[Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
Hello, world! I'm considering running Asterisk 1.2.4 on Solaris 10 on a Sun Fire X2100 server or two (Opteron CPU, nForce 4 chipset), and apparently this works. I've read that the Zaptel package won't work on anything other than Linux, since it's intended to hook into the Linux kernel in the form of a kernel module. This concerns me, since I've read that ztdummy, the timing-source component of Zaptel, is required for the music-on-hold and conferencing functions of Asterisk to function. So, with this in mind, is there any way to run a complete Asterisk solution on Solaris 10 (including music-on-hold and conferencing)? If so, how? Thanks in advance! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
Hello, Mark! At 06:33 AM 02/20/2006, you wrote: >Please forgive the question, but what is the rationale behind using Solaris >over Linux as an asterisk hosting platform? Because of a few reasons, actually: (1) The remote hardware management options available for the X2100 work better (or only, I'm not sure which) under Solaris, and they seem to *really* kick ass. Plus, being Sun-engineered, the X2100 should keep working until it's completely obsolete, and then some. (2) I know someone who knows Solaris inside-out and backwards, blindfolded, while hung upside-down, and codes Bourne shell and C in his sleep; this is vaguely reminiscent of www.chucknorrisfacts.com. I'm quite sure this will come in handy when (not if) something breaks, giving him the opportunity to make some money and giving me the opportunity to reduce my downtime. :) (3) I'd like to learn Solaris, and being SysV-based like Linux, it shouldn't be too much of a stretch. -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
Hello all, I really appreciate the replies I've gotten about this so far (especially the support for wanting to run it on Solaris!). The core issue seems to have been missed, though -- is there any way to run a complete Asterisk solution on Solaris 10 (including music-on-hold and conferencing)? This probably comes down to a few issues: - Is ztdummy (a component of Zaptel) *really* required for MoH and conferencing support? - I've heard rumblings about "zaprtc" being a potential replacement. Is it a *real* replacement? Will it work on Solaris 10? If not, what will? - I *know* people have got to be running Asterisk on Solaris 10 (but I don't know who they are, unfortunately!). If you happen to be a member of that esteemed clique, could you please let me know how you got ztdummy working, or what you used as a replacement? I really don't see people going without MoH and conferencing in a "real" setup. Thanks again! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada Date: Sun, 19 Feb 2006 23:45:01 -0500 To: asterisk-users@lists.digium.com From: Alexander Burke <[EMAIL PROTECTED]> Subject: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100) Hello, world! I'm considering running Asterisk 1.2.4 on Solaris 10 on a Sun Fire X2100 server or two (Opteron CPU, nForce 4 chipset), and apparently this works. I've read that the Zaptel package won't work on anything other than Linux, since it's intended to hook into the Linux kernel in the form of a kernel module. This concerns me, since I've read that ztdummy, the timing-source component of Zaptel, is required for the music-on-hold and conferencing functions of Asterisk to function. So, with this in mind, is there any way to run a complete Asterisk solution on Solaris 10 (including music-on-hold and conferencing)? If so, how? Thanks in advance! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell PowerEdge 2850
Hello, Klaus! At 06:23 PM 02/20/2006, you wrote: Both riser cards only have 64 Bit PCI slots. I think 64 bit is always 3.3 Volt - isn't it? Nope! http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 Grab a copy of Asterisk: The Future of Telephony from the link above, and open it to page 16 (PDF page 34 or thereabouts). There's a great illustration of the possibilities. Plus, it's a great book to have on hand; I bought a paper copy before I knew it was available online, but I still would have bought it, because it's nice to have. Kudos to O'Reilly for setting it free, too! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple TDM400P's in a single machine
