four drops were on the phone side of the network and the other two were on the regular data side. It was a long process, a huge challenge, a total success and they were completely happy at the end, but as everyone knows pharm companies have VERY deep pockets. My budget was a million or so and I came in well under. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On Wed, Oct 29, 2008 at 10:48 AM, Drew Gibson [EMAIL PROTECTED] wrote: Bill Michaelson wrote: I'm wondering how prevalent the practice of physically segregating voice and data networks is in the Real World. What are the factors that typically lead to such a decision? DIscussions of pros and cons are most welcome by me. Experiences, anybody? We chose to go with a segregated network and certainly don't regret the choice. Voice and data are on separate ports at the desk, avoiding QoS issues completely and reducing confision amongst users who still expect separate Phone and Computer plugs on the wall. The traffic does run through the same switches and inter-switch trunks but always on distinct VLANs. My experience with connecting the desktop computer through the phone has been very poor. Audio breaks up when the computer does large data transfers. Yes, Sir. I'll just look that up in our datab...baba.bas.ss..ss..se In addition our users require gigabit to the desktop. The phones are 100Mb. Worst part is the few Cisco phones we have insist on searching for VLAN (which doesn't exist) for 5 minutes on startup. Hopefully they will be replaced through attrition but despite being over-priced, over-featured and proprietary, Cisco do build robust kit. Sigh. regards, Drew Drew, Disable CDP on the phone and that will go away. I know you said you're not using VLANs but... You can use CDP and set your voice-vlan on Cisco switches. Or... you can install cdp-tools on a Linux box and have it advertise a voice vlan for you! http://gpl.internetconnection.net/ I added the voice vlan support to cdp-tools. ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On Wed, 29 Oct 2008, Bill Michaelson wrote: I'm wondering how prevalent the practice of physically segregating voice and data networks is in the Real World. What are the factors that typically lead to such a decision? DIscussions of pros and cons are most welcome by me. Customer budget and choice... I've installed in-line phones where there has been one drop point per desk, and installed a separate LAN for phones for a customer who wanted it. If I were dealing with anything less than a small office, or they needed Gb to the desktop, I'd get them to run a 2nd line for phones and put them on separate switches. My biggest in-line client has 25 desks and were on a tight budget when they moved offices, so they have phones in-line with their PCs (diskless Linux workstations!) and we did some tests at install time and couldn't see any issues at the time (or hear any issues!) That was just over a year ago, and I was in-touch recently for an anual review and everything was going just fine for them. Not had a pressing need to ever use VLANS (but I typically don't deal with clients who want/need that) but putting all the VoIP devices on one bank of switches and data on another works very well. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
In my experience most of the serious QoS issues arise in relation to the Internet pipe (if the provider is IP, and outside the network), not the LAN. Of course, LANs can be heavily contended, but are not in most organisations, especially as gigabit cores are getting increasingly common even in smaller mid-size and small organisations. I would pay most attention to the router(s), unless your PSTN connectivity is TDM and on-premise. Drew Gibson wrote: Bill Michaelson wrote: I'm wondering how prevalent the practice of physically segregating voice and data networks is in the Real World. What are the factors that typically lead to such a decision? DIscussions of pros and cons are most welcome by me. Experiences, anybody? We chose to go with a segregated network and certainly don't regret the choice. Voice and data are on separate ports at the desk, avoiding QoS issues completely and reducing confision amongst users who still expect separate Phone and Computer plugs on the wall. The traffic does run through the same switches and inter-switch trunks but always on distinct VLANs. My experience with connecting the desktop computer through the phone has been very poor. Audio breaks up when the computer does large data transfers. Yes, Sir. I'll just look that up in our datab...baba.bas.ss..ss..se In addition our users require gigabit to the desktop. The phones are 100Mb. Worst part is the few Cisco phones we have insist on searching for VLAN (which doesn't exist) for 5 minutes on startup. Hopefully they will be replaced through attrition but despite being over-priced, over-featured and proprietary, Cisco do build robust kit. Sigh. regards, Drew -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What syntax to send user:pass in SIP Dial string?
Hi All, I'm trying to get the user:pass embedded in a SIP Dial string instead of calling a SIPuser in sip.conf: Regular way, exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]|30|) Where the 'sipuser' is a context on sip.conf [sipuser] fromuser=sipuser What I would like to do is embed the username:password in the Dial string, something like this: exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]:[EMAIL PROTECTED]|30|) doesn't work though, can't create sip channel. I'm not sure if this can be done? Any guidance will be appreciated. JR -- - JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. --snip-- In almost all cases it is much better to have two seperate networks. This may be impractical in some smaller installs, but in any office setting we always do this. The only reason I can think of not to is to eliminate the cost of the second cable. --snip-- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
I'm pretty sure they meant two logical networks. At least, I hope they did. David Gibbons wrote: Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. --snip-- In almost all cases it is much better to have two seperate networks. This may be impractical in some smaller installs, but in any office setting we always do this. The only reason I can think of not to is to eliminate the cost of the second cable. --snip-- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec not in channel variables
Did you try show translation
On Wed, Oct 29, 2008 at 3:55 PM, Stanisław Pitucha [EMAIL PROTECTED]wrote:
Hi,
I'm trying to access audionativeformat / other codec variables in the
hangup handler of a call (with ${CHANNEL(audioreadformat)}), but I get no
response. Also 'core show channel ...' doesn't list those variables. Are
they always set by asterisk, or only in some scenarios? It's a simple
SIP-SIP call with audio passing through asterisk, same codecs on both sides.
I see that with ast-1.4.11.
Thanks for ideas,
Stan
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet
Maybe you have a Codec issue? On Tue, Oct 28, 2008 at 11:20 PM, Benny Amorsen [EMAIL PROTECTED][EMAIL PROTECTED] wrote: Lincoln King-Cliby [EMAIL PROTECTED] writes: Periodically I'm seeing calls placed from the 7961s through anything on the PBX that requires digit entry (the Auto Attendant, Voicemail, etc.) 'randomly' drop; extension-to-extension calls extension-to-PSTN, and PSTN-to-extension calls never have any issues whatsoever. Nor have I been able to duplicate the issues hopping around auto attendants on an inbound PSTN call. I am not sure this is relevant in the 1.4.x versions, but here goes anyway: In Asterisk 1.2.x it could sometimes happen that Asterisk believed the path to a server was so good, that it would only allow 1 ms for answers to be received. It would do all its retransmissions in less than 200ms, and then it would complain about no reply to critical packet. Anyway, you can adjust the minimum timer with the configuration option t1min in sip.conf. I would recommend setting it to at least 100 (it is in ms) and perhaps 500 would help for you. It is also highly possible that your issue is completely different. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to bind a SIP channel to an IP
On Mon, Oct 27, 2008 at 7:12 PM, srinivas Antarvedi [EMAIL PROTECTED] wrote: Hello members, Mysetup: Asterisk 1.4 Phones:Polycom501 I wanted to register my polycom phones only from a fixed IP(on LAN ) i tried following scenarios and my results are described as follows 1)sip.conf [xxx] host=192.168.0.15 result is after some time the registration expires and i was unable to receive calls on my channel... 2)sip.conf [xxx] defaultip=192.168.0.15 i) result is after some time the registration expires and i was unable to receive calls on my channel ii)it is even allowing me to register from another ip address say 192.168.0.16 3)sip.conf [xxx] host=dynamic defaultip=192.168.0.15 in this case i dont have any problems and it was working fine... can anybody helpme out to bind the phones to a particular ip if not is it possible to do at all just give me a hint so that i will work on Look out some examples here http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf ram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] openser+asterisk
On Mon, Oct 27, 2008 at 11:53 AM, jordan pan [EMAIL PROTECTED] wrote: Hi everyone, I want to use the openser and asterisk to create a system ,who can give me a detail example about it,i found it have some complicated. Thanks in advance. http://www.mail-archive.com/asterisk-users@lists.digium.com/msg60425.html Look the above link for your requirement Ram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Headset Recommendation
Does anyone have a recommendation for a headset that plugs into the Mic/Line-out port on a PC? Ideally something like the Plantronics SupraPlus. I'd prefer Monaural instead of stereo, and cheap in price but not in quality. Thanks for any suggestions... Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: [EMAIL PROTECTED] This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet
Benny and Mark, Thank you for your replies. I tried adding t1min=500 to sip.conf per the suggestion below and since doing that haven't been able to reproduce the issue. If it comes back, I'll do the SIP debug per Mark's suggestion and post the results here. (Mark, per your question the Auto Attendant and Voicemail are on the same box) Thanks again for the quick help! Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC http://www.thecontrolworks.com/ Crestron Authorized Independent Programmer -Original Message- From: Benny Amorsen [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 28, 2008 5:20 PM To: Lincoln King-Cliby Cc: 'asterisk-users@lists.digium.com' Subject: Re: Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet snip In Asterisk 1.2.x it could sometimes happen that Asterisk believed the path to a server was so good, that it would only allow 1 ms for answers to be received. It would do all its retransmissions in less than 200ms, and then it would complain about no reply to critical packet. Anyway, you can adjust the minimum timer with the configuration option t1min in sip.conf. I would recommend setting it to at least 100 (it is in ms) and perhaps 500 would help for you. It is also highly possible that your issue is completely different. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sendmail for Voicemail
I use PostFix and MailHop Outbound from Dyndns.com. They will accept your outgoing email on multiple ports to help with the blocking problem. It's $15/year for a limited number of messages. Todd On Oct 28, 2008, at 7:39 PM, [EMAIL PROTECTED] wrote: When I send email from my local asterisk machine, my IP address get's RBL'd. Asterisk is my only reason for running sendmail, so to keep it simple, I tried to make my ISP's mail server a 'smart host' (relaying to a trusted mail server) but my ISP doesn't allow ANY kind of relaying these days. I imagine there are many like me who are not sendmail experts who want to send Asterisk Voicemal. Can someone direct me to the quick, dirty and secure way to send mail from my asterisk box? The good news is that I'm on a Fixed IP on a registered network with working reverse in-addr.arpa lookups, and as you might have guessed, all mail would originate from the local host. Suggestions? Thanks! -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Intergrating vicidial with trixbox
Hello, I am searched the net for tutorials on how I can Integrate vicidial with trixbox. I can't find any. Anyone who knows where I can get one? James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED] title:Lead Consultant tel;work:+254-722-490994 tel;home:+254-722-490994 tel;cell:+254-722-490994 url:www.agile.co.ke version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec not in channel variables
- michel freiha [EMAIL PROTECTED] wrote: Did you try show translation That shows a table of times taken by translation... I'm asking about codecs used by a channel on a certain call. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intergrating vicidial with trixbox
