Re: [asterisk-users] Call Waiting problems
On 3/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote: If you are using sip then you should look for the call-limit option in sip.conf file. Using IAX. Is that a problem? ___ --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] Call Waiting problems
Situation, simple home setup: * Trixbox 2.0 * Feature Codes installed * GNet PA-168V based ATA * Cheesy cordless analogue phone From what I gather, dialing *70 from the handset should activate Call Waiting. All it seems to do is change the message The person at extension is on the phone to ring ring The person at extension is unavailable. The person speaking on the phone at the time of the second incoming call hears no indication that another call is incoming. Part of the problem is that I have no idea how the feature should work when it's functional. Could someone help me troubleshoot this, or point me in the right direction? It seems as though, as a very basic feature, not a lot of documentation is written about it. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible, horrible firewall issues in * to * setup
Okay, I *think* I got it, but I must be missing something. Here is
what the files say on the various boxen:
On *1:
[401]
type=friend
secret=password
qualify=yes
port=4569
notransfer=yes
host=dynamic
dial=IAX/401
context=from-internal
[601]
type=friend
secret=password
qualify=yes
port=4569
notransfer=no
host=dynamic
dial=IAX/601
context=from-internal
On *2:
iax.conf:
[601]
type=friend
disallow=all
context=from-internal
canreinvite=yes
allow=ulaw
[asterisk-1]
username=601
type=peer
secret=777
qualify=yes
host=asterisk-1.someplace.net
disallow=all
context=from-internal
canreinvite=yes
allow=ulaw
register=601:[EMAIL PROTECTED]
extensions.conf:
[outrt-003-CallA1]
exten = _4XXX,1,Macro(dialout-trunk,1,${EXTEN:1},,)
exten = _4XXX,n,Macro(outisbusy,)
So now, of course, I can call from *2 to extension 401 on *1 (by
dialing 4401) without a problem, but I still cannot seem to call from
*1 to extensions on *2. It's complaining about there not being a route
to the given extension, which makes sense I guess. I don't know how to
create a proper outbound route on *1 to *2 since I don't have a trunk
to direct it to, just a registration. I'm sure I'm lacking something
fundamental here - any help would be greatly appreciated.
Thanks!
On 11/24/06, Tim Panton [EMAIL PROTECTED] wrote:
On 22 Nov 2006, at 22:21, Lachek Butalek wrote:
My mission is to get one * box to dial another * box' extensions. I
have set this up previously without any issues by making a simple IAX
trunk/extension pair on the two boxes and create a dial plan with a
prefix like 9|XXX to select an extension on the other box.
My problem is that I now have to do this with extremely restrictive
firewalls thrown into the mix - firewalls I have no control over.
Basically, the setup is:
*1 --- FW1 --- (Internet) --- FW2 --- FW3 --- *2
I have control over firewall 1 and 3, but not 2. Using port forwarding
(4569 UDP) on FW1, I have been able to make calls from *2 to *1. My
problem lies with making calls the other way, as I have no way of port
forwarding on FW2.
If FW2 and FW3 permit outbound UDP and associated replies you won't
need to.
(even if they NAT them).
Set up 4569 on FW1 to go to *1
Add *2 as a peer (and user) in iax.conf on *1
Do _nothing_ with FW3
Set up *2 to _register_ with *1
The repeated registration from 2 to 1 will keep the any
NAT's and port maps open and tell 1 how to reach 2.
(IAX is great)
Tim.
