[Asterisk-Users] TFTP Secondary Ports
Im publishing tftp through my firewall to support external Cisco 7960 sip phones. I know that the primary port is 69 for tftp. However, tftp also uses secondary ports ranging from 1,0XX to 30,XXX. ( A broad range) In an effort to limit the secondary ports that are opened, some Windows based tftp server such as the winagents product allows you to limit the range of secondary ports that are used allowing you to somewhat tighten firewall publishing rules. Does anyone know how to do this using the linux tftp server? Thanks, Chad ___ 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] Has Sixtel gone under?
If you have an account you can try: http://control.sixtel.net This works and they seem to be adding some features. My service still works. However sixtel has been unable to tell me how much $ is available for use. I'm not too confident at this point. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Espinoza Sent: Tuesday, August 02, 2005 4:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Has Sixtel gone under? That's always been the site at that url. On 8/2/05, Tony Hoyle [EMAIL PROTECTED] wrote: Carlos Chavez wrote: I have been using Sixtel from the beginning of the year and service was getting worse and worse. Yesterday I tried to access the website to get the CDR and I got an error saying that the domain no longer exists. I checked the whois and it says that the domain is on hold. Have they finally folded? http://www.sixtel.net/voip/ doesn't look too promising... Tony ___ 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] invalid extension dilemma
In the example below if I dial valid extension 1000, the Invalid context plays pbx-invalid as it is included with _7 context. Include voicemail in the main context. ___ 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] same extension on multiple sip phones?
I have a need to have the two sip phones register with the same extension (at least I think I have the need :) Consulting the wiki about the dialplan and the dial application reveals that you can dial several phones at once, or in series, whichever you wish. Dial(SIP/2000SIP/2001) will do the former Dial(SIP/2000,15) Dial(SIP/2001,30) will do the latter. Now you just have to remember which is which, former and latter :) ___ 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] same extension on multiple sip phones?
One way to do this would be to create a call queue with the two sip phones as separate extensions connected to the one logical extension (the queue). The other, and possibly simpler way to do it is to use Dial(SIP/extensionSIP/extension) to ring both sip phones at the same time. Regardless, you can't have two sip phones try to register to the same account. It's all in the dialplan. Aaron Picht -Original Message- From: Kevin Hanson [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 02, 2005 10:19 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] same extension on multiple sip phones? I have a need to have the two sip phones register with the same extension (at least I think I have the need :) A client wants an incoming call to ring at the receptionists desk and also at their desk. If the receptionist is in it will be answered there and put on hold followed by a Joe, you have a call on line 1. Is there a way to do this w/ asterisk? I've played with two phones with same sip registration and it seems the last one to register is the one asterisk recognizes. Thanks, Kevin ___ 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] ASTCC: different incriments
Please see comments inline. Rusty Shackleford wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Tuesday, July 26, 2005 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ASTCC: different incriments How can I fulfill that? *Billing Increments* Continental USA: six (6) second increments. International: thirty (30) seconds minimum and six (6) seconds thereafter. Mexico: sixty (60) seconds minimum and six (6) seconds thereafter. The billing increment is set in the brands table. When you create cards, this value is copied into the inc column in the cards table. (I'll spare us the rant on normalization here...) :-) Increments should only come out of the brands table IMHO, but I'm not sure how to fix it without causing breakage. Could be I just don't have time. :-) The per call minimum is set in the includedseconds column, in the routes table. This value, along with the value of the connectcost column for a given record (route) is used to compute the cost of the call. In ASTPP, I have added support to have different rates depending on the brand. It would not be that hard to port to ASTCC but I'm not sure how to do it without breaking existing installations. So, in theory, you set all your cards for 6 second increments, and you set your routes to 6, 30, or 60 includedseconds. That's the theory, but the stock ASTCC code has a bug in the way it makes this computation. Darren has reopened the bug report. On this subject, does anybody have feedback on the bug? They want external testing feedback before doing anything with it. http://bugs.digium.com/view.php?id=4479 Darren Wiebe [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] How to let ZAPHFC work with and act on different incoming MSNs?
