Re: [Asterisk-Users] Asterisk + XEN does it make sense?
Kevin Steil a écrit : I use VMWare, but will start testing XEN...I use VMWare to slice up some nice big servers to provide dedicated hosted PBXes. We also use the VMs for easy deployment and is a vital part of our DR Plan... Which version of VMWare are you using? Are you having any issues with it? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announcement: Snom 360 with integrated XML O bjects
The XML minibrowser is available in a special image only.
http://www.snom.com/minibrowser/firmware/
In future images it will be in by default.
On Friday 27 January 2006 16:15, Colin Anderson wrote:
Aha, I see it's 4.1, cool. So I just have to do a straight upgrade to 5.0
and I have this new toy to play with, correct?
-Original Message-
From: Sven Fischer (support) [mailto:[EMAIL PROTECTED]
Sent: Friday, January 27, 2006 5:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Announcement: Snom 360 with integrated XML
O bjects
Hi.
Use massdeployment for putting the licenses on to your phones.
There is a setting called license_url you can use like the firmware
update
URL, the macro {mac} will be replaced by the MAC address of the phone. So
if
you provide the setting like this:
license_url: http://yourwebserver/{mac}.txt
all phones will download and install their licenses from this directory
automatically.
Ok, you need to send us a list of all MAC addresses and we will send you
the
needed licenses.
BTW which firmware is already on the phones ? If it is 4.something and they
are working, the license is already on the phones. With firmware 4.0 you
need
to have a license on the phone.
Best regards,
Sven
On Friday 27 January 2006 06:01, Colin Anderson wrote:
Is there any plans for a site license or some way to deploy the license a
little more elegantly? I have a lot of 360's!
I'm excited about this feature - it enables me to deploy some solutions
that I have been promising to my endusers. The two I have in mind are
Outlook calendar push to the display, and Outlook contact pull to the
directory. Some other ones will involve caller ID lookup to our CRM. If I
make progress along these lines, I will post results to the list.
-Original Message-
From: Christian Stredicke [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 26, 2006 9:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Announcement: Snom 360 with integrated XML
Objects
As far as the licence is concerned that is something that we introduced
in the 4.0 image and this is not against our customers (which would be
stupid). It shall protect us from clones.
The jump to the 5.0 is not about this licensing stuff, we just changed
the ramdisk and freed up more memory. I know this is not very pleasant,
and we cross fingers that this is the last time we have to do something
like this. But running out of memory is also not very pleasant!
Especially when new cool features ask for more memory!
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Colin Anderson
Sent: Thursday, January 26, 2006 2:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Announcement: Snom 360 with
integrated XML Objects
that is *very* cool. However, I am somewhat concerned about
being forced to license the firmware (even if it is free) can
you comment for the list the rationale behind forcing a
license and how this might affect Snom users who, say, want
to DOWN grade their firmware?
ps is there a timetable for supported, formally released
firmware version?
-Original Message-
From: Hirosh Dabui [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 26, 2006 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Announcement: Snom 360 with
integrated XML Objects
-BEGIN PGP SIGNED MESSAGE-
Hash: RIPEMD160
Dear user,
the new snom 360 is able to use services from standard web servers.
Users can deploy customized client services with snom 360 and
interact with other users via the keypad. The snom 360 will
use HTTP protocol from standard web servers, like Apache.
Typical services are:
~ 1. To-do lists
~ 2. Stock Information
~ 3. Weather
~ 4. Provisioning
~ 5. Agenda
~ 6. Telephone directory
For further information go to
http://snom.com/wiki/index.php/Xmlobjects
Note: *That is a pre-release, probably the software is still unstable*
Best regards,
Hirosh Dabui
- --
snom technology AG
Dipl.-Ing. Hirosh Dabui
PGP Key-ID: 0x30A34758
mailto:[EMAIL PROTECTED]
http://snom.com
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FCXMUdN9loiwy948EO8th9U=
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Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
On Sun, Jan 29, 2006 at 05:24:12PM +1000, Rob Thomas wrote: The question is somewhat ludicrous, and I'm slightly surprised that no-one has sat down and done the maths about bandwidth utilization. So I did. To handle 5000 calls coming in over a PRI, you'd need 210 or so T1s or 170 E1's. Yes. All of those would generate 320Mega BYTES of data per second (eg, 32Gigabit/sec) No. Using G711A (ie, worst case bandwidth wise): it's 64kbit/s not 64Kbyte/s so it's 320Megabits per seconds Using g729 it's a lot less. And even if it was 320Megabytes/s that equals a little over 3Gbps not 32. There is no way possible that you're going to pump that amount of data through a PC. Don't care about codecs and dialplans, PC's just don't have that sort of internal bandwidth from peripherals. Well, with your above miscalculations, no. But 3.2Gbps is possible with a few boxes. And there are PCs with at least four different PCI-X buses, that's 4GB/s so it might even be possible with one machine. Anyway, it's still a lot of calls. If you do, honestly, need to handle 5k calls, you'd probably have to have a bank of Cisco 5850s doing the termination - With a max of 5 DS3 (1 DS3 = 28 T1's) into each one, you'll need 8, or probably 9 as you'd want to have one as a hot spare. Each of those DS3's would go into some beefy switching fabric (note, that each one is producting 225mbit) and you'd have some sort of asterisk box with huge internal bandwidth handling each one. Cross connect all 9 asterisk boxes via 10Gbit networks (note, you'll need PCI-16x 10g cards) and have a pair of voicemail servers. I'd suggest a pair of big Sun boxes. Then, of course, you have the issue of getting the calls _out_ of the asterisk machines. You've just doubled your bandwidth requirements, so you'll need to double up on the asterisk machines, and split the network up further. I'd take a guess that you could do it under USD$1million (just for hardware) but I wouldn't be surprised if it was USD$10million. I'm happy to sell you any of this 8-) --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vic Sent: Sunday, 29 January 2006 1:16 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question Hi, Zoa, yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them. We will also need an IVR function as well. I am not up to speed on Asterisk yet, so, I am a little bit confused by all the different ways of doing it. Someone is talking about IAX: I think it can only be used between Asterisk servers, right? In this particula rscenario we are getting calls as SIP directly from carrier, so we will not need to do any conversion (I think). We just route the calls to the destination, that's it. Any suggestions on how to proceed? Can Asterisk do it? I read somewhere that it takes about 30 MHz per one voice channel, so if we want to have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines... Not going to fly with our people. Or do 30 MHz are only necessary for transcoding? In other words, if it comes in as SIP and we keep it that way, can we make it a bt more feasible number? Zoa [EMAIL PROTECTED] wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! --- - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717
[Asterisk-Users] How many digium cards per server ?
Hello, How many digium cards is supported per asterisk server ? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] english snom support forums ?
What about starting such a thing on groups.yahoo.com? CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, January 28, 2006 6:32 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] english snom support forums ? Is there a forum for snom support in english? There are some very active snom forums but they appear to be entirely german language only. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nagios and Asterisk
I put a .call file under asteriskserver:/var/spool/asterisk/outgoing via scp (with keys) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darrell Long Is anyone using Asterisk (and Festival) to make calls to appropriate persons (techs, etc. ) when Nagios generates a particular type of alert? If so, I would love to hear how people are doing it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] intel 536 EP as x100p clone?
Hi.. I have one intel 536 EP. Does it possible use it as x100p clone for asterisk? I tried today with no luck :(.. Here is what I did : - plugged the card the card is recognised as (lspci -vv): 00:09.0 Communication controller: Intel Corp. 536EP Data Fax Modem Subsystem: Intel Corp.: Unknown device 1000 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32, Cache Line Size 08 Interrupt: pin A routed to IRQ 12 Region 0: Memory at e200 (32-bit, non-prefetchable) [size=4M] Capabilities: [e0] Power Management version 2 Flags: PMEClk- DSI- D1- D2+ AuxCurrent=375mA PME(D0-,D1-,D2+,D3hot+,D3cold-) Status: D0 PME-Enable+ DSel=0 DScale=0 PME- - make and make install zaptel driver - modprobe zaptel (there's no output) - modprobe wcfxo and get these following messages: ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wcfxo Is there any possibilities using this card with asterisk? I've searched wiki and there I found that people are success using Intel 537EP as x100p clone. While mine is merely Intel 536EP, but I think both are modems made by Intel.. Maybe there's a way making it function like x100p too like someone in asterisk channel (irc) told me :).. Thanks.. Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How many digium cards per server ?
hi I know people in here use 4 boards, but I believe the only real limitation is the number of PCI slots in your computer. Obviously, with 4x4xE1/T1 boards (480 B channels) you need a pretty fast monster to drag it around due to the way the zaptel/asterisk works. So the actual limit is probably mostly restricted to how fast your PC is. I know other telco engines easily drag 16 E1's, but I am not sure Asterisk can do that even without echo cancel? You have to test... Jan [EMAIL PROTECTED] wrote: Hello, How many digium cards is supported per asterisk server ? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream Budgetone mass deployment?