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Re: [asterisk-users] Terrible, horrible firewall issues in * to * setup
Correction: those 'secret' lines are of course all supposed to say '777'. :)
On 11/24/06, Lachek Butalek [EMAIL PROTECTED] wrote:
Okay, I *think* I got it, but I must be missing something. Here is
what the files say on the various boxen:
On *1:
[401]
type=friend
secret=password
qualify=yes
port=4569
notransfer=yes
host=dynamic
dial=IAX/401
context=from-internal
[601]
type=friend
secret=password
qualify=yes
port=4569
notransfer=no
host=dynamic
dial=IAX/601
context=from-internal
On *2:
iax.conf:
[601]
type=friend
disallow=all
context=from-internal
canreinvite=yes
allow=ulaw
[asterisk-1]
username=601
type=peer
secret=777
qualify=yes
host=asterisk-1.someplace.net
disallow=all
context=from-internal
canreinvite=yes
allow=ulaw
register=601:[EMAIL PROTECTED]
extensions.conf:
[outrt-003-CallA1]
exten = _4XXX,1,Macro(dialout-trunk,1,${EXTEN:1},,)
exten = _4XXX,n,Macro(outisbusy,)
So now, of course, I can call from *2 to extension 401 on *1 (by
dialing 4401) without a problem, but I still cannot seem to call from
*1 to extensions on *2. It's complaining about there not being a route
to the given extension, which makes sense I guess. I don't know how to
create a proper outbound route on *1 to *2 since I don't have a trunk
to direct it to, just a registration. I'm sure I'm lacking something
fundamental here - any help would be greatly appreciated.
Thanks!
On 11/24/06, Tim Panton [EMAIL PROTECTED] wrote:
On 22 Nov 2006, at 22:21, Lachek Butalek wrote:
My mission is to get one * box to dial another * box' extensions. I
have set this up previously without any issues by making a simple IAX
trunk/extension pair on the two boxes and create a dial plan with a
prefix like 9|XXX to select an extension on the other box.
My problem is that I now have to do this with extremely restrictive
firewalls thrown into the mix - firewalls I have no control over.
Basically, the setup is:
*1 --- FW1 --- (Internet) --- FW2 --- FW3 --- *2
I have control over firewall 1 and 3, but not 2. Using port forwarding
(4569 UDP) on FW1, I have been able to make calls from *2 to *1. My
problem lies with making calls the other way, as I have no way of port
forwarding on FW2.
If FW2 and FW3 permit outbound UDP and associated replies you won't
need to.
(even if they NAT them).
Set up 4569 on FW1 to go to *1
Add *2 as a peer (and user) in iax.conf on *1
Do _nothing_ with FW3
Set up *2 to _register_ with *1
The repeated registration from 2 to 1 will keep the any
NAT's and port maps open and tell 1 how to reach 2.
(IAX is great)
Tim.
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Re: [asterisk-users] Terrible, horrible firewall issues in * to * setup
Two apologies to be made: #1: Sorry for all the spamming. #2: Sorry for all the top-posting. Gmail doesn't lend itself well to email lists since it neatly tucks all the quoted text away (out-of-sight, out-of-mind for the poster), leaving a mess for those whose email clients do not. Again, my apologies. ___ --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] Terrible, horrible firewall issues in * to * setup
My mission is to get one * box to dial another * box' extensions. I have set this up previously without any issues by making a simple IAX trunk/extension pair on the two boxes and create a dial plan with a prefix like 9|XXX to select an extension on the other box. My problem is that I now have to do this with extremely restrictive firewalls thrown into the mix - firewalls I have no control over. Basically, the setup is: *1 --- FW1 --- (Internet) --- FW2 --- FW3 --- *2 I have control over firewall 1 and 3, but not 2. Using port forwarding (4569 UDP) on FW1, I have been able to make calls from *2 to *1. My problem lies with making calls the other way, as I have no way of port forwarding on FW2. My initial thought was to set up a reverse SSH tunnel from *2 to *1, which would have worked fine if SSH would tunnel UDP (latency is a different matter altogether). I found a software called Zebedee (http://www.winton.org.uk/zebedee/) which claims to do UDP tunneling, and is able to do it in reverse, but I can't for the life of me get it to work. Before I try further with Zebedee, I thought it wise to ask the * community if there is a standard solution in this particular case, or perhaps if I'm attempting the impossible. Any input is greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to find good VOIP provider.