On Wed, Aug 03, 2005 at 12:38:17AM +0200, Michel Koenen wrote: I have this working with a Teles ISA card, see config below (numbers are changed because I dont want everybody to call me;-) ) In modem.conf ZapHFC is configured in zapata.conf, not in modem.conf, right? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] port forwarding ip to ip sip calls
I've got two pa1688 phones that I want to set up to communicate between branch offices without a gatekeeper. Both phones will be behind a firewall and I want to use port forwarding so the phones can communicate. Are you using these phones with SIP? Why not try IAX2? I tested the phones behind a firewall on the same network segment and there were no problems at all using sip. However, I then moved the phones into situ and port forwarded udp on 5060 and 1 - 2 at both branch offices firewalls. I set the rcp port to 1 and the sip port to 5060. The phones were able to ring each other, however, there was no sound on both ends. Can some one please tell me which ports I have to open in order to make communications between the two branch offices using these phones. Or share a config or suggest another protocol so I can make this happen. Check for nat=yes and canreinvite=no in sip.conf ___ 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] IAX2, can't receive calls
I have IAX2 (FWD) partially working. I can place calls from my Asterisk box but I cam unable to receive them (comes back as busy). I have my firewall forwarding the udp ports 5060, 4569, 5036 and 1 thru 2 to my asterisk server. I think I have the firewall correctly setup as I can forward other services to their appropriate servers. I have no mail box on the one account (the one I'm testing to). I've followed the FWD instructions but I've had no luck. what does iax2 show register and iax2 show peers show wrt FWD? ___ 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.conf question
On Tue, Aug 02, 2005 at 05:47:50PM -0400, Tim King wrote: # It must be in the module loading order # Span 1: WCTDM/1 Wildcard TDM400P REV I Board 2 fxoks=1 fxoks=2 fxoks=3 fxoks=4 # Span 2: WCTDM/2 Wildcard TDM400P REV I Board 3 fxsks=5 fxsks=6 fxoks=7 fxoks=8 # Global data loadzone = us defaultzone = us Is this creating a problem because of the two FXO ports being in the middle of the FXS ports? This should not be a problem. Have you changed the order since you last run genzaptelconf? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Nat Transversal
the extension register ok on asterisk server , but not audio is transmited on answer a call look for canreinvite=no in sip.conf ___ 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] sip ata's
Hello. I have a linux and two sip-ata's, a sipura 2002 and a GS ht-386. I also have three sipphone numbers. I can connect the atas to the sipphone accounts and I get a dial tone and I can call my house and it says, Thank you for using SipPhone... Using asterisk, I have the ata's registering to my computer and I register two sipphone numbers with my computer. When I pick up the phone I don't get a dialtone. I can use kphone and call a sipphone and the logs come back saying I have phone on hook, phone is off the hook, and one phone rings usually, one comes back busy (in log). I pick-up the phone and nobody is there and then the asterisk-voicemail kicks in. I guess I have two questions: Where is the dial-tone? I noticed I compiled phone sounds but my ata has a dial-tone when its not serviced. My grandstream 386 has 2 fxs's. One of them clicks on and off and on and off when I pick up the receiver even though it rings when I call it. I have it set up the same as the other port as best as I can. I think it may be a setting on the 386 that I'm not seeing. Is there anyone aware of what causes this? I also noticed that when the call is handled by asterisk there is an invite. Is this a reinvite and where do the canreinvite/reinvites go? ___ 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] port forwarding ip to ip sip calls
can you give me more details ? like : are you using one asterisk server in public ip and two phones behind NAT or two asterisk servers both are behind NAT and haveing phones connected locally one with each other... after that i can help u - Original Message - From: Oliver Bode [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 02, 2005 4:56 PM Subject: [Asterisk-Users] port forwarding ip to ip sip calls Hi, I've got two pa1688 phones that I want to set up to communicate between branch offices without a gatekeeper. Both phones will be behind a firewall and I want to use port forwarding so the phones can communicate. I tested the phones behind a firewall on the same network segment and there were no problems at all using sip. However, I then moved the phones into situ and port forwarded udp on 5060 and 1 - 2 at both branch offices firewalls. I set the rcp port to 1 and the sip port to 5060. The phones were able to ring each other, however, there was no sound on both ends. Can some one please tell me which ports I have to open in order to make communications between the two branch offices using these phones. Or share a config or suggest another protocol so I can make this happen. Thanks, Oliver ___ 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] TFTP Secondary Ports
On Tue, Aug 02, 2005 at 10:46:17PM -0700, Chad Brown wrote: I'm publishing tftp through my firewall to support external Cisco 7960 sip phones. I know that the primary port is 69 for tftp. However, tftp also uses secondary ports ranging from 1,0XX to 30,XXX. ( A broad range) In an effort to limit the secondary ports that are opened, some Windows based tftp server such as the winagents product allows you to limit the range of secondary ports that are used allowing you to somewhat tighten firewall publishing rules. The secondary ports are not determained by the server. Rather, they are set by the client, IIRC. File transfers simply re-use the existing socket that was used to connect in the first place. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] port forwarding ip to ip sip calls
hi but i don't think IAX2 is good, because with IAX2 RTP packets goes via IAX servers as mini packets not directly from one client to other client so for a big implementation it may consume more bandwith then that of a SIP solution rest is up to the user... - Original Message - From: Wilson Pickett [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 02, 2005 11:07 PM Subject: Re: [Asterisk-Users] port forwarding ip to ip sip calls I've got two pa1688 phones that I want to set up to communicate between branch offices without a gatekeeper. Both phones will be behind a firewall and I want to use port forwarding so the phones can communicate. Are you using these phones with SIP? Why not try IAX2? I tested the phones behind a firewall on the same network segment and there were no problems at all using sip. However, I then moved the phones into situ and port forwarded udp on 5060 and 1 - 2 at both branch offices firewalls. I set the rcp port to 1 and the sip port to 5060. The phones were able to ring each other, however, there was no sound on both ends. Can some one please tell me which ports I have to open in order to make communications between the two branch offices using these phones. Or share a config or suggest another protocol so I can make this happen. Check for nat=yes and canreinvite=no in sip.conf ___ 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] TFTP Secondary Ports
