Re: [asterisk-users] Flash IDE
Does it have to be a flash device? I have an 8GB flash drive that is really a small hard disk that plugs into and is powered by a USB port. The device is 3cmx3cmx0.5cm, silent, fast and wears out like a hard disk not flash memory. It doesn't stick out (and so get knocked off) because the USB connecter is hinged. Something like this might be better for your (though more expensive). On the other hand, flash devices able to accommodate the needs of Asterisk cost almost nothing. Why not use two devices one that is updated in real-time and that is backed up periodically (say overnight). If one begins to fail then you can switch over the periodic back up and used a new device for backup. Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Sandro Sent: 12 September 2007 13:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Flash IDE You could read the archives from a week or 2 ago under the heading: Build your own appliance Yap... read it, thanks I use these deices, but I unload them entirely into RAM. Fine.. I though about that too but what about: - if power fails? - how/when to write changes to DOM? If you're sticking a normal disctibution on it, I'd suggest dumping the DOM and getting a laptop type IDE/SATA drive and using that instead. It's not silent, but will be very quiet. Yeah, the trouble is that device we bough do not have space for HDs.. :) Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends. aspxmkt=en-us ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is is possible yo play a recording into a channel
We record calls using Asterisk. We can listen to music-on-hold. We can play pre-recorded messages to a caller. So is it possible to inject an existing recording into a conversation? From time-to-time we want to be able to review all or part of a past conversation with a third party with them. I imagine there will need to be some mechanism to select the recording file which is to be played but making the assumption such a mechanism exists, how can the sound associated with any given recording be heard by the parties to a call? Is it a case of creating a channel that plays a file as one of the parties to a conference call? If you have done this any direction you can offer will be appreciated. Bill Seddon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIPURA SPA-2000 webserver dead after firmwareupgrade
Has anyone seen something like that and is there a fix? A google search didn't turn up any apparent hits. I have seen exactly this problem. Even IVR failed to work. Got an RMA from the supplier and they exchanged with no questions. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Prior Sent: May 11, 2005 2:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIPURA SPA-2000 webserver dead after firmwareupgrade I just got a refurb Sipura SPA-2000 and was able to assign it an IP address with DHCP and ping the device, but then I ran the firmware upgrade utility to bring it up to spa2k-2.0.13g which seemed to work just fine, but after it rebooted I cannot connect to its webserver for configuration. I can still ping the unit. When I use the built in voice menu it reads back the right IP address, webserver port, and claims the webserver is enabled, but I can't connect to port 80 on the device and running the firmware upgrade utility says that it cannot connect to the unit either. Has anyone seen something like that and is there a fix? A google search didn't turn up any apparent hits. Thanks Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@Home Installation Problems
David If your machine has been used for Windows, book from a DOS floppy and use FDISK to remove the partitions and try again. I've never tried to install on a machine with existing partitions but never had a failure on a machine without them. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Fulton-Howard Sent: March 10, 2005 5:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] [EMAIL PROTECTED] Installation Problems I just found out about Asterisk (yes, from the Slashdot posting), and I would like to set up my old computer as a dedicated box for my house using [EMAIL PROTECTED] However, when I try to install from the bootable CD, it gets to 54% of copying the image to the hard drive and then says that an error occurred because I have run out of disk space. The machine has a 10.2 GB hard drive, so I don't think space should be a problem since the image is being copied over from a CD... right? If not, what else could be the problem? Also, I noticed that when I boot with an XP CD to look at the partitions, the first one is about 800 MB, the scond one is 9 GB, and the third one is a couple hundred MB. I would assume it's trying to use the 9 GB one and the 800 MB one is the swap, right? If not, is there any command-line parameter I could use at the beginning of setup to fix things? If it helps, my specs are as follows: 450 MHz PII 256 MB SDRAM i440BX-based motherboard nVidia TNT AGP video card 3Com 3C905TX NIC Thanks, David Fulton-Howard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Manager API - multi Originate calls
read inplaces that you use originate command and wait for an event back, does that mean you cannot place another originate until the event comes back ? Not in my experience. Originate will not send an event to the caller until either the intended caller (that is the extension used in Originate) has picked up their phone or a timeout occurs because the intended caller does not pick up their phone. You can send as many originate requests as you like but they will fail if more than one uses the same extension at the same time. The issue you will face is determining which event generated by Asterisk belongs to which origination request. For this reason, the Manager API allows you to specify an ActionID on any command. An ActionID is any string of characters that you use to uniquely identify each command use issue. Asterisk will include the ActionID with each related event so you know which events to respond to and which to ignore. You will see many events generated by Asterisk only some of which will relate to your command. The others will be events that Asterisk raises (for example when a phone registers) or events in response to commands issues by other Manager API users and at the command line. Take a look at Nicolas Gudinos Flash Operator Panel (www.asternic.org) as it used the manager API extensively (albeit through a proxy) and will typically make many requests via the Manger API. Is it true that multiple API connections to Asterisk Manager API will crash it (thinking of alternative way to crack the nut) Again, not in my experience. Lyquidity Solutions Limited +44 (0) 208 241 0500 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Owen hosted Sent: March 02, 2005 12:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Manager API - multi Originate calls Been researching connecting over TCP\IP to Asterisk Manager API to initiate several concurrent calls to dial out. Prefer not to generate ASCII .call files. Question : I read inplaces that you use originate command and wait for an event back, does that mean you cannot place another originate until the event comes back ? Is it true that multiple API connections to Asterisk Manager API will crash it (thinking of alternative way to crack the nut) All help would be welcome - thanks Stephen Owen sip:[EMAIL PROTECTED] IM:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Manager Message: Originate failed beinggenerated when callee does not pick up
I am getting "Message: Originate failed" even the phone is ringing on the other end of the line. Originate will ring your own extension first and when you pick up, call the other number. If you don't pick up your extension, you will receive the message you see. Bill Seddon From: [EMAIL PROTECTED] on behalf of Thomas MillerSent: Mon 28/02/2005 17:33To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Manager "Message: Originate failed" beinggenerated when callee does not pick up I am using the Manager tooriginate calls. I am getting "Message: Originate failed" even the phone is ringing on the other end of the line. How can I reliably know if the phone on the other end of the line is receiving the call? Thanks, Tom Do you Yahoo!?Yahoo! Mail - 250MB free storage. Do more. Manage less. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Behind NAT
Title: [Asterisk-Users] Asterisk Behind NAT Alternatively, I'm open to any suggestions that would work Like you I read about and NAT and the problems. After a few days unsuccessful battling I gave up. Instead of using SIP directly, we've taken SIP numbers with a VoIP service provider and receive calls using IAX from the VoIP provider. I guess you could do the same yourself: have an instance of Asterisk outside your firewall holding just SIP definitions and a simple dialplan to direct calls to and an Asterisk instance within a firewall using IAX that has a complete dialplan. I'm sure the VoIP providers that offer SIP-IAX and IAX-SIP, such as the one we use, are doing more and that there are some gotchas. But its an idea. Bill Seddon From: [EMAIL PROTECTED] on behalf of sammy ominskySent: Mon 2/28/2005 9:03 PMTo: Asterisk UsersSubject: [Asterisk-Users] Asterisk Behind NAT Hi all,I've done quite a bit of reading, and I see that it's going to bedifficult, but as a last-ditch effort before implementing a suggestionI don't like at all, I figured I'd ask...Has anyone successfully put an asterisk box on an internal networkbehind a NAT device and been able to connect with SIP from outside?The real point behind all this is to implement QoS for the voicetraffic, and putting a third box in front of the asterisk and NAT boxeshas been deemed "too expensive".Currently, asterisk has a public IP, as does the NAT box behind whichall the office machines sit. If it can be done, the NAT box would bethe best place to do the QoS, so why not ask, right?Alternatively, I'm open to any suggestions that would work. I've beenhanded this challenge on my first day on a new job... :/Thanks,---sambo___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@Home
Title: [EMAIL PROTECTED] Is there a forum for [EMAIL PROTECTED] Its a great Asterisk option but I have some questions and this forum doesnt seem the right place to ask. Bill Seddon Lyquidity Solutions Limited ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Determine IP addres of a AIP/IAX user
Niels
Are there any other possibility to store the
SIP/IAX callers IP address on every call?
Run the command database
show at the Asterisk command line (CLI). I believe this shows the information you
require. If it does, you can access the information within the dialplan using
DBGet()
Bill Seddon
Lyquidity Solutions Limited
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: February 26, 2005 10:32 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Determine IP addres of a AIP/IAX user
Hello all!
Is there any possibility
to determine the IP address of a caller in my dialplan?
I would like to have a
predefined channel variable like ${CALLER_IP} but it seems it doesnt
exist (http://www.voip-info.org/wiki-Asterisk+Variables)
.. is this list complete?
Are there any other
possibility to store the SIP/IAX callers IP address on every call?
Thanks
Niels
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RE: [Asterisk-Users] CallTransfer
Herman Do you mind sharing where in the configs you have changed this ? Have a look at the options that are controlled by settings in features.conf. If these are what you want to control, I think you will find the options obvious. Bill Seddon Lyquidity Solutions Limited -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Herman Cremer Sent: February 24, 2005 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallTransfer Do you mind sharing where in the configs you have changed this ? -Herman On Thu, 2005-02-24 at 12:40, Mark Benson wrote: I get the impression that the transfer/flash/recall etc etc buttons don't always work - it seems to depend on what phone/firmware you are using. And possibly the version of asterisk. I am using BT102s and some generic voip phone. On the BT102 the transfer button will put the call on hold and give you a new line to call an extention with, however nothing happens when I call an extention. On the generic voip phone the transfer button does nothing. I have resorted to using # for blind xfer and *2 for attended xfer. Herman Cremer wrote: Hi I was wondering if there are any special settings that I need to be able to transfer calls. Whenever I press the 'recall' button, I just here a click, and no ring-tone to transfer. in my debug log I get this : -- Feb 24 09:09:27 DEBUG[19216]: Exception on 10, channel 1 Feb 24 09:09:27 DEBUG[19216]: Got event Pulse Start(14) on channel 1 (index 0) Feb 24 09:09:27 DEBUG[19216]: Exception on 10, channel 1 Feb 24 09:09:27 DEBUG[19216]: Got event Event 65585(65585) on channel 1 (index 0) Feb 24 09:09:27 DEBUG[19216]: Pulse dial '1' -- Any ideas ? Herman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Inheriting variables
Julian For example, I want to store the zap channel number into SourceChannel on the incoming call, and make it available to the called agent channel. If global variables are not suitable maybe the registry will work. You may have looked at DBGet and DBPut to retrieve and store arbitrary values. For example you might store values using the channel id as the key and when the call ends delete the values. Database values are available to all channels but you can have your dialplan only work with database values related to its own channel or channels that are related in someway. Bill Seddon Lyquidity Solutions Limited -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: February 24, 2005 3:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Inheriting variables Jonathan Hobbs wrote: What you want is SetGlobalVar, which sets a variable that is available to any channel. No, I only want the variable available to the original channel and all connected channels. I don't want it available to any channel. For example, I want to store the zap channel number into SourceChannel on the incoming call, and make it available to the called agent channel. Thanks anyway. Julian. hth Jonathan - Original Message - From: Asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: February 24, 2005 10:31 AM Subject: [Asterisk-Users] Inheriting variables I'm trying to set a channel variable and make it available to another channel: I thought that if I SetVar(_SomeVariable=SomeValue) or SetVar(__SomeVariable=SomeValue) then SomeVariable would be available in the destination channel. However __SomeVariable, _SomeVariable and SomeVariable are all blank. The scenario: Agents logon to the queue using callbacklogin. From what I can gather the following happens: 1) Incomming Call - answered, plays welcome message added to Q 2) Local channel calls the agent, plays announcement 3) incomming call connected to agent where would I be able to retreive the inherited variables ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mixing sound files?
is there anyway to mix these two files? Use the soxmix utility. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Muhammad Muzzamil Luqman Sent: February 23, 2005 11:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] mixing sound files? As soon as a call hits the asterisk a menu is played Press 1 for ... and 2 for... I have got the speech in different mp3 file and the music in different mp3 file. is there anyway to mix these two files? Kindest Muhammad Muzzamil Luqman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] List tips for new subscribers --sorry for 2nd post, missed this.