Hello! I am considering mass deployment of Budgetones 102. According to their website, remote provisioning (configuration via TFTP) is possible. Anyone has experience with this? Is this really working? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] changing displayed call info on snom 360
That INFO must be inside the extsting dialog, maybe that was the problem. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Blundell Sent: Monday, January 30, 2006 10:16 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] changing displayed call info on snom 360 Several of my SIP users are in the habit of diverting all their calls to an assistant when they're out of the office. When these calls ring on the assistant's phone, she wants to be able to tell which number they've been forwarded from so that she can say Joe Blow's phone or whatever when she picks up the call. The assistant's phone is a snom 360, which normally just displays the number of the calling party while it's ringing. Snom's FAQ page at http://www.snom.com/wiki/index.php/FAQs suggests that I can send a SIP INFO message to the phone to change the displayed call information. I did a few experiments with a hacked chan_sip.c, but wasn't able to produce any visible effect on the phone. Does anybody have any experience making this snom feature work with Asterisk, or know of any other way to influence the information that's displayed on the phone? Thanks Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nagios and Asterisk
On Mon, Jan 30, 2006 at 10:08:00AM +0100, Mimmus wrote: I put a .call file under asteriskserver:/var/spool/asterisk/outgoing via scp (with keys) There are chances asterisk will read the file before the sshd has completed writing it. Normally you need to move it into the spool directory. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How many digium cards per server ?
On Mon, Jan 30, 2006 at 10:39:11AM +0100, [EMAIL PROTECTED] wrote: hi I know people in here use 4 boards, but I believe the only real limitation is the number of PCI slots in your computer. Is this 4 E1 boards? Obviously, with 4x4xE1/T1 boards (480 B channels) you need a pretty fast monster to drag it around due to the way the zaptel/asterisk works. So the actual limit is probably mostly restricted to how fast your PC is. I know other telco engines easily drag 16 E1's, but I am not sure Asterisk can do that even without echo cancel? You have to test... I'd say one of the things you will run into is the number of interrupts coming at you with so many cards. Personally I recommend one card per box. The card goes for $1800(I buy Sangoma with echo cancel) while the computer goes for $800 (HP Proliant 145). So the computer is really the small expense here and with one card per box I instead have several boxes and thus better redundancy. One card per box also guarantess I won't have any interrupt problems and it's capable of transcoding. Kristian. -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How many digium cards per server ?
Hello, How many digium cards is supported per asterisk server ? hi I know people in here use 4 boards, but I believe the only real limitation is the number of PCI slots in your computer. Obviously, with 4x4xE1/T1 boards (480 B channels) you need a pretty fast monster to drag it around due to the way the zaptel/asterisk works. So the actual limit is probably mostly restricted to how fast your PC is. I know other telco engines easily drag 16 E1's, but I am not sure Asterisk can do that even without echo cancel? You have to test... I recently tested a dual xeon64 3.2 with two sangoma A104 cards, and that could just handle the load. I beleive the digium cards will require about the same load, since it is mainly zaptel eating cpu... -- Roy Sigurd Karlsbakk [EMAIL PROTECTED] --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nagios and Asterisk
Is anyone using Asterisk (and Festival) to make calls to appropriate persons (techs, etc. ) when Nagios generates a particular type of alert? If so, I would love to hear how people are doing it. yeah... do that on ALL the services and hosts, and see how long it takes before your colleagues beat you to death :) -- Roy Sigurd Karlsbakk [EMAIL PROTECTED] --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_snmp
Hello, Is there an app_snmp for asterisk-1.2.3 ? Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announcement: Snom 360 with integrated XML O bjects
On Mon, 30 Jan 2006, Sven Fischer (support) wrote: The XML minibrowser is available in a special image only. http://www.snom.com/minibrowser/firmware/ In future images it will be in by default. Is the xml minibrowser firmware equivalent to 5.0+xml, 5.2+xml? It's not obvious. Also, might want to look at http://www.voip-info.org/wiki/view/snom+360 -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] adress book
Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone mass deployment?
On Mon, 2006-01-30 at 11:42 +0200, Dmitry Ivanov wrote: I am considering mass deployment of Budgetones 102. According to their website, remote provisioning (configuration via TFTP) is possible. Anyone has experience with this? Is this really working? It does work, yes, though I think you need to configure the TFTP server address manually on each phone. Personally I'd be a bit wary of mass Budgetone deployment for other reasons, but the remote configuration stuff shouldn't be a problem. Grandstream use basically the same configuration file system for the Budgetones as they do on the Handytones and the GXP-2000. Obviously you need some way to make the files in the first place: when we deployed our GXP-2000s I ended up writing a little Python script to create the Grandstream config files (and the associated Asterisk config entries) from input data in a Gnumeric spreadsheet. p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Serusers] adress book
Hello Joao, I'm using SER and Asterisks-based system, with centralized LDAP backend. To access LDAP I use SOAP and DSML.This is now used for every provisioning/management/billing/ivr activity in the system. In future I plan to have centralized phonebook based in LDAP. I think that having centralized LDAP directory and accessing it from clients via SOAP/XML is a best option. If security is an issue, SOAP/XML Digital Signature and Encryption could be used here. Maybe we will live until times when hardware vendors will support SOAP clients in their phones, or at least XML browsers or some sort of thin clients. I see SNOM is doing something in XML - maybe worth checking. I think that will be the soft-phone manufacturers that will first adapt idea of central phonebook, based on SOAP message exchanges with centralized LDAP directory servers. -- Regards, Arek Bekiersz Joao Pereira wrote: Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playing music while transfering
Hi, Does anyone know how to play music to a caller while you dial a second call? Once the second calls has answered, i'd like to music to stop, and the calls to be bridged. Thanks Dan Journo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime Queue not realtime anymore in Asterisk 1.2.3?!
Heya, I upgraded from 1.2.0 to 1.2.3, and when a new call comes in on a realtime queue, the queue settings and members are not updated anymore! Only a reload of Asterisk seems to update the settings. Is this a bug or is there some way to solve this? Cheers, Frank ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Serusers] adress book
Yes its how I started (ldap+radius( But it depends what you want to do. 1) If you want to have nice display in softphone (or hardware phone with LCD) of global system phonebook and/or private phonebook - I'm sorry, no vendor is supporting this. I was trying to convince few videophone manufacturers to support XML or SOAP, but there are other reasons they won't do it now (they all go proprietary). 2) If you want to just implement a SIP-like phonebook functionality, like you hook-off, you press # and number of phonebook entry and system dials for you, then voila. You can do it yourself. Just write clever SER module, or PHP script and use Ldap. -- Regards, Arek Bekiersz Voipers Portugal wrote: Hi, I am using SER with centralized LDAP backend which is accessed by RADIUS. Maybe it could work out for you. Jose Simoes On 1/30/06, *Arek Bekiersz* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello Joao, I'm using SER and Asterisks-based system, with centralized LDAP backend. To access LDAP I use SOAP and DSML.This is now used for every provisioning/management/billing/ivr activity in the system. In future I plan to have centralized phonebook based in LDAP. I think that having centralized LDAP directory and accessing it from clients via SOAP/XML is a best option. If security is an issue, SOAP/XML Digital Signature and Encryption could be used here. Maybe we will live until times when hardware vendors will support SOAP clients in their phones, or at least XML browsers or some sort of thin clients. I see SNOM is doing something in XML - maybe worth checking. I think that will be the soft-phone manufacturers that will first adapt idea of central phonebook, based on SOAP message exchanges with centralized LDAP directory servers. -- Regards, Arek Bekiersz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
Using G711A (ie, worst case bandwidth wise): it's 64kbit/s not 64Kbyte/s so it's 320Megabits per seconds That will only do if you talk a lot with your mother in law! ;-) For the rest of the conversation (those with both speaking): 5000 * 64k * 2 = 640M It should in theory work with a 1Gbits Ethernet, but you would be counting on ca 65% utilization. I would normally plan with 30-40 % utilization and you need 2 for redundancy anyway. Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can you disable Forward on a Polycom phone?