hey chad, just a heads up tftp is one of the worst protocols to use when your behind a nat or firewall it drove me pretty crazy a while ago. Carlos AlcantarRace Technologies, Inc.101 Haskins WaySouth San Francisco, CA 94080P: 650.246.8900F: 650.246.8901E: carlos at race.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad BrownSent: Tuesday, August 02, 2005 10:46 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] TFTP Secondary Ports Im publishing tftp through my firewall to support external Cisco 7960 sip phones. I know that the primary port is 69 for tftp. However, tftp also uses secondary ports ranging from 1,0XX to 30,XXX. ( A broad range) In an effort to limit the secondary ports that are opened, some Windows based tftp server such as the winagents product allows you to limit the range of secondary ports that are used allowing you to somewhat tighten firewall publishing rules. Does anyone know how to do this using the linux tftp server? Thanks, Chad ___ 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] CISCO 7960 with Asterisk
Hi, We are trying to set up an asterisk configuration using some 7960 Cisco Telephone. We need to deploy those in our company and we also need to see on the screen who is on line or not. After making a research on the web, we thing that we have to use MGCP or sccp. Does anybody have the last firmware of Cisco 7960 to work either in SCCP or MGCP? Rgds, Nicolas ___ 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] FXO PCI Master abort (What does it take)
I'm similarly exacerbated over the FXO PCI Master Abort thing. Right now, I'm totally stuck! I dont have much more info to give, but I'm SURE somebody on this list is running a X101P card (ambient md3200), on linux. I can't see how they can have failed to come across the same problem - since I've now tried 3 different kernels, 2 different snapshots of zapatel, 2 different H/W platforms, and 2 different cards Can somebody at least say that they have it working with no problem? I've seen a number of these questions go un-answered.. are people who get these errors simply giving up on Asterisk? Cheers Mark. On 2 Aug 2005, at 10:06, Mark Burton wrote: Hi, I have the following configuration, which doesn't seem to work, any help much appreciated Linux 2.6.11 used to run asterisk CVS version of zaptel X101P So far, so easy. However, whenever I turn the machine on with the card in, I get FXO PCI Master abort errors. Depending on the way it feels, either these are repeated till /var/log/ is full, or I get one and then the thing hangs. This may, or may not, have something to do with a message Uhhuh. NMI received. Dazed and confused, but trying to continue You probably have a hardware problem with your RAM chips I have tried all 4 combinations of a) stock debian builds of zaptel, and cvs head versions b) an old pentium 2 machine, and a new (ish) P4 mahcine In all cases with the same result. I have also tried the new machine with linux 2.6.8 -- yup -- same result... I've mucked with the IRQ's till they dont conflict.. no change... So, I'm clearly deluded as everybody else seems to have no problem. Can anybody help - what silly thing have I done? Cheers Mark. ___ 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] SIP WEB Phone (Wanna implement Click to Call)
Hello Tim, I am definitely interested in testing it. Please contact me off the list. Best Regards, Boris. If anyone is interested I'm (slowly) developing a GPL'd Java applet that works as an IAX softphone. I should have a test version out at the end of the week for a limited number of testers. Tim. ___ 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] TFTP Secondary Ports
I understand. However, Im successfully managing this without any problems using a Windows tftp server by www.winagents.com. This software allows you to limit secondary transfer connections to a range of IPs. Therefore you only need to open up port 69 and the range you specify. Everything just works! I would like to move the solution to Linux for a couple reasons. However, It looks like the default tftp server does not support this feature and that is why you were going crazy. The number of ports you must open is ridiculous for tftp. However, I just found a seemingly robust linux version with firewall support offered by weirdsolutions. It looks promising. http://www.weirdsolutions.com/ Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Sent: Wednesday, August 03, 2005 12:10 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] TFTP Secondary Ports hey chad, just a heads up tftp is one of the worst protocols to use when your behind a nat or firewall it drove me pretty crazy a while ago. Carlos Alcantar Race Technologies, Inc. 101 Haskins Way South San Francisco, CA 94080 P: 650.246.8900 F: 650.246.8901 E: carlos at race.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Brown Sent: Tuesday, August 02, 2005 10:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TFTP Secondary Ports Im publishing tftp through my firewall to support external Cisco 7960 sip phones. I know that the primary port is 69 for tftp. However, tftp also uses secondary ports ranging from 1,0XX to 30,XXX. ( A broad range) In an effort to limit the secondary ports that are opened, some Windows based tftp server such as the winagents product allows you to limit the range of secondary ports that are used allowing you to somewhat tighten firewall publishing rules. Does anyone know how to do this using the linux tftp server? Thanks, Chad ___ 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] Installing a TE100P (Digium) card over Suse 9.2..
Hi everybody, I managed to install card over Suse 9.2, I substituted Zaptel drivers and compiled them. Now "ztcfg" says I have one card with correctly configured 31 channels, but red led on back of card doesn't flash. Suse 9.2 has detected the card as a Tiger Jet card, since the chip on it is a Tiger 320. The second card configuration is still waiting for configuration, but I think this can be bad for Zaptel drivers. Has someone done something like this? Regards Mauro Zanin ___ 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] Music on Hold: CPU Intensive Monster
Hi Matthew, i found the following link very usefull: http://www.orderlyq.com/asteriskqueues.html#moh It is an alternativ to mpg123. It works very fine for me. Regards Matthew Boehm wrote: OK. So I did a test last night. All of asterisk's threads where using 0.0% CPU. I made 1 call to our call queue. CPU jumped to average of 9% and stayed around that for the 2 minutes I was in the queue just listening to music on hold. MOH is in MP3 format and I'm using format_mp3. Phone was linksys PAP2-NA using G729. Can I reasonably assume that the 9% was decoding the MP3, then encoding G729? I tried using Anthm's RAW format but that actually made things worse. I tried going back to mpeg321 and asterisk still used the same amount of CPU. Any ideas for getting processor usage down on MOH? -Matthew ___ 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] same extension on multiple sip phones?