Colin wrote: A lot of good sensible stuff. Well done Colin. Bill Seddon Lyquidity Solutions Limited -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: February 23, 2005 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] List tips for new subscribers --sorry for 2nd post, missed this. This list is for discussions among users of Asterisk, not a getting started hotline for beginners. Beginners learn by reading documentation and examining the sample files included. Mmm, I (respectfully) disagree. One of the unstated objectives of mechanisms like this list is to evangelize the platform. Obviously, we all want it to do well. You can only do that by creating mindshare. You create mindshare by winning people over. You win people over in a technical context by helping them wrap their heads around the concept and implement it. Sometimes, this involves hand-holding, as I do with my boss, my boss' boss etc every day. Look, there are two kinds of people (on the list): One that can deal with the technical implementation of Asterisk and have no problem with it, and the other kind, that get fired up about the *concept* but are short of the chops to make it happen. We ignore the second kind at our own peril (I actually should have said you guys instead of we because I am in-between the two types). I can see a scenario where if the platform becomes inaccessible to PHB / noob types because of things like attitude, Asterisk will be relegated to also-ran status with such illustrious company as the Amiga, which still has an incredibly vocal minority that insists that Amiga still r00lz, but nobody listens to them and considers them crackpots who should Just Get Over It. We *know* Asterisk is a category-redefining platform. We *know* it is Insanely Great. But ticking people off with brusque answers and flames will *not* win the hearts and minds of potential adopters. *1-2 *1 As far as the previous black box comment goes (where an implementor doesn't want someone to know about the inner workings, so they can charge $$$), there is a certain truth to that, and, while GPL allows for that, it is completely contrary to the spirit of the platform and makes the Asterisk community no better than (insert your favorite telecom player whipping boy here) *2 Are you so fussy about how your inbox or whatever is displayed that you are willing to alienate a potential adopter because he top posts or uses HTML? Come on you guys, who cares? What if you pissed off the CTO of a Fortune 500 and he ruled out an Asterisk rollout because he took your flames personally? (sounds like a stretch, but I find it plausible) You didn't do too much to help the platform that day, did you? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Able to tell if phone is registered?
I was wondering if I could do something with regexten, but seemed like a messy solution. Is there any other way? I don't mind getting my hands dirty with perl or C, but I wouldn't really know where to start ... A suggestion is to take a look at the commands available at the CLI and determine if the information they provide would meet your requirement. In particular look at the command database show. This will/should show the phone registered and you can query the contents of the database from the dialplan using DBGet(). If this does not meet your needs you could try some of the other command to see if they do provide the information you are looking for. For example, the CLI command sip show peers. Information such as that from the sip show peers command is not available (so far as I'm aware) from the dialplan. However, if one or more CLI commands do provide the information you are looking for and you are prepared to write a little code, you can use the Manager API to call these commands and write the status into the registry database or into variables. It would be inefficient to have to run a manager command periodically. Fortunately, Asterisk provides some event information to processes listening on the Manager API port. The events broadcast include register/unregister actions. By monitoring the events your code can respond to the events of interest and, for example, update status variables dynamically. For more information about the Manager API see: http://www.voip-info.org/tiki-index.php?page=Asterisk%20manager%20API Bill Seddon Lyquidity Solutions Limited -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Pushor Sent: February 23, 2005 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Able to tell if phone is registered? Kevin, Thanks for the reply, and thats what I am doing now, but I would really like to know in advance if a particular phone has been registered before taking a certain action. I was wondering if I could do something with regexten, but seemed like a messy solution. Is there any other way? I don't mind getting my hands dirty with perl or C, but I wouldn't really know where to start ... Thanks, Tim Kevin P. Fleming wrote: Tim Pushor wrote: I have a new asterisk setup running at home and am very happy with it. One thing that I am trying to do is to take various actions in the dialplan *if* a particular phone is registered/authenticated/connected. For example, if someone dials *me* and is shows that I am connected via my softphone, to try it instead of my deskphone (and possibly notifiy the user in advance that it is taking that action). Just go ahead and try to send the call there... it will fail immediately if it is not registered, with a DIALSTATUS of CHANUNAVAIL. If you are using CVS HEAD, you can use regexten to accomplish something similar, but I wouldn't recommend it as it's about to change :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial (Local/.....)
Mondial Software Limited 020 7043 2795 www.mondialsoftware.com Click here to view our presentation of Cash Controller showing its forecasting and automated bank reconciliation features -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: February 16, 2005 12:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dial (Local/.) On Wed, 16 Feb 2005 03:40:38 +0330, mohammad [EMAIL PROTECTED] wrote: I saw several examples of Dial app with the format: Dial(Local/..) Anybody knows what the Local technology means? Did you try the WiKi? Or Google? http://www.google.com/search?q=asterisk+local -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] n priority
Oh, that's cool. I presume it will only assign the priorities monotonically increasing from the last assigned priority? Are there docs anywhere on this? I checked the archives and voip-info.org... I guess the changelog in CVS might have it, eh? Yes, it really is cool. n does increase priorities monotonically but you can do two things that make creating extension logic a lot easier. The first is that you can set labels: exten = s,n(Start),Answer As well as making the dialplan more readable, labels can be the target of gotos: exten = s,n,Goto(Start) The second feature that makes using priorities easier is that arbitrary increments can be defined: exten = s,n+2,Dial(...) Maybe its utility is not so obvious but inconjunction with labels it can be pretty handy. One of the things that needs to be done often is handle the +101 priority to handle the failure condition of an application like Dial(). Without n, this is a pain in the neck because not only must the Dial() priority change, so must the corresponding +101 priority. With n priority increments, the following expression is possible: exten = s,n(MainDial),Dial(...) ; Dial the main numbers for this context ... ... exten = s,MainDial+101,Voicemail(u100) Now when new dial plan instructions are added before (MainDial) there is no need to update any priorities. By the way, the +101 priority can also takes a label. I'm not sure what practical value it has but it does help make the dialplan more readable: exten = s,MainDial+101(MainDialNotAnswered),Voicemail(u100) Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: February 08, 2005 12:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] n priority On Mon, Feb 07, 2005 at 09:04:25PM -0700, Kevin P. Fleming wrote: What are the n priorities in the above? I thought the priorities had to be explicitly set on each exten line... That's a feature of CVS HEAD... it allows Asterisk to compute the priorities for you. Some of us have been using it so long we've forgotten how to do it the old (hard) way :-) Oh, that's cool. I presume it will only assign the priorities monotonically increasing from the last assigned priority? Are there docs anywhere on this? I checked the archives and voip-info.org... I guess the changelog in CVS might have it, eh? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Manager API
Has anyone tried the API before? Of course. Here is a snippet taken from Nicolas Gudino's op_server.pl to originate a call. my $comando = Action: Originate\r\n; $comando .= Channel: $originate\r\n; $comando .= Exten: $canal\r\n; $comando .= Context: $meetme_context\r\n; $comando .= Priority: 1\r\n; $comando .= \r\n; send_command_to_manager($comando); Mondial Software Limited 020 7043 2795 www.mondialsoftware.com Click here to view our presentation of Cash Controller showing its forecasting and automated bank reconciliation features -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: February 04, 2005 7:41 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Manager API Hi, I try to use Manager API to originate a call from a channel to an existing extension. Based on sip channel show command, the Manager initiates a call to the channel only. It doesn't generate a call to the extension. So the originate call API of Manager is failed. I think I pretty much follow the API description at http://www.voip- info.org/wiki-Asterisk+Manager+API+Action+Originate. Has anyone tried the API before? Thanks. Jason -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outlook Integration
would it use the telnet utility or some sort of .call file thing? I can't speak for Matt but ours uses the Manager API and issues an originate command action. I believe that FOP uses the same mechanism. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Adams Sent: February 02, 2005 5:35 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Outlook Integration Are you going to be making this one available to all. I am not sure if or how it is possible, but maybe you would be able to have it so that if you right click on the contact, it has an option to iniate a call from there. If I may ask, trying to think how the thing you are making will interact with my asterisk server, would it use the telnet utility or some sort of .call file thing? Dan On Wed, 2 Feb 2005, Matt Riddell wrote: Dan Adams wrote: Is this think you have written something that the others of os can use/help with? I'm working on one at the moment too. But with mine it's a separate program, but you can drag and drop contacts from outlook. Coming soon! -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outlook Integration
Only works with their system, but there is no modifications to Asterisk for it to run? We've also written a system tray based call monitor that can call out using information from Outlook (or other sources). Our application uses the Manager API to receive from and send commands to Asterisk as does the Nicolás Gudiños FOP - no change to Asterisk required. I imagine PBXTray also uses the Manager API. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: February 01, 2005 9:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Outlook Integration Unfortunately the PBXtray app only works with our systems, and we cannot sell it separately. It is not released under the GPL because there are no modifications to Asterisk or any related software for it to run. Let me get this straight... Only works with their system, but there is no modifications to Asterisk for it to run ? I don't get this one ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on MS Virtual Server
Were the pops/drops/buzzes a problem only with communications via a telephony card? We ran Asterisk in a VPC instance (Redhat 8.0) for 3 months while we evaluated Asterisk. The only reason we had to move to a version of Linux running directly on hardware was a need to run X101P cards. We had no sound issues except when the host machine was printing. The host ran (still does) Windows 2000 Server, has an AMD Duron processor and 1GB RAM. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Courtnage Sent: January 30, 2005 6:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk on MS Virtual Server Hi Greg, On Sun, 2005-30-01 at 09:42 -0500, Greg Boehnlein wrote: It works, but you will have timing issues and very poor audio quality. I've run linux both under Vmware as well as running it under CoLinux directly on windows w/ no emulation neccessary. All emulation / virtualization layers have a problem whereby they are not able to keep up with the interrupt frequency that good quality audio requires, so you will hear pops, drops and buzzes in your conversation. Were the pops/drops/buzzes a problem only with communications via a telephony card? I'm curious to know if one could run * in a vmware session (only VoIP trunks), and if a telephony card is required, run another instance of * (and the card drivers) on the host OS. Thanks Ryan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FWD and IAX2
Any suggestions? Anyone got something similar to work? Mondial Software Limited 020 7043 2795 www.mondialsoftware.com Click here to view our presentation of Cash Controller showing its forecasting and automated bank reconciliation features -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chamberland-Larose, Guillaume Sent: January 28, 2005 7:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] FWD and IAX2 Hi, I had a FWD account set up with asterisk (using SIP) and it was working fine both ways. I switched to IAX2 and now I can't get incoming calls from FWD. People who call my FWD number get a 480 - user is not online message without any traffic reaching my box. I can call FWD numbers fine over IAX2. It seems fwd isn't trying to place the call over IAX2 because it thinks I'm not online. *CLI iax2 show registry Host UsernamePerceived Refresh State 65.39.205.121:4569xxx xxx.xxx.xxx.xxx:4569 60 Registered It looks like I'm registered though, and I can even call my own number fine. Other can't. :s Any suggestions? Anyone got something similar to work? Thanks, Guills ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA-2000
I also have a SPA-2000 and it works just fine. However there a many options available in admin mode, mainly on the Line 1/2 tabs and I've no idea what many of then do. Is there a document somewhere that describes them? The quick start guide that comes with the device only covers the basics and the documentation on the Sipura web site is the same document. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luki Sent: January 22, 2005 8:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-2000 Any links on how to get a sipura 2000 to connect to asterisk remotely over the internet ? In my experience, getting the Sipuras (1000, 2000 or 3000) to connect to * over the Internet is a piece of cake -- just make sure you have the NAT related options turned on, if your SPA is behind a NAT. 1) NAT support parameters on the SIP tab 2) NAT Mapping Enable on the Line 1/2 tab (I never needed NAT keep alive) Since you can type in a host name for the server, having a dynamic IP for * is not a problem, just sign up for a free DynDNS service somewhere. The bigger questions are: 1) is your * also behind a NAT? 2) Behind a firewall? 3) Does * accept connection packets from the outside (both SIP and RTP) Getting your * configured so it accepts outside connections is key; one you get your * setup working, configuring a SPA isn't all that bad. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] controlling recording