I would contact the vendor. James Mutuku wrote: Hello, I am searched the net for tutorials on how I can Integrate vicidial with trixbox. I can't find any. Anyone who knows where I can get one? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On Oct 29, 2008, at 8:21 AM, Alex Balashov wrote: In my experience most of the serious QoS issues arise in relation to the Internet pipe (if the provider is IP, and outside the network), not the LAN. Of course, LANs can be heavily contended, but are not in most organisations, especially as gigabit cores are getting increasingly common even in smaller mid-size and small organisations. I would pay most attention to the router(s), unless your PSTN connectivity is TDM and on-premise. I would agree with this as long as you have a decent LAN. We have about 60 computer workstations and 85 phones on our network. The entire thing is Gigabit. Each phone (with a few exceptions that we are running new cable to rectify) has a dedicated ethernet port, no sharing. We are NOT however separating the data/voice networks. They are on one VLAN. We may segment later, but only if the need arises. Right now we have no problems. I should point out that all of our switches have 2+ gigabit links back to the master switch. We've never had a problem with the phones other than related to the outside world (telco side). I won't argue that best practice would probably be to VLAN off the phones, but if you don't have a massive network and are fully gigabit smart switches etc with good cabling, then keeping the two networks merged should not be a problem. I do wholly recommend multiple drops per workstation though. In a day when I can buy CAT 6 cable for 10 cents a foot, there is really just no reason not to be doing multiple drops in new installs. Daniel -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On Wednesday 29 October 2008 10:22:43 David Gibbons wrote: A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. --snip-- In almost all cases it is much better to have two seperate networks. This may be impractical in some smaller installs, but in any office setting we always do this. The only reason I can think of not to is to eliminate the cost of the second cable. --snip-- The concern is almost never one of taking bandwidth away from the desktop, but one of the desktop taking bandwidth (especially by introducing latency) away from the phone. Though, as you pointed out, a good QOS and VLAN policy will make that usually unnecessary. Folks do have to contend with customers who won't spring for anything but el cheapo network switches, and that's where a completely separate physical network makes sense. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intergrating vicidial with trixbox
I noticed that the vicidial site has documentation available which probably covers the topics required. However, I also see that they want $50-$100 to download the docs. Seems harsh. Ron Byer Jr. NetWeave Integrated Solutions, Inc. +1.732.786.8830 x120 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Wednesday, October 29, 2008 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Intergrating vicidial with trixbox I would contact the vendor. James Mutuku wrote: Hello, I am searched the net for tutorials on how I can Integrate vicidial with trixbox. I can't find any. Anyone who knows where I can get one? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG. Version: 7.5.549 / Virus Database: 270.8.4/1753 - Release Date: 10/28/2008 9:20 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sendmail for Voicemail
On Tue, 28 Oct 2008, [EMAIL PROTECTED] wrote: When I send email from my local asterisk machine, my IP address get's RBL'd. Asterisk is my only reason for running sendmail, so to keep it simple, I tried to make my ISP's mail server a 'smart host' (relaying to a trusted mail server) but my ISP doesn't allow ANY kind of relaying these days. So how do you normally send email? How did you send this one? I imagine there are many like me who are not sendmail experts who want to send Asterisk Voicemal. One solution would be to switch from sendmail to something you know - but I'm guessing you're using some canned asterisk solution which comes with sendmail? Can someone direct me to the quick, dirty and secure way to send mail from my asterisk box? The good news is that I'm on a Fixed IP on a registered network with working reverse in-addr.arpa lookups, and as you might have guessed, all mail would originate from the local host. So the fixed IP with reverse DNS isn't helping you get by the RBLs - a lot of which know the ISP end-user ranges... Your ISP must allow some sort of email relaying to let you send email from your desktop - unless they're forcing you to use a webmail solution? Do they actively block outbound port 25? If they genuinely don't provide email relaying, then you might have to enlist the services of an indepedant ISP and relay via their servers - this will almsot certinly involve some sort of authentication - usually SMTP-AUTH, which is very possible in sendmail, but might not work in a pre-canned version. I've used and worked with sendmail for more years than I care to remember, but it seems here that the issue isn't neccessarily with sendmail, but with your ISP.. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
Daniel Hazelbaker wrote: I would agree with this as long as you have a decent LAN. We have about 60 computer workstations and 85 phones on our network. The entire thing is Gigabit. Each phone (with a few exceptions that we are running new cable to rectify) has a dedicated ethernet port, no sharing. We are NOT however separating the data/voice networks. They are on one VLAN. We may segment later, but only if the need arises. Right now we have no problems. I should point out that all of our switches have 2+ gigabit links back to the master switch. We've never had a problem with the phones other than related to the outside world (telco side). I won't argue that best practice would probably be to VLAN off the phones, but if you don't have a massive network and are fully gigabit smart switches etc with good cabling, then keeping the two networks merged should not be a problem. I do wholly recommend multiple drops per workstation though. In a day when I can buy CAT 6 cable for 10 cents a foot, there is really just no reason not to be doing multiple drops in new installs. Yep. It really depends on how much activity there is on the LAN. Sure, if you've got a large network with 250 megabits of bursty traffic swinging through there consistently, or doing large amounts of multicast, then partitioning the voice off on a dedicated VLAN makes a lot of sense. One problem for which I've never found a satisfactory solution is busy call centers where the agents are all on softphones (to save money, of course). The IT staff are never willing to accommodate the relatively large amount of reconfiguration required to do VLAN trunking into every agent's workstation in order to provide a feed into the voice VLAN for the softphone. I'm not a Windows guy, so I don't even know how easily or readily Windows supports 802.1q trunking, although surely it does in principle. Either way, it's something that seemingly nobody ever wants to do. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On Wed, 29 Oct 2008 11:50:31 -0500, Tilghman Lesher wrote On Wednesday 29 October 2008 10:22:43 David Gibbons wrote: A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. --snip-- In almost all cases it is much better to have two seperate networks. This may be impractical in some smaller installs, but in any office setting we always do this. The only reason I can think of not to is to eliminate the cost of the second cable. --snip-- The concern is almost never one of taking bandwidth away from the desktop, but one of the desktop taking bandwidth (especially by introducing latency) away from the phone. Though, as you pointed out, a good QOS and VLAN policy will make that usually unnecessary. Folks do have to contend with customers who won't spring for anything but el cheapo network switches, and that's where a completely separate physical network makes sense. I was under the (very possibly mistaken) impression that by running the desktop through the phone the phone will keep the PC from doing such terrible things. At least that seems to be the case in the installations I have done with Polycom phones to date. My largest is a 150 phone installation at a major resort. No VLANs, no QOS, and pretty much everything runs back to mid-range Linksys managed switches. I only have quality issues when they try to use the Internet connection for long distance, which in the Virgin Islands is spotty at best ;) I think much too big a deal is being made here and over-engineering is at work. But then my installations are not into heavy LAN use. I suppose as always it depends on the situation. -- Jeff LaCoursiere JB Telenet, LLC 6501 Redhook Plaza, box 395 St Thomas, USVI 00802 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] Flash player for call recordings - 8khz
Hello, I'm trying to find simple MP3 player in flash, to integrate it with call recordings. My requirements would be: * simple UI * buffering (would be nice) * slider * volume control * support of 8kHz stereo mp3 * javascript access to seek/position * free for any use (GPL, MPL, MIT, BSD) So far I've found that JWplayer[1] does great with my recordings. However it's not small in size, as there's video player and playlists - none of which i need. Also it should be paid, even if i use it internally in company, which i don't like. Also I found niftyPlayer[2], which would be perfect, however it creates chipmunk sound out of 8kHz recordings, so i wonder what's the difference. Could anybody share their experience with call recordings? What i have found to be useful: * Record everything in G.711 (as it's native codec, thus less transcoding and more quality) * Do a nightly (or per request) conversion to stereo MP3's preserving 8kHz. sox -M is great for this. Resulting files have smaller size than in GSM or WAV format, so you can keep more recordings. References: [1] http://www.jeroenwijering.com/?item=JW_FLV_Player [2] http://www.varal.org/media/niftyplayer/ Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
David Gibbons wrote: Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. EVER? What about Gigabit networks with 10/100 phones? While some Gigabit phones are available, gigabit POE switches are not cheap, while non-POE gigabit switches are pretty cheap and most business class desktops these days come with gigabit network connections. In a new wiring install I almost always insist on two jacks per location rather than relying on pass-thru connectors on phones. Try giving a few users gigabit access to an Exchange server, then taking it away. They will certainly not be happy! Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
David Gibbons wrote: Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. --snip-- In almost all cases it is much better to have two seperate networks. This may be impractical in some smaller installs, but in any office setting we always do this. The only reason I can think of not to is to eliminate the cost of the second cable. --snip-- That's two _logically_ separate networks. The key point is that the last yard cable to the phone is not shared with the computer. The issue is not a lack of bandwidth but that the phone has to try and get its little packets inserted between the massive packets of a database lookup or file transfer in a timely manner (latency and jitter). You might get away with a single logical network on a smaller site or a larger one with very light traffic. QoS is not required on lightly loaded links and will do nothing for you on over loaded ones. I only use it on WAN links where bandwidth is more expensive. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Blank Voicemail.Conf after Password Change
Hi, For a few weeks now, our asterisk server has been experiencing something very odd. From time to time, voicemail.conf would go blank. We finally tracked it down to happening when someone attempts to change their password. It seems the file is touched, but not written to, and we're left with a blank voicemail file. Permissions seem to be fine: -rw-rw-r-- 1 asterisk asterisk 12707 2008-10-29 12:14 /etc/asterisk/voicemail.conf Asterisk is running as asterisk: 24560 ?Ssl 409:34 /usr/sbin/asterisk -U asterisk We're at a loss of what is going wrong, and how to resolve it. Meanwhile, *I don't think it's possible to not allow password changes (correct me if I'm wrong). *Using externpass to run a script that will copy a backup voicemail file to voicemail.conf could potentially work, but it won't save password changes. Nothing generated from voicemail is showing up in the asterisk logs, nor does the console show any error after changing a password. Any assistance on this strange behavior is much appreciated! Thank you, Leah Newmark VoIP Programmer Capalon Communications ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best Sales 2008!