U can use this way in extensions.conf:
exten = 2,1,Dial(${BRUNO_FGA}${GIORGIO_FGA},${RING_TIME}) ; supp-tecnico
Bruno
Kevin Hanson wrote:
I have a need to have the two sip phones register with the same
extension (at least I think I have the need :)
A client wants an incoming call to ring at the receptionists desk and
also at their desk. If the receptionist is in it will be answered
there and put on hold followed by a Joe, you have a call on line 1.
Is there a way to do this w/ asterisk? I've played with two phones
with same sip registration and it seems the last one to register is
the one asterisk recognizes.
Thanks,
Kevin
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
BRUNO DE LUCA
Tel. +39 02 9350 4780 (102)
FGA Software
20017 Rho - Via Puccini, 8
E-Mail :
[EMAIL PROTECTED]
Internet:
http://www.fgasoftware.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] invalid extension dilemma
u can use this:
exten = i,1,Playback(invalid_selection)
exten = i,2,Goto(inbound_menu,_X.,1)
Bruno.
Joseph wrote:
Ho do you folks solve the problem with invalid extension when someone
dials a wrong number?
For example if somebody dial prefix _7 I want to allow tall free
numbers from that line but not a long distance. However, if somebody
dial
wrong number I want to play invalid extension instead of congestion.
In the example below if I dial valid extension 1000, the Invalid
context plays pbx-invalid as it is included with _7 context.
[goto-dialout]
exten = _9.,1,SetMusicOnHold(loud)
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _9.,3,Hangup()
exten = _71800XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _71866XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _71877XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _71888XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _7NXX,1,SetMusicOnHold(loud)
exten = _7NXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _7NXX,3,Hangup()
include = invalid
[invalid]
exten = _.,1,NoCDR()
exten = _.,2,Playback(pbx-invalid)
exten = _.,3,Hangup()
[voicemail]
exten = 1000,1,NoCDR()
exten = 1000,2,Answer()
exten = 1000,3,VoicemailMain(${CALLERIDNUM})
exten = 1000,4,Hangup()
--
BRUNO DE LUCA
Tel. +39 02 9350 4780 (102)
FGA Software
20017 Rho - Via Puccini, 8
E-Mail :
[EMAIL PROTECTED]
Internet:
http://www.fgasoftware.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] port forwarding ip to ip sip calls
Ashish Raikwar wrote: can you give me more details ? like : are you using one asterisk server in public ip and two phones behind NAT or two asterisk servers both are behind NAT and haveing phones connected locally one with each other... after that i can help u - Original Message - From: Oliver Bode [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 02, 2005 4:56 PM Subject: [Asterisk-Users] port forwarding ip to ip sip calls No gatekeeper, no asterisk. These phones can communicate by simply dialing the ip address of the other phone - well that's how it worked when I was on the same network segment. ___ 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] Polycom Soundpoint 500
Try to control the file in the server... i have seen that this phone change the server file in an wrong way... Bruno. Brent Davidson wrote: I have a Polycom Soundpoint IP 500 that I have been using with Asterisk for a few weeks. It has been working OK, no major problems other than a freeze up every now and then, until today. The power apparently went out last night and for some reason the phone appears to be working but I keep getting the following errors repeating over and over in my Asterisk log file (IP's X'ed out): Aug 2 15:48:49 NOTICE[11606]: chan_sip.c:9405 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]:5060' failed for 'XX.XX.XX.XX' Aug 2 15:48:50 NOTICE[11606]: chan_sip.c:9299 handle_request_subscribe: Failed to authenticate user 7202 sip:[EMAIL PROTECTED]:5060;tag=CD6D3F82-1211688D for SUBSCRIBE Aug 2 15:48:52 NOTICE[11606]: chan_sip.c:9299 handle_request_subscribe: Failed to authenticate user 7202 sip:[EMAIL PROTECTED]:5060;tag=CFBF905B-DD972A1A for SUBSCRIBE Aug 2 15:48:53 NOTICE[11606]: chan_sip.c:9299 handle_request_subscribe: Failed to authenticate user 7202 sip:[EMAIL PROTECTED]:5060;tag=24939F70-451E5F93 for SUBSCRIBE Aug 2 15:48:55 NOTICE[11606]: chan_sip.c:9405 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]:5060' failed for 'XX.XX.XX.XX' Aug 2 15:48:56 NOTICE[11606]: chan_sip.c:9299 handle_request_subscribe: Failed to authenticate user 7202 sip:[EMAIL PROTECTED]:5060;tag=2E59724E-73F0A849 for SUBSCRIBE The phone has two lines, extension 7202 and 7203. I don't receive any messages regarding 7203, and the two sip profiles are identical in the sip.conf file (with teh exception of substituting 7202 for for 7203) and I have retyped the password into the phone more times than I can count. Now the odd thing is that the phone can make and receive calls, they are just very choppy when calling IAX extensions. When the calls go to/from the Polycom from/to a Zap channel, the calls are perfectly clear. I am completely lost at this point. Any ideas? Thanks, Brent Davidson ___ 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 -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.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] Gmail and the list
Gmail users, I had the similar problem, but I discovered that all my mail for 30-31 july was delivered into my junk folder. Then I selected them all and move then to the inbox. Since then I have been receiving mail from the list goksie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Wednesday, August 03, 2005 4:02 AM To: Michel Koenen; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Gmail and the list or the list server is only making it part of the way into the lists of addresses. I am using gmail and even got some mail last weekend when it was all but dead. Like slasdot, early subscribers benefit :) On 8/1/05, Michel Koenen [EMAIL PROTECTED] wrote: Same here, nothing is coming in anymore on my gmail address neither. I read your posting by going to the web version of the list. Maybe gmail is blocking mail from the list or is it really some configuration setting in the list itself ? ___ 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 -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ 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.org beta site up!