Matt:If you use centralized configuration of your Polycom, using XML files; you can disable the call forwarding by setting the divert.fwd.x.enabled attribute of the fwd XML element to 0.For more information and extra attributes you can check the 4.6.2.3.1 section of the IP 300 Admin Guide, available in the Polycom website.Saludos desde Argentina, Tomás.2006/1/24, Matt Darnell [EMAIL PROTECTED]: Matt,Wouldn't they have to actually enter a forwarded number for the forward to activate? I've hit the forward button myself many times after a call ends, and the phone asks you for a new number to forward to. Douglas.You are correct you have to enter something as the contact and then press enable..The users must panic and just press buttons to make it go away. I really wish the re-map buttons worked, that would be an easy way out - or if the screen had forward active inverted like when you have DND active. If I find something I will let you know.Aloha,Matt ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playing music while transfering
Hi, the server play the standard music on hold while you are calling the second person. Bridging the calls is an option of the phone, although once bridged, the music will stop straight away. Dan Journo wrote: Hi, Does anyone know how to play music to a caller while you dial a second call? Once the second calls has answered, i'd like to music to stop, and the calls to be bridged. Thanks Dan Journo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web interface
That manager looks really awesome! There is 1 problem.. I only took 1 semester of German 15 years ago. Looked all over the page for the English button, but I could not find one. I did wake up 10 minutes ago, so I could still be blind. I will rephrase the statement.. AMP hands down is STILL the best FREE asterisk manager... Cheers, /Zac ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re: [Asterisk-Users] Web interface
Sorry my vote goes to AMP for sure. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan-Michael. Guenther (in-put GbR) Sent: Monday, 30 January 2006 2:58 AM To: asterisk-users@lists.digium.com Subject: Re: Re: [Asterisk-Users] Web interface Hi, the Asterisk PBX Manager is STILL the best... though a few are catching up quickly ;-))) (Another absolutely subjective opinion) http://www.in-put.de/voice-over-ip/asterisk-pbx-manager.html Stefan AMP hands down is STILL the best... though a few are catching up quickly On Mon, 2006-01-30 at 01:29 +, Strain Jer wrote: I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] txfax application problem
I wonder if anyone can tell me which version of spandsp is used in iaxmodem?
I would like to use the app_rxfax application WITH iaxmodem.
Regards Allan Gee
Phone: +27 21 4644400 Ext. 103
www.equation.co.za
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Philip
Edelbrock
Sent: 26 January 2006 02:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] txfax application problem
Jeff Herring wrote:
would you care to share with the list your installation procedure
and configuration files associated with your iaxmodem and hylafax
installation alongside asterisk?
Sure! Some things, I'm sure, could use improvement, but this is working
for me:
Get iaxmodem: https://sourceforge.net/projects/iaxmodem
You need libiax2 and spandsp-0.0.3 (yes, the devel one not the other
installed. Both are included in iaxmodem (in the lib directory),
however I grabbed a slightly newer spandsp-0.0.3 from the spandsp site.
Make sure spandsp and libiax2 are found by your system (usually by doing
a 'ldconfig').
Build and install iaxmodem.
iaxmodem wants a config file at /etc/iaxmode-cfg.something. Mine looks
something like:
# cat /etc/iaxmodem-cfg.ttyIAX
device /dev/ttyIAX
port4569
refresh 60
server YOUR.SERVER.IP.HERE
peernameiaxmodem
secret YOUR_SECRET_HERE
cidname John Doe
cidnumber 8005551212
codec slinear
swapbytes true
Now, create an entry for the iax channel in your Asterisk config. Mine
looks something like this (in iax.conf):
[iaxmodem]
type=friend
username=iaxmodem
secret=YOUR_SECRET_HERE
context=faxout
host=dynamic
auth=md5,plaintext,rsa
Notice that the context is 'faxout' in my extensions.conf. Here's what
the relevent contexts are in my extensions.conf:
[fax]
exten = s,1,Dial(IAX2/iaxmodem)
[faxout]
exten = _.,1,Dial(Zap/g2/${EXTEN})
Notice I also have 'fax' which is incoming. That's a context for my zap
channel (a dedicated fax line). From zapata.conf:
group = 2
faxdetect=both
faxdetect=incoming
faxdetect=outgoing
faxdetect=yes
rxgain=0.0
txgain=0.0
context=fax
channel = 4
OK, what this all does thus far: It sets up a serial port, /dev/ttyIAX
in this case, which looks like a fax-modem that is connected to the
provided iax channel. You can point minicom at it and play with it if
you want. Calls coming into Zap-4 will automaticly go to iaxmodem and
'ring' on the /dev/ttyIAX serial device. Faxes going out on iaxmodem
automatically go our on the same Zap channel (although doesn't have to).
Now, run iaxmodem (e.g. iaxmodem ttyIAX), after you've already got
asterisk going, to get it registered. Make sure it registers and things
look OK (iax2 show peers). You could even try to call it or dial out w/
minicom (using /dev/ttyIAX as the modem device).
Now, you can set up hylafax. I installed from RPM, which was pretty
easy following the directions. Run through its set up and get the email
addresses and those relevent things set. Instead of setting up a new
modem config, however, I edited and then copied the supplied one out of
the iaxmodex distro (config.ttyIAX).
Get hylafax going (/etc/rc.d/init.d/hylafax start).
Now, here's the only stumbling block that I had: In order for things to
work, faxgetty needs to be running! The hylafax service doesn't do this
for you, you need to set it up yourself. The easiest way is to add it
to your /etc/inittab. I added it to mine like this (the new line is the
last here, the rest were already there and included for context):
# Run gettys in standard runlevels
1:2345:respawn:/sbin/mingetty tty1
2:2345:respawn:/sbin/mingetty tty2
3:2345:respawn:/sbin/mingetty tty3
4:2345:respawn:/sbin/mingetty tty4
5:2345:respawn:/sbin/mingetty tty5
6:2345:respawn:/sbin/mingetty tty6
7:2345:respawn:/usr/sbin/faxgetty ttyIAX
You may need to restart or 'telinit q' or something to get the changes
noticed by init.
Now you can try sending and receiving faxes. For fun, I created a new
account on the Asterisk server and put a .procmailrc file there which
passes emails to that account to hylafax:
SUBJECT=`formail -xSubject:`
:0 c
* ^Subject: [EMAIL PROTECTED]
|/usr/bin/faxmail -d $SUBJECT
How it works: send a mail to the account and in the subject line put
[EMAIL PROTECTED], and it will fax the email to 555-1234 with 'attn Joe' on
the cover page. Slick... although it seems to take more work to convert
non-text (e.g. images), which I haven't attempted yet.
OK, so there's the crash course. I hope it helps.
Phil
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To
Re: [Asterisk-Users] Web interface
On Mon, Jan 30, 2006 at 06:05:57AM -0600, Zac Amsler wrote: That manager looks really awesome! There is 1 problem.. I only took 1 semester of German 15 years ago. Looked all over the page for the English button, but I could not find one. I did wake up 10 minutes ago, so I could still be blind. I will rephrase the statement.. I figure that http://www.thirdlane.com/opensource.htm#manager -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
On Mon, Jan 30, 2006 at 12:49:15PM +0100, [EMAIL PROTECTED] wrote: Using G711A (ie, worst case bandwidth wise): it's 64kbit/s not 64Kbyte/s so it's 320Megabits per seconds That will only do if you talk a lot with your mother in law! ;-) For the rest of the conversation (those with both speaking): 5000 * 64k * 2 = 640M Indeed you are correct, I'll defend myself with stating that I presumed we were talkin full duplex ;) It should in theory work with a 1Gbits Ethernet, but you would be counting on ca 65% utilization. I would normally plan with 30-40 % utilization and you need 2 for redundancy anyway. Though now you're wrong ;) 65% isn't correct. If you're counting both in and out traffic you'll have to assume that the Gigg card is capable of 1Gbps in each direction thus 2Gbps in total and 640M of 2000G is about 30% or just as much as 320M is of 1G. I don't know the average packet size of a voice RTP packet but I guess it's quite small. Being a network guy I've dealt quite a lot with software routers and a normal Linux machine can forward about 500kpps, and this is mere forwarding if you run this via Asterisk you should probably split that by ten. -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone mass deployment?
On Monday 30 January 2006 13:03, Phil Blundell wrote: Personally I'd be a bit wary of mass Budgetone deployment for other reasons, but the remote configuration stuff shouldn't be a problem. What reasons do you mean? Grandstream use basically the same configuration file system for the Budgetones as they do on the Handytones and the GXP-2000. Obviously you need some way to make the files in the first place: when we deployed our GXP-2000s I ended up writing a little Python script to create the Grandstream config files (and the associated Asterisk config entries) from input data in a Gnumeric spreadsheet. I have created dynamic CGI-like TFTP server so I will create config files on-the-fly. Now we use this system (dynamic tftp server and Perl CGI script) for country-wide Sipura 3000 configuration. BTW, if anyone is interested I can send sources of this TFTP server. -- The PSTN will never be a slave to you. You must be a slave to it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web interface
On 1/30/06, Strain Jer [EMAIL PROTECTED] wrote: I was curious which one is best suited for asterisks. Thanks The 'best' depends on your personal flavor I guess. However I'm impressed by voiceone (http://www.voiceone.it/), didn't use it yet, but will surely look into it sooner or later. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP-H323 translation
Hello, I would like to find an appropriate solution for SIP to H323 translation (vice versa would be great too!), in an environment where there's going to be 100+ concurrent calls: has anyone succesfully implemented such a translator/gateway, e.g. using Opal+OpenH323/Asterisk or anything else? Any idea of the requisites or issues that could be faced?Thank you!Tim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] txfax application problem
Allan Gee wrote: I wonder if anyone can tell me which version of spandsp is used in iaxmodem? IAXmodem version 0.0.8 uses spandsp development snapshot 20051220... so it's 0.0.3, more recent than pre6. I would like to use the app_rxfax application WITH iaxmodem. This will be possible so long as you can get both app_rxfax and iaxmodem to function properly with the same spandsp installation... or unless you can get each to use a different installation of spandsp. That said, there should be no functionality in app_rxfax that you would miss in using IAXmodem+HylaFAX. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] backgrounddetect and busy
Seems like backgrounddetect is missing some checks (perhaps)? Using an analog TDM04B card how do I tell if the line is busy? The call (from a call file) is always given a state of answered immediately after calling even though the person has not yet answered the call. My message to the person starts talking right away (again even though the person has not answered the all) and ... I cant tell if the line is busy on the TDM card. How is this done on a TDM04B? two states to detect. 1) When the person actually answers the call so they dont miss part of the message 2) That the line isnt busy Thanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and LCS ?