Voxee.com supports both SIP and IAX2, as well as GSM, iLBC, uLAW, aLAW and G729 codecs. Their rates to international locations are based on 6/30 billing, and vary a lot from location to location. You can view their rates here: http://www.voxee.com/rates.xls You may also want to look at VoipStunt.com, although I believe they are SIP-only. Their rates are very low, though - perhaps you could use them as a complement to another provider for normally high-priced locations. On 6/15/06, Nikolay Pavlov [EMAIL PROTECTED] wrote: Hi, guys. May be someone could give me advise? I am trying to find good VOIP provider ONLY for OUTGOING calls with low per channel cost and cheap rates on Eastern Europe, Turky and xUSSR. Should support g729 or g723 codecs, SIP or IAX connectivity. -- = = Best regards, Nikolay Pavlov. = = ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Easiest (best?) linux distribution for dedicated Asterisk box?
While it's not the way I do it personally, I've been told the absolutely easiest setup is [EMAIL PROTECTED] It is based on CentOS so security updates should be coming down at a decent pace and I imagine this could be set up to occur automatically. At that point, you pretty much have a fire-and-forget Asterisk install with most of the bells and whistles. My personal choice is Slackware - combined with a 3rd party package managing tool called Swaret you can set up a very slim, simple and secure box in a very short amount of time and have do security patch deployments automatically, much like CentOS/Redhat/etc. I'm personally running Asterisk with FreePBX on an Apple PowerMac G3 266MHz using Slackintosh, with great results so far. Your system should be fine, though I'd probably dump some more RAM into it if I were you - especially if you want to run [EMAIL PROTECTED], which uses a MySQL database and the Apache webserver, among other things. On 6/13/06, John Klimek [EMAIL PROTECTED] wrote: First off, I'm sorry for sending so many messages to the list-serv. Hopefully this will be my last for a while! I was going to use my WRT54G router as a small Asterisk box, but I forgot that I had a spare eMachines computer (Intel Celeron 633 MHz, 20GB HD, 64mb RAM). Will this machine work OK for a very simple dedicated home Asterisk box? Also, what is easiest linux distribution to use and install? All I want is a simple Asterisk box that I can telnet into and have voicemail, music-on-hold (MP3), etc... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreePBX 2.1.0: Manually rewriting extensions_additional.conf
Yes, I have learned my lesson and now know to add code to _custom instead of _additional. I also realize I can manually update the MySQL database. However, that does not solve my problem - FreePBX is updating the MySQL database fine, and it appears as if all my changes are sticking, until I actually look at the extensions_additional.conf file and realize no changes have been committed at all. In fact, the timestamp on the file hasn't even changed from when I made my manual edit. My question is, is there a way to forcibly tell FreePBX to dump all its MySQL database configuration settings into the /etc/asterisk files even if they have been manually modified? If FreePBX does this by default (overwriting any custom changes to the _additional files) then where might I start looking for the reason it's not doing it? Thanks. On 6/8/06, Tom Vile [EMAIL PROTECTED] wrote: you put custom code in the _custom.conf files not in the _addtional.conf files. You can also add the changes directly to the Mysql database as well and they will be included in the _addtional.conf files. On 6/8/06, Lachek Butalek [EMAIL PROTECTED] wrote: Figuring I knew what I was doing (I didn't - surprise) I added a totally unnecessary line in /etc/asterisk/extensions_additional.conf a couple of days ago. Troubleshooting a dialing rule issue, I'm now realizing that FreePBX is updating its database with the new settings but is not rewriting/updating extensions_additional.conf with the changes I'm making. I've tried renaming the file, changing its ownership, changing its permissions, restarting the portal, all without any success. Web resources on this issue claim the opposite problem - that custom changes to extensions_additional.conf will be automatically rewritten every time FreePBX/AMP is updated. If that was true, I'd be done - unfortunately, it seems this is not the case. I really don't want to reinstall FreePBX and redo my entire configuration again... :( Any assistance would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MWI on the PA168V in IAX mode?