Matt Brooks wrote: I am just emailing to inform you guys that a new website has been created for asterisk.org. You can find the beta site up at http://beta.asterisk.org. It utilizes the drupal portal framework and Looking very good and much easier to navigate! Great work! Cheers, Kristof ___ 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] same extension on multiple sip phones?
Kevin, can I make a suggestion that you look at ring groups (possibly
even download [EMAIL PROTECTED] - as you can implement ring groups really
easy using AAH).
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Bruno De Luca
Sent: Wednesday, 3 August 2005 3:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] same extension on multiple sip phones?
U can use this way in extensions.conf:
exten = 2,1,Dial(${BRUNO_FGA}${GIORGIO_FGA},${RING_TIME}) ;
supp-tecnico
Bruno
Kevin Hanson wrote:
I have a need to have the two sip phones register with the same
extension (at least I think I have the need :)
A client wants an incoming call to ring at the receptionists desk
and
also at their desk. If the receptionist is in it will be answered
there and put on hold followed by a Joe, you have a call on line
1.
Is there a way to do this w/ asterisk? I've played with two phones
with same sip registration and it seems the last one to register is
the one asterisk recognizes.
Thanks,
Kevin
___
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
--
BRUNO DE LUCA
Tel. +39 02 9350 4780 (102)
FGA Software
20017 Rho - Via Puccini, 8
E-Mail :
[EMAIL PROTECTED]
Internet:
http://www.fgasoftware.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
[Asterisk-Users] DND Indication
Hi, Has anyone come up with a clever way of indicating DND is activated? I've thought of stutter dial tone and using the mwi, but have no idea how to implement these. I'm using Budgetones. My concern is that users will activate the DND, then forget about it not realizing that they are not receiving calls. Thanks, G ___ 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] TFTP Secondary Ports
Chad Brown wrote: I'm publishing tftp through my firewall to support external Cisco 7960 sip phones. I hope the files requested by the Cisco phones don't contain username / password information. Passing that in cleartext is just so wrong ;-) -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] app_intercept
Hi, Can anyone give me any information at all to get app_intercept working? I've found these pages, but there is just not enough for me to get it going. http://www.pbxfreeware.org/archives/2005/06/new_download_--.html and http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002692 Thanks, G ___ 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] How to let ZAPHFC work with and act on different incoming MSNs?
I have this working with a Teles ISA card, see config below (numbers are changed because I dont want everybody to call me;-) ) In modem.conf ZapHFC is configured in zapata.conf, not in modem.conf, right? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is Yes, I know but I gave the modem.conf example to show you how it is working with the Teles card. The question was how to get the same thing working with zaphfc. In the mean time I spent some more time to experiment and I found out that with zaphfc I can make use of DID to get the same results. This page gave the solution: http://voip-info.org/tiki-index.php?page=Asterisk+tips+did I just had to set immediate=no and overlapdial=yes to get it working. There is only one tiny issue left: the MSN via modem.conf is delivered as extension 402901 to the dial plan, while the MSN via zapata.conf is delivered with an extra 0 prepended so 0402901. This means that I cannot make use of the same context when using both cards. Does anybody know how to preprocess the extension before it is send to the dialplan context so that the MSN is always presented the same regardless of via which channel it is coming in? Best regards, Michel ___ 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] Astcc Charging \ Matching Pattern Problem - SOLVED
Darren's suggestion did the trick, thanks. Keep up the good work!!! Ade.Darren Wiebe [EMAIL PROTECTED] wrote: You should have your pattern set to ^4207. Then the pattern has to start with 4207. The way my setup would be is ^0114207.Darren Wiebe[EMAIL PROTECTED]Ade Agbero wrote: Astcc applies a charge for Czech Republic - Mobile Code - 4207 to a call destined for UK Landline 44207. It appears Astcc uses the first matching pattern of 4207 it finds in the routes table instead of continuing to search through the routes table until it comes to 44207 for UK. Any ideas on how to resolve this problem. Thanks, Ade. How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. *Get Yahoo! Photos* ___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 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 To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre.___ 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] Re: What does pbx-wilcalu.so do and why does it keep crashing my * box?