About this mailing: You are receiving this e-mail because you subscribed to MSN Featured Offers. Microsoft respects your privacy. If you do not wish to receive this MSN Featured Offers e-mail, please click the "Unsubscribe" link below. This will not unsubscribe you from e-mail communications from third-party advertisers that may appear in MSN Feature Offers. This shall not constitute an offer by MSN. MSN shall not be responsible or liable for the advertisers' content nor any of the goods or service advertised. Prices and item availability subject to change without notice. ©2008 Microsoft | Unsubscribe | More Newsletters | Privacy Microsoft Corporation, One Microsoft Way, Redmond, WA 98052 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
Kristian Kielhofner wrote: On Wed, Oct 29, 2008 at 10:48 AM, Drew Gibson [EMAIL PROTECTED] wrote: Worst part is the few Cisco phones we have insist on searching for VLAN (which doesn't exist) for 5 minutes on startup. Hopefully they Drew, Disable CDP on the phone and that will go away. I know you said you're not using VLANs but... You can use CDP and set your voice-vlan on Cisco switches. Or... you can install cdp-tools on a Linux box and have it advertise a voice vlan for you! http://gpl.internetconnection.net/ I added the voice vlan support to cdp-tools. ;) I tried out the cdp-tools some time ago (it may have been on your recommendation, Kristian) but with no success. Is it possible to disable CDP on the 7940 (image_version : P0S3-08-2-00)? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On Oct 29, 2008, at 10:10 AM, Darrick Hartman wrote: David Gibbons wrote: Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. EVER? What about Gigabit networks with 10/100 phones? While some Gigabit phones are available, gigabit POE switches are not cheap, while non-POE gigabit switches are pretty cheap and most business class desktops these days come with gigabit network connections. In a new wiring install I almost always insist on two jacks per location rather than relying on pass-thru connectors on phones. Try giving a few users gigabit access to an Exchange server, then taking it away. They will certainly not be happy! I always considered myself to be rather tight on budget, but maybe I have more money available than most. We use the SGE2000P LinkSys Gigabit, Managed, PoE switches and they work great. I get them for about $800, which is just under $200 more than the non-PoE version. I don't find that to be an excessive price since most decent managed non- PoE switches are in the $500-$600 range (I'm sorry, I just can't bring myself to buy a D-Link or NetGear Gigabit managed switch for $300 to run my entire network on, maybe they are fine but they always struck me as a small player so to speak). Daniel Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
Fair enough, I guess I was concentrating on this line in Jerry's message :) The only reason I can think of not to is to eliminate the cost of the second cable. I believe you're mistaken about the QOS though. QoS is not required on lightly loaded links and will do nothing for you on over loaded ones. QOS will absolutely allow voice traffic to pass with priority over heavily loaded links -- this is in fact the reason that it would be implemented. Obviously giving priority to the voice traffic on these heavily loaded links serves to mitigate both latency and jitter. The concern is almost never one of taking bandwidth away from the desktop, but one of the desktop taking bandwidth (especially by introducing latency) away from the phone. Agreed -- but with VLAN tagging and QOS, the issue of how much bandwidth the desktop uses and/or needs becomes moot since the phone is given priority. Dave David Gibbons wrote: Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. --snip-- In almost all cases it is much better to have two seperate networks. This may be impractical in some smaller installs, but in any office setting we always do this. The only reason I can think of not to is to eliminate the cost of the second cable. --snip-- That's two _logically_ separate networks. The key point is that the last yard cable to the phone is not shared with the computer. The issue is not a lack of bandwidth but that the phone has to try and get its little packets inserted between the massive packets of a database lookup or file transfer in a timely manner (latency and jitter). You might get away with a single logical network on a smaller site or a larger one with very light traffic. QoS is not required on lightly loaded links and will do nothing for you on over loaded ones. I only use it on WAN links where bandwidth is more expensive. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Current Open Source Billing Package
After spending a couple hours scanning for an open source (non- commercial) billing package yesterday I am underwhelmed. Almost all of the packages listed on the WIKI appear to be defunct, for several years now. I will be happy to get a login and edit them out if that is the proper method to do so. My requirements are very minimal and at this point unless I have missed something will just write my own. I do not do calling cards. I have no near term need for the package to actually talk with asterisk at all, other than to import the CDR either via files or as a login to MySQL. I do have monthly recurring charges which need to be included monthly. I do occasionally have need to one off (manual) billing charges. Rating for calls would be nice but not mandatory ( we have very minimal International). Ability to export to an accounting package a plus. Ability to generate hard copy Invoices and/or email them to the cust. Ability to generate a list of current Invoices. Runs on Linux. All in all not a very complex set of requirements, but the few packages that seem to be currently offered generally do not fit the bill. Yes there are many commercial packages, but unless they are very minimal in cost I have no interest in them. So my question is, have a missed a golden nugget out there? tia Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom - we are puzzled
I would get a PCAP trace from the phone to see what is going on on the cable. CS -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Ronald Wiplinger (Lists) Gesendet: Dienstag, 28. Oktober 2008 23:01 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Snom - we are puzzled we have installed asterisk and snom with PUBLIC IPs (IP/25) on one DSL line we have for our office a different ADSL with one IP shared. Two identical setup snom 360 (except the user name) with two public IP addresses are connected at the hub to the server / DSL line phone A can call B, B cannot call A, because A is not registered!!! We disconnect A and setup a softphone (on the ADSL line with stun) and it works. How can I track down this problem. bye R. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intergrating vicidial with trixbox
Hello, The paid VICIDIAL user manuals do not cover installing on Trixbox. Mostly because it can be very difficult to install VICIDIAL on Trixbox due to the many different versions of Trixbox and the dialplan complexity of Trixbox.(also I want to mention that there are FREE versions of the VICIDIAL manuals, and all admin-based documentation is in the open-source codebase) We do not recommend putting VICIDIAL on the same machine as Trixbox, mostly due to the performance hit of just running trixbox which effectively cuts the functionaly capacity of the machine in half. We recommend using IAX trunks to connect a separate VICIDIAL machine to your Trixbox machine, that way you can still use your trixbox phones and inbound DIDs if needed with VICIDIAL while still allowing VICIDIAL to efficiently dial out through it's own trunks if you like, all without messing with the internals of the Trixbox-generated dialplan and utilities. MATT--- On 10/29/08, Ron Byer Jr. [EMAIL PROTECTED] wrote: I noticed that the vicidial site has documentation available which probably covers the topics required. However, I also see that they want $50-$100 to download the docs. Seems harsh. Ron Byer Jr. NetWeave Integrated Solutions, Inc. +1.732.786.8830 x120 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Wednesday, October 29, 2008 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Intergrating vicidial with trixbox I would contact the vendor. James Mutuku wrote: Hello, I am searched the net for tutorials on how I can Integrate vicidial with trixbox. I can't find any. Anyone who knows where I can get one? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG. Version: 7.5.549 / Virus Database: 270.8.4/1753 - Release Date: 10/28/2008 9:20 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Current Open Source Billing Package
- Jerry Jones [EMAIL PROTECTED] escribió: After spending a couple hours scanning for an open source (non- commercial) billing package yesterday I am underwhelmed. Almost all of the packages listed on the WIKI appear to be defunct, for several years now. I will be happy to get a login and edit them out if that is the proper method to do so. My requirements are very minimal and at this point unless I have missed something will just write my own. I do not do calling cards. I have no near term need for the package to actually talk with asterisk at all, other than to import the CDR either via files or as a login to MySQL. I do have monthly recurring charges which need to be included monthly. I do occasionally have need to one off (manual) billing charges. Rating for calls would be nice but not mandatory ( we have very minimal International). Ability to export to an accounting package a plus. Ability to generate hard copy Invoices and/or email them to the cust. Ability to generate a list of current Invoices. Runs on Linux. All in all not a very complex set of requirements, but the few packages that seem to be currently offered generally do not fit the bill. Yes there are many commercial packages, but unless they are very minimal in cost I have no interest in them. So my question is, have a missed a golden nugget out there? Check a2billing: http://www.asterisk2billing.org/cgi-bin/trac.cgi Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail: [EMAIL PROTECTED] www : http://www.manta.telconet.net SIP : [EMAIL PROTECTED] Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Current Open Source Billing Package