Asterisk talks SIP-UDP and LCS talks SIP-TCP. A guy called hjlee or Hyoungjoo Lee has been working on a TCP-patch for Asterisk but this patch is not yet in the official Asterisk and he cannot work on it any longer. Hoe good is it ? Also there is SER or SIP Express Router from iptel.org, does this do what I need and how do I do it ? and does in the conversion in both direction etc ? My investigation on LCS and Asterisk in getting somewhat pessimistic, is that correct or there a solution available at this time ? Also a similar request from someone else on this subject has not been answered to here in the last few days, does this mean LCS-Asterisk is a lost game ? Thanks, Ton den Hartog -- I want to fly like an eagle, in the sky , fly fly fly, like an eagle in the sky Homepage http://www.tonh.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fail over to Pri on VoIP connection failure
All, Thanks for the help. Checking on and changing the route based on dialstatus is the way to go. Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone mass deployment?
On Mon, 2006-01-30 at 15:14 +0200, Dmitry Ivanov wrote: On Monday 30 January 2006 13:03, Phil Blundell wrote: Personally I'd be a bit wary of mass Budgetone deployment for other reasons, but the remote configuration stuff shouldn't be a problem. What reasons do you mean? Just that, from my limited experience of Budgetones, they seem to be generally a bit buggy. But if they work OK in your environment, there's probably no reason not to use them. p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISAC Codec Support
Well, skype. but i was tweaking some code. This is more a question for lab usage. On 1/28/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote: Erick Perez wrote: Besides the codecs that * supports. Is there any ISAC implementation for asterisk available? This is to be used mainly with softphones, i haven't seen any hardphones that support this codec. Which softphone supports it? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP-H323 translation
I have found * with the ooh323 channel to be best for this. On Mon, 2006-01-30 at 15:23 +0200, [EMAIL PROTECTED] wrote: Hello, I would like to find an appropriate solution for SIP to H323 translation (vice versa would be great too!), in an environment where there's going to be 100+ concurrent calls: has anyone succesfully implemented such a translator/gateway, e.g. using Opal +OpenH323/Asterisk or anything else? Any idea of the requisites or issues that could be faced? Thank you! Tim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
64kb/s channel will require a bandwidth of about 90kb/s using ONE voice packet per UDP packet. The overhead is a bit low if you put TWO voice packets in ONE UDP packet. check this out. http://web1.egvrn.net/tokata/VoIP%20Bandwidth%20Consumption.pdf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kristian Larsson Sent: Monday, January 30, 2006 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question On Mon, Jan 30, 2006 at 12:49:15PM +0100, [EMAIL PROTECTED] wrote: Using G711A (ie, worst case bandwidth wise): it's 64kbit/s not 64Kbyte/s so it's 320Megabits per seconds That will only do if you talk a lot with your mother in law! ;-) For the rest of the conversation (those with both speaking): 5000 * 64k * 2 = 640M Indeed you are correct, I'll defend myself with stating that I presumed we were talkin full duplex ;) It should in theory work with a 1Gbits Ethernet, but you would be counting on ca 65% utilization. I would normally plan with 30-40 % utilization and you need 2 for redundancy anyway. Though now you're wrong ;) 65% isn't correct. If you're counting both in and out traffic you'll have to assume that the Gigg card is capable of 1Gbps in each direction thus 2Gbps in total and 640M of 2000G is about 30% or just as much as 320M is of 1G. I don't know the average packet size of a voice RTP packet but I guess it's quite small. Being a network guy I've dealt quite a lot with software routers and a normal Linux machine can forward about 500kpps, and this is mere forwarding if you run this via Asterisk you should probably split that by ten. -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gateways
I have a DrayTek Vigor3300V gateway with 8 FXO ports. I am trying to
configure asterisk to dial out on the gateway. I have one of the FXO ports
configured on sip account 100. If I dial the sip account then the router
gives me dial tone, with which I can dial a number. Unfortunately this is
not the behaviour I desire. I want to setup the FXO port as a trunk with out
it giving me a dial tone. I have tried the Dial(sip/100, D(${NUMBER}))
command, but it doesn't work. Any ideas please.
Corne.
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[Asterisk-Users] Manage api- Matching 'Newchannel' event with the 'Originate' command
Title: Manage api- Matching 'Newchannel' event with the 'Originate' command Hi all, When the 'Originate' command is issued with 'Async' open set to 'yes', I got the response right away with the correct 'ActionID'. What follows is the 'Newchannel' event with a 'Channel' ID, but their is no 'ActionID' to tie it back to the command. How do you guys deal with this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
5k+ simultaneous calls (in/out) are becoming normal with the kind of call centers being opened in my country during the past 24 months (Panama, Central America). Take Dell Corp. for example. the call center they have here is about 3k people taking/making calls (internal, to/from US, Europe, Asia). Other Call Centers are in that figure too. For me, this thread seems a good learning point to calculate how to do that with asterisk. Thanks to the people who answered here. On 1/30/06, Kristian Larsson [EMAIL PROTECTED] wrote: On Mon, Jan 30, 2006 at 12:49:15PM +0100, [EMAIL PROTECTED] wrote: Using G711A (ie, worst case bandwidth wise): it's 64kbit/s not 64Kbyte/s so it's 320Megabits per seconds That will only do if you talk a lot with your mother in law! ;-) For the rest of the conversation (those with both speaking): 5000 * 64k * 2 = 640M Indeed you are correct, I'll defend myself with stating that I presumed we were talkin full duplex ;) It should in theory work with a 1Gbits Ethernet, but you would be counting on ca 65% utilization. I would normally plan with 30-40 % utilization and you need 2 for redundancy anyway. Though now you're wrong ;) 65% isn't correct. If you're counting both in and out traffic you'll have to assume that the Gigg card is capable of 1Gbps in each direction thus 2Gbps in total and 640M of 2000G is about 30% or just as much as 320M is of 1G. I don't know the average packet size of a voice RTP packet but I guess it's quite small. Being a network guy I've dealt quite a lot with software routers and a normal Linux machine can forward about 500kpps, and this is mere forwarding if you run this via Asterisk you should probably split that by ten. -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
64kb/s channel will require a bandwidth of about 90kb/s using ONE voice packet per UDP packet. The overhead is a bit low if you put TWO voice packets in ONE UDP packet. And what is 'one voice packet'? Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Set caller id on Swedish PRI (euroisdn)
Hi,
I have a problem with setting outgoing caller id to nothing (secret)
on our Wildcard TE405P connected to a swedish euroisdn line. Caller ID
seems to work fine when connecting the same line to a Ericsson PBX - so
something must be wrong in my settings, but I don't know what.
I've tried:
exten = _*70X.,1,Set(CALLERID(name)=) exten =
_*70X.,2,Set(CALLERID(num)=) exten =
_*70X.,3,Dial(Zap/g0/${EXTEN:3}|60|T)
But the result is always that the caller id is our main number
(A-number).
Here is an from zapata.conf:
[channels]
language=se
context=from-pstn
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=400
rxgain=1.0
txgain=-4.0
group=0
callgroup=1
pickupgroup=1
immediate=no
overlapdial=no
channel = 1-15,17-31,63-77,79-93
group=1
channel = 94-108,110-124
group=2
context=from-internal
signalling=pri_net
channel = 32-46,48-62
Regards,
Jan
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RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
One voice packet is about 20ms of sampling depending on the codec. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Monday, January 30, 2006 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question 64kb/s channel will require a bandwidth of about 90kb/s using ONE voice packet per UDP packet. The overhead is a bit low if you put TWO voice packets in ONE UDP packet. And what is 'one voice packet'? Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set caller id on Swedish PRI (euroisdn)
Check with your service provider. Hälsningar /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Manage api- Matching 'Newchannel' event with the 'Originate' command
Hello, It's kind of sad, but the only way to have the events of a specific call in EVERY instance have a unique tag to follow is to use the CallerIDname field. Call variables(that you can set at Origination) and variable inheritance have gotten much better in the last year, but they still don't work in all cases for call identification in the Manager API output. CallerIDname is able to be set at almost any stage of a call and is printed with most Manager API outputs for channel events. Also, on the USA PSTN network you cannot send out custom CallerIDname anyway so this makes it a little less painful in terms of causing crazy callerIDnames being sent out. This is how we track calls with the astGUIclient Asterisk Central Queue System(ACQS): http://astguiclient.sourceforge.net/acqs.html Hope that helps, MATT--- On 1/30/06, Wai Wu [EMAIL PROTECTED] wrote: Hi all, When the 'Originate' command is issued with 'Async' open set to 'yes', I got the response right away with the correct 'ActionID'. What follows is the 'Newchannel' event with a 'Channel' ID, but their is no 'ActionID' to tie it back to the command. How do you guys deal with this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Budgetone mass deployment?