Mark Phillips schrieb: I downloaded the latest CVS a few days ago. It all compiled nicely on my new AAH platform. However, it won't start up. Investigation of my log files produces this; Jul 26 22:59:18 VERBOSE[31473] logger.c: [pbx_wilcalu.so] Jul 26 22:59:18 VERBOSE[31473] logger.c: [pbx_wilcalu.so] Jul 26 22:59:18 WARNING[31473] loader.c: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Jul 26 22:59:18 WARNING[31473] loader.c: Loading module pbx_wilcalu.so failed! I think I got this error when I updated from stable 1.0.7 to 1.0.9 via CVS and had not deleted everything in the asterisk modules directory before installation. Regards, Gunde ___ 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] 7970 SIP
Nkm [EMAIL PROTECTED] : On 8/2/05, Darren Wright wrote: Can anyone point me to the location of the 7970 SIP image? I'm logged There's no SIP firmware for 7970, only SCCP firmware. Am I right? Sergio ___ 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] Full T38 sip Faxing now Available
Carlos [EMAIL PROTECTED] lazily top-posted: Has anyone got a response from this? It was just spam. Forget it. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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] call does not hangup after client quits
Hi, I'm seeing a problem where if I place a call, then forcibly quit or turn off the client the call stays active. The frames counters stop so its apparent the client has gone away but the call remains active. Asterisk is CVS-HEAD 23-Jun-05 What is supposed to happen in this scenario? 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] Database querie
Hi Guys Just a quick question. Does * write directly into PGSQL database like MySQL? Kind Regards Terry Wade Mobile: +27 82 802-5750 Office: +27 11 784-7642 Fax: +27 11 388-0855 Linux is like a Wigwam - No gates, no windows, Apache inside Disclaimer and Confidentiality Warning This message is intended for the addressee only. If you are not the intended recipient of this message, you are notified that any distribution, use of or copying of this communication is strictly prohibited. If you have received the communication in error, please notify the sender immediately. The views and opinions expressed in this message are those of the individual sender of this message and do not necessarily represent the views and opinions of ActiCom. Consequently, ActiCom does not accept responsibility for such views and opinions and this message should not be read as representing the views and opinions of ActiCom without subsequent written confirmation. Each page attached hereto must also be read in conjunction with this disclaimer. ___ 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] LG Goldstar GDK-186/162 question on voicemail
Are there any other GDK users out there with Asterisk? Ive got all the integration working, except voicemail. Does anybody know a way of disabling the forward to voicemail on a per extension or per DDI basis (I can disable the voicemail hunt group but then I cant light the MWI indicators as it seems that only ports marked in the voicemail group can issue the MWI on/off commands). Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk___ 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.org beta site up!
Kristof Hardy [EMAIL PROTECTED] wrote: Matt Brooks wrote: I am just emailing to inform you guys that a new website has been created for asterisk.org. You can find the beta site up at http://beta.asterisk.org. It utilizes the drupal portal framework and Looking very good and much easier to navigate! Great work! Well, at least the new website doesn't say that I can register today to participate in an event that took place last June (see www.asterisk.org). -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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] app_dbodbc for asterisk stable 1.09
Hi, Has anyone manage to comile app_dbodbc or ast_data with the latest stable release (1.09). If so can you give some guidence on howto do it as I have trouble getting either working. Umar ___ 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] Is there an upper extension limit to Asterisk?
Hello I have an application for Asterisk which could involve potentially 5000 or more extensions. Possibly this number of people making calls. All calls would be internal. Could enough hardware be thrown at the problem to make this work? Anyone setup an installation of this size? Any comments on how to size it, etc? Angus ___ 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] Has Sixtel gone under?