I do understand that this not free, but BillMax (www.billmax.com) supports all of your requirements plus includes the source code. I think you can get a demo that supports under 100 accounts for free... at least you used to be able to. j On Wed, 29 Oct 2008, Jerry Jones wrote: After spending a couple hours scanning for an open source (non- commercial) billing package yesterday I am underwhelmed. Almost all of the packages listed on the WIKI appear to be defunct, for several years now. I will be happy to get a login and edit them out if that is the proper method to do so. My requirements are very minimal and at this point unless I have missed something will just write my own. I do not do calling cards. I have no near term need for the package to actually talk with asterisk at all, other than to import the CDR either via files or as a login to MySQL. I do have monthly recurring charges which need to be included monthly. I do occasionally have need to one off (manual) billing charges. Rating for calls would be nice but not mandatory ( we have very minimal International). Ability to export to an accounting package a plus. Ability to generate hard copy Invoices and/or email them to the cust. Ability to generate a list of current Invoices. Runs on Linux. All in all not a very complex set of requirements, but the few packages that seem to be currently offered generally do not fit the bill. Yes there are many commercial packages, but unless they are very minimal in cost I have no interest in them. So my question is, have a missed a golden nugget out there? tia Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *** Handled by Will's new toy *** ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
Alex Balashov wrote: Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com I'm pretty sure they meant two logical networks. At least, I hope they did. Unfortunately, I was indeed referring to two physical networks. Cabling, switches, everything, all the way back to the TDM connection to the PSTN. David Gibbons wrote: Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. I agree, especially about QoS design intent. But I posted my question as a sanity check, and there seems to be no shortage of opinions. Now mine: I can think of two valid reasons to physically segregate the networks: 1) Insurance. I.e., to eliminate the possibility that otherwise properly configured QoS mechanisms become broken, either by accident, incompetence, or badly-designed or rogue software or hardware - or are otherwise handled carelessly as Jerry Jones suggested. But this is not a compelling argument to me in any but the most critical scenarios such as public-safety applications, etc. 2) Customer preference. If you need the business, then the customer is always right. You might not have adequate credibility with the customer or influence over the design decision, and if a customer in such a situation gets it in their heads that voice and data can't coexist on wires, then it can't. There is a variety of opinions, but no general consensus about where QoS failures typically occur, when they occur. I'm wondering if anyone has anyone has ever experienced QoS issues caused by contemporary Polycom phones like IP330s that had workstations hanging off their builtin switches? If you did, were you able to identify the cause, and was it due to any inherent failure of the phone, such as not marking packets or prioritizing dispatch correctly? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On Oct 29, 2008, at 12:30 PM, David Gibbons wrote: Fair enough, I guess I was concentrating on this line in Jerry's message :) The only reason I can think of not to is to eliminate the cost of the second cable. I believe you're mistaken about the QOS though. QoS is not required on lightly loaded links and will do nothing for you on over loaded ones. QOS will absolutely allow voice traffic to pass with priority over heavily loaded links -- this is in fact the reason that it would be implemented. Obviously giving priority to the voice traffic on these heavily loaded links serves to mitigate both latency and jitter. The concern is almost never one of taking bandwidth away from the desktop, but one of the desktop taking bandwidth (especially by introducing latency) away from the phone. Agreed -- but with VLAN tagging and QOS, the issue of how much bandwidth the desktop uses and/or needs becomes moot since the phone is given priority. Dave David Gibbons wrote: Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. --snip-- In almost all cases it is much better to have two seperate networks. This may be impractical in some smaller installs, but in any office setting we always do this. The only reason I can think of not to is to eliminate the cost of the second cable. --snip-- That's two _logically_ separate networks. The key point is that the last yard cable to the phone is not shared with the computer. The issue is not a lack of bandwidth but that the phone has to try and get its little packets inserted between the massive packets of a database lookup or file transfer in a timely manner (latency and jitter). You might get away with a single logical network on a smaller site or a larger one with very light traffic. QoS is not required on lightly loaded links and will do nothing for you on over loaded ones. I only use it on WAN links where bandwidth is more expensive. Allow me to clarify. Yes I do advocate seperate cable runs for phones and computers. Do not care if they both use a single switch as long as they are VLANd on seperate paths, either port based or tag based. And before everyone starts up again - :) - let me say that YES, I do install single cable fully integrated systems - when I manage the network. If I remember the OP was looking for real world examples and guidance. In the real world, just last week I picked up a new customer, drove 6 hours to a branch office of theirs that kept complaining about voice performance, and threw out the hub I found they had installed when they moved into their brand new building. Had a nice new switch - which I was told about - for their pc's. But all phones were on a hub - which I had not been told about. The new switch had been sent down to plug the phones into, but yeah. So in the real world I really like the KISS principle. Of course if there are qualified data folk ALWAYS makeing sure network is setup properly then feel free to disregard. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is anyone using * for 2 way video conferencing?
Hi, One of my clients, wants to use * box to run weekly meetings between remote locations over the internet. What would be the best configuration for this? We are talking about two conference rooms. I am referring to the actual hardware/software and bandwidth requirements for this to work well. I have run two software video phones and I had marginal results with it when displayed on large LCDs, delay and blockines ware the problems I have run into ... Sincerely, Robert Augustyn http://www.linqone.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: Thecus N7700
Not directly related to Asterisk but I'm sure one or two of you will get hot and bothered over this. :-) http://deancollinsblog.blogspot.com/2008/10/thecus-n7700.html Regards, Dean Collins Cognation Inc [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). From: Dean Collins [mailto:[EMAIL PROTECTED] Sent: Wednesday, 29 October 2008 2:10 PM To: Dean Collins Subject: [Dean Collins] Thecus N7700 Mmmm Yummy, look what the lovely DHL delivery guy just dropped off. http://4.bp.blogspot.com/_jmYevHrBr6M/SQigUUbUyfI/Axg/JnwClKJA4 Xk/s320/2008_10290022.JPG The Thecus N7700 http://www.thecus.com/products_over.php?cid=10pid=82 is a brand new 7 drive Raid NAS server. http://1.bp.blogspot.com/_jmYevHrBr6M/SQigUiqYnDI/Axo/NxLBLSmGc Pg/s1600-h/2008_10290023.JPG Offers iSCSI direct disk support on Raid 0,1,5,6,10 (even multiple raid groups). http://2.bp.blogspot.com/_jmYevHrBr6M/SQigUkp9ubI/Axw/1jGvopsi2 y8/s320/2008_10290024.JPG Allows network seperate configurable LAN and WAN ports. The eSata drive can be used for offloading onto another disk/tape solution. In addition there is a PCI slot that I've been told is there to create 'multiple N7700 clusters' though I haven't seen it done and not sure what the limitations are but I've been told you can stack 7 of them in a cluster http://1.bp.blogspot.com/_jmYevHrBr6M/SQigU1FL7dI/Ax4/_o0rfx9cX Ig/s320/2008_10290025.JPG http://3.bp.blogspot.com/_jmYevHrBr6M/SQigVWhn0oI/AyA/L9XxSgVHo kQ/s320/2008_10290026.JPG Unfortunately I'm still waiting for the 7 x Seagate 1.5 Tb drives ST31500341AS drives to arrive so at the moment it's just sitting there looking all hot and pretty. Although I'm using it for my personal home use i'm sure it will be right at home in a SME without the high pricepoints for a Pillar or similar NAS solution. If you are looking for a solid NAS and those puny 2 and 4 drive boxes are not enough for you :) then you should check out the N7700. Cheers, Dean P.S. This was one of the first 50 or so to land in the country this week so you might find it hard to get your own. Also they are coming out with an rackbased version in about 2 months if you want to wait for it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
I can think of two valid reasons to physically segregate the networks: 1) Insurance. I.e., to eliminate the possibility that otherwise properly configured QoS mechanisms become broken, either by accident, incompetence, or badly-designed or rogue software or hardware - or are otherwise handled carelessly as Jerry Jones suggested. But this is not a compelling argument to me in any but the most critical scenarios such as public-safety applications, etc. or you wish to eliminate service runs - that is unless they are always billable and your customers do not mind you informing them they messed up again and that is why they ahd issues. This is ok once or twice but some customers just cant control things and IF possible to reduce areas where problems could arise why not. 2) Customer preference. If you need the business, then the customer is always right. You might not have adequate credibility with the customer or influence over the design decision, and if a customer in such a situation gets it in their heads that voice and data can't coexist on wires, then it can't. True - just refer to my earlier examples. it is definately smarter at times to walk away. There is a variety of opinions, but no general consensus about where QoS failures typically occur, when they occur. I'm wondering if anyone has anyone has ever experienced QoS issues caused by contemporary Polycom phones like IP330s that had workstations hanging off their builtin switches? If you did, were you able to identify the cause, and was it due to any inherent failure of the phone, such as not marking packets or prioritizing dispatch correctly? No. Well other than the port going dead or flaky. But the switch had best be up to the task. I find in installs where customer is looking for inexpensive phones, they tend to want very inexpensive - and normally unmanaged switches. I will not install an unmanaged switch for other than a residential install. Plus even in fairly large installs where they are hitting an ITSP and traversing say a Watchguard firewall, the firewall will honor marked packets but cannot itself run diffserv and apply a tag. In this case the users pc's are in total control and all that corporate data and voip gets to compete with users streaming music et al to their desktops. In this case unless there is a local voip server even their inside calls will suffer. But the proper solution is to always have a firewall/router than can properly dispatch the packets to the WAN. Have a couple Juniper firewalls I hope to try in a couple weeks to see how they perform. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDP (was Re: network design philosophy and practice)