We do this routinely as a service for our customers. How many phones do you need to provision? Do you already have the phones? Contact me off list and I can help you out. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Dmitry Ivanov [mailto:[EMAIL PROTECTED] Sent: Monday, January 30, 2006 4:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Grandstream Budgetone mass deployment? Hello! I am considering mass deployment of Budgetones 102. According to their website, remote provisioning (configuration via TFTP) is possible. Anyone has experience with this? Is this really working? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to do anonymous outbound calling
Hi, I'm wanted to do working anonymous calling with my sip provider. To do it, I use SetCallerPres(prohib). The problem: The "fromuser=" parameter overide the value of "CallerID(number)" and do it don't working. Anyone had an idea? Tank's Loic Foucault ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Budgetone mass deployment?
I had a problem with the scripts you can bulk generate, they are linked to the MAC address you initially put in, so if the phone packs in you can't just rename the file. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Blundell Sent: 30 January 2006 13:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream Budgetone mass deployment? On Mon, 2006-01-30 at 15:14 +0200, Dmitry Ivanov wrote: On Monday 30 January 2006 13:03, Phil Blundell wrote: Personally I'd be a bit wary of mass Budgetone deployment for other reasons, but the remote configuration stuff shouldn't be a problem. What reasons do you mean? Just that, from my limited experience of Budgetones, they seem to be generally a bit buggy. But if they work OK in your environment, there's probably no reason not to use them. p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How many digium cards per server ?
How many free intreuppts do you have, its hard to get more than two to work in most systems. How many PSTN lines are you trying to support? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, January 30, 2006 1:00 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How many digium cards per server ? Hello, How many digium cards is supported per asterisk server ? Regards Harry __ _ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question on SIP Domains and registration
Hello, I have a situation where I need to differentiate between registrations by users where there might be clashes on the left hand side (username) portion of the SIP From URI. (for a multi-domain virtual hosting system) It seems that only the username portion is used for SIP authentication, and this means that I have clashes. I need to find a way that asterisk can differentiate between [EMAIL PROTECTED] and [EMAIL PROTECTED] Does anyone have any idea? Asterisk is being used as the PSTN gateway, where all registrations take place on OpenSER. Thanks. -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID over analog?
I've some DID's that I'm using for in-bound faxing, but I'm having some trouble with getting that working perfectly on my T1. So I'm thinking of pointing them to an analog line. Will the DID's simply come in over the analog, presumably sending the DID digits via DTMF? Or is that not something that'll work? Thanks, -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and LCS ?
Also there is SER or SIP Express Router from
iptel.org, does this do what I need and how do I do it ?
Yes.
Converting from TCP-to-UDP is simple; in ser.cfg put:
# Forward to Asterisk
if (method == INVITE) {
if (uri =~ sip:[EMAIL PROTECTED]) {
log(1, Forwarding to Asterisk\n);
rewritehostport(192.168.1.12:5060);
}
}
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
and
does in the conversion in both direction etc ?
Yes but I tried only from LCS to Asterisk.
You need to put a static route in LCS (remember to put asterisk server and
IP in host authentication).
Bye
Mimmus
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[Asterisk-Users] Most Popular FREE SoftPhone for Windows
Hi all. I am trying to find out what the most popular soft phone for Windows is for use with Asterisk. SIP or IAX? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need to recompile * after changing zap echo method?
Dearest List, I guess I missed this point: Is it true that if you change the echo canceler in zconfig.h, and then recompile/install your zap modules, that for this to be taken into effect by * you must then recompile/install *? I would have figured that the zap echo cancellation method was independent of *, and I don't recall seeing any docs mentioning either way. Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cant compile asterisk #error You need newer libpri
Trying to compile asterisk (again) from scratch. I seem to be still experiencing the effects fro Jan 25 where I get no sip to sip audio. I have tried upgrading to 1.2.3 which has made no change in the problem. I am starting over and now trying to compile/install /trunk zaptel libpri asterisk following the instructions to grab the source trees: # svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk # svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel # svn checkout http://svn.digium.com/svn/libpri/trunk libpri I cannot get zaptel or libpri to compile successfully I get this error with libpri: chan_zap.c:66:2: #error You need newer libpri chan_zap.c: In function `zt_call': chan_zap.c:2134: warning: passing arg 2 of `pri_sr_set_useruser' discards qualifiers from pointer target type chan_zap.c: In function `zap_send_keypad_facility_exec': chan_zap.c:2258: warning: implicit declaration of function `pri_keypad_facility' chan_zap.c: In function `zt_hangup': chan_zap.c:2528: warning: passing arg 2 of `pri_call_set_useruser' discards qualifiers from pointer target type chan_zap.c:2537: warning: passing arg 2 of `pri_call_set_useruser' discards qualifiers from pointer target type make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/usr/src/libpri/channels' make: *** [subdirs] Error 1 And I get this with zaptel: chan_zap.c:66:2: #error You need newer libpri chan_zap.c: In function `zt_call': chan_zap.c:2134: warning: passing arg 2 of `pri_sr_set_useruser' discards qualifiers from pointer target type chan_zap.c: In function `zap_send_keypad_facility_exec': chan_zap.c:2258: warning: implicit declaration of function `pri_keypad_facility' chan_zap.c: In function `zt_hangup': chan_zap.c:2528: warning: passing arg 2 of `pri_call_set_useruser' discards qualifiers from pointer target type chan_zap.c:2537: warning: passing arg 2 of `pri_call_set_useruser' discards qualifiers from pointer target type make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/usr/src/zaptel/channels' make: *** [subdirs] Error 1 Vontage:/usr/src/zaptel# I have been careful to get rid of all existing asterisk stuff executables modules and libraries etc. off thesystem before trying to compile this. The release compiles just fine but I st ill get no audio with sip-sip calls if I compile the release version. Thanks Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How many TDM2400P's will a server take?
How many TDM2400P cards can I safelly install in one PC? I'm loking for answers from whoever has a working scenario with * and a number of cards higher than one. Thx, Juan __ Mensagem enviada usando o Webmail da ViaLink ver. 2.7.8 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom 501 horrible echo
One thing I was pondering: you are not, by chance, using the same sip.cfg between version 1.4.1 and version 1.6.2 are you? The file has changed significantly between these versions, and certain acoustic settings that worked with 1.4.1 may not work with 1.6.2 (Not to mention that ipmid.cfg and sip.cfg were merged in the 1.5.x release). That has got to be the problem! I'll let you know how the results go. Upgrading to the correct sip.cfg fixed the problem. The Polycoms are back to their great speakerphone-ness. A gotcha is that the new sip.cfg now contains ntp settings. You'll need to modify these to fit your timeserver setup. -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] How many digium cards per server ?
Hi Harry, How many IRQ do you have ? Be carefull for power supply is it is several TDM2460E (all FXS ports) ! It is better to use a seconf power supply... Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL PROTECTED] Envoyé : lundi 30 janvier 2006 10:00 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] How many digium cards per server ? Hello, How many digium cards is supported per asterisk server ? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to recompile * after changing zap echo method?
Hi there, Don't think you have to recompile * if you've already compiled * with zaptel before. (chan_zap.so exists) Should just have to rebuild zaptel, install the module, and do a ztcfg Good luck, - Original Message - From: Brent Torrenga [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, January 30, 2006 11:34 AM Subject: [Asterisk-Users] Need to recompile * after changing zap echo method? Dearest List, I guess I missed this point: Is it true that if you change the echo canceler in zconfig.h, and then recompile/install your zap modules, that for this to be taken into effect by * you must then recompile/install *? I would have figured that the zap echo cancellation method was independent of *, and I don't recall seeing any docs mentioning either way. Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Polycom 501 horrible echo
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- Upgrading to the correct sip.cfg fixed the problem. The Polycoms are back to their great speakerphone-ness. A gotcha is that the new sip.cfg now contains ntp settings. You'll need to modify these to fit your timeserver setup. Heh, found this out too. I have an option time-offset for the phone, but it's not being used. Had to change the appropriate sip.cfg setting for the phone. Regards, --- Gavin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SER redirect
Check http://www.voip-info.org/wiki/view/Asterisk+at+large Or sipedu http://mit.edu/sip/sip.edu/ Plenty of examples /Vel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sharon Sent: Friday, January 27, 2006 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SER redirect hello, can someone help me with ser redirect to asterisk. any help appreciated. Thanks, AA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Set caller id on Swedish PRI (euroisdn)
Your settings are fine. Debug PRI to make sure SETUP message is OK (which
probably is) and then check with you PRI provider that callerid is enabled
on that PRI - E1.