I just checked my account via https://secure.inetm.net and my balance is visible where it always has been on the billing activity page. *shrug* -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Brown Sent: August 3, 2005 01:55 To: Erik Espinoza; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Has Sixtel gone under? If you have an account you can try: http://control.sixtel.net This works and they seem to be adding some features. My service still works. However sixtel has been unable to tell me how much $ is available for use. I'm not too confident at this point. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Espinoza Sent: Tuesday, August 02, 2005 4:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Has Sixtel gone under? That's always been the site at that url. On 8/2/05, Tony Hoyle [EMAIL PROTECTED] wrote: Carlos Chavez wrote: I have been using Sixtel from the beginning of the year and service was getting worse and worse. Yesterday I tried to access the website to get the CDR and I got an error saying that the domain no longer exists. I checked the whois and it says that the domain is on hold. Have they finally folded? http://www.sixtel.net/voip/ doesn't look too promising... Tony ___ 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
[Asterisk-Users] SIP phone procedural question
Hello A lot of my customers have people who are in the office most of the time but occasionally wish to work from home. So they may have a sip phone which is extension 208 in the office. When they work from home they can of course plug in a sip phone into their broadband connection and work with that. But it would be ideal if they could be same extension as phone in office. If they try to register as same sip user - eg extn 208 - will it work. Then problem is phone on their desk will still ring p***ing all their colleagues off. How do people deal with this sort of thing. Ideally, would want person to be able to easily switch from office to home but use same extension. Or does sip somehow deal with this? Is there a standard sip way of dealing with this? Angus ___ 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] Two questions about Asterisk Call Center
Hello, routing based on DNIS is dependant on what your telco sends you. Usually on Robbed-bit T1s(RBS) they will send you ANI and DNIS together separated by stars like this: *7275551212*1234* (where 7275551212 is the ANI[callerID] and 1234 is the DNIS[last 4 digits of the number dialed]) In Asterisk this shows up all as the exten and you need *NXXNXX*1234 in your dialplan. If you have PRI T1s then you can usually receive both the CallerID and the full 10-digit number dialed from the carrier and you will get the full number dialed as the extension, so 8881231234 in your dialplan. Collecting wrapup codes is another thing. This means you need a database for the calls coming in and in case of Asterisk that means tinkering with the code. There are several add-ons that add this functionality to Asterisk and some of them cost money, just do a search for queues and agents in Asterisk on google. Or you could go with a package like Aheeva or VICIDIAL that have GUI interfaces and allow you a great deal more interoperability with other systems and the ability for the agent to enter more info. MATT--- -Original Message- From: Tielin Xu [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 02, 2005 2:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Two questions about Asterisk Call Center Hi: I am new at Asterisk. Does anyone know how to define the call routing based on DNIS as our conventional ACD to route a call in Asterisk? Second, how do I collect Wrap-Up code for agents in Asterisk? Many thanks. Tielin ___ 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] Full T38 sip Faxing now Available
Kevin Walsh wrote: Carlos [EMAIL PROTECTED] lazily top-posted: Has anyone got a response from this? It was just spam. Forget it. I have an account with them, just waiting for a suitable ATA to arrive. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo 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
[Asterisk-Users] Anyone know of an open source sip video phone like eyebeam available?
I just wondered - might save me some development effort! Angus ___ 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] Minimum CPU required for 60 calls
On Tuesday 02 August 2005 16:50, [EMAIL PROTECTED] wrote: I know that a 3GHz P4 box with 1GB ram, Intel 815 chipset can handle 120 ... Excellent description of a specific benchmark snipped ... Of course, I can't answer the question as to minimum CPU - I only have the CPU that I have. May I ask what your dialplan / scripting looked like to generate this kind of load? I could figure something out but I bet it'd not work nearly as well... -A. ___ 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] Re: two UA with the same usr/pwd
On Tuesday 02 August 2005 14:35, Michael D Schelin wrote:
Rich is correct. Example: Night security guards may need to catch an
inbound calls that could ring at more than one station. Maybe one is
doing rounds and the other is at another desk off site. Sometimes call
forwarding is too slow. There are many reasons why this could be used.
You haven't described a single scenario that would require the same
authentication information from two UAs. Ringing multiple extensions would
solve all of them and there's no call forwarding involved. Try again.
Example: This is how my personal DID gets to me:
exten = 2914574,1,Dial(${ANDREWHOME}${BENSHAW}/${EXTEN},16,rT)
exten = 2914574,n,Dial(${ANDREWHOME}${ANDREWCELL},16,rT)
exten = 2914574,n,VoiceMail([EMAIL PROTECTED],sua)
exten = 2914574,n,Macro(handle-hangup)
It rings my house and a private DID at my office simultaneously, then
continues ringing my house and also dials out a Zap channel to my cell,
finally dropping off to unavailable voicemail.
My home Asterisk server gets this call and dials
(Zap/1Zap/2Zap/3IAX2/andrew-btSIP/xten) -- there are no delays and I can
pick up the call from any of those extensions.
-A.
___
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] How to config incall ?I have a E400p card
asterisk-users How to config incall ?I have a E400p card but How to config incall ? thanks a lot. E400P - Quad Span E1 Card outcall can set: # more extensions.conf [default] include = from-sip [from-sip] exten = 200,1,Dial(Zap/1); exten = 200,2,Hangup dev2002 [EMAIL PROTECTED] 2005-08-03 ___ 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] Re: two UA with the same usr/pwd