John We have no experience of providing a tool like the one you describe, but I'd guess that AGI using sox/soxmix to edit recorded files should be able to get you where you want. The Trim options of soxmix provides the ability to, for example, take the first x.y seconds of a recorded file so that you can splice a file in accordance with a users instructions. A second or any subsequent part of the recording can then be made and concatenated to the first. We use sox and soxmix in similar ways to allow some of our voice recordings to be replaced. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Hammen Sent: January 21, 2005 1:25 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] controlling recording Thanks for your quick replies - I had no idea this list was so responsive! One more issue for you - we are trying to build a custom voicemail application and so far everything works great by using AGI to play prompts, get input and record files. However one of our customers has asked for some pretty advanced functionality, specifically the ability to: * stop in the middle of recording (so far so good, escape character) * fast forward/rewind the recording as it is so far (!) * overdub (i.e. rerecord) starting from a desired point in the existing sound file (!!!) So, it's pretty clear that AGI's RECORD FILE function as it stands is not going to cut it, so I'm assuming that do anything like this means writing a new custom app_something.c, does that sound right? Anybody have any tips or comments on how ridiculously difficult this might be? Thanks again, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] failed to compile zaptel on redhat
then it looks like the does not have the full kernel sources installed ...or isn't running the 2.6 kernel. I had the same problem with the CVS on Friday (but not from the week before). It turns out that moduleparam.h is included as part of a bug fix on 2.6 but instead of being #ifdef'd for 2.6 and later, the inclusion was absolute causing compilations of zaptel on earlier Linux kernels to fail. The advice I received was: The culprit is bugfix #3334, it is supposed to fix a 2.6 kernel issue but ended up messing up Zaptel on 2.4. I have edited: pciradio.c tor2.c torisa.c wcfxo.c wct1xxp.c wct4xxp.c wctdm.c wcte11xp.c zaptel.c ztdummy.c ztdynamic.c and changed: #include linux/moduleparam.h to: #ifdef LINUX26 #include linux/moduleparam.h #endif Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Goddard Sent: January 16, 2005 9:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] failed to compile zaptel on redhat On Sunday 16 January 2005 04:29, Steven Critchfield wrote: linux/moduleparam.h is actually part of the kernel source. It is created when you config and compile the kernel. It holds the version symbols needed to properly link the new drivers into the kernel. No, it is part of the virgin kernel sources and defines the kernel modules parameters api. If he does not have it, then it looks like he does not have the full kernel sources installed. I suggest you find a kernel compile howto that is at least understanding of anything specific to the brokenness of RedHat and follow the suggestions found within. [...] Xu, Duo wrote: why linux/moduleparam.h is missing in the source? I saw it in 2.6 source. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel in HEAD broken?
Soren, thanks for the information and advice. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Soren Rathje Sent: January 14, 2005 1:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaptel in HEAD broken? Bill Seddon wrote: Are there new instructions for compiling the zaptel driver in HEAD? I compiled the zaptel driver from HEAD successfully last weekend but trying to compile the current driver for another machine results in the error: zaptel.c:45:31: linux/moduleparam.h: No such file or directory If I go to compile zaptel on the machine that compiled successfully last weekend, the same error occurs. So far as I can tell, I don't have a file called moduleparam.h anywhere on either machine. Yeah, the culprit is bugfix #3334, it is supposed to fix a 2.6 kernel issue but ended up messing up Zaptel on 2.4. I have edited: pciradio.c tor2.c torisa.c wcfxo.c wct1xxp.c wct4xxp.c wctdm.c wcte11xp.c zaptel.c ztdummy.c ztdynamic.c and changed: #include linux/moduleparam.h to: #ifdef LINUX26 #include linux/moduleparam.h #endif and now it compiles on 2.4... /Soren ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel in HEAD broken?
Are there new instructions for compiling the zaptel driver in HEAD? I compiled the zaptel driver from HEAD successfully last weekend but trying to compile the current driver for another machine results in the error: zaptel.c:45:31: linux/moduleparam.h: No such file or directory If I go to compile zaptel on the machine that compiled successfully last weekend, the same error occurs. So far as I can tell, I don't have a file called moduleparam.h anywhere on either machine. Bill Seddon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Manager API !!!!!!!!!
Has anyone had any success with the Manager API ?
Yes, lots of success. I think we found that tracking the status of an
extension is the most reliable way to monitor extension state. We do issue
a PeerStatus request at the beginning but that's pretty time and resource
intensive. Since the Manager is going issue status events anyway, we
monitor them and save the resources.
The second is anyone know what ActionID is ?
Yes, ActionID is a value you can use when issuing a command. It there so
that you can be sure you respond to your own responses not to someone else's
or that you respond to an response instance in the correct way. In a
multi-threaded app you might have several actions outstanding so you will
need to know what response corresponds to which command.
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon
Sent: January 13, 2005 4:56 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Manager API !
Hello all
Has anyone had any success with the Manager API ?
I am trying to check an extension status without too much luck I have
the following
?php
$fp = fsockopen(127.0.0.1, 5038, $errno, $errstr, 30);
if (!$fp) {
echo $errstr ($errno)br /\n;
} else {
$out = Action: Login\r\n;
$out .= UserName: simon\r\n;
$out .= Secret: xx\r\n\r\n;
fwrite($fp, $out);
$in = Action: ExtensionState\r\n;
$in .= Exten: 4367\r\n\r\n;
$in .= Context: office\r\n\r\n;
$in .= ActionID: \r\n\r\n;
$in .= Action: Logoff\r\n\r\n;
fwrite($fp,$in);
while (!feof($fp)) {
echo fgets($fp, 128);
}
fclose($fp);
}
?
The first prob is it ignores the Context and just goes to default.
The second is anyone know what ActionID is ?
Best Regards
Simon
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RE: [Asterisk-Users] asterisk won't release line
If you'd like to make a call, please hangup and try again, which is preceded by the fast busy tone. I have the same experience. I believe it is because the telco (Telewest in my case) does not use or provide Disconnect Supervision. Not being a telecoms engineer and as I dont know one I can cant test the veracity of this belief. I have called Telewest to ask them to enable disconnect supervision but no one seemed to know anything about it. Bill Seddon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Knight Sent: January 13, 2005 6:12 PM To: asterisk Subject: [Asterisk-Users] asterisk won't release line Hi all, I have an issue with my system and the phone company (Alltel). If someone goes into any mailbox on the server to leave a message, once they hang up on their side the system is not detecting this and will sit there and record until you get the standard message from the phone company If you'd like to make a call, please hangup and try again, which is preceded by the fast busy tone. Since I never had this problem with any of my equipment in another state, I feel it's either the wiring in the building OR it is a problem in the phone companies equipment. Any hints or suggestions would be greatly appreciated. As a side note I've tried this with a simply answering machine and it acts the same way. Thanks, Ken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Has there been a change to t functionality in HEAD
Over the weekend we moved our Asterisk upto HEAD so that we can take advantage of the new n priority and associated labels in the dialplan (fantastic). However I think there is a change to the way t option (in the Dial command) works. Before reporting it I thought I'd see if anyone else has noticed it. The use of t is supposed to allow the callee to transfer a call. On Friday (before upgrade) if worked fine. After upgrade pressing the # key does nothing. However put the caller on hold, get the caller back and *then* press the # key and transfer works. A quick check shows that the T option (allows the caller to transfer) continues to work OK. Also, without the t the callee is unable to transfer at all. So it appear that in HEAD the t and T options are being recognised by Asterisk but that t is not usable until the callee presses flash. If you have any experience with transfer when using HEAD I'd welcome your feedback. Bill Seddon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Execute dialplan command at startup
How can Asterisk be configured to execute some number of dialplan commands when it is started or restarted? I want to be able to populate the registry (using DBPut() commands) to store some information each time Asterisk starts. Such information could, of course, be stored in a database and perhaps that will be the long term objective. In the meantime I'm hoping that it is possible to use the built-in database and be able to run some kind of autostart context. Does such a facility exist? Bill Seddon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Setup Documentation
Title: Message An alternative is http://www.asteriskdocs.org/ . Its only getting going but there is some getting started stuff. The challenge of the wiki is that its relatively fragmented and assumes a level of knowledge. I found the wiki much more useful when I had a running system and wanted to find expert ideas on how to achieve a specific objective. Regards Bill Seddon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Brock Sent: January 10, 2005 3:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk Setup Documentation Phil, http://www.voip-info.org/wiki-Asterisk is a good place to start, and will point you to most resources. Paul From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Menico Sent: 10 January 2005 15:20 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Asterisk Setup Documentation Hello all: Can anyone help mewith finding the best locations for getting setup and other documentation for *. Thank you. Phil Menico www.xtend.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Execute dialplan command at startup
I tried it, and it works, but it is hardly an ideal solution I've setup the following file: Channel: SIP/1003 Context: from-sip-internal Extension: 338 Priority: 1 And it does almost exactly what I want so Oliver/Niksa, thanks for the idea. However the call file mechanism seems to insist upon a channel which implies a phone has to ring before the extension (containing the dialplan actions I want to execute) is called. Is it right that the channel must be included in the call file? If so, is there a dummy channel available that will always answer. I've had no joy getting the application option to work but it, too, seems to require a channel parameter to be present in the file. Thanks Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Niksa Baldun Sent: January 10, 2005 6:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Execute dialplan command at startup I tried it, and it works, but it is hardly an ideal solution. If you, for example, forget yourself and execute 'restart now' from * console, .call file won't be generated and you may spend hours wondering what went wrong. I believe the lack of some kind of 'autostart' context is a major flaw in design, but then again, I guess it could easily be implemented. Peer Oliver Schmidt wrote: Bill Seddon wrote: How can Asterisk be configured to execute some number of dialplan commands when it is started or restarted? [..] In the meantime I'm hoping that it is possible to use the built-in database and be able to run some kind of autostart context. Does such a facility exist? Without getting into details, I would create a call file in the outgoing spool directory of asterisk within the asterisk startup script which calls a specific application. Haven't tried, but should easily work. Let me know, how it works. rgds pos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P random hangups - Please help with suggestions
Both of the X100Ps seem to randomly hang-up both incoming and outgoing calls. I think most people who use X100P cards (clone or originals) have had your experience. So far as I can tell, the cause is always an interrupt problem. Specifically that affected X100P cards share an interrupt with one or more other devices. Have you checked for shared interrupts using the command: cat /proc/interrupts to see if any interrupts are shared? In my case we have 3 X100P. At first hang-ups occurred all the time on all cards. A quick review showed that all three cards were sharing interrupts: one with eth0 device and the other two with each other. After some trial and error I now have a workable system. Two of the cards work fine however one, it seems, must share with eth0 and continues to hangup. The rules seem to be: 1) If at all possible, try to arrange to have the X100P cards on their own interrupt. 2) If this is not possible, have them share with devices that are not used or don't generate many interrupts. For example one of my cards shares with an unused USB device and this seems to be fine. Finally, choose a motherboard that offers flexibility in the use of interrupts. Being a thrifty sort, I bought the cheapest mother board I could. It appears that one of the compromises the board designers made has been to share three interrupts across 5 PCI *and* the built-in eth0 device. As a consequence it is hard (impossible) to arrange use of the slots so that there is no interrupt sharing. By contrast, a friend is reusing an old HP machine and each PCI slot (and so each X100P) seems to have its own interrupt. In any case there are no problems - presumably because there are no interrupt conflicts. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vassilis Konstantinou Sent: January 09, 2005 9:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] X100P random hangups - Please help with suggestions This one is driving me crazy. So any suggestions will be very welcome. My setup: Suse Linux 9.0 (Pentium 4, 1GB) Asterisk: current stable (1.0.3?) - tried the head CVS before Christmas but did not fix it 2 X100P clones - one for a UK BT line, one connected to an ATA186 configured for a UK BT Broabband-Voice service (MGCP) 1 ATA186 (SIP) connected to two dect internal phones (configured as extensions 5000-5001) The problem: Both of the X100Ps seem to randomly hang-up both incoming and outgoing calls. There is no fixed dureation but it always happens. Sometimes as soon as a call is answered and sometimes at any point up to 10-15 minutes. All calls through my true VOIP lines (I use sipcall in UK and fwd) are fine and never disconnect during the call. The X100Ps seem to detect the real hangup properly (of course). Things I have tried: 1) The latest CVS (up to early December). No change 2) The current stable. No change 3) Playing with the rxgain in the zapata.conf file (no change) 4) Using the Loopstart instead of Kewlstart. No false hangups here BUT as expected lots of line noise. Is this a good clue to what is happening? Are there any parameters I can tweak to make the Kewlstart driver a bit more reliable? Please help. This is driving me (and the people using the system crazy). Vassilis My current zapata.conf is attached below: ;; ; Zapata telephony interface ; ; Configuration file [channels] ; ; Default language ; group = 1 language=en ; ; Default context ; context=incoming switchtype=national usedistinctiveringdetection=no useincomingcalleridonzaptransfer=yes rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=no callwaitingcallerid=no threewaycalling=no transfer=yes cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes rxgain=0.0 txgain=1.5 ; ; Specify whether the channel should be answered immediately or ; if the simple switch should provide dialtone, read digits, etc. ; immediate=no ;musiconhold=default callprogress=no progzone=uk ; usecallerid = yes ; we want Caller*ID support cidsignalling = v23 ; UK (BT) Caller*ID uses the V.23 std cidstart = history ; use the Zaptel history (X100P) busydetect=no signalling=fxs_ks channel = 1 ;BT Broadband Voice - Uses US ID and busysignal on Hangup busydetect=yes busycount=6 cidstart = ring ; ring starts Caller*ID cidsignalling = bell ; Cid US signalling=fxs_ks channel = 2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo
RE: [Asterisk-Users] New 'n' priority
Thanks for your replies. I'd read somewhere that the n priority was after
1.0 but I now understand that this means after 1.0 and subsequent stable
patches. Thanks again and I shall download HEAD.