/Vel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, January 30, 2006 3:43 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Set caller id on Swedish PRI (euroisdn)
Hi,
I have a problem with setting outgoing caller id to nothing (secret)
on our Wildcard TE405P connected to a swedish euroisdn line. Caller ID
seems to work fine when connecting the same line to a Ericsson PBX - so
something must be wrong in my settings, but I don't know what.
I've tried:
exten = _*70X.,1,Set(CALLERID(name)=) exten =
_*70X.,2,Set(CALLERID(num)=) exten =
_*70X.,3,Dial(Zap/g0/${EXTEN:3}|60|T)
But the result is always that the caller id is our main number
(A-number).
Here is an from zapata.conf:
[channels]
language=se
context=from-pstn
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=400
rxgain=1.0
txgain=-4.0
group=0
callgroup=1
pickupgroup=1
immediate=no
overlapdial=no
channel = 1-15,17-31,63-77,79-93
group=1
channel = 94-108,110-124
group=2
context=from-internal
signalling=pri_net
channel = 32-46,48-62
Regards,
Jan
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Re: [Asterisk-Users] How many TDM2400P's will a server take?
Juan Carlos Castro y Castro wrote: How many TDM2400P cards can I safelly install in one PC? I'm loking for answers from whoever has a working scenario with * and a number of cards higher than one. Depends on the specs of the server. For example, a quad Xeon will be able to service many more interrupts/card/channels than a 500 mHz Celeron. :-) Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Live CD?
I would love to run Asterisk on an old laptop, in a mostly solid state configuration, with no HD. The laptop is slow (Pentium 233), and I need PCMCIA support (for my network card). Are any of you aware of a live CD that might work? Thanks, Dave ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP domain support for authentication and virtual hosting
Hi, Can someone explain to me or point me in the direction of documentation for the domain support feature in 1.2.x? Specifically I need to be able to have sip users who are authenticated as [EMAIL PROTECTED] Thanks.. -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DID over analog?
I've some DID's that I'm using for in-bound faxing, but I'm having some trouble with getting that working perfectly on my T1. So I'm thinking of pointing them to an analog line. Will the DID's simply come in over the analog, presumably sending the DID digits via DTMF? Or is that not something that'll work? There is no real technical reason why DID digits can't be sent over an analog pstn line, however I do not know of any telco in the US that actually supports that. Twenty-plus years ago, that would be the only way of handling DID's (since digital facilities didn't exist at that level), but the implemention usually involved E M trunking over analog circuits. I believe asterisk does support at least some level of E M trunking, but not sure if it applies to the TDM04b-type cards. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DID over analog?
Ken, Analog DID's are a bit backwards compared to normal POTS lines. I don't know about outside the US, but here (in California, specifically) I've done a few analog DID installs on some NEC PBX equipment. The trick is that with an analog DID line, the CPE provides the battery to the telco (i.e. the DID phone line plugs into an FXS port on your equipment instead of an FXO port). The DTMF's come from the telco but only because the telco goes offhook and sends the DTMF's down the line, and then connects the call. If you point a DID number at a regular POTS line then when a call rings in it simply rings in like a regular analog phone line. Unless the carrier can provide some form of DNIS on an analog line I believe you'd need one POTS line per DID number. Or you could order one DID trunk group and point all of your DID numbers to it and use FXS ports to handle those incoming calls... -MC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Monday, January 30, 2006 8:22 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DID over analog? I've some DID's that I'm using for in-bound faxing, but I'm having some trouble with getting that working perfectly on my T1. So I'm thinking of pointing them to an analog line. Will the DID's simply come in over the analog, presumably sending the DID digits via DTMF? Or is that not something that'll work? Thanks, -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller Holds (how to ignore Drop Call events from callee?)
Hi, In Brazilian POTS, the caller holds the callee line hijacked - the call is not droped until de caller hungs up. I really don't know the right english term to describe this behavior. My problem is that its most standard here, and there are a number of hacks that count on this. For example, to avoid collect calls, you can setup your PABX to pickup the call, hung and pickup again in the first 1000ms (its called double pickup). If its a collect call, the telco will drop the call. When I'm dialing (my setup is PSTN only) to some PABX that implements double pickup, asterisk receives a Drop Call event and byebye... How can I ignore Drop call events from the callee?? TIA, -- Paulo Scardine ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Most Popular FREE SoftPhone for Windows
I prefer IDEFISK. -Kerry From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave MorrowSent: Monday, January 30, 2006 8:35 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Most Popular FREE SoftPhone for Windows Hi all. I am trying to find out what the most popular soft phone for Windows is for use with Asterisk. SIP or IAX? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cant compile asterisk #error You need newer libpri
In article [EMAIL PROTECTED], Steve Gladden [EMAIL PROTECTED] wrote: I am starting over and now trying to compile/install /trunk zaptel libpri asterisk following the instructions to grab the source trees: # svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk # svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel # svn checkout http://svn.digium.com/svn/libpri/trunk libpri But from the error descriptions below, I bet you didn't do the exact commands above. My guess is that after the first command, you did up-arrow, backspaced over asterisk and typed zaptel, BUT forgot to change the other asterisk between svn/ and /trunk. So you ended up with three copies of asterisk, instead of asterisk, zaptel and libpri.. Also, make sure you do make install in zaptel and libpri before you try to compile asterisk. Cheers Tony I cannot get zaptel or libpri to compile successfully I get this error with libpri: chan_zap.c:66:2: #error You need newer libpri chan_zap.c: In function `zt_call': chan_zap.c:2134: warning: passing arg 2 of `pri_sr_set_useruser' discards qualifiers from pointer target type chan_zap.c: In function `zap_send_keypad_facility_exec': chan_zap.c:2258: warning: implicit declaration of function `pri_keypad_facility' chan_zap.c: In function `zt_hangup': chan_zap.c:2528: warning: passing arg 2 of `pri_call_set_useruser' discards qualifiers from pointer target type chan_zap.c:2537: warning: passing arg 2 of `pri_call_set_useruser' discards qualifiers from pointer target type make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/usr/src/libpri/channels' make: *** [subdirs] Error 1 And I get this with zaptel: chan_zap.c:66:2: #error You need newer libpri chan_zap.c: In function `zt_call': chan_zap.c:2134: warning: passing arg 2 of `pri_sr_set_useruser' discards qualifiers from pointer target type chan_zap.c: In function `zap_send_keypad_facility_exec': chan_zap.c:2258: warning: implicit declaration of function `pri_keypad_facility' chan_zap.c: In function `zt_hangup': chan_zap.c:2528: warning: passing arg 2 of `pri_call_set_useruser' discards qualifiers from pointer target type chan_zap.c:2537: warning: passing arg 2 of `pri_call_set_useruser' discards qualifiers from pointer target type make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/usr/src/zaptel/channels' make: *** [subdirs] Error 1 Vontage:/usr/src/zaptel# I have been careful to get rid of all existing asterisk stuff executables modules and libraries etc. off thesystem before trying to compile this. The release compiles just fine but I st ill get no audio with sip-sip calls if I compile the release version. Thanks Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DID over analog?