Oh well. It would have been a nice feature, but with Asterisk's voicemail-to-email it's not really a necessity. Thanks for the information! On 6/7/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: http://www.aredfox.com/eqa.htm#line_10 Check this Dan On 08/06/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote: Thomas Kenyon wrote: Lachek Butalek wrote: I've gotten nothing from http://bbs.atcom.cn on this so far. Perhaps someone on the list has experience with this. Is there a way to get MWI support for PA168V-based ATAs? Afaik, none of the aredfox ATA firmware images support MWI, one reason I've never bought one. Apparently some IP phones based on the PA168V chip has this support already (Atcom AT-320 for example) Uses a PA168S. by configuring Asterisk with 'mailboxdetails=yes' in iax.conf. On my ATA, however, it does nothing. I think just stating mailbox=number will work too. 1. There is a PA1688 mailing list on Yahoo: http://groups.yahoo.com/group/pa1688/ 2. What firmware version do you have? Latest is 1.51 3. The PA1688 chip is being replaced by the AR1688, so new products will use those instead. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FreePBX 2.1.0: Manually rewriting extensions_additional.conf
Figuring I knew what I was doing (I didn't - surprise) I added a totally unnecessary line in /etc/asterisk/extensions_additional.conf a couple of days ago. Troubleshooting a dialing rule issue, I'm now realizing that FreePBX is updating its database with the new settings but is not rewriting/updating extensions_additional.conf with the changes I'm making. I've tried renaming the file, changing its ownership, changing its permissions, restarting the portal, all without any success. Web resources on this issue claim the opposite problem - that custom changes to extensions_additional.conf will be automatically rewritten every time FreePBX/AMP is updated. If that was true, I'd be done - unfortunately, it seems this is not the case. I really don't want to reinstall FreePBX and redo my entire configuration again... :( Any assistance would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI on the PA168V in IAX mode?
I've gotten nothing from http://bbs.atcom.cn on this so far. Perhaps someone on the list has experience with this. Is there a way to get MWI support for PA168V-based ATAs? Apparently some IP phones based on the PA168V chip has this support already (Atcom AT-320 for example) by configuring Asterisk with 'mailboxdetails=yes' in iax.conf. On my ATA, however, it does nothing. Any insights are appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P fails to initialize
Thanks to everyone for their tips and suggestions. I finally got the card working by using the YellowDog Linux kernel from ppckernel.org. There must have been some setting in the kernel config that made a difference because the card suddenly started working after that, even after a kernel recompile of 2.6.16.18 using the same config. On 6/3/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Jun 03, 2006 at 11:15:57PM +0200, Woodoo People .pGa! wrote: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.5 Echo Canceller: MG2 Failed to initailize DAA, giving up... wcfxo: probe of :00:0e.0 failed with error -5 these lines means, your x100p is not initialized - therefore cannot be used by zaptel. the problem below, reported by ztconfig. as the usage of zaptel device is following: modprobe zaptel modprobe module_of_card (like wcfxo) if it found, ztcfg if you see the device in /proc/zaptel/1 (or 2 or so) you can start asterisk and enjoy the device. you can believe me, this problem is in relation with irq sharing. (as i meet with that problem every time i have installed more than one card in a box - what i did more than 20 times) In one at least one board the problem was resolved by adding the boot parameter 'pci=noacpi' , IIRC. Which is, indeed, something in the neighbourhood of IRQ assignments. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I guess my server capacity is ok
What process is taking up 100% CPU? Is it Asterisk processes or something else? Also, is the load spread out over multiple processes, or do you have one or two processes taking up 90% or more of your total? You also have dual CPUs (and hyperthreading, which to FC3 should look like 4 CPUs if I'm not mistaken) - is the 100% CPU usage across all two (or four) processors, or is it only CPU1 that peaks at 100%? Have a look at Last Used CPU in top. What load are the other CPUs at? I don't have personal experience running that large of an installation, but I imagine your system specs would allow you to handle more simultaneous calls than 50, even though you're doing some transcoding. On 5/30/06, Goke Aruna [EMAIL PROTECTED] wrote: can someone overthere help? the server specs are as follows HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM, running fedora core 3 asterisk-1.2.5 ss7-0.8.3d. using sip as advised to receive calls from another gateway in US. using g729 in transcoding way. however, I noticed the call hit the 51 active calls which is 102channels, I run top to check the system resources usage and i discovered that the cpu is 100% used. asterisk, sip, ss7 never crashed throughout. however, since transcoding takes alot of system resources.. how can I use g729 in passthru mode. and I guess disabling hyperthreading will save me more system resouces. I will be glad, if you can give me more info on system management cos i think with that system, it should able to handle at least five E1's. I say thank you for finding time to reply my mail. goksie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: X100P fails to initialize