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Fineberg
Sent: January 08, 2005 2:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] New 'n' priority
Christopher L. Wade wrote:
Bill Seddon wrote:
Can anyone point me to documentation covering the new 'n' priority.
I've
downloaded and have working v1-0-2. But when I attempt to use the n
priority - for example:
exten = s,1(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten =
s,n(dial),Dial(SIP/1001)
Asterisk generates errors containing the text pbx_load_module: Invalid
priority.
Can anyone help or show me what I'm doing wrong?
Bill Seddon
'n' and 's' as well as labels only work in CVS HEAD, its not in any of
the 'stable' versions.
-Chris
Well I'm using v1.0.3 and previously v1.0.2 and they certainly know
about 's'. Haven't tried 'n' yet.
Adam
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RE: [Asterisk-Users] Channel Variable
Does anyone know how to get the channel ID on the
other side of the call?
Assaf, I dont know if there is such
an ID available. However if there is not, the value you want is pushed
out in one of the events that Asterisk publishes to AGI connections when a call
is constructed. As it result it ought to be possible to write an AGI
script using, say, Perl to capture this value and write it back as a Dialplan
variable.
Bill Seddon
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Assaf Benharoosh
Sent: January 08, 2005 12:27 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Channel
Variable
Hi all,
Does anyone know how to get the channel ID on the other side
of the call?
For example: When SIP/50 calls SIP/21, and the call is
answered by SIP/21 I get:
SIP/21-6735 answered SIP/50-b456
${CHANNEL} will show me SIP/50-b456.
Is there a parameter or a workaround to get the SIP/21-6735
part?
Thanks.
Assaf Benharoosh
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RE: [Asterisk-Users] virtual pbx
Is it possible to set asterisk up as a virtual pbx like in apache and virtual host? You have provisioning that may address some or all of your needs. Mark Spencer talks about other possible deployment options in response to questions at the end of this presentation... http://graphics.cs.uni-sb.de/VCORE/Publications/mark_spencer/mark.smil Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres E. Moya Sent: January 08, 2005 4:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] virtual pbx Is it possible to set asterisk up as a virtual pbx like in apache and virtual host? If so can someone point me to the right direction. I would also like to setup asterisk with some type of redundancy, I have searched the lists and googled but havent really found anything, I would be willing to put together a paper if I had the info and make it available, through asterisk doc project or voip-info.org or which ever. Any help would be greatly appreciated. Thanks in advance, Andres Moya ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT help with rmdir pls
rmdir will only remove empty folders and --ignore just prevents error messages being displayed. Run the command: rm -rf * in the asterisk root folder and then execute rmdir Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Middleton Sent: January 08, 2005 11:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OT help with rmdir pls I am tying to clear down an asterisk source directory before CVS'ing a new version the --ignore... option is being used but its still not being deleted, can anyone give me some clues. Sorry I'm new to Linux, as if you havent guessed. Googling hasnt helped so far Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ringing an extension on multiple phones
You can Dial() extension SIP/line1SIP/line2 Yes, and if the multiple extensions that ring are members of the same group then any one of the phones can pickup the call. So the next question is: how does the receptionist put the system into group ring mode. The answer is to have the receptionist call a nominated number such as **221 (enable group ringing) and **222 (to disable group ringing). When the receptionist calls **221 a global variable (or an entry in the registry is created) is made to contain a value that indicates group ringing is in effect. When **222 is called, calls ring on the operator extension. We use a similar approach to have support calls forwarded to mobile phones out of office hours. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Listas Sent: January 07, 2005 6:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Ringing an extension on multiple phones You can Dial() extension SIP/line1SIP/line2 even more you can and that will call both extensions only after a 5 seconds timeout exten = xxx,1,Dial(SIP/line1,5) exten = xxx,2,Dial(SIP/line1SIP/line2,10) etc... that's if I understood what ou needed... bye, M. - Original Message - From: Scott Henderson [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 1:45 PM Subject: [Asterisk-Users] Ringing an extension on multiple phones I am using Cisco 7960 phones and have had a request to have the receptionist phone ring on multiple phones just in case she is not around. Call pickup is the theory here but the issue is that not all the people that need to hear the ring would here the receptionist phone ring so I think I need to have a second line appearance on the phones in question so that line will ring. Can this be done or is there a better way. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New 'n' priority
Can anyone point me to documentation covering the new 'n' priority. I've
downloaded and have working v1-0-2. But when I attempt to use the n
priority - for example:
exten = s,1(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n(dial),Dial(SIP/1001)
Asterisk generates errors containing the text pbx_load_module: Invalid
priority.
Can anyone help or show me what I'm doing wrong?
Bill Seddon
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RE: [Asterisk-Users] voiptalk.org IAX service - user experiences
Paul wrote: Having all sorts of nightmares getting IAX working from voiptalk.org Originally I did too but it was all my fault. We have been using VoIP Talk for about 3 months and have no complaints. Getting outbound IAX (from PBX to PSTN via VoIPTalk) is straight forward the guide on their web site is accurate. Provided you have bought IAX credits you should be able to use IAX successfully. To receive calls via IAX you must have a number (either a free 0870 number or a paid for geographical). You must also ask VoIPTalk support to add the IP address of your * server to the telephone number. This IP address can be the address of your gateway if you are using NAT. Bill Seddon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Record() problem
Kobus We have a similar requirement and our solution has been to minimise our dependence upon hardware wherever possible. We have two PSTN lines that are serviced by * and a pair of X101P cards but these are for emergency use, for example if our broadband connection were to be unavailable. For our main telephony needs we use www.voiptalk.com as a VoIP provider that supports IAX (by making sure we use IAX we know there will be no SIP/RPT/NAT issues). We get the benefit of fxo technology without having to buy and support it. We can make up to 25 outgoing calls and receive up to 5 calls simultaneously. Clearly not sustainable if the company were to grow significantly but effective and great value for money now and the immediate future. Out other investment has been in HandyTone ATAs into which we plug DECT phones so that they be used anywhere within the office. So far it has proved remarkably successfully. By the way, we started using Libretel and FWD but the FWD element has proved too unreliable. To be fair, they say that their IAX service is experimental. Moving to a commercial IAX provider has made a big difference. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kobus Wolvaardt Sent: December 24, 2004 4:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Record() problem Hi all, I want to install 4 to 6 lines for our new office and hook it into an * system. I want as little trouble as possible, what fxo hardware do you recommend? I see that poeple on the list are complaining about digium tdm400 cards...? Are grandstream phones stable and easy to setup? Any problems with *? Thanks, Kobus Wolvaardt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Record() problem
You syntax for the command is incorrect. See http://www.voip-info.org/wiki-Asterisk+cmd+record. Record is an application to be executed from within the dialplan. So the channel it will record is implicit and cannot be explicitly stated as one of the parameters. If you want to originate and record a call automatically, you will have to do this via AGI. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Me Sent: December 24, 2004 6:38 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Record() problem Any idea why this: Record(IAX2/[EMAIL PROTECTED]/5, /tmp/whatever.gsm|6|25) Would result in this: WARNING[3293201]: app_record.c:117 record_exec: No extension found Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] My Boss wants background music!!!!
On Fri, Dec 17, 2004 at 07:54:19AM +0100, Wilson Pickett wrote: I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. This is done in a conventional PBX that he wants, but I can use the Asterisk PBX if it can How about a desktop application that plays streamed music from a server but cuts out when the respective user's phone rings? Detecting the ringing state of a specific device from, say, a desktop running Windows or Linux AGI is trivial. Since playing streamed music on Linux or Windows is also trivial there should be no problem meeting your requirements. Bill Seddon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] My Boss wants background music!!!!