Hello, Marc! At 06:24 PM 02/20/2006, you wrote: Can someone give me a definite answer as to wether or not you can reliably run multiple TDM400P's in the same machine? I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing key system, but I have seen several threads suggesting that this is not a supported configuration According to "Asterisk: The Future Of Telephony": IRQ latency Interrupt request (IRQ) latency is basically the delay between the moment a peripheral card (such as a telephone interface card) requests that the CPU stop what it's doing and the moment when the CPU actually responds and is ready to handle the task. Asterisk's peripherals (especially the Zaptel cards) are extremely intolerant of IRQ latency. Linux has historically had problems with its ability to service IRQs quickly; this problem has caused enough trouble for audio developers that several patches have been created to address this shortcoming. So far, there has been some mild controversy over how to incorporate these patches into the Linux kernel. Because the Digium cards require so much, it is generally recommended that only one Digium card be run in a system. If you require more connectivity than a single card can provide, either replace your existing card with one of higher density, or add another server to your environment.* * Many people report that Sangoma cards are more robust when it comes to dealing with unpredictable motherboard chipsets, and thus can handle sharing motherboard IRQ resources. Regardless, it is still worth considering using multiple servers, as the redundancy that can be gained from this strategy can quickly offset the cost. -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Download "Asterisk: The Future Of Telephony"
Hello, list! I'm hosting a mirror of the book "Asterisk: The Future Of Telephony" by O'Reilly Press, published under the Creative Commons license; I believe this license allows me to do this, but if I'm mistaken, please let me know. I've taken the liberty of fixing the page numbers so Acrobat is now aware of the correct number of each page, and shrinking the filesize with Acrobat's "Reduce File Size" tool (while still maintaining compatibility with Acrobat 4.0, apparently). I bought a paper copy before I knew the book was available online, but it's good enough that even had I known it was available online, I still would have bought it on paper. You're welcome to download it and keep it on hand -- it makes for EXCELLENT reading: http://www.alexburke.ca/asterisk-tfot.pdf -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple TDM400P's in a single machine
Marc: At 06:24 PM 02/20/2006, you wrote: I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing key system, but I have seen several threads suggesting that this is not a supported configuration This bad boy might be what you need: http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM2400P&tab=details If not, consider an external channel bank: http://www.voipsupply.com/product_info.php?products_id=868 http://www.voipsupply.com/product_info.php?products_id=781 It would be great if you could let the list know which route you take, and the success (or lack thereof) that you have with it! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Download "Asterisk: The Future Of Telephony" [More Info]
One thing I forgot to mention: I also cropped the registration and cut marks off the sides of the pages. If you want the uncropped version, get: http://www.alexburke.ca/asterisk-tfot-uncropped.pdf Sorry about the excessive noise, but I figured I should mention this. Date: Mon, 20 Feb 2006 18:55:50 -0500 To: asterisk-users@lists.digium.com From: Alexander Burke <[EMAIL PROTECTED]> Subject: Download "Asterisk: The Future Of Telephony" Hello, list! I'm hosting a mirror of the book "Asterisk: The Future Of Telephony" by O'Reilly Press, published under the Creative Commons license; I believe this license allows me to do this, but if I'm mistaken, please let me know. I've taken the liberty of fixing the page numbers so Acrobat is now aware of the correct number of each page, and shrinking the filesize with Acrobat's "Reduce File Size" tool (while still maintaining compatibility with Acrobat 4.0, apparently). I bought a paper copy before I knew the book was available online, but it's good enough that even had I known it was available online, I still would have bought it on paper. You're welcome to download it and keep it on hand -- it makes for EXCELLENT reading: http://www.alexburke.ca/asterisk-tfot.pdf -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