Michael Collins wrote: Analog DID's are a bit backwards compared to normal POTS lines If you point a DID number at a regular POTS line then when a call rings in it simply rings in like a regular analog phone line. Unless the carrier can provide some form of DNIS on an analog line I believe you'd need one POTS line per DID number. Or you could order one DID trunk group and point all of your DID numbers to it and use FXS ports to handle those incoming calls... Mr. Collins's explanation is correct, the only fly in the ointment is that while FXS ports will electrically connect to an analog DID trunk, I don't think anyone has done the work to support analog DID signalling on an FXS port. A workable (but not inexpensive) approach is to get an analog DID trunk and then use a converter that handles the DID signalling and converts the line into a standard telco trunk appropriate for connecting to an FXO port. When the call comes in the converter rings the FXO port and once Asterisk answers, delivers the DID digits as DTMF tones. You can find DID/DTMF converters from these suppliers: http://www.exacom.com/html/vertical_markets/did_solutions.htm http://www.ctpx.com/html/vp_2000__product_series.html Good luck! g. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting the two servers
Hi All, I want to setup the interconnectionm between two servers, both having sip clients behind firewalls. I want the calls from any of the servers to land on any of SIP clients on the other. I am looking for dial out plans with the sample configuration files . Thanks,. satish ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DID over analog?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Monday, January 30, 2006 9:22 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DID over analog? I've some DID's that I'm using for in-bound faxing, but I'm having some trouble with getting that working perfectly on my T1. So I'm thinking of pointing them to an analog line. Will the DID's simply come in over the analog, presumably sending the DID digits via DTMF? Or is that not something that'll work? Thanks, -Ken One would have to think that fixing the T1 issue is a far better solution, have you tried asking the questions related to the T1 fax problems? Analog DID trunks are problematic at best, and not supported as far as I have seen in asterisk. Most reliable DID trunks are 4 wire, not 2 wire. They require a special DID trunk interface and I have not seen one for asterisk. While there are 2 wire DID trunks form some telcos, they are a joke. ISDN BRI (2B+D) is also a viable solution as multiple numbers can be routed to an ISDN BRI line - call your telco and see if they will do multiple numbers on ISDN and then look at the capi cards for use with asterisk like the DIVA series from EICON. I have no personal experience with them but many others do. Damon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] About Extensions
Im trying to detect before entering in Meetme , which dtmf has been entered. I did a Background(file) and go to a context where i define a exten = _X.,1,Meetme() I have detected that with (1.2.1) when 1 is entered and conference 1 must be created, extensions say it is not possible and gave a fail. Other case as 01,0001, 1001,etc, works fine. What could be wrong?. Is there any other way to do that. I want to to detect # as send key, but with background and exten it seems not to work. Regards Alberto ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [asterisk-dev] SIP domain support for authentication and virtual hosting
Barry Flanagan wrote: Hi, Can someone explain to me or point me in the direction of documentation for the domain support feature in 1.2.x? Specifically I need to be able to have sip users who are authenticated as [EMAIL PROTECTED] For authentication, we only look at 1) Whether the domain is a domain we host 2) If the user (regardless of domain) is a user we know The user namespace is common for all hosted domains, so you can't have [EMAIL PROTECTED] *and* [EMAIL PROTECTED] /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Kirk IP600
Hi all, has anyone tryied to configure asterisk with Kirk IP600 Dect-IP gateway? Could it works using the skinny channel ? Thanks Giordano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [asterisk-dev] SIP domain support for authentication and virtualhosting
Barry Flanagan wrote: Hi, Can someone explain to me or point me in the direction of documentation for the domain support feature in 1.2.x? Specifically I need to be able to have sip users who are authenticated as [EMAIL PROTECTED] For authentication, we only look at 1) Whether the domain is a domain we host 2) If the user (regardless of domain) is a user we know The user namespace is common for all hosted domains, so you can't have [EMAIL PROTECTED] *and* [EMAIL PROTECTED] /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How many TDM2400P's will a server take?
And you should consider how many FXS's you're running. More than one card with all FXS's will require a turbo fan to cool and if they all ring you'll need a decent power supply to handle the power draw.Rob On 1/30/06, Steven Ringwald [EMAIL PROTECTED] wrote: Juan Carlos Castro y Castro wrote: How many TDM2400P cards can I safelly install in one PC? I'm loking for answers from whoever has a working scenario with * and a number of cards higher than one. Depends on the specs of the server. For example, a quad Xeon will beable to service many more interrupts/card/channels than a 500 mHzCeleron. :-)Steve___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.3 CentOS 4.x RPMS
Hi! I'm trying to install the RPMS, in the installation document the following module is not mentioned: perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm But the RPM is in the CentOS 4 directory. On CentOS 4 the rpm is even already present albeit an older version: [EMAIL PROTECTED] rpms]# rpm -qa | grep -i perl-dbd perl-DBD-MySQL-2.9004-3.1 Update of the stock rpm doesn't work: [EMAIL PROTECTED] rpms]# rpm -Uvh perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm warning: perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm: V3 DSA signature: NOKEY, key ID 0f0d98a3 Preparing...### [100%] file /usr/share/man/man3/Bundle::DBD::mysql.3pm.gz from install of perl-DBD-mysql-3.0002-1.RHEL4.LSE conflicts with file from package perl-DBD-MySQL-2.9004-3.1 etc. What should I do with the new module, is it safe to ignore (not install) the LSE rpm? Thanks!! On Thu, 26 Jan 2006, Andrew McRory wrote: Available in the usual place. ftp://ftp.linuxsys.com/pub/releases/CentOS-4.0 This release includes minor spec changes, spandsp 0.0.2pre23, a new Sangoma wanpipe RPM for use with the LSE kernel rpm and an AMP installation document. Best Regards, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7940 not reading SIP image file
I have a 7940 trying to connect to an existing running system. tftp is configured and running normal. (NOTE: I know there is a later SIP version but this is the one I have) I see the phone bootup and ask for OS79XX.TXT which has [EMAIL PROTECTED] src]# cat /tftpboot/OS79XX.TXT P0S3-05-1-00 Jan 30 18:40:07 snorkel in.tftpd[3875]: RRQ from 192.168.1.96 filename OS79XX.TXT Jan 30 18:40:07 snorkel in.tftpd[3875]: RRQ from 192.168.1.96 filename OS79XX.TXT Jan 30 18:40:07 snorkel in.tftpd[3876]: RRQ from 192.168.1.96 filename SEP0015FA66E8DE.cnf.xml Jan 30 18:40:07 snorkel in.tftpd[3876]: RRQ from 192.168.1.96 filename SEP0015FA66E8DE.cnf.xml Jan 30 18:40:07 snorkel in.tftpd[3877]: RRQ from 192.168.1.96 filename XMLDefault.cnf.xml Jan 30 18:40:07 snorkel in.tftpd[3877]: RRQ from 192.168.1.96 filename XMLDefault.cnf.xml Jan 30 18:40:07 snorkel in.tftpd[3878]: RRQ from 192.168.1.96 filename SEP0015FA66E8DE.cnf Jan 30 18:40:07 snorkel in.tftpd[3878]: RRQ from 192.168.1.96 filename SEP0015FA66E8DE.cnf Jan 30 18:40:07 snorkel in.tftpd[3879]: RRQ from 192.168.1.96 filename SEPDefault.cnf Jan 30 18:40:07 snorkel in.tftpd[3879]: RRQ from 192.168.1.96 filename SEPDefault.cnf [EMAIL PROTECTED] src]# I see the phone asking for the files. [EMAIL PROTECTED] src]# ls /tftpboot/P0S3-05-1-00.* /tftpboot/P0S3-05-1-00.bin /tftpboot/P0S3-05-1-00.sbn my SIPDefault.cnf has the image [EMAIL PROTECTED] src]# grep -i image /tftpboot/SIPDefault.cnf # Image Version image_version: P0S3-05-1-00 [EMAIL PROTECTED] src]# Why is the phone not grabbing the SIP version? Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kirk IP600
Hi! Yes, it works (sort of) but I still have some issues. When using more than 2 handsets some of them do not always ring on an incoming call. This might be because I use only 2 Kirk handsets and the rest are Siemens, maybe it's the driver I created a howto for it, you can find it here: http://www.ecem-it.nl/hardware/Asterisk-Kirk-IP600.txt Let me know if you find any errors / omissions, or the solution to the ringing problem :) On Mon, 30 Jan 2006, Giordano Grandis wrote: Hi all, has anyone tryied to configure asterisk with Kirk IP600 Dect-IP gateway? Could it works using the skinny channel ? Thanks Giordano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: How many TDM2400P's will a server take?
Ah -- for all intents and purposes, assume I can obtain the most kickass PC server hardware in the known Universe. So -- any real-life experiences out there? How many TDM2400P cards can I safelly install in one PC? I'm loking for answers from whoever has a working scenario with * and a number of cards higher than one. Thx, Juan __ Mensagem enviada usando o Webmail da ViaLink ver. 2.7.8 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] How many TDM2400P's will a server take?
A PICMG card equiped with Pentium M CPU permits to reduce dragsticaly the power consumption. As this, you can use more power from your PSU for the interface cards. But, for several TDM2460E/B cards with a heavy traffic charge (many simultaneous rings), I believe that it could be better to use a second separate PSU for the cards. The peak consumption is about 120 W on the 12Vcc branch if all the 24 FXS are ringing together. I think that only an industrial PC can provide a so high power level without risk during years not a small office server. You must also think to add further fans to cool the box AND the FXS modules wich have small heatsinks for the hot components. Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Rob Lith Envoyé : lundi 30 janvier 2006 19:47 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] How many TDM2400P's will a server take? And you should consider how many FXS's you're running. More than one card with all FXS's will require a turbo fan to cool and if they all ring you'll need a decent power supply to handle the power draw. Rob On 1/30/06, Steven Ringwald [EMAIL PROTECTED] wrote: Juan Carlos Castro y Castro wrote: How many TDM2400P cards can I safelly install in one PC? I'm loking for answers from whoever has a working scenario with * and a number of cards higher than one. Depends on the specs of the server. For example, a quad Xeon will be able to service many more interrupts/card/channels than a 500 mHz Celeron. :-) Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone mass deployment?