Care to share a trivial example with us? Sure. Here's a how-to. If you'd like a working example, let me know. Open a TCP connection to * either directly of via, say the sampleproxy.pl example (http://www.popvox.com/simpleproxy.pl). Login using the manager API by sending the text string on the connection: Action: login\n\n After this, that trick is to monitor events from *. Asterisk will send events to all manager connections. Events will arrive as text on the connection in the form: Event: some event type\nsome event text\n\n The event text will contain event type specific information in the format: field: value\n For example: Event: Newstate Channel: SIP\1002-1a23 You can monitor Newstate events to review the state of a call to an extension (channel). For example it might be ringing or dialled. * will send a Event: Link event when a call is answered and an Event: Unlink event when the call is terminated. Eventually * will send Event: Down indicating that the call channel is on-hook. You can download an application that illustrates this functionality and is written in C# from: http://www.yottadot.org/download.php?op=viewsdownloadsid=10 There examples of playing streamed music in C# such as: http://www.codeproject.com/cs/media/nbass.asp Bill Seddon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.0.1 Too many open files
There should be no reason that I hit my limit of open files on this machine. Restarting asterisk immediately solved the problem, so I'm leaning towards a leak, however, I didn't have the opportunity, in the moment, to check and see how many files and what type were open. This is a problem that I've also run into but I know it cannot be anything to with files system limits - at least in the obvious sense. So I'll describe my experience and it might help someone identify the root cause. When I set up my Asterisk box, it was inside our firewall. With the system working, I moved it outside the firewall but SIP phones would not connect and the console reported the chan_sip errors described. I was able to use TELNET, VNC and could call the PBX using IAX (FWD). A few days later and without, apparently, doing anything, the PBX began to work. Last weekend the errors chan_sip errors started again. I rebooted the Asterisk server. I rebooted the firewall and router. No change. So I brought the server back inside the firewall and it works just fine. Go figure. So the error cannot be anything to do with open file limits. The difference appears to be related to the site of PBX server with respect to the Bill Seddon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Kind of off-topic: VoIP services and multiplecallers
I guess it is going to depend upon your provider. Our providers do support multiple calls. We use one provider for inbound calls and one provider for outbound calls. We can have up to five simultaneous inbound calls (why five?) and up to 25 simultaneous outbound calls. We can't even begin to test these limits so I can't confirm them. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanwar Ranbir Sandhu Sent: December 06, 2004 8:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Kind of off-topic: VoIP services and multiplecallers Hello Everyone, I've been running Asterisk as our PBX for several months now, and recently I've been thinking about using one of the VoIP providers to lower our phone bill. I know that VoIP providers can supply their customers with a local number and/or virtual numbers, and then that number/account can be used with Asterisk (well, it depends on the provider and whether or not their service is compatible with Asterisk). However, I have a question: can more than one person make/receive a call at the same using one VoIP line? In the analog world this obviously isn't possible, but with VoIP, it's just bandwidth: the more callers, the more bandwidth. I keep saying to myself it should be possible, but the VoIP providers don't provide any information about this. If five people in the office all need to use their phones at the same time, would I need five VoIP lines, or would I only need one VoIP line? Am I over-thinking this? Thanks in advance, Ranbir -- Kanwar Ranbir Sandhu Systems Aligned Inc. www.systemsaligned.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK available SIP phone?
Mike I use Asterisk at home and have bought a couple of HandyTones ATAs. The DECT phones are plugged in and work really well. The ATAs are £56 from Goods2World (though the additional one I've just bought didn't work and is being returned) and about the same from VoIPTalk (who are out of stock currently) The only downside to the ATA is that I've not yet worked out how to have CallerID displayed on the DECT phones. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Dent Sent: November 21, 2004 9:39 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] UK available SIP phone? Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Top posting
I don't recall Tom asking you to adhere to his rules No, he didn't. But I'd guess that readers of his email who are, perhaps, less sophisticated users of email lists will have been concerned about the attitude he projected and be unnecessarily anxious about posting questions in the wrong way. Since there is no wrong way I just wanted to try to project a more flexible attitude. This list has been and continues to be enormously helpful to me and it would be a shame that anyone was put off learning about Asterisk because they believe there is some need to post questions (or there own experiences) in some particular and mysterious way. As you say, those who demand a specific format (it seems an overly pompous requirement to me) do not have to answer. But others will nonetheless. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: November 11, 2004 11:51 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Top posting On November 11, 2004 05:17 am, Bill Seddon wrote: Oh, that's a great idea, Tom. Let's have everyone operate to your exacting standards. I can appreciate that not everyone did their degree in mail list etiquette and have lives to live and so want to be economical with their time. I don't recall Tom asking you to adhere to his rules; he gave what his conditions were for his help. My conditions are very similar. If you aren't going to make half an effort to give me the information I need to help you in a manner which makes it easy for me to help you, I ain't gonna try. You haven't paid anything for my time so I will dole it out any way that suits me. Capiche? So for my part I scan emails top, bottom or otherwise posted and reply if I think I have a contribution to make or something to learn (in my experience knowledgeable people are often extremely busy and brief). Correct. You seem to have more time to help, or more of a willingness to expend energy to help than Tom. If so, good. If not, no matter, it's your time anyway. :-) Clearly if something has become illegible or doesn't include relevant information, it's not going to garner any attention or convey any useful information. But in my experience most posts on this list are good enough and some tolerance goes a long way. Personally I will top-post if I am giving a message to the effect of Your solution was bang-on, my problem is gone, thanks for the help -- Otherwise I will put replies inline with the blocks of text that they go with. Either way, I trim any irrelevant bits. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Top posting
Oh, that's a great idea, Tom. Let's have everyone operate to your exacting standards. I can appreciate that not everyone did their degree in mail list etiquette and have lives to live and so want to be economical with their time. So for my part I scan emails top, bottom or otherwise posted and reply if I think I have a contribution to make or something to learn (in my experience knowledgeable people are often extremely busy and brief). Clearly if something has become illegible or doesn't include relevant information, it's not going to garner any attention or convey any useful information. But in my experience most posts on this list are good enough and some tolerance goes a long way. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Ivar Helbekkmo Sent: November 11, 2004 9:38 AM To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Top posting George Gardiner [EMAIL PROTECTED] writes: So that I can understand the almost religious fervour on this point could someone please explain to me why top posting is so hated!! Because there's such an enormous amount of communication one would like to take part in, and not enough time. The easier it is to quickly discover a) whether each item is interesting, and b) what is the exact context of the item, and of its constituent parts, the more interesting material we can actually read. Therefore, top posting and bottom posting are equally bad; the ideal is an easily readable text that's placed into its proper context by short quotes of the relevant bits of previous communication. (Note: *short* quotes. If the reader wants the full text of the previous message, retrieving that message takes but a moment, so there's no need to quote it all.) For my own part, I have taken to ignoring anything that is badly formatted, top posted, bottom posted, or otherwise makes it difficult to quickly get into the flow of the communication. My default is to move on; only if your posting quickly establishes that it is, in fact, interesting to me, will I read it. To put it bluntly: if you can't be bothered to make an effort to communicate, what you say can't be very important. ;-) -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: ZapTel problems
Isn't it supposed to be: Channel = 1 Just been through this exact same problem myself. If it is really the same, you should be able to get Asterisk working again by commenting out the channel = xxx lines from Zapata.conf. OK, zap channels will not work but you will know it maybe the same problem. If it is, make you are loading zaptel and wcfxo by using lsmod. My problem turned out to be that I was editing a copy of zaptel.conf in /etc/asterisk rather than the correct one in /etc If zaptel and wxfxo is loaded, you should check that ztcfg is loading the channels correctly by running ztcfg -tvvv My guess is that you will see that there are no channels created by the config utility. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Hartmann Sent: November 04, 2004 8:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FW: ZapTel problems Good Day, Well I had the box up and running accepting an incoming pstn line. Then made a change to the zap conf file and now asterisk will not start 001 Had the box up and running 002 went to change the zapata.conf to include my pstn lines 003 and then rebooted and asterisk will not load. 004 005 Not sure if it is a co-incidence or not 006 007 008 my zapata.conf file 009 010 [channels] 011 language=en 012 013 context=from-pstn 014 signalling=fxs_ks 015 faxdetect=incoming 016 usecallerid=yes 017 echocancel=yes 018 echocancelwhenbridged=no 019 echotraining=800 020 group=0 021 channel=1 022 channel=2 023 024 context=default 025 signalling=fxo_ks 026 channel=3 027 channel=4 028 029 030 031 Zaptel Configuration 032 == 033 034 035 4 channels configured. 036 037 038 Nov 5 13:50:14 pc-11 kernel: Zapata Telephony Interface Registered on major 196 039 Nov 5 13:50:17 pc-11 kernel: Freshmaker version: 71 040 Nov 5 13:50:17 pc-11 kernel: Freshmaker passed register test 041 Nov 5 13:50:17 pc-11 kernel: Module 0: Installed -- AUTO FXS/DPO 042 Nov 5 13:50:17 pc-11 kernel: Module 1: Installed -- AUTO FXS/DPO 043 Nov 5 13:50:17 pc-11 kernel: Module 2: Installed -- AUTO FXO (FCC mode) 044 Nov 5 13:50:17 pc-11 kernel: Module 3: Installed -- AUTO FXO (FCC mode) 045 Nov 5 13:50:17 pc-11 kernel: Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules) 046 047 048 when i start asterisk -vc 049 [chan_phone.so] = (Linux Telephony API Support) 050 == Parsing '/etc/asterisk/phone.conf': Found 051 == Registered channel type 'Phone' (Standard Linux Telephony API Driver) 052 [chan_zap.so] = (Zapata Telephony w/PRI) 053 == Parsing '/etc/asterisk/zapata.conf': Found 054 == Unregistered channel type 'Tor' 055 == Unregistered channel type 'Zap' [EMAIL PROTECTED] asterisk]# Ouch ... error while writing audio data: : Broken pipe 057 058 059 Not sure where to go is this a zap issue or is it a co-incidence? 060 Any idea what I should have in my config file for a single digium 4 port card with 2 fxo and 2 fxs thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with own IP address
If a server running Asterisk has its own IP address while SIP phones are behind a NAT router on a different IP address, the phones should be able to contact and register with the Asterisk server, right? I'm sure the answer is yes, because I can register my SIP phones with other Asterisk installations. But since it's not working for me, and I gone to the trouble of getting a dedicated IP address for the Asterisk box, I thought I'd check. Bill Seddon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Suggestion re: SIP/NAT/*
Karl Are you saying it is nonsense that there difficulties using Asterisk and SIP behind a NAT server. Or are you saying it is nonsense that SIP and NAT are dangerous together? Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl Brose Sent: October 29, 2004 5:49 PM To: Benjamin on Asterisk Mailing Lists; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Suggestion re: SIP/NAT/* NONSENSE Benjamin on Asterisk Mailing Lists wrote: On Thu, 28 Oct 2004 14:45:46 -0600, Ryan Courtnage [EMAIL PROTECTED] wrote: Yep, you can do this, just requires some port forwarding and special considerations in sip.conf. You are missing the point. There is no *solution* to SIP NAT traversal. All there is are *workarounds*, otherwise known as bad and rather dangerous hacks. Whether it works or not is highly dependent on external factors that you don't usually control. It also depends on the type of NAT/PAT your router is using, ie the router's particular NAT/PAT implementation. The fact remains that SIP NAT traversal setups are highly insecure and unreliable. Consider this to be the equivalent of locking your apartment with duct tape. It may work for you, but you wouldn't recommend it to anyone else UNLESS you wish them harm. Now, this is valid for single NAT situations and it is even more valid for double NAT situations. If you want to do this properly without duct tape, then you will have the three choices I mentioned: - If you must use SIP, don't use NAT - If you must use NAT, use IAX - If you must use both SIP and NAT, build a tunnel Anything else is improper and unprofessional. rgds benjk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto-Login/Auto Answer
Following contributions from respondents on this list, we're using a handytone ATA and an entry phone handset (no dial pad) as a batphone. We've set the ATA autodial value to an extension and created an entry in both the sip.conf (for the handytone login/context specification) and extension.conf for the autodial extension. Not elegant but works well (for us). Hope this response is of some value Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolás Gudiño Sent: October 26, 2004 2:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Auto-Login/Auto Answer Hello, I think that I may have been confusing everything. I am looking for a batphone or hotline compatible ATA. I think this would work, anyone know of one? I am hoping that a Sipura 2000 will work but I can't figure out what dialplan would match if a user simply picks up the phone. Try the sipura faq http://www.sipura.com/Documents/faq/Section_2.html#5 -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] manager interface to barge
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nicolás Gudiño Sent: 20 October 2004 18:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] manager interface to barge Hello, On Wed, 20 Oct 2004 09:48:43 -0600, TELUX [EMAIL PROTECTED] wrote: Can the Manager interface be used to barge my phone into an existing conversation? You need to use manager redirect and meetme. Check out my Flash Operator Panel, it lets you barge on calls. http://www.asternic.org -- Nicolás Gudiño Buenos Aires - Argentina Thats interesting, can you explain a bit more how that is done. I would like to implement something simillar without using the Flash operator. Umar Hey, Umar It seems a bit cheeky asking the guy who wrote Flash Operator Panel how to get something done so you don't have to use it. I'm sure Nicolas will reply but it might be helpful to him to learn from you why FOP doesn't work for you. If its a feature thing, maybe it a feature he can add and we all win. Bill Seddon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with FWD?