Hello, Steve! At 03:55 AM 02/21/2006, you wrote: ztdummy was only used for timing. Linux 2.6 provides this function in the kernel and I assume Solaris already has timing functions there. Page 36 of Asterisk: The Future Of Telephony (O'Reilly Press) states that you either require a Digium PCI card to provide clocking, or ztdummy if you "lack the PCI hardware required to provide timing". It goes on to mention that a UHCI USB controller was required pre-2.6 but now that there's a 1kHz clocking source in the kernel, ztdummy will attach to that instead, thus eliminating the requirement for the UHCI USB controller. While it doesn't explicity say so, it seems to very strongly imply that either a PCI card or ztdummy are *required* for some Asterisk functionality (namely music-on-hold and conferencing, apparently). Is this actually not the case? Just for reference, here's the section in question, verbatim (copy-and-paste from the PDF): The ztdummy Driver In Asterisk, certain applications and features require a timing device in order to operate (Asterisk wont even compile them if no timing device is found). All Digium PCI hardware provides a 1-kHz timing interface. If you lack the PCI hardware required to provide timing, the ztdummy driver can be used as a timing device. On Linux 2.4 kernel based distributions, ztdummy must use the clocking provided by the UHCI USB controller. The driver looks to see that the usb-uhci module is loaded and that the kernel version is at least 2.4.5. Older kernel versions are incompatible with ztdummy. On a 2.6 kernelbased distribution, ztdummy does not require the use of the USB controller. (As of v2.6.0, the kernel now provides 1-kHz timing with which the driver can interface; thus, the USB controller hardware requirement is no longer necessary.) The default Makefile configuration does not create ztdummy. To compile ztdummy, you must remove a comment marker from the Makefile. Open it in your favorite text editor and look for the following line: MODULES=zaptel tor2 torisa wcusb wcfxo wctdm \ ztdynamic ztd-eth wct1xxp wct4xxp wcte11xp # ztdummy Remove the hash* (#) symbol from in front of ztdummy, save the file, and compile Zaptel as usual. -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended rack-mountable server anyone?
Hello, Mitchel! At 07:41 AM 02/21/2006, you wrote: I've been doing a lot of research into a decent server for Asterisk but I seem to be running and circles and now I am turning to you. The issue I have is it needs to be rack mountable (so a Dell SC430 isn't going to work) and preferably have 3 pci ports. The problem that I seem to be running into is that when I look at servers from Dell or IBM or the like they only seem to support PCI-X which (from what I understand) does not support the Digium cards that we already have and that they still make. So if anyone has a suggestion or has a server they rather prefer for it's reliability, expandability, etc, please recommend it! As I understand it, PCI-X is fully backwards-compatible with PCI (as in the presence of a PCI card on a PCI-X bus will cause that bus to drop back to regular PCI mode). If you want something super-reliable which can run Linux, Solaris, or Windows, and you require three PCI slots, this may interest you: http://www.sun.com/servers/entry/x4200/ (Click on the "Gallery" link for pretty pictures.) I'm seriously considering two X2100s (because I don't need four disks or any PCI cards): http://www.sun.com/servers/entry/x2100/ These boxes will run Solaris, Linux, or (ack) Windows, and their remote monitoring/management support is second to none. -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Download "Asterisk: The Future Of Telephony" [More Info]
Hello, Bob! At 01:32 PM 02/21/2006, you wrote: Speaking of this book, where can I get it? Obviously I can read the pdf, but I lack the facility to print it in any usable fashion. The labor and materials that I have spent on trying to print it thus far probably outweighs the cost of the silly thing. Is it only available online, or do you think Barnes and Noble, Borders, etc might have it? Oh, I wouldn't print the whole thing; the price of the paper copy doesn't make it cost-effective to run one off... unless you happen to work at a place with a nice laser printer and a spiral-binding machine, I guess! Any reputable book seller should be able to order it by its ISBN (0596009623). I bought my paper copy from Amazon, and had it in a week. It *is* a real book -- the PDF that was released is (most of) exactly what went to the book printing company -- the markings in the corners are alignment marks, and the vertical and horizontal lines in the margins are the cut marks for binding. The table of contents and index are missing, probably because they're fairly useless in a file you can do full-text searches on, and also probably to make counterfeiters actually have to do some work. -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie config help? Wellgate 3701a
Hello, Martin! At 02:50 AM 02/26/2006, you wrote: I got my new Welltech 3701a, 1FXS,1FXO gateway. If you do give up with it (isn't Engrish documentation fun?), you may wish to take a look at the Sipura SPA-3000. I have one but haven't put it to use yet. I've heard *many* good things about it, though! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Web-Based Voicemail?