On Wed, Oct 29, 2008 at 1:28 PM, Drew Gibson [EMAIL PROTECTED] wrote: I tried out the cdp-tools some time ago (it may have been on your recommendation, Kristian) but with no success. Is it possible to disable CDP on the 7940 (image_version : P0S3-08-2-00)? regards, Drew Hmmm... I guess I'd like to know why it didn't work for you but in the meantime it's pretty easy to disable CDP. I've never used 8.x but up through 7.x there was an option to disable CDP in the setup menu on the phone. Because CDP discovery is the first thing these phones do, there isn't a way (at least not one that's practical) to disable it in a config file.* * Classic chicken or the egg... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP ACCOUNT CODE not included in CDR when SIP Status is Unknown
Perhaps this is an issue with the SIP registration? Any idea why Asterisk accepts the call if qualify fails? Please help with this strange issue. When sip show peers returns status Unknown the CDR does not include the accountcode even though the call is correctly processed. I'm using A2 Billing and it uses the accountcode to determine the authentication. Asterisk version 1.4.21.2 I'm calling from a Quintum device. I'm very puzzeled. Name/username HostDyn Nat ACL Port Status 1532497439/1532497439 (Unspecified)D 0UNKNOWN The SIP settings are: [1532497439] type=friend host=dynamic username=1532497439 secret=wspiov8729 accountcode=1532497439 callerid=90002 regexten=90002 amaflags=billing context=OutboundWS disallow=all allow=g729 trunk=yes qualify=6000 qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 directrtpsetup=no Thanks Shaun___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is anyone using * for 2 way video conferencing?
On Wed, 29 Oct 2008, Robert Augustyn wrote: Hi, One of my clients, wants to use * box to run weekly meetings between remote locations over the internet. What would be the best configuration for this? We are talking about two conference rooms. I am referring to the actual hardware/software and bandwidth requirements for this to work well. I have run two software video phones and I had marginal results with it when displayed on large LCDs, delay and blockines ware the problems I have run into ... I've been playing with video phones over the past month or 2. You've got 3 choices: Bottom-end is Xlite, etc. soft-phones. Desktop videophones - currently Grandtream GXV3000 and ATL4000's. Top of the range Polycom video conferencing units. Starting with the top-of the range ones - these just work Don't even need an Asterisk box. Expensive though - I did one help setup a pair of these, one in the UK, the other west-coast US. Both with 42 plasma screens. Very nice, worked very well. Very expensive. More recently I've been using Grandstream GXV 3000's. For the price; Fantastic. They do have audio and video outputs too - I have connected one up to my 32 flat-screen TV and it worked satisfactorily. Picture quality is as good as the bandwidth you allow it to use and they can go from 1 to 30 frames per second. It uses about 128Kb/sec by default, but you can crank it up to 2 or 3 times that. The Polycoms I think were using about 225Kb/sec. I've used the Grandstreamw with XLite - XLite using the same codec, so same screen picture size. More or less just worked when I got the codecs to match. So the big issue is the Internet - you're using a lot more bandwidth, so need a better link. I found with the Polycoms that the VPN we were using was introducing a lot of Jitter to the link which degraded picture quality - turned off encryption and it was fine (cheap Draytek routers doing encryption in software) Right now, I'm using them in a more domestic setting than business - I know more about the Internet in hte UK, so all sites I'm experimenting with have good ADSL conections and 3 of us are on the same ISP, so minimising traffic over the public Internet. So there you go - hope this helps! Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift installation problems
Hi, I have tried installing app_swift on both mac os x and ubuntu now
and am getting the same error. I must be missing something, as I have
tried multiple versions and everytime do sudo make install i get:
if ! [ -f /etc/asterisk/swift.conf ]; then \
install -m 644 swift.conf.sample /etc/asterisk/swift.conf ; \
fi
if [ -f app_swift.so ]; then \
install -m 755 app_swift.so /usr/lib/asterisk/modules ; \
fi
and when i do just sudo make, it spits out a ton of junk, this is at
the end:
/usr/lib/gcc/i486-linux-gnu/4.2.4/include/stddef.h:214: error:
declaration for parameter ‘size_t’ but no such parameter
app_swift.c:451: error: expected ‘{’ at end of input
make: *** [app_swift.o] Error 1
Im not sure whats going on here, i have setup asterisk and gotten it
configured with the x-lite soft phone, so i know that is working. I
am ultimately trying to use adhearsion to integrate with my rails
app. I have also installed cepstral voices and these work in the
terminal so i am confident that is also installed correctly. Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is anyone using * for 2 way video conferencing?
Thank you. What units from Polycom line did you use? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Wednesday, October 29, 2008 4:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is anyone using * for 2 way video conferencing? On Wed, 29 Oct 2008, Robert Augustyn wrote: Hi, One of my clients, wants to use * box to run weekly meetings between remote locations over the internet. What would be the best configuration for this? We are talking about two conference rooms. I am referring to the actual hardware/software and bandwidth requirements for this to work well. I have run two software video phones and I had marginal results with it when displayed on large LCDs, delay and blockines ware the problems I have run into ... I've been playing with video phones over the past month or 2. You've got 3 choices: Bottom-end is Xlite, etc. soft-phones. Desktop videophones - currently Grandtream GXV3000 and ATL4000's. Top of the range Polycom video conferencing units. Starting with the top-of the range ones - these just work Don't even need an Asterisk box. Expensive though - I did one help setup a pair of these, one in the UK, the other west-coast US. Both with 42 plasma screens. Very nice, worked very well. Very expensive. More recently I've been using Grandstream GXV 3000's. For the price; Fantastic. They do have audio and video outputs too - I have connected one up to my 32 flat-screen TV and it worked satisfactorily. Picture quality is as good as the bandwidth you allow it to use and they can go from 1 to 30 frames per second. It uses about 128Kb/sec by default, but you can crank it up to 2 or 3 times that. The Polycoms I think were using about 225Kb/sec. I've used the Grandstreamw with XLite - XLite using the same codec, so same screen picture size. More or less just worked when I got the codecs to match. So the big issue is the Internet - you're using a lot more bandwidth, so need a better link. I found with the Polycoms that the VPN we were using was introducing a lot of Jitter to the link which degraded picture quality - turned off encryption and it was fine (cheap Draytek routers doing encryption in software) Right now, I'm using them in a more domestic setting than business - I know more about the Internet in hte UK, so all sites I'm experimenting with have good ADSL conections and 3 of us are on the same ISP, so minimising traffic over the public Internet. So there you go - hope this helps! Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dealing with progress codes
Thanks for the reply!
I've played around with R to solve this (probably should have mentioned
that), however I wasn't able to make it work. The message is still played
(this message is from the provider). It will move to the next line in the
dialplan, but as soon as users hear the message they hang up.
Since the progress code comes before actual audio is played to the caller
there has to be a way of catching this and dealing with it in the dialplan,
but nothing I've tried so far works.
On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez [EMAIL PROTECTED] wrote:
Try using a R or r on the Dial command, the R option is better for you in
my opinion.
i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
The R option is going to generate a ring tone when the callee indicates
ringing and is going wait for an Answer. As Progress is just for early
media, you wont get that message.
For more info on the Dial command see:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
On Tue, Oct 28, 2008 at 6:56 PM, arkda [EMAIL PROTECTED] wrote:
Some additional information.
I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an
unusual result:
[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical Response)
This occurs about a second after the user hangs up on the error message
being played from the provider. I have a feeling it's trying to execute the
next step in the dialplan but unable since the caller hung up.
Thoughts, criticism, insults all welcome!
On Tue, Oct 28, 2008 at 12:53 PM, arkda [EMAIL PROTECTED] wrote:
Hi,
I've ran into an issue with a PRI provider in a major metropolitan area
that I haven't needed to deal with before and I was hoping someone might
have some insight on how to handle this within the Asterisk dialplan.
At this location users can't always tell if a number is long distance or
not (there are a lot of area codes and prefixes in the vicinity).