I think I saw someone suggesting there are problems with FWD IAX registrations. In case Ed Guy or one of the other FWD people is watching the list, we're also having horrible problems receiving and sending calls via FWD. The symptom is that the registration will take forever to succeed and when it does is lost the next time a registration request is made. If a call is made in the few seconds after a registration succeeds, the call quality is extremely low. It's not a bandwidth or connectivity issue here because we can make calls via other IAX/SIP providers successfully. Bill Seddon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem building Zaptel from CVS v1-0
When trying to build the zaptel source from the v1-0 label, I get an error compiling the new file wcte11xp.c: parse error before spinlock_t This error doesn't occur when compiling the zaptel source from HEAD nor did it occur in the source from Sept 19th. Can anyone suggest the right course of action to take to be able to work with a stable version? For example v1-0-1 compiles just fine (doesn't include the new file) is this an OK stable version to use? I ask because the advice on the Asterisk download pages is to use v1-0. Thanks Bill Seddon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP peers in MySQL Database
I'm sure you've considered it, but having distributed asterisk services dependent upon one instance of SQL Server at remote location always being available seems a weak point in the design. If the SQL Server node is not available, all asterisk users will be affected. Have you considered using one master sql server instance with local msde instances (no license issues) and use replication services to ensure each slave copy is updated as needed? It may make for a more robust solution in a multi-node environment. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of harry gaillac Sent: October 13, 2004 10:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP peers in MySQL Database I don't try the perl script. here is what I expect from asterisk and sql database for example. one asterisk pbx per office, several offices,one sql server.I want to admin all sip conf offices from sql server I create one sip table per office on my database server. each pbx office get his sip conf from sql server. If i add or remove sip clients on my sql server how pbx office update his sip conf ? Harry --- Matthew Boehm [EMAIL PROTECTED] a écrit : How do you update many pbx ? crontab ? How often are you needing to update them? Hourly? Daily? I only have 1 * box so I currently use the perl script method on our prod server. I'm using the RealTime on our dev server. RealTime will deffinatly be easier once it has become stablized. You will be able to have multiple Sip tables in 1 database server that can handle multiple * machines. Be patient.. -Matthew Best regards Harry NB: everybody should be able to find a full documentation about Asterisk features not in mailing list. I look at voip-info. --- Matthew Boehm [EMAIL PROTECTED] a écrit : Yes you are wrong. You seem to be combining two different methods of getting SIP info out of a database. Pick 1. I use the perl script right now so here is how to do that: In order to use the perl script which can support 'ALL' sip abilities, use this table: CREATE TABLE sip_perl ( id INT(11) DEFAULT -1 NOT NULL, keyword VARCHAR(20) NOT NULL, data VARCHAR(50) NOT NULL, flags INT(1) DEFAULT 0 NOT NULL, PRIMARY KEY (id,keyword) ); Then, insert a new row for each sip parameter keeping the 'id' the same for each phone: INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'account', '3038', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'callerid', 'Cytel 2814494000', 1); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'nat', 'yes', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'context', 'cytel-internal', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'type', 'friend', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'mailbox', '[EMAIL PROTECTED]', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'secret', '3038joshdana', 0); INSERT INTO `sip_perl` (`id`, `keyword`, `DATA`, `flags`) VALUES (3038, 'host', 'dynamic', 0); Edit the perl script to match. Then run the perl script. It should create/overwrite whatever file you set in it and produce a new .conf Go into sip.conf and add a #include line for this new file. Matthew - Original Message - From: harry gaillac [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 11, 2004 6:42 PM Subject: Re: [Asterisk-Users] SIP peers in MySQL Database I read the perl script. here is table structure for table `sipfriends` CREATE TABLE `sipfriends` ( `name` varchar(40) NOT NULL default '', `secret` varchar(40) NOT NULL default '', `context` varchar(40) NOT NULL default '', `username` varchar(40) default '', `ipaddr` varchar(20) NOT NULL default '', `port` int(6) NOT NULL default '0', `regseconds` int(11) NOT NULL default '0', PRIMARY KEY (`name`) ) TYPE=MyISAM; I would like asterisk retrieve all sipfriends variables from database. I wish to add other variables for each sip clients like qualify, nat, ... in sipfriends table but sip code channel don't seem to be able to support others variables. may be i'm wrong ? best regards harry --- Matthew Boehm [EMAIL PROTECTED] a écrit : It is possible to use 1 database for many asterisk boxes. You can do this with the retreive script I mentioned. By adding another column to the database to indicate which * server that phone
RE: [Asterisk-Users] Control Panel
Peter Here are a few comments that may help though you maybe aware of them already. The act of delivering the information you want to a web page can be distributed quite extensively. There are three server components to consider: Asterisk Manager API Webserver Separating Asterisk and the web server is, perhaps obvious. That is, the Manager API operations can be called from any process that has TCP/IP access to the Asterisk server. This might be a process on the same machine as Asterisk or another machine completely. The separation of the Manager API is included as, like the Flash operator panel, access to the Manager API can be through a proxy. The proxy can be on the server running Asterisk, on the web server or on a third server. The advantage of the proxy is that Asterisk is servicing just one connection with the proxy rather than lots of independent user connection. Of course the web server script will communicate with asterisk via the proxy. As you may know, a simple proxy (based on the Flash operator panel proxy) is available from the Wiki at http://www.voip-info.org/wiki-Asterisk+Manager+Proxy I guess what I'm saying is that the load can be distributed widely (and specifically away from your asterisk server). With suitable distribution of the three server components, there should be almost no additional load on the asterisk server. Finally, you mention using javascript to request updates every 10 seconds. However there is a degree of complexity and maintenance with that option. Obviously you know your requirements, but a possible alternative is to present the call status information in an IFRAME within a page on your CRM web site. You can then use the meta tag: meta http-equiv=refresh content=10 To update the page in a traditional way every 10 seconds and no special client-side code is required. Good luck Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Osborne Sent: October 12, 2004 8:10 PM To: Asterisk Users Mailing List Subject: [Asterisk-Users] Control Panel Hi All, I would like to add a panel to our in-house web based CRM software that displays the status of people's extensions similar to the way the Asterisk Flash Control Panel does but I would like to use either straight html or dynamically generated GIF images. First off, I assume I can connect to the Asterisk Manager and get the status of each extension but I would like to keep the page's up-to-date (without reloading the page) by adding some Javascript to keep updating a dynamically generated image that displays the status every 10 seconds or so. Naturally I would cache the generated images so that the server isn't generating an image for every user. Does this sound reasonable or will it incur too much load on the Asterisk server. My Asterisk server is a PIII 800mhz with 128mb RAM, TDM400 with 4 FXO's and about 6 SIP phones. Ideas are more than welcome, Thanks, Pete ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] windows messenger
Asterisk doesn't support MSN9 the protocol Windows Messenger (and MSN Messenger) uses to communicate with a messenger server such as MSN or Windows 2003 running the Live Conferencing server. It should be possible to write an MSN9 server independently of Asterisk since the information needed by such a server is available via the Manager API. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shabanip Sent: October 11, 2004 4:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] windows messenger is it possible to windows messenger clients of an asterisk server to chat (text chat) with each other? what about the status presence? is it possible to each windows messenger client of an asterisk server to see the presence on other clients? if not, what is missing in asterisk? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: An old problem still hanging around?
Having just run the command sip show channels I get a list of channels even though there is no one on the phone (we only have 4 so it's easy to tell). Here is what I get: Peer User/ANRCall ID Seq (Tx/Rx) Format 192.168.0.22 (None) 4c81ac8e90c 00101/0 UNKN 192.168.0.22 (None) 984ee48048d 00101/0 UNKN 192.168.0.22 (None) 200d9d37123 00101/0 UNKN Is this normal? Why just one phone (a Grandstream Handytone ATA)? Running sip show channel 984ee48048d I get the output below so it seems active: * SIP Call Direction: Incoming Call-ID:[EMAIL PROTECTED] Our Codec Capability: 524302 Non-Codec Capability: 1 Their Codec Capability: 0 Joint Codec Capability: 0 Format UNKN Theoretical Address:192.168.0.22:5060 Received Address: 192.168.0.22:5060 NAT Support:RFC3581 Our Tag:1190462248 Their Tag: SIP User agent: Need Destroy: 0 Last Message: Promiscuous Redir: No Route: N/A DTMF Mode: rfc283 Here's a quote from a post to an earlier question by someone seeing a similar list in Jun/Jul this year. QUOTE This behavior was observed by several people for a short period of time and then seemed to have disappeared with a cvs versions starting around 1.390 - 1.394 (chan_sip.c) according to my observations (more like a guess actually) couldn't exactly pinpoint the patch that stopped it. /QUOTE My version of asterisk is from HEAD on 2004-09-19. Should I be concerned? Thanks Bill Seddon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] American vs English
Can you let me know what messages were omitted? Thanks Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: September 22, 2004 11:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] American vs English Folks, A few people have made me aware of some omissions in my files (not my fault, they weren't in the Script from the Wiki) which I shall be tackling this weekend. Whilst I'm making the files are there any other files you want? IVR's etc. If so make sure I have a script sent by email. -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] English vs American voice files
Steve Can you offer some recommendations regarding the sox arguments to use? My use of sox for down sampling is limited to this kind of command: sox in.wav -r 8000 out.gsm Are there other arguments that will give better sound from compressed formats? Thanks Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: September 20, 2004 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] English vs American voice files [EMAIL PROTECTED] wrote: On 20 Sep 2004 at 12:38, Andreas Sikkema wrote: [EMAIL PROTECTED] wrote: Initially we recorded using 16 bit/8K sampling on the basis that this is what is required by Asterisk but that was really terrible. So we're sampling at higher rates on the basis that we can use sox to change it as necessary. Any thoughts on what we can do to make the recordings sound sharper? We've found that downbsampling with sox resulted in significantly lower quality files as those downsampled with Cool Edit. Dithering in Cool Edit maybe? Matt Riddell sox offers several ways to change sampling rates. The poorest one is really quite poor. The best should not be distinguishable from any other good converter over a telecphone line. Dithering is completely irrelevant for telephony. It too LoFi to notice. :-) Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] English vs American voice files
Thanks for your suggestions. I took at look at man sox and could understand something less than every other word. A document written by and for sound engineers I think. -Original Message- From: James Cloos [mailto:[EMAIL PROTECTED] Sent: September 20, 2004 6:18 PM To: Bill Seddon Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [Asterisk-Users] English vs American voice files Bill == Bill Seddon [EMAIL PROTECTED] writes: Bill My use of sox for down sampling is limited to Bill this kind of command: Bill sox in.wav -r 8000 out.gsm You really want to use the polyphase app in sox for resampling. It is significantly slower than the other options, but that is irrelevant here. So try: sox in.wav -r 8000 out.gsm polyphase Use out.sl for a slinear file (then rename it to whatever.snl), out.ul for a mu-law file, out.al for an alaw file, etc. Cf show file formats at the * cli. If you are mostly sending the audio out over a zap channel, you may as well use ulaw or alaw -- whichever the pstn in your area uses. If primarily voip, you can match whichever codec you use or just make it slinear (.sw in sox; .sln for *) and convert it on the fly. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk presence utility
I've spent a couple of evenings writing a presence utility in C# so that a window, listing the currently registered SIP phones, can be displayed and a user can see who is on the phone and who is not. It uses the Manager API and works well, updating the display as API events are received. But what I want to be able to do is add to the list of displayed users when a phone registers and remove them when a phone unregisters. However the Manager API does not seem to generate event messages for these events. Is this correct or have a I missed an option somewhere? Certainly the register and unregister event is displayed on the Asterisk command line. It is possible to have the utility run the sip show peers command periodically and update its list based on the results of the command. However that means polling the server periodically, comsuming resources, instead of being notified just once. Any insight gratefully received. Bill Seddon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Martin Croome
Does anyone know how to get in touch with Martin Croome, author of a C# library for the Asterisk Manager API? Bill Seddon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] English vs American voice files
My wife has been recording the text published on the wiki. A couple of questions for you: 1) One of the recordings says please enter the full 10 digit number starting with the area code. Any opinions on whether this should be changed for the UK and, if so, to what? 2) The recordings seem dull on playback even though we are recording using a good quality microphone with matching impedance. Initially we recorded using 16 bit/8K sampling on the basis that this is what is required by Asterisk but that was really terrible. So we're sampling at higher rates on the basis that we can use sox to change it as necessary. Any thoughts on what we can do to make the recordings sound sharper? Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Linus Surguy Sent: September 18, 2004 8:28 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] English vs American voice files Ah, this brings up an interesting point. I've noted that BT are calling # square rather than hash. What do the other providers call it back in Blighty? 'Hash' is by far the most common used. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How To get response of command from another socket
You are correct that you will not be able to see a command response directly. However Asterisk will generate events that will be received by any manager api connection and it is these events that you will want to monitor from one or both connection windows. For example, when you originate a call, Asterisk will generate a newexten event that will contain the name of the originating device and some other stuff. If as a result of the origination, macros need to be evaluated, each macro evaluation will generate an event. Eventually there will be an event representing the dial() command from the dialplan followed by a ringing event. At the same time, an event showing that the called party device has been called will be generated. At some point the call may be answered and at this time a link event is generated showing the two parties connected in a call. Finally one or other of the parties will hangup and hangup events (one for each connected device) are generated. So although you cannot see the response generated by an action in one connection from another, you can pretty much infer the response from the events. In your case, a successful action response in one window will generate a set of events that can be seen in another window. Bill Seddon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of vrushank Sent: September 20, 2004 12:02 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] How To get response of command from another socket hi i logged on to manager API from other terminal by telnet IPADDR 5038 now logged in with username mark let's say this connection Window A now i opened another connection with Manager API with same usename lets say this window B now if i give a command like originate,Redirect through window A connection , can i able to see its response:success/failure Originate:failed/succesfully queued.. in another window B i think its not possible to see a response of command from another socket of the same user is it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] English vs American voice files
ATT plugin is quite good
I've listened to them all and to me, the Rhetorical stuff stands out.