Hello, list! After Googling and checking out the voip-info wiki, I haven't had much luck in locating a decent web-based voicemail system for Asterisk to check your VM while you're away from the office without using a phone. Can anyone make any recommendations for such packages/applications? Thanks in advance! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Over WiFi
Hello, Dumpexec! At 12:35 PM 02/26/2006, you wrote: Is there a sort of high grade cat5 cable that can propagate signals for up to 1Km? No. The standard is 100m per leg, maximum, even with STP (shielded twisted-pair) cable. You could go to multimode fiber to get 2km, but you'd have to find another way to power your device. Sorry! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prepaid / postpaid solution
At 05:03 PM 02/26/2006, you wrote: I want to match the user from the users callerid. All users have DIDs. You probably shouldn't do that for security reasons -- rather, match them according to the SIP username/password pair they provide when they register. -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] courtesy message calling mobile phones
At 12:07 PM 02/27/2006, you wrote: Can you explain this? What country? In this case it's not asterisk but the telco that has to do the Answer. To every mobile? or just that provider? My knowledge of SS7 is limited, but this has to do with opening the audio path before a call-answered event (which never comes), or even before a call-alerting event. This is also the case where a SIT is generated, and a message like "the number you have reached is not in service" is played for those not hardcore enough to know the specific error from the sound of the SIT alone. :) -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Custom Extension halting execution upon caller hanging up
Hello, list!
I'm having some trouble with [EMAIL PROTECTED] 2.7(?), Asterisk 1.2.5, inasmuch as
my custom extension is not continuing execution when the caller hangs
up. (Please excuse the sterilized output.)
Here's how it's supposed to go:
exten => 2,8,Monitor(wav,${TIMESTAMP})
exten => 2,9,Dial(SIP/Provider/8005551212)
exten => 2,10,Macro(record-cleanup)
If the caller hangs up before the callee does, execution of the
custom extension halts and does not continue to priority 10
(record-cleanup), where sox is used to reverse the audio files and
then mix them then reverse them again so they'll be in sync (since
inbound audio only starts from call-answered but outbound audio
starts from the beginning of ringback).
Asterisk provides this debug output to the console (internal
extension 101 is the caller):
-- Called Provider/8005551212
-- SIP/Provider-993d is making progress passing it to SIP/101-1666
-- SIP/Provider-993d answered SIP/101-1666
The call proceeds normally, but then Asterisk spits this out the
moment the caller hangs up first:
== Spawn extension (custom-extension, 2, 9) exited non-zero on
'SIP/101-1666'
How can I prevent the extension from bailing before I have a chance
to clean up the recording?
Thanks in advance!
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
___
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[Asterisk-Users] Custom extension halting execution upon caller hanging up
Hello, list!
I'm having some trouble with [EMAIL PROTECTED] 2.7(?), Asterisk 1.2.5, inasmuch as
my custom extension is not continuing execution when the caller hangs
up. (Please excuse the sterilized output.)
Here's how it's supposed to go:
exten => 2,8,Monitor(wav,${TIMESTAMP})
exten => 2,9,Dial(SIP/Provider/8005551212)
exten => 2,10,Macro(record-cleanup)
If the caller hangs up before the callee does, execution of the
custom extension halts and does not continue to priority 10
(record-cleanup), where sox is used to reverse the audio files and
then mix them then reverse them again so they'll be in sync (since
inbound audio only starts from call-answered but outbound audio
starts from the beginning of ringback).