>From what I understood Zaptel was ported to the Mac quite some time ago. http://lists.digium.com/pipermail/asterisk-users/2004-October/060872.html Also, TerraSoft sponsored an Asterisk port to YellowDog Linux on PPC - from what I gather, with full Zaptel support. http://www.voip-info.org/wiki/view/Asterisk+Linux+Yellow+Dog I didn't think it's be necessary to run YDL to get Zaptel to work. Does anyone know if this is the case? From what I understand, as long as zaptel compiles it's up to udev and the kernel to do the remaining hardware detection and resource assignment, which should be a distribution-agnostic process. Is it possible that this particular chip used in this particular X100P clone is not supported on LinuxPPC? It is a Motorola 6508 chip, identical to the SM56 winmodem. A picture of the card is here: http://cgi.ebay.ca/Its-Real-X100P-FXO-card-oem-for-Digium-Asterisk-pbx_W0QQitemZ9730539695QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem#ebayphotohostingAppreciative for any help. On 5/25/06, Lachek Butalek [EMAIL PROTECTED] wrote: So I took a chance with an X100P knock-off on eBay. I'm running Asterisk + FreePBX on a PowerMac G3 (beige desktop) using Slackintosh 10.2 and kernel 2.6.16.16. Everything has been fine up until now. I compile the 1.2.5 Zaptel drivers without a problem, get the udev configuration in, modprobe zaptel, and finally modprobe wcfxo. At this point, I get the message:ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wcfxodmesg gives me:Zapata Telephony Interface Registered on major 196Zaptel Version: 1.2.5 Echo Canceller: MG2Failed to initailize DAA, giving up... wcfxo: probe of :00:0e.0 failed with error -5syslog tells me:May 25 21:28:20 asterisk kernel: Failed to initailize DAA, giving up...May 25 21:28:20 asterisk kernel: wcfxo: probe of :00:0e.0 failed with error -5 ztcfg -vv says:Channel 01: FXS Kewlstart (Default) (Slaves: 01)1 channels configured.ZT_CHANCONFIG failed on channel 1: No such device or address (6)and lspci -vv tells me this:00: 0e.0 Communication controller: Motorola: Unknown device 5608 Subsystem: Motorola: Unknown device Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 24 Region 0: I/O ports at fe000800 [size=256] Region 1: Memory at 81803000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=0mA PME(D0+,D1-,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- One of the reasons I'm running on a PowerMac is specifically because I've been told the X100P cards work well on this type of hardware, since it tends to not have problems with flawed PCI buses and IRQ sharing, but I'm starting to have my doubts. If anyone has experience with X100P cards on PPC, or have any other insights, it would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P fails to initialize
So I took a chance with an X100P knock-off on eBay. I'm running Asterisk + FreePBX on a PowerMac G3 (beige desktop) using Slackintosh 10.2 and kernel 2.6.16.16. Everything has been fine up until now. I compile the 1.2.5 Zaptel drivers without a problem, get the udev configuration in, modprobe zaptel, and finally modprobe wcfxo. At this point, I get the message:ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wcfxodmesg gives me:Zapata Telephony Interface Registered on major 196Zaptel Version: 1.2.5 Echo Canceller: MG2Failed to initailize DAA, giving up... wcfxo: probe of :00:0e.0 failed with error -5syslog tells me:May 25 21:28:20 asterisk kernel: Failed to initailize DAA, giving up...May 25 21:28:20 asterisk kernel: wcfxo: probe of :00:0e.0 failed with error -5 ztcfg -vv says:Channel 01: FXS Kewlstart (Default) (Slaves: 01)1 channels configured.ZT_CHANCONFIG failed on channel 1: No such device or address (6)and lspci -vv tells me this:00:0e.0 Communication controller: Motorola: Unknown device 5608 Subsystem: Motorola: Unknown device Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 24 Region 0: I/O ports at fe000800 [size=256] Region 1: Memory at 81803000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=0mA PME(D0+,D1-,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- One of the reasons I'm running on a PowerMac is specifically because I've been told the X100P cards work well on this type of hardware, since it tends to not have problems with flawed PCI buses and IRQ sharing, but I'm starting to have my doubts. If anyone has experience with X100P cards on PPC, or have any other insights, it would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-1001 behind NAT - Internet Asterisk box -- BOUNTY!