Probably why they think it should always be sold. Maybe if more people
badgered them...
Well, boyo, I can't do a welsh accent.
But I used to live down't pit sor if tha wants a good Yorkshire accent let
mi knorr. Al si thee.
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: September 17, 2004 2:05 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] English vs American voice files
I thought about the TTS route. MS have a fairly good set that allows other
peoples engines to be added. The ATT plugin is quite good.
Perhaps I'll start there and post a few for you all to try.
Still no Taff speakers :-{
Bill Seddon said:
I agree! Rhetorical (www.rhetorical.com) have a really good
Text-to-speech
system (good in the sense that its voice rendition is quite good). Much
better than Festival or Cephstral (IMHO). The advantage of a good TTS is
that it is possible to have control over exactly what's said, it can be
changed easily and the voice talents never tire.
Anyway, wanting to see if they would permit me to create a set of voice
file using their system (you can do it from their web page) I've
corresponded with them. However they want me to buy a licence. Shame. I
thought it would be some business development for them
Failing that, it would be good to get a set of files in a modern Southern
England voice, alongside Scottish and Irish voices.
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: September 17, 2004 12:06 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] English vs American voice files
Hi folks,
Does anyone have any English voice files rather than American voice
files. I know that Digium and Alison Smith have worked hard to provide a
library of sounds etc but this doesn't work for my UK client.
Ideally I'm looking for female files but I'll settle for male ones.
If not then I'd be happy to start one (I used to be a radio announcer in a
past life). They would obviously be done on a request basis initially. At
some point the library would become usefull to someone other than the file
requestors.
Could some clever wag that deals with the language bits of * create some
other languages like British, Aussie, SouthAfrican. I'd also be looking
for Welsh too (anyone here speak Taff?) How about Georgie (I'm kidding
about that one).
All these modes of English are more than just a dialect. My 7 or so years
as an Ex-Pat in the US have taught me that American really is a valid
language. Whilst most of us English speakers can cope with American we'd
be a bit suprised when calling a VM system in Slough, Cooperpedy or
Pretoria only to be spoken to in American.
Am I just ranting here or does someone get my point?
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/
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RE: [Asterisk-Users] English vs American voice files
My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C code available but if you already have the kit... Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: September 17, 2004 2:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] English vs American voice files Looks like I've drawn the short straw here. I do have the facilities and so can do a Male Southern England recording but I'm still stuck for female (which seems to be customers preference). I also have the techincal know how as well as a web server. OK folks, I'll start with the common things like numbers and the VM system stuff etc. I'll post a link when I have something to start with. Get a lot of call for French in New Zealand do you Matt? Mark [EMAIL PROTECTED] said: On 16 Sep 2004 at 20:37, Mark Phillips wrote: No disrespect to Alison (whom I know is a Canadian) intended but her British accent is exactly that; British. It's very easy to hear that she's not from Chipping Sodbury. Also, do you really have the budget to spend on having all the relevant files recorded at $12 a time. That works out to a lot of money! Hate to state the obvious, but why don't you just record them yourselves. Options: 1) Beg, borrow or steal a microphone 2) Download the list of filenames, prompts contained 3) Record all prompts in one go, 1 after another with a 1-2 second gap between 4) Use some free audio editing software to snip the big file into little files, and save each one as the correct filename (albeit with .wav as the extension). 5) If you feel up to it, run a batch process over them to bring them all close to 0db 6) Use sox to convert to gsm files 7) Provide the resulting sound files as a free download from your website so that others don't have to do the same thing. I can help you with any step from 2 on (unless you want to come to one of my 2 studios here in New Zealand and borrow a microphone). Really the hardest part is splitting the files, but it only takes around a hour for the full set (I'm lucky, my wife who I recorded for the French prompts had also done School of Audio Engineering and so was able to use Wavelab to do the snipping etc. The other option is to just use the telephone and the asterisk dialplan to record the prompts, but I would say this would take rather a bit longer (unless you made a script that would record the first file, press # to confirm, record next file etc). Drop me a line if you need a hand with any of the above, should you devide to record them yourself. Matt Riddell (New Zealand Digium Distribution/Custom Software) http://www.sineapps.com/downloads.php (French Prompts) http://www.sineapps.com/news.php (asterisk news) -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] English vs American voice files
I agree! Rhetorical (www.rhetorical.com) have a really good Text-to-speech system (good in the sense that its voice rendition is quite good). Much better than Festival or Cephstral (IMHO). The advantage of a good TTS is that it is possible to have control over exactly what's said, it can be changed easily and the voice talents never tire. Anyway, wanting to see if they would permit me to create a set of voice file using their system (you can do it from their web page) I've corresponded with them. However they want me to buy a licence. Shame. I thought it would be some business development for them Failing that, it would be good to get a set of files in a modern Southern England voice, alongside Scottish and Irish voices. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: September 17, 2004 12:06 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] English vs American voice files Hi folks, Does anyone have any English voice files rather than American voice files. I know that Digium and Alison Smith have worked hard to provide a library of sounds etc but this doesn't work for my UK client. Ideally I'm looking for female files but I'll settle for male ones. If not then I'd be happy to start one (I used to be a radio announcer in a past life). They would obviously be done on a request basis initially. At some point the library would become usefull to someone other than the file requestors. Could some clever wag that deals with the language bits of * create some other languages like British, Aussie, SouthAfrican. I'd also be looking for Welsh too (anyone here speak Taff?) How about Georgie (I'm kidding about that one). All these modes of English are more than just a dialect. My 7 or so years as an Ex-Pat in the US have taught me that American really is a valid language. Whilst most of us English speakers can cope with American we'd be a bit suprised when calling a VM system in Slough, Cooperpedy or Pretoria only to be spoken to in American. Am I just ranting here or does someone get my point? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Clarification - FAX on local network
A potential reason for the difference could be if Asterisk uses UDP (which I think that I've read somewhere it does). TCP is a protocol that demands that transmitted packets are numbered and that the receiver both request the re-send of packets that appear to be missing and order any packets that are received out of order (the frame header includes a frame number). In this way, there is a fairly reasonable chance that the data sent will be received. After all, where would this message be (which will be sent and received using TCP) if sme of th chractrs wer misig. By comparison, UDP was designed for use in environments where the transmission control is not so important. By reducing or removing the need for transmission control, more frames can be squeezed into the available bandwidth potentially providing higher data rates. However, as data can be lost UDP is only suitable for use in applications where the loss of some data is not likely to be materially important. An example is voice transmission. Others include video or web conferencing. This is why SIP phones include features like missing packet interpolation. So I'd guess that Asterisk uses UDP, that Hylafax uses TCP and that sending a TIF image via Asterisk is asking for trouble unless Asterisk can, under prescribed circumstances, use TCP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Howard Sent: September 14, 2004 7:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Clarification - FAX on local network On 2004.09.14 11:10 Marty Mastera wrote: Ok, ok, I know there has been plenty of discussion on asterisk and fax - from this I understand: 1)First and foremost, use g.711 ulaw Yes, the codec must be lossless. 2)Packet loss, etc...makes faxing over the internet unreliable I'm not sold on this theory yet. I don't think that it's so much a matter of packet loss (this shouldn't occur regularly), but rather of latency. Transmitting packets over a network, and in particular the internet, can result in latency delays that could, in theory, pose a problem for FoIP, but I've heard of so many people successfully doing FoIP with equipment other than Asterisk (i.e. using Cisco VoIP equipment), that I tend to believe that the reliability factor is more a consequence of SIP or the equipment used (Asterisk and, in my case, a Sipura SP-2000). I have used a HylaFAX system connected to an SP-2000 for both sending and receiving faxes. The fax call comes in to an FXO on the Asterisk server, which directs it over a small LAN to the Sipura. This arrangement works tolerably well, but it's quite noticeably less-reliable than when the HylaFAX server is connected directly to the PSTN. I don't know if the Sipura is to blame or if it's Asterisk, or if it's SIP design. Although I've used a number of IAXys for voice, I haven't tried using one instead of the Sipura for fax. (Testing that would be able to eliminate or isolate SIP+Sipura.) Lee. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP on Handhelds
We've installed and used PPC X-Lite on an iPAQ with 802.11b. While the sound quality of the iPAQ user was OK (not great but OK) the sound quality as heard by the other caller was very poor. If you try using a softphone on a PPC, I'd be interested to hear of your experience. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley E. Siler Sent: September 11, 2004 12:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP on Handhelds Thank you! Found the link here... http://www.freewareppc.com/communication/xlite.shtml -Original Message- From: Iassen Hristov [mailto:[EMAIL PROTECTED] Sent: Friday, September 10, 2004 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP on Handhelds As far as I can tell, both SJphone and X-Lite offer PocketPC versions. --On Friday, September 10, 2004 4:25 PM -0700 Wiley E. Siler [EMAIL PROTECTED] wrote: I think the Bluetooth requirement may be where that hangs up. I want to be able to setup an handsfree headset too. I am thinking I will either write a sip based client in .net using the RTC API or implement the IAX model you reference here. Thank you! Wiley -Original Message- From: Matt Gibson [mailto:[EMAIL PROTECTED] Sent: Friday, September 10, 2004 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP on Handhelds If you're not opposed to running linux the sharp zaurus 6100 has 802.11b built in http://www.sharpusa.com/products/TypeLanding/0,1056,112,00.html and there's a client available to connect to iax on voip-info.org. I know you asked for SIP, but.. this is all that's avail i can find :) http://www.kauss.org/Stephan/ziaxphone/ matt Wiley E. Siler wrote: Does anyone know if SIP will/is support on handheld PCs such as the iPaq or Axiom? With their integrated 802.11b and Bluetooth it seems like a solution to provide a wireless based sip phone for any user would be possible. Handoff between access points might be problematic but most users I know would be using their PDA phone in an airport with free wireless or at the local cafe, etc, etc... Can anyone with experience in this department let me know if they think this idea is possible? Thanks, Wiley - - -- The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer - -- - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FWD