Additionally, users are required by the provider to dial the full 10 digit
number even if a call is local since a local call could be for a few
different area codes and prefixes. The problem is the provider requires a 1
in front of the number for long distance calls, but errors out if the call
has a 1 in front and the call is local.
As a result, users are complaining that they are constantly having to
redial with or without the 1. I've tracked down this behavior when a call
fails:
-- Executing [EMAIL PROTECTED]:1] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:2] GotoIf(SIP/user9-b696fb58,
1?5) in new stack
-- Goto (internal,5551515121,5)
-- Executing [EMAIL PROTECTED]:5] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:6] Answer(SIP/user9-b696fb58,
) in new stack
-- Executing [EMAIL PROTECTED]:7] Set(SIP/user9-b696fb58,
CALLERID(num)=555222) in new stack
-- Executing [EMAIL PROTECTED]:8] Set(SIP/user9-b696fb58,
CALLERID(name)=HiThere) in new stack
-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/user9-b696fb58,
--out the pri--) in new stack
-- Executing [EMAIL PROTECTED]:10] Dial(SIP/user9-b696fb58,
Zap/G2/15551515121) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G2/15551515121
-- Zap/22-1 is proceeding passing it to SIP/user9-b696fb58
-- PROGRESS with cause code 31 received
-- Zap/22-1 is making progress passing it to SIP/user9-b696fb58
-- Hungup 'Zap/22-1'
== Spawn extension (internal, 5551515121, 10) exited non-zero on
'SIP/user9-b696fb58'
The above call was a call that is considered local by the provider. The
caller is then redirected to a message (by the provider) saying 'You do not
need to dial a one or zero...' and the message repeats indefinitely.
I'd like to figure out how to handle this in the dial plan so users do
not even know anything happened. To test to see if I could stop the call
progress and reroute it I've tried this so far:
exten = _NX,1,Set(GROUP(default)=dialpool)
exten = _NX,2,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}19]?5)
exten = _NX,3,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED]
)}18]?BLOCK)
exten = _NX,4,NoOp
exten = _NX,5,Set(GROUP(default)=dialpool)
exten = _NX,6,Answer()
exten = _NX,7,Set(CALLERID(num)=${CLR})
exten = _NX,8,Set(CALLERID(name)=HiThere)
exten = _NX,9,NoOp(--out the pri--)
; Primary Dialout
exten = _NX,10,Dial(Zap/G2/1${EXTEN})
exten = _NX,11,GotoIf,($[${HANGUPCAUSE} = 31]?YAY)
exten = _NX,12,Hangup()
; Call limiter
exten = _NX,n(BLOCK),Answer()
exten = _NX,n(BLOCK),Playback(all-circuits-busy-now)
exten = _NX,n(BLOCK),Playback(pls-try-call-later)
exten = _NX,n(BLOCK),Hangup()
; 1
Re: [asterisk-users] Is anyone using * for 2 way video conferencing?
I've been playing with video phones over the past month or 2. You've got 3 choices: Bottom-end is Xlite, etc. soft-phones. Desktop videophones - currently Grandtream GXV3000 and ATL4000's. Top of the range Polycom video conferencing units. Starting with the top-of the range ones - these just work Don't even need an Asterisk box. Expensive though - I did one help setup a pair of these, one in the UK, the other west-coast US. Both with 42 plasma screens. Very nice, worked very well. Very expensive. Define expensive and what models? Thanks! j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dealing with progress codes
From zapata.conf.sample:
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn't seem to work
; with all telcos.
;
; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
; inband: Signal Busy/Congestion using in-band tones
;
; priindication = outofband
arkda wrote:
Thanks for the reply!
I've played around with R to solve this (probably should have mentioned
that), however I wasn't able to make it work. The message is still played
(this message is from the provider). It will move to the next line in the
dialplan, but as soon as users hear the message they hang up.
Since the progress code comes before actual audio is played to the caller
there has to be a way of catching this and dealing with it in the dialplan,
but nothing I've tried so far works.
On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez [EMAIL PROTECTED] wrote:
Try using a R or r on the Dial command, the R option is better for you in
my opinion.
i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
The R option is going to generate a ring tone when the callee indicates
ringing and is going wait for an Answer. As Progress is just for early
media, you wont get that message.
For more info on the Dial command see:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
On Tue, Oct 28, 2008 at 6:56 PM, arkda [EMAIL PROTECTED] wrote:
Some additional information.
I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an
unusual result:
[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical Response)
This occurs about a second after the user hangs up on the error message
being played from the provider. I have a feeling it's trying to execute the
next step in the dialplan but unable since the caller hung up.
Thoughts, criticism, insults all welcome!
On Tue, Oct 28, 2008 at 12:53 PM, arkda [EMAIL PROTECTED] wrote:
Hi,
I've ran into an issue with a PRI provider in a major metropolitan area
that I haven't needed to deal with before and I was hoping someone might
have some insight on how to handle this within the Asterisk dialplan.
At this location users can't always tell if a number is long distance or
not (there are a lot of area codes and prefixes in the vicinity).
Additionally, users are required by the provider to dial the full 10 digit
number even if a call is local since a local call could be for a few
different area codes and prefixes. The problem is the provider requires a 1
in front of the number for long distance calls, but errors out if the call
has a 1 in front and the call is local.
As a result, users are complaining that they are constantly having to
redial with or without the 1. I've tracked down this behavior when a call
fails:
-- Executing [EMAIL PROTECTED]:1] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:2] GotoIf(SIP/user9-b696fb58,
1?5) in new stack
-- Goto (internal,5551515121,5)
-- Executing [EMAIL PROTECTED]:5] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:6] Answer(SIP/user9-b696fb58,
) in new stack
-- Executing [EMAIL PROTECTED]:7] Set(SIP/user9-b696fb58,
CALLERID(num)=555222) in new stack
-- Executing [EMAIL PROTECTED]:8] Set(SIP/user9-b696fb58,
CALLERID(name)=HiThere) in new stack
-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/user9-b696fb58,
--out the pri--) in new stack
-- Executing [EMAIL PROTECTED]:10] Dial(SIP/user9-b696fb58,
Zap/G2/15551515121) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G2/15551515121
-- Zap/22-1 is proceeding passing it to SIP/user9-b696fb58
-- PROGRESS with cause code 31 received
-- Zap/22-1 is making progress passing it to SIP/user9-b696fb58
-- Hungup 'Zap/22-1'
== Spawn extension (internal, 5551515121, 10) exited non-zero on
'SIP/user9-b696fb58'
The above call was a call that is considered local by the provider. The
caller is then redirected to a message (by the provider) saying 'You do not
need to dial a one or zero...' and the message repeats indefinitely.
I'd like to figure out how to handle this in the dial plan so users do
not even know anything happened. To test to see if I could stop the call
progress and reroute it I've tried this so far:
exten = _NX,1,Set(GROUP(default)=dialpool)
exten = _NX,2,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}19]?5)
exten = _NX,3,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED]
)}18]?BLOCK)
exten = _NX,4,NoOp
exten = _NX,5,Set(GROUP(default)=dialpool)
exten = _NX,6,Answer()
exten = _NX,7,Set(CALLERID(num)=${CLR})
exten = _NX,8,Set(CALLERID(name)=HiThere)
exten
Re: [asterisk-users] Dealing with progress codes
I left something out on that last message, sorry.
With r, not R, it will mask the message with ringing. I could then fail it
over to another dial out, however from testing I've found that my users
expect something to happen within 30 seconds (voicemail, pickup, etc.) The
worse-case scenario would be using r a time of 60 seconds. I've been
thinking of implementing this as a temp fix, but not something I want to
leave in place.
On Wed, Oct 29, 2008 at 5:46 PM, arkda [EMAIL PROTECTED] wrote:
Thanks for the reply!
I've played around with R to solve this (probably should have mentioned
that), however I wasn't able to make it work. The message is still played
(this message is from the provider). It will move to the next line in the
dialplan, but as soon as users hear the message they hang up.
Since the progress code comes before actual audio is played to the caller
there has to be a way of catching this and dealing with it in the dialplan,
but nothing I've tried so far works.
On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez [EMAIL PROTECTED]wrote:
Try using a R or r on the Dial command, the R option is better for you in
my opinion.
i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
The R option is going to generate a ring tone when the callee indicates
ringing and is going wait for an Answer. As Progress is just for early
media, you wont get that message.
For more info on the Dial command see:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
On Tue, Oct 28, 2008 at 6:56 PM, arkda [EMAIL PROTECTED] wrote:
Some additional information.
I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an
unusual result:
[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical Response)
This occurs about a second after the user hangs up on the error message
being played from the provider. I have a feeling it's trying to execute the
next step in the dialplan but unable since the caller hung up.
Thoughts, criticism, insults all welcome!
On Tue, Oct 28, 2008 at 12:53 PM, arkda [EMAIL PROTECTED] wrote:
Hi,
I've ran into an issue with a PRI provider in a major metropolitan area
that I haven't needed to deal with before and I was hoping someone might
have some insight on how to handle this within the Asterisk dialplan.
At this location users can't always tell if a number is long distance or
not (there are a lot of area codes and prefixes in the vicinity).
Additionally, users are required by the provider to dial the full 10 digit
number even if a call is local since a local call could be for a few
different area codes and prefixes. The problem is the provider requires a 1
in front of the number for long distance calls, but errors out if the call
has a 1 in front and the call is local.