Stupid not-quite-an-answer - if you're willing to pay money for a fix, why not buy an IAX2 compatible FXS to replace the SPA-1001 with? It will traverse NATs without a problem. I'm using a GNet VP168I (same as the PA168V) and it works fine even behind a NAT which is itself behind a corporate firewall... Just a thought.On 5/17/06, Eric Lyons [EMAIL PROTECTED] wrote: I'm still unable to get my SPA-1001 to work behind NAT with an Asterisk box out on the Internet.It works fine to my local [EMAIL PROTECTED] box.I've tried... many things.I'm willing to pay a $250 bounty (PayPal preferred) to anyone who can help me get it working.Any Sipura experts out there? Eric.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP adapters to connect PSTN lines to SIP phones
I have no hands-on experience with this product, but it seems that the Mediatrix 1204 would do the trick for you:http://www.twacomm.com/catalog/model_MEDIATRIX-1204.htm There are probably lots of other ones as well out there. As the previous poster indicated, you're looking for 4-port FXO adapters with SIP support.On 5/15/06, Alex Robar [EMAIL PROTECTED] wrote: Mike,Yes, this is absolutely possible. You're just looking for FXO adapters. Off the top of my head, look for Linksys/Sipura, Grandstream Handy Tone, and Aastra devices for this purpose.Alex On 5/15/06, Mike [EMAIL PROTECTED] wrote: Hi, I have a question on VoIP adapters. As far as I understand, those adapters are usually used to connect DSL/Cable access to a normal phone (Internet to Adapter, then to PSTN phones). I want to know if you can use those adapters to do the opposite: connect a few lines (1-4 let`s say) to the adapters, then deliver via SIP to an Asterisk box. (I know I could use a TDM400 and Asterisk, but I want to avoid needing a computer at all on site and deliver via SIP to another site). So, is this possible, and with which product? Thank you, Mike ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP adapters to connect PSTN lines to SIP phones
Sorry, didn't mean to direct you to a vendor site - here's the mfg's website:http://www.mediatrix.com/products_devices.php?prodid=13 On 5/15/06, Lachek Butalek [EMAIL PROTECTED] wrote: I have no hands-on experience with this product, but it seems that the Mediatrix 1204 would do the trick for you: http://www.twacomm.com/catalog/model_MEDIATRIX-1204.htm There are probably lots of other ones as well out there. As the previous poster indicated, you're looking for 4-port FXO adapters with SIP support. On 5/15/06, Alex Robar [EMAIL PROTECTED] wrote: Mike,Yes, this is absolutely possible. You're just looking for FXO adapters. Off the top of my head, look for Linksys/Sipura, Grandstream Handy Tone, and Aastra devices for this purpose.Alex On 5/15/06, Mike [EMAIL PROTECTED] wrote: Hi, I have a question on VoIP adapters. As far as I understand, those adapters are usually used to connect DSL/Cable access to a normal phone (Internet to Adapter, then to PSTN phones). I want to know if you can use those adapters to do the opposite: connect a few lines (1-4 let`s say) to the adapters, then deliver via SIP to an Asterisk box. (I know I could use a TDM400 and Asterisk, but I want to avoid needing a computer at all on site and deliver via SIP to another site). So, is this possible, and with which product? Thank you, Mike ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