Asterisk provides this debug output to the console (internal
extension 101 is the caller):
-- Called Provider/8005551212
-- SIP/Provider-993d is making progress passing it to SIP/101-1666
-- SIP/Provider-993d answered SIP/101-1666
The call proceeds normally, but then Asterisk spits this out the
moment the caller hangs up first:
== Spawn extension (custom-extension, 2, 9) exited non-zero on
'SIP/101-1666'
How can I prevent the extension from bailing before I have a chance
to clean up the recording in priority 10?
Thanks in advance!
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
___
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Re: [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)
Erik: At 01:17 PM 03/22/2006, you wrote: On 3/22/06, Andrew D Kirch <[EMAIL PROTECTED]> wrote: > Andrew D Kirch > Indianapolis, United States Well if that isn't one of the most bizarre emails I've seen come across this list. It's a spoof of a typical Nigerian "419" scam email. Rather well done, too. :) Thanks for the laugh, Andrew! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7970 SIP Firmware; SIP 8.2 for 7940/7960
Hello! I'm hearing about this 7970 SIP firmware. I'm a Cisco Registered Partner with full access to the Cisco Software Center, and yet I can't find it. Can someone enlighten me as to where to get it? Is it also available/applicable to the 7971G-GE? Did you know that on March 10, SIP 8.2 was released for the 7940/7960? Has anyone tried it yet? If so, what are people's opinions? Thanks in advance! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7940/7960 SIP 8.2 Freely Downloadable
Just in case anyone here hadn't noticed, Cisco is apparently making 7940/7960 SIP 8.2 firmware freely downloadable by anyone: http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960 username: anonymous password: your email address -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Waiting before executing System command
Hello, all!
I'm having a problem with the following snippet that executes upon hangup:
exten => h,n,Wait(5)
exten => h,n,System(mv /some/file /some/other/dir/)
Wait() doesn't want to seem to wait! So instead I tried:
exten => h,n,System(sleep 5; mv /tmp/${CALLFILENAME}
/var/spool/asterisk/outgoing/)
This only executes sleep, not mv. How can I make it wait before
moving the file?
Thanks in advance!
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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Re: [asterisk-users] Waiting before executing System command
Hello, Moses!
At 09:20 PM 27/10/2006, you wrote:
what about
exten => h,n,System(mycommand /some/file /some/other/dir/)
Where "mycommand" is your custom shell script to sleep before moving the file.
That would work, but I'm trying to avoid kludges like that. Hence my
question about doing it entirely within the dialplan.
Any ideas?
On 10/27/06, Alexander Burke <[EMAIL PROTECTED]> wrote:
Hello, all!
I'm having a problem with the following snippet that executes upon hangup:
exten => h,n,Wait(5)
exten => h,n,System(mv /some/file /some/other/dir/)
Wait() doesn't want to seem to wait! So instead I tried:
exten => h,n,System(sleep 5; mv /tmp/${CALLFILENAME}
/var/spool/asterisk/outgoing/)
This only executes sleep, not mv. How can I make it wait before
moving the file?
Thanks in advance!
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
___
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[asterisk-users] Off-Site Extensions That Would Show As In-Use?
Hello, list! I'd like to create an extension that points to an offsite location (a number on the PSTN), the purpose of which would be to see if that offsite location is still on a call forwarded to it by Asterisk. This way a receptionist could choose to transfer calls to a mobile phone only if it's finished with the last call the receptionist forwarded to it. If I configure a custom extension with the destination SIP/TrunkName/NXXNXX, the calls transfer fine but don't show as busy using the Flash Operator Panel (as an example). Any thoughts? Thanks in advance, Alex -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Site Extensions That Would Show As In-Use?
Dovid B wrote: Are you trying to get FOP to monitor the SIP account that you are using to dial the cell phone on ? The SIP extension, yes. So, as long as a call that has been forwarded to that cell phone is still in progress, that extension should still show busy. Thanks again, Alex -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