As a result, users are complaining that they are constantly having to
redial with or without the 1. I've tracked down this behavior when a call
fails:
-- Executing [EMAIL PROTECTED]:1] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:2] GotoIf(SIP/user9-b696fb58,
1?5) in new stack
-- Goto (internal,5551515121,5)
-- Executing [EMAIL PROTECTED]:5] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:6] Answer(SIP/user9-b696fb58,
) in new stack
-- Executing [EMAIL PROTECTED]:7] Set(SIP/user9-b696fb58,
CALLERID(num)=555222) in new stack
-- Executing [EMAIL PROTECTED]:8] Set(SIP/user9-b696fb58,
CALLERID(name)=HiThere) in new stack
-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/user9-b696fb58,
--out the pri--) in new stack
-- Executing [EMAIL PROTECTED]:10] Dial(SIP/user9-b696fb58,
Zap/G2/15551515121) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G2/15551515121
-- Zap/22-1 is proceeding passing it to SIP/user9-b696fb58
-- PROGRESS with cause code 31 received
-- Zap/22-1 is making progress passing it to SIP/user9-b696fb58
-- Hungup 'Zap/22-1'
== Spawn extension (internal, 5551515121, 10) exited non-zero on
'SIP/user9-b696fb58'
The above call was a call that is considered local by the provider. The
caller is then redirected to a message (by the provider) saying 'You do not
need to dial a one or zero...' and the message repeats indefinitely.
I'd like to figure out how to handle this in the dial plan so users do
not even know anything happened. To test to see if I could stop the call
progress and reroute it I've tried this so far:
exten = _NX,1,Set(GROUP(default)=dialpool)
exten = _NX,2,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}19]?5)
exten = _NX,3,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED]
)}18]?BLOCK)
exten = _NX,4,NoOp
exten = _NX,5,Set(GROUP(default)=dialpool)
exten = _NX,6,Answer()
exten =
[asterisk-users] Dimensioning a telephony system based on openser!
Hi, I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to dimensioning this system to work with this requisites: - 1 users; - 100 VoIP to VoIP calls simultaneously capacity; - 30 VoIP to PSTN calls simultaneously capacity; Can anyone point me some ideas of how can i design such a system (how many servers, how to distribute the services among them, etc.). I have this prototype mounted with VMWare, so i think that even making tests with sipp aren't going to be reliable. Thanks in advance, Nuno ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
Nuno Marques wrote: Every calls should pass through mediaproxy so that i can account them. You can do accounting without handling media. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
Without mediaproxy? Only based on SIP messages? 2008/10/29 Alex Balashov [EMAIL PROTECTED] Nuno Marques wrote: Every calls should pass through mediaproxy so that i can account them. You can do accounting without handling media. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] XML Cisco config file
Well guys I got it, I started up again making the xml file according to this: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP#Downgradingthefirmware And... voila ! 7911G working with Asterisk and firmware 8.4.0!!! if anybody need the xml, let me know :) 2008/10/28 Lincoln King-Cliby [EMAIL PROTECTED] I'm not sure if it's the only issue but you're going to have issues with *phonelabel***Etiqueta_del_telefono*/phonelabel*** The text within the phonelabel tag is a maximum of 11 or 12 characters (I can't remember off the top of my head), if it's longer than that--I count 21 characters in the example, the phone will reject the entire configuration file more or less silently (it is logged in the phone's debug log at http://phone ip address/ but there's no display on the phone itself). That sounds like at least part of what's happening in your case. -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC http://www.controlworks.com Crestron Authorized Independent Programmer -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *César García *Sent:* Tuesday, October 28, 2008 6:46 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] XML Cisco config file Hello guys, anybody here that can help me checking out this xml file, cause I am traying to configure some cisco 7911G phones to asterisk and I can't get it done thanks a paste of the file is here: http://pastebin.ca/1239083 -- http://celord.blogspot.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://celord.blogspot.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
Yes. There are some liabilities with that in that the signaling messages may be incomplete (i.e. you may miss a BYE) and this is the usual reason given for doing media proxying for more accurate accounting. But the latency, bandwidth consumption, and increased complexity and cost associated with doing it on a large scale does not justify it, in my opinion. SIP-only accounting is good enough most of the time. Nuno Marques wrote: Without mediaproxy? Only based on SIP messages? 2008/10/29 Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Nuno Marques wrote: Every calls should pass through mediaproxy so that i can account them. You can do accounting without handling media. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
Ok... Maybe you're right. I've read somewhere that this service is needed for taping reasons (policy and other law enforcements). If it's needed whe can just turn it on for that specific number, right? But answering to my question, can you point me some ideas refering about equipment that i should use? BR Nuno 2008/10/29 Alex Balashov [EMAIL PROTECTED] Yes. There are some liabilities with that in that the signaling messages may be incomplete (i.e. you may miss a BYE) and this is the usual reason given for doing media proxying for more accurate accounting. But the latency, bandwidth consumption, and increased complexity and cost associated with doing it on a large scale does not justify it, in my opinion. SIP-only accounting is good enough most of the time. Nuno Marques wrote: Without mediaproxy? Only based on SIP messages? 2008/10/29 Alex Balashov [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] Nuno Marques wrote: Every calls should pass through mediaproxy so that i can account them. You can do accounting without handling media. --Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] XML Cisco config file
Post it on the wiki! Im sure Ill need it someday From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of César García Sent: October 29, 2008 6:54 PM To: Asterisk Users List Subject: Re: [asterisk-users] XML Cisco config file Well guys I got it, I started up again making the xml file according to this: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configurati on+files+for+SIP#Downgradingthefirmware And... voila ! 7911G working with Asterisk and firmware 8.4.0!!! if anybody need the xml, let me know :) 2008/10/28 Lincoln King-Cliby [EMAIL PROTECTED] I'm not sure if it's the only issue but you're going to have issues with phonelabelEtiqueta_del_telefono/phonelabel The text within the phonelabel tag is a maximum of 11 or 12 characters (I can't remember off the top of my head), if it's longer than that--I count 21 characters in the example, the phone will reject the entire configuration file more or less silently (it is logged in the phone's debug log at http://phone ip address/ but there's no display on the phone itself). That sounds like at least part of what's happening in your case. -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC http://www.controlworks.com/ http://www.controlworks.com Crestron Authorized Independent Programmer _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of César García Sent: Tuesday, October 28, 2008 6:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] XML Cisco config file Hello guys, anybody here that can help me checking out this xml file, cause I am traying to configure some cisco 7911G phones to asterisk and I can't get it done thanks a paste of the file is here: http://pastebin.ca/1239083 -- http://celord.blogspot.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://celord.blogspot.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
SIP-only accounting is good enough most of the time. Does not work in production environment. Specially when you are charging per second or per minute. Works only if some one is offering unmetered only service or just doing it for fun. If it metered service like calling cards, termination or metered DID etc, then this can be really bad. My 2 cents. -Jai Buy unmetered SIP DID www.didforsale.com On Wed, Oct 29, 2008 at 3:56 PM, Alex Balashov [EMAIL PROTECTED]wrote: Yes. There are some liabilities with that in that the signaling messages may be incomplete (i.e. you may miss a BYE) and this is the usual reason given for doing media proxying for more accurate accounting. But the latency, bandwidth consumption, and increased complexity and cost associated with doing it on a large scale does not justify it, in my opinion. SIP-only accounting is good enough most of the time. Nuno Marques wrote: Without mediaproxy? Only based on SIP messages? 2008/10/29 Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Nuno Marques wrote: Every calls should pass through mediaproxy so that i can account them. You can do accounting without handling media. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
Jai Rangi wrote: SIP-only accounting is good enough most of the time. Does not work in production environment. Really? Next time I will consult with your authority on what works and does not work in production environments before implementing for large-scale billing solutions that are perfectly functional, and indeed, very much in production. By the way, there are, of course mitigating strategies to minimise risk. Dialog-stateful modules can end the dialog after a certain timeout, you can send periodic re-invites with an SDP offer to probe the endpoints, etc. It is far wiser than introducing a point of failure, a source of latency, and a source of huge bandwidth and processing cost into the call path when you don't need it. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_swift installation problems
What version of Asterisk and what version of app_swift?
On 29 Oct 2008, at 15:10, [EMAIL PROTECTED] wrote:
Hi, I have tried installing app_swift on both mac os x and ubuntu now
and am getting the same error. I must be missing something, as I have
tried multiple versions and everytime do sudo make install i get:
if ! [ -f /etc/asterisk/swift.conf ]; then \
install -m 644 swift.conf.sample /etc/asterisk/swift.conf ; \
fi
if [ -f app_swift.so ]; then \
install -m 755 app_swift.so /usr/lib/asterisk/modules ; \
fi
and when i do just sudo make, it spits out a ton of junk, this is at
the end:
/usr/lib/gcc/i486-linux-gnu/4.2.4/include/stddef.h:214: error:
declaration for parameter ‘size_t’ but no such parameter
app_swift.c:451: error: expected ‘{’ at end of input
make: *** [app_swift.o] Error 1
Im not sure whats going on here, i have setup asterisk and gotten it
configured with the x-lite soft phone, so i know that is working. I
am ultimately trying to use adhearsion to integrate with my rails
app. I have also installed cepstral voices and these work in the
terminal so i am confident that is also installed correctly.