Steve Try re-registering the IAX option from your FWD account. I had exactly the same problem until Thursday when, out of frustration I re-registered. When I reported this resolution here in response to another list member's question, Ed Guy, the FWD support person, responded to my post to me to let me know there had been a provisioning issue sometime the week before and some of the accounts may need re-registering. Hope this works for you. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Maroney Sent: September 11, 2004 5:30 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FWD Im trying to get IAX to work between my * and FWD. I activated my iax2 account on iax.fwdnet.net and I get the output: Registered to '65.39.205.121', who sees us as 68.14.203.254:4569 when I start asterisk. I tried used the Call Me tool on fwdnet.net but I dont get any calls even though the Call Me tool says everything looks ok. Can someone call my FWD number and just leave me a message if i dont answer. FWD Number is 474538. My * box is configured to ring one of my extentions. Thank you, Steve Maroney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Audio level in compressed wav files
Brian Take a look at sox (type man sox at the command prompt if sox it installed for details on the options available). There is a vol argument that allows you to adjust the gain. If this is what you need, you can call sox after record (using the system command) to adjust the gain. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Cuthie Sent: September 11, 2004 7:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Audio level in compressed wav files Anybody know an easy way to adjust audio level of recordings made in Asterisk (using the 'record' application)? I've noticed that recordings using the wav format are about twice the level of those made using WAV or wav49. Unfortunately, the wav recordings are uncompressed and about 10 times the size of the other formats. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Simple question about SIP community
Have you had chance to look at Jeff Pulver's Communicator? This is a soft-phone, currently in beta, that allows you to bring together your contacts from MSN, ICQ, AOL and, importantly from your point of view, add contacts that are SIP users. I've not tried it yet with asterisk, but now you have asked the question, I'll try it out... It certainly detects FWD presence so I think it might work with Asterisk. If it doesn't I'll ask put it forward as a suggestion. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird Sent: September 09, 2004 8:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Simple question about SIP community On Sep 9, 2004, at 8:53 AM, Marcello Lupo wrote: we have a community of people on an * box that use SIP softphones to talk each other. Can you suggest me the quickest and simple way to let someone know who is online without have to call one by one the persons to look if they are present or not?? Something the user list in Microsoft Messenger. I was thinking on some sort of web page that can check the registration of the sip clients on the asterisk but want to know if already exist to avoid to reinvent the wheel. thanks, The generic term for this is 'presence'. Everyone seems to agree that it's important, but I'm not aware of anyone actively working on it for Asterisk. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iaxy vs sipura
I run Asterisk on Redhat 8.0 with a VM hosted by Microsoft's Virtual PC which, in turn, runs on Windows 2000 Server. Works like a charm. Can't use Zaptel cards but that's OK for me. I can put it into standby any time and it takes only a few seconds to start up the VM from its saved state and at that time the Linux session (and Asterisk) is available once again. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists Sent: September 10, 2004 2:03 PM To: Andy Powell Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iaxy vs sipura On Fri, 10 Sep 2004 14:05:09 +0200, Andy Powell [EMAIL PROTECTED] wrote: At the risk of stating the obvious if you have a laptop not running MacOSX (ie perhaps running windows) download my asterisk live! cd ( http://www.automated.it/asterisk/ ), burn it and test it on your laptop and bung it in your laptop case along with your iaxy/sipura/whatever and errm... problem solved.. :D Certainly an option, but most business folks will want to have their Outlook contacts and Excel spreadsheets in front of them when they are on the phone. Dual boot environments are not ideal in those situations. Imagine you're talking to some guy on the phone about prices and he tells you I cant' tell you what the discounts are right now because I would have to shut down the phone system to open Excel. However, you could use VMware on an Intel notebook to run both Windoze and Linux concurrently. This wouldn't be ideal for a real PBX for performance reasons, but since all you are going to use Asterisk for is to be a gateway for one single user, it's probably ok in this particular scenario. I remember there was a guy in Romania who reported he had VMware with Windoze and Asterisk on Linux running as a home PBX on his PC and it seemed to be alright. If you'd combine such a setup with a Windoze GUI tool that will start and stop the Linux environment and Asterisk at the push of a button, then you'd have a fairly convenient and workable SIP/IAX gateway solution for travelling biz folks. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] incomming call rejected using IAX2 with FWD
I've also had problems receiving inbound IAX calls from FWD. And without any outbound problems. I've three numbers and fortunately one worked while two did not. The one working number encouraged me to keep trying to work out a reason for the difference. In the end (today) I re-activated IAX for both of my FWD non-working accounts and it started working again. If you have not already tried re-activating your IAX it might be one thing to try. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: September 09, 2004 10:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] incomming call rejected using IAX2 with FWD Works for me, follow the instructions closer. :) Storm D. J. Petersen wrote: Hi, I cannot seem to accept incoming calls from FWD using IAX2. I followed the directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing calls fine using IAX via FWD. When someone calls me from FWD I get the following message: Chan_iax2.c:5251 socket_read: Reject connect attempt from 65.39.205.121 Any ideas? I had similar problems getting inbound from FWDI had to do two things: 1) Register with FWD (pretty standard, but I'll be thorough) ; Let FWD know where to send inbound calls register = 484162:[EMAIL PROTECTED] 2) Create the FWD entry in IAX.conf (note that I found the name of the entry makes a differenceie iaxfwd) [iaxfwd] type=user context=fwd-in auth=rsa inkeys=freeworlddialup Make sure that the context you specify for incoming calls exists and has an extension entry matching your FWD number...also I didn't try it without the RSA keys, but my configs use them and it works, so if you aren't currently using them you might try that...I can't remember exactly, but I think that the keys are included in the keys directory by default as shipped with Asterisk... Hope that helps, feel free to call me via FWD (484162 and x200) if I can help, it would be good to have someone to call me to test Marty ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie: Only allow authenticated users to call
I'm wondering if you are confusing two ideas. It has to be possible for anyone to be able to call you just like they can on an ordinary POTS line. Registration is for those who need to appear in some sense internal to the PBX. Using dialplan contexts you can offer very different functionality to callers who are registered versus those who are just calling. For example, you might assign all registered users to a context call internal and provide access to all the dial plans. You might set the context of all non-registered callers to an external dialplan context. The internal context might provide access to all the telephony services an internal user might expect (eg dial 7 to get to voicemail automatically). The external context might direct a caller to the operator or to a voice prompt. Optionally, you might provide an extension for voicemail so that external employees calling from home or a client site can get to their messages. Clearly the caller will need to be prompted for a voice mail box and password but that's covered by the voicemail system. Bill Seddon Lyquidity Solutions -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Jensen Sent: September 08, 2004 9:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie: Only allow authenticated users to call I made the observation that I'm able to make a call with my SIP client (kphone) even when I'm not registered/authenticated. Of course, when I'm not registered at asterisk, people can't call me, but it's still a huge security hole, that unregistered Clients can make calls. Is there a way to tell asterisk to only allow registered clients making calls? I know about the Anti Ex Girlfriend function, but this is not what I want. Regrads, Henry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help Running am-main.pl Perl/CGI on Apache Server
Another reason can be a missing or incomplete addhandler line in /etc/httpd/conf/httpd.conf. The line should look like: Addhandler cgi-script .cgi .pl My Redhat installation was missing the .pl and when trying to run perl based cgi scripts also generated the same 500 error. Correcting the line fixed my problem. Bill Seddon Lyquidity Solutions 16 Lynton Road New Malden SURREY KT3 5EE +44 (0) 208 336 2556 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shilliday, Jim Sent: September 04, 2004 8:36 PM To: Shekhar Prasad; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Help Running am-main.pl Perl/CGI on Apache Server I had the same problem -- make sure these three lines are at the beginning of each of your .pl's (right after the comment section): use CGI; my $q = CGI-new(); print $q-header(); That fixed it for me. Jim Shilliday IT Director Equal Justice Center 1315 Walnut St. Suite 400 Philadelphia PA 19107 215-238-6970 -Original Message- From: Shekhar Prasad [mailto:[EMAIL PROTECTED] Sent: Saturday, September 04, 2004 2:19 PM To: [EMAIL PROTECTED] Cc: Taemoor Abbasi Subject: [Asterisk-Users] Help Running am-main.pl Perl/CGI on Apache Server Hi all, I've installed Asterisk on Linux Red Had 9. Now, I was trying to set up a GUI based system for the PBX. I downloaded some packages, but I have to have Perl running CGI scripts through the webserver. It does not allow me to. I am able to run a basic script that just just prints out html messages and nothing else. However, when I try to run am-main.pl or config.pl or any other cgi-bin scripts that came with Asterisk, I get a 500 error: Premature End of Script Header. I've tried changing the chmod to -x or 755 and also have included the header required to display the html script. However, no luck. I am able to run it in command line, but not through the Mozilla browser. Is this enough information I've provided? You can contact me directly if you need more information. shekhar . prasad at gmail . com shekhar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] incomming call rejected using IAX2 with FWD
Just by way of giving you some encouragement, I too run a version downloaded on about the same day and use Telappliant successfully. I did get a rejected error until I converted one of my accounts to IAX from the default SIP. I don't think there is anything special about my setup and the * PBX is behind a NAT firewall. Also use FWD via their IAX2 service. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: August 28, 2004 4:02 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] incomming call rejected using IAX2 with FWD All, I am experiencing this problem with an IAX link to the UK provider TelAppliant. chan_iax2.c:5251 socket_read: Rejected connect attempt from 217.14.132.162 Not sure what is causing this, however, it seems to have started sine I downloaded Asterisk CVS-HEAD-08/19/04-19:55:53. Not sure if this is a red herring., but, seems coincidental. Again any idea's ? Best regards Steve Beaumont -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Storm D. J. Petersen Sent: 09 January 2100 15:13 To: Asterisk-Users Subject:[Asterisk-Users] incomming call rejected using IAX2 with FWD Hi, I cannot seem to accept incoming calls from FWD using IAX2. I followed the directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing calls fine using IAX via FWD. When someone calls me from FWD I get the following message: Chan_iax2.c:5251 socket_read: Reject connect attempt from 65.39.205.121 Any ideas? Thanks, S. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