I have 3 Grandstream Budge-Tone 100 with Firmware 1.0.7.11beta, and they are working very well since more then 4 month now. Guenther Davao City, Philippines, Planet Earth, 32.1 °C Phil Blundell wrote: On Mon, 2006-01-30 at 15:14 +0200, Dmitry Ivanov wrote: On Monday 30 January 2006 13:03, Phil Blundell wrote: Personally I'd be a bit wary of mass Budgetone deployment for other reasons, but the remote configuration stuff shouldn't be a problem. What reasons do you mean? Just that, from my limited experience of Budgetones, they seem to be generally a bit buggy. But if they work OK in your environment, there's probably no reason not to use them. p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Gateways
Hi,
I didn't have this gateway, But on welltech 4fxo gateway. You can just dial
SIP/[EMAIL PROTECTED] Even the gateway didn't register to the
asterisk server.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Corne
Vermeulen
Sent: Monday, January 30, 2006 10:15 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Gateways
I have a DrayTek Vigor3300V gateway with 8 FXO ports. I am trying to
configure asterisk to dial out on the gateway. I have one of the FXO ports
configured on sip account 100. If I dial the sip account then the router
gives me dial tone, with which I can dial a number. Unfortunately this is
not the behaviour I desire. I want to setup the FXO port as a trunk with out
it giving me a dial tone. I have tried the Dial(sip/100, D(${NUMBER}))
command, but it doesn't work. Any ideas please.
Corne.
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[Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't
I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,d4,ami fxsks=25 And in zapata.conf, I have: group=2 language=en context=from-pstn signalling=fxs_ks channel=25 I have one analog line plugged in for testing. If I dial that analog number, the inbound call arrives, and it works great. However, when I place an outbound call, I get the following output: -- Called g2/5148346 -- Zap/25-1 answered SIP/412-9b72 However, my number never rings. After about 30 seconds, I get a message saying my call could not be completed as dialed. Almost like it didn't get all of the digits. Is there a way to inject a pause before dialing? Any other thoughts? Any help is greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help with iaxy - one way sound
I have an iaxy the is across a 802.11b link from my asterisk server. signal strength is good and it has been working fine there for about a year. Friday night lightning took out the x100p card in my asterisk server and I just got it all working again last night. Lucky I had a spare. Since I got every thing setup I have been having problems with the iaxy - I really doubt that it is lightning related - but probably something I am missing. I can call the phone from any of the other phones on the server and it works. I can also call any of the other phones or features on the asterisk server and it works - including dtmf. But if an incoming call rings to it, it rings - you can answer and you can hear the calling party but they can't hear you. It is in a ring group and all the other phones in the ring group work fine. I have reprovisioned it both from asterisk and with the winiaxprov utility. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec preference selection?
Hi; I'm trying to implement what is known by Cisco Callmanager as regions: Specify that when phones from zone A call to phones in zone B, use g729, but if they call to zone C, use g711. Any ideas on how to achieve this? Thanks! Francisco Sedano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web interface
Hello Zac, There is 1 problem.. I only took 1 semester of German 15 years ago. Looked all over the page for the English button, but I could not find one. I did wake up 10 minutes ago, so I could still be blind. the language of the module is influenced by the language you choosed for webmin. That's why there is no button in the PBX Manager to choose the language. I will rephrase the statement.. AMP hands down is STILL the best FREE asterisk manager... ACK ;-)) Bye, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: help with redirect from SER
hello all,
i have a problem, and i'm tearing my hair out...any assistance is appreciated. I am trying to redirect from SER to Asterisk, both on the same machine. In 1.09 I didnt need to set up a peer for SER, just autocreatepeer=yes, and rewritehostport from SER as below, and asterisk accepted the requests without a problem. When I updated to
1.23 requests from SER to asterisk die quietly, no matter how verbose my asterisk is. It's as if the requests dont exist at all.
My setup is as follows: asterisk and SER on the same box, SER running on 5060 and asterisk on 5070.
All i want is a simple redirect from SER to asterisk, in ser.cfg thusly:
if (uri == sip:[EMAIL PROTECTED]) { log(1, Forwarding to Voicemail\n); rewritehostport(myIP:5070);
route(1); break;
}
and in SIP.conf (this is what i have after some hours of trying, but it doesnt seem to be helping):
bindaddr=myIP
bindport=5070disallow=all ; Disallow all codecsallow=ulawallow=alawallow=ilbcallow=gsmdtmfmode=rfc2833autocreatepeer=yesinsecure=port,invite
[SER]type=friendhost=myIPfromdomain=myDomaincontext=mycontextcanreinvite=noinsecure=very
if anyone can help i'd me most grateful. I originally thought it would be as simple as changing port to bindport in sip.conf. Oh, how wrong i was.
thanks,
yair
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Re: [Asterisk-Users] Grandstream Budgetone mass deployment?
On Mon, 30 Jan 2006, Dmitry Ivanov wrote: I have created dynamic CGI-like TFTP server so I will create config files on-the-fly. Now we use this system (dynamic tftp server and Perl CGI script) for country-wide Sipura 3000 configuration. BTW, if anyone is interested I can send sources of this TFTP server. you know you can provision sipura 3000 via http, right? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone mass deployment?
On Mon, 30 Jan 2006, Phil Blundell wrote: Budgetones as they do on the Handytones and the GXP-2000. Obviously you need some way to make the files in the first place: when we deployed our GXP-2000s I ended up writing a little Python script to create the Grandstream config files (and the associated Asterisk config entries) from input data in a Gnumeric spreadsheet. does your python script generate the binary format grandstream files or do you still need to use their closed-source tool? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] help with iaxy - one way sound
At first glance it sounds like a routing issue. Are you IAX to your phones, but SIP to your tisp provider? Any change on your asterisk box / firewall / ISP / TISP since then? My first guess is that when you replaced the card you changed a network / iptables setting etc. on your asterisk box. The lightning might be a red herring. Michelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Litwiller Sent: Monday, January 30, 2006 2:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] help with iaxy - one way sound I have an iaxy the is across a 802.11b link from my asterisk server. signal strength is good and it has been working fine there for about a year. Friday night lightning took out the x100p card in my asterisk server and I just got it all working again last night. Lucky I had a spare. Since I got every thing setup I have been having problems with the iaxy - I really doubt that it is lightning related - but probably something I am missing. I can call the phone from any of the other phones on the server and it works. I can also call any of the other phones or features on the asterisk server and it works - including dtmf. But if an incoming call rings to it, it rings - you can answer and you can hear the calling party but they can't hear you. It is in a ring group and all the other phones in the ring group work fine. I have reprovisioned it both from asterisk and with the winiaxprov utility. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help with iaxy - one way sound
All the phones, the iaxy and the server are on a 192.168.3.* network and the only outside interface currently is the x100p card. I do have an account with a voip provider but I haven't got that setup up yet since rebuilding the server on Sunday. So there shouldn't be any routing issues coming into play here. Technical Support wrote: At first glance it sounds like a routing issue. Are you IAX to your phones, but SIP to your tisp provider? Any change on your asterisk box / firewall / ISP / TISP since then? My first guess is that when you replaced the card you changed a network / iptables setting etc. on your asterisk box. The lightning might be a red herring. Michelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Litwiller Sent: Monday, January 30, 2006 2:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] help with iaxy - one way sound I have an iaxy the is across a 802.11b link from my asterisk server. signal strength is good and it has been working fine there for about a year. Friday night lightning took out the x100p card in my asterisk server and I just got it all working again last night. Lucky I had a spare. Since I got every thing setup I have been having problems with the iaxy - I really doubt that it is lightning related - but probably something I am missing. I can call the phone from any of the other phones on the server and it works. I can also call any of the other phones or features on the asterisk server and it works - including dtmf. But if an incoming call rings to it, it rings - you can answer and you can hear the calling party but they can't hear you. It is in a ring group and all the other phones in the ring group work fine. I have reprovisioned it both from asterisk and with the winiaxprov utility. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.3 CentOS 4.x RPMS
Remco Barende [EMAIL PROTECTED] wrote: Hi! I'm trying to install the RPMS, in the installation document the following module is not mentioned: perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm But the RPM is in the CentOS 4 directory. On CentOS 4 the rpm is even already present albeit an older version: [EMAIL PROTECTED] rpms]# rpm -qa | grep -i perl-dbd perl-DBD-MySQL-2.9004-3.1 Update of the stock rpm doesn't work: [EMAIL PROTECTED] rpms]# rpm -Uvh perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm warning: perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm: V3 DSA signature: NOKEY, key ID 0f0d98a3 Preparing... ### [100%] file /usr/share/man/man3/Bundle::DBD::mysql.3pm.gz from install of perl-DBD-mysql-3.0002-1.RHEL4.LSE conflicts with file from package perl-DBD-MySQL-2.9004-3.1 etc. What should I do with the new module, is it safe to ignore (not install) the LSE rpm? Thanks!! Remy, It should be safe to ignore the package. Also please copy my email address on any question regarding my RPM's. Thanks, Andrew McRory - President/CTO Linux Systems Engineers, Inc. - http://www.linuxsys.com Located in beautiful Tallahassee, Florida Office 850-224-5737 Office 850-575-7213 Mobile 850-294-7567 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dlink DVG-3004S ?
Anyone have one of these yet? http://www.dlink.com/products/?pid=451 -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