Thanks.___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 'Asterisk is not thread safe' message
I recently built Asterisk from scratch on Ubuntu (Ubuntu 4.2.3-2ubuntu7). Everything seemed to build ok, but when I start Asterisk, I get the message: Warning! Asterisk is not thread safe. Is this anything to be concerned about? How can I make it go away? Is there an alternative threading library I can link against? Thanks! Joe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
Really? Yes, Specially when your service is metered, I don't know how some once justify good enough billing. Dealing with 500 customer calling every day for billing inquiries can turn out to be much more expensive then all other expenses. Next time I will consult with your authority on what works and does not work in production environments before implementing for large-scale billing solutions that are perfectly functional, and indeed, very much in production. No Need to be so contemptuous. On Wed, Oct 29, 2008 at 4:37 PM, Alex Balashov [EMAIL PROTECTED]wrote: Jai Rangi wrote: SIP-only accounting is good enough most of the time. Does not work in production environment. Really? Next time I will consult with your authority on what works and does not work in production environments before implementing for large-scale billing solutions that are perfectly functional, and indeed, very much in production. By the way, there are, of course mitigating strategies to minimise risk. Dialog-stateful modules can end the dialog after a certain timeout, you can send periodic re-invites with an SDP offer to probe the endpoints, etc. It is far wiser than introducing a point of failure, a source of latency, and a source of huge bandwidth and processing cost into the call path when you don't need it. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!
By good enough I really did mean good enough, not sort-of kind-of okay. Jai Rangi wrote: Really? Yes, Specially when your service is metered, I don't know how some once justify good enough billing. Dealing with 500 customer calling every day for billing inquiries can turn out to be much more expensive then all other expenses. Next time I will consult with your authority on what works and does not work in production environments before implementing for large-scale billing solutions that are perfectly functional, and indeed, very much in production. No Need to be so contemptuous. On Wed, Oct 29, 2008 at 4:37 PM, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Jai Rangi wrote: SIP-only accounting is good enough most of the time. Does not work in production environment. Really? Next time I will consult with your authority on what works and does not work in production environments before implementing for large-scale billing solutions that are perfectly functional, and indeed, very much in production. By the way, there are, of course mitigating strategies to minimise risk. Dialog-stateful modules can end the dialog after a certain timeout, you can send periodic re-invites with an SDP offer to probe the endpoints, etc. It is far wiser than introducing a point of failure, a source of latency, and a source of huge bandwidth and processing cost into the call path when you don't need it. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Current Open Source Billing Package
Jerry Jones wrote: After spending a couple hours scanning for an open source (non- commercial) billing package yesterday I am underwhelmed. Almost all of the packages listed on the WIKI appear to be defunct, for several years now. I will be happy to get a login and edit them out if that is the proper method to do so. My requirements are very minimal and at this point unless I have missed something will just write my own. I do not do calling cards. I have no near term need for the package to actually talk with asterisk at all, other than to import the CDR either via files or as a login to MySQL. I do have monthly recurring charges which need to be included monthly. I do occasionally have need to one off (manual) billing charges. Rating for calls would be nice but not mandatory ( we have very minimal International). Ability to export to an accounting package a plus. Ability to generate hard copy Invoices and/or email them to the cust. Ability to generate a list of current Invoices. Runs on Linux. All in all not a very complex set of requirements, but the few packages that seem to be currently offered generally do not fit the bill. Yes there are many commercial packages, but unless they are very minimal in cost I have no interest in them. So my question is, have a missed a golden nugget out there? tia Jerry Have a look at astpp (www.astpp.org) along with OSCommerce. This should do what you're looking for and you do not need to link to Asterisk, etc. Darren Wiebe [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'Asterisk is not thread safe' message
On Wednesday 29 October 2008 19:10:49 joe mcguckin wrote: I recently built Asterisk from scratch on Ubuntu (Ubuntu 4.2.3-2ubuntu7). Everything seemed to build ok, but when I start Asterisk, I get the message: Warning! Asterisk is not thread safe. Is this anything to be concerned about? How can I make it go away? Is there an alternative threading library I can link against? It's a very serious error, in that Asterisk detected that your mutexes are not recursive. They need to be recursive-enabled, or locking will simply not work correctly. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dealing with progress codes
Arka:
I thought you would reroute the call with (or without) the leading one, so,
just Dial again.
This will work and your users wont notice a BIG difference if the call is
answered. The problem is if the call is not answer, because if you have a
busy number, then your users will get something like ring, ring...ring,
beep,beep
For a better solution I would recommend you to get at least your local
prefixes and use the correct dial string with patterns. This can be achieved
with a script.
On Wed, Oct 29, 2008 at 6:15 PM, arkda [EMAIL PROTECTED] wrote:
I left something out on that last message, sorry.
With r, not R, it will mask the message with ringing. I could then fail it
over to another dial out, however from testing I've found that my users
expect something to happen within 30 seconds (voicemail, pickup, etc.) The
worse-case scenario would be using r a time of 60 seconds. I've been
thinking of implementing this as a temp fix, but not something I want to
leave in place.
On Wed, Oct 29, 2008 at 5:46 PM, arkda [EMAIL PROTECTED] wrote:
Thanks for the reply!
I've played around with R to solve this (probably should have mentioned
that), however I wasn't able to make it work. The message is still played
(this message is from the provider). It will move to the next line in the
dialplan, but as soon as users hear the message they hang up.
Since the progress code comes before actual audio is played to the caller
there has to be a way of catching this and dealing with it in the dialplan,
but nothing I've tried so far works.
On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez [EMAIL PROTECTED]wrote:
Try using a R or r on the Dial command, the R option is better for you in
my opinion.
i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
The R option is going to generate a ring tone when the callee indicates
ringing and is going wait for an Answer. As Progress is just for early
media, you wont get that message.
For more info on the Dial command see:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
On Tue, Oct 28, 2008 at 6:56 PM, arkda [EMAIL PROTECTED] wrote:
Some additional information.
I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an
unusual result:
[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical
Response)
This occurs about a second after the user hangs up on the error message
being played from the provider. I have a feeling it's trying to execute the
next step in the dialplan but unable since the caller hung up.
Thoughts, criticism, insults all welcome!
On Tue, Oct 28, 2008 at 12:53 PM, arkda [EMAIL PROTECTED] wrote:
Hi,
I've ran into an issue with a PRI provider in a major metropolitan area
that I haven't needed to deal with before and I was hoping someone might
have some insight on how to handle this within the Asterisk dialplan.
At this location users can't always tell if a number is long distance
or not (there are a lot of area codes and prefixes in the vicinity).
Additionally, users are required by the provider to dial the full 10 digit
number even if a call is local since a local call could be for a few
different area codes and prefixes. The problem is the provider requires a
1
in front of the number for long distance calls, but errors out if the call
has a 1 in front and the call is local.
As a result, users are complaining that they are constantly having to
redial with or without the 1. I've tracked down this behavior when a call
fails:
-- Executing [EMAIL PROTECTED]:1] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:2] GotoIf(SIP/user9-b696fb58,
1?5) in new stack
-- Goto (internal,5551515121,5)
-- Executing [EMAIL PROTECTED]:5] Set(SIP/user9-b696fb58,
GROUP(default)=dialpool) in new stack
-- Executing [EMAIL PROTECTED]:6] Answer(SIP/user9-b696fb58,
) in new stack
-- Executing [EMAIL PROTECTED]:7] Set(SIP/user9-b696fb58,
CALLERID(num)=555222) in new stack
-- Executing [EMAIL PROTECTED]:8] Set(SIP/user9-b696fb58,
CALLERID(name)=HiThere) in new stack
-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/user9-b696fb58,
--out the pri--) in new stack
-- Executing [EMAIL PROTECTED]:10] Dial(SIP/user9-b696fb58,
Zap/G2/15551515121) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G2/15551515121
-- Zap/22-1 is proceeding passing it to SIP/user9-b696fb58
-- PROGRESS with cause code 31 received
-- Zap/22-1 is making progress passing it to SIP/user9-b696fb58
-- Hungup 'Zap/22-1'
== Spawn extension (internal, 5551515121, 10) exited non-zero on
'SIP/user9-b696fb58'
The above call was a call that is considered local by the provider. The
caller is then redirected to a message (by the provider) saying 'You do
not
need to dial a one or
Re: [asterisk-users] Dial() - any way to limit waiting for a RINGING state?
Think more deeply, I understand this is a user forum - but it doesn not mean that all question must be newbie. RINGING state meand until I REALLY get a notification from destination device (SIP for instance) that call have been accepted by the destination and it have returned a RINGING - other. If it does - I would wait more for an answer - I would like to return the user switched off I think there is no solution yet, and Dial() hacking would be a one. On Wednesday 29 October 2008 18:41, Vinícius Fontes wrote: Sure it is: exten = blah,1,Dial(SIP/blah,30) Where 30 is the time in seconds the application will wait before quitting and setting the DIALSTATUS variable to NOANSWER. Atenciosamente, Vinícius Fontes Núcleo de Tecnologias Convergentes Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Convergent Technologies Core Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Anton [EMAIL PROTECTED] escreveu: Hello! Just trying to find out how to limit waiting for a RINGING state for an initiated call by Dial() - This is necessary since I want to inform the CALLER that destination is not available if RINGING state was not received within, say 20 seconds. This applies for mostly SIP and IAX2 calls - Is that possible without hacking the app_dial? Regards, Anton. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