All of these postings about ringing two (or more) phones is well known and fairly well understood by everyone. The issue that everyone seems to want to ignore in the postings is the busy lamp field functionality of key systems (not pbx's). I'm not the OP and I've been around * and sip phones for over two years. The issue that the OP was asking about (as have many many others over the same two years) is that associated with a lamp (led) indicating when a specific extension is in use, AND, being able to press the button associated with that lamp and truly pick up that specific extension. Over my 20+ years of being a technical engineer for a very large telco, the best example that I've seen over and over again is that of an executive and secretary where neither one can see the other. The executive will typically yell for the secretary to pick up the line that is on hold and finish handling a call. The secretary can't tell which of the executive's six lines are on hold, can't use call pickup (cause the phone ain't ringing), can't use directed call pickup (since the secretary doesn't know which of the six lines is on hold), and suggestions to train the executive will likely involve seeking employment. Personally, I understand that sip phones need more functionality then is currently engineered into firmware (with few exceptions), but the posters asking for this functionality don't know that. The typical response on this list is sure you can ring two phones, or, that's the difference between a pbx and a key system. Neither one of those responses cut the mustard as the first response doesn't answer the original question, and the second response is based on historical pbx vs key system definitions that do not hold true with current day competitive pbx functionality. So, the bottom line answer to the OP's original question really is that asterisk AND current sip phones cannot emulate key system or competitive pbx functionality, but that certainly does not imply the functionality can't be added to both at some future time. Lots of people would really be thrilled if that could happen sooner rather then later. That functionality is considered/assumed to exist in all telephone systems by non-technical business users, regardless of what us technical types think. Therefore, asterisk does not meet the typical business user's expectations. Regardless of what you call it, this functionality is available in Asterisk through other avenues.. Ring two devices with one extention.. rtfm. Michael D Schelin wrote: Rich is correct. Example: Night security guards may need to catch an inbound calls that could ring at more than one station. Maybe one is doing rounds and the other is at another desk off site. Sometimes call forwarding is too slow. There are many reasons why this could be used. Rich Adamson wrote: Regardless of what has (or has not) been implemented in asterisk, there is a very valid business reason for wanting an extension number to ring on multiple phones and to determine the status of an extension from multiple phones. Business have needed (and implemented) that for years. Having such an implementation in asterisk would definitely be a major plus (regardless of what our definitions of a pbx and keysystem happen to be). Many people seem to want this feature. I think they are just confused. I've never actually heard of a good reason to let multiple devices register with the same username/secret. Most of the time they want a call to ring on multiple devices and they are trying to make a device == extension, which is not correct. A device is a device and an extension is an extension and they are not the same thing and there is no 1-to-1 mapping between them. Victor Alvarez wrote: I really think this matter deserves attention. I have been asked many times about it. Regards, Victor. Hello, I can understand why asterisk is designed to not to allow two UAs with the same usr/pwd, http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html, but I have to find a solution for this. My first option is use SER as an extension end of Asterisk, to allow more than one SIP endpoint to register with the same details http://www.voip-info.org/wiki-Asterisk+at+large. I wonder if there is another way to do this. Of course, I am talking about a SIP proxy behaviour, simultaneous registration, both phones ringing at the same time and first to answer gets the call. Kind regards, Victor. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] IAX2, can't receive calls
Wilson Pickett wrote: I have IAX2 (FWD) partially working. I can place calls from my Asterisk box but I cam unable to receive them (comes back as busy). I have my firewall forwarding the udp ports 5060, 4569, 5036 and 1 thru 2 to my asterisk server. I think I have the firewall correctly setup as I can forward other services to their appropriate servers. I have no mail box on the one account (the one I'm testing to). I've followed the FWD instructions but I've had no luck. what does iax2 show register and iax2 show peers show wrt FWD? mozart*CLI iax2 show registry Host UsernamePerceived Refresh State 65.39.205.121:4569xx 69.142.122.219:456960 Request Sent mozart*CLI iax2 show peers Name/UsernameHost Mask Port Status fwd2/xx 69.90.155.70(S) 255.255.255.255 4569 Unmonitored 1 iax2 peers [0 online, 0 offline, 1 unmonitored] Hmm, I do seem to have a problem but this is not what caused my post (but I do have to fix that). I seem to be loosing registration for about 60 seconds or so (it's registered, it's unregistered). You're not going to beleive this, it turns out the problem is with my BT101 and/or Asterisk config related to the 101. I thought I had the 101 properly configured but when I switched to extenstion 2210 (a different extension) it started working. For some reason the BT101 isn't registered (seems none of my SIP phones are). Let me do some more work and I'll post a new message under the proper heading for that problem. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] How to test E400p card without E1 lines?thanks a lot
asterisk-users E400P - Quad Span E1 Card How to test E400p card without E1 lines?thanks a lot May I loop the card? how to do ? dev2002 [EMAIL PROTECTED] 2005-08-03 ___ 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] TFTP Secondary Ports
Just a data point... tftp works just fine in RHv9 and FC3 with remote 7960's. Images, config files, etc, get transferred correctly every time, and the 7960's are between elcheapo firewall boxes. If you really want to restrict who can access the tftp server, run one of the firewall app's on the linux server. I understand. However, Im successfully managing this without any problems using a Windows tftp server by www.winagents.com. This software allows you to limit secondary transfer connections to a range of IPs. Therefore you only need to open up port 69 and the range you specify. Everything just works! I would like to move the solution to Linux for a couple reasons. However, It looks like the default tftp server does not support this feature and that is why you were going crazy. The number of ports you must open is ridiculous for tftp. However, I just found a seemingly robust linux version with firewall support offered by weirdsolutions. It looks promising. http://www.weirdsolutions.com/ Chad --- --- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Sent: Wednesday, August 03, 2005 12:10 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] TFTP Secondary Ports hey chad, just a heads up tftp is one of the worst protocols to use when your behind a nat or firewall it drove me pretty crazy a while ago. Carlos Alcantar Race Technologies, Inc. 101 Haskins Way South San Francisco, CA 94080 P: 650.246.8900 F: 650.246.8901 E: carlos at race.com -- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Brown Sent: Tuesday, August 02, 2005 10:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TFTP Secondary Ports Im publishing tftp through my firewall to support external Cisco 7960 sip phones. I know that the primary port is 69 for tftp. However, tftp also uses secondary ports ranging from 1,0XX to 30,XXX. ( A broad range) In an effort to limit the secondary ports that are opened, some Windows based tftp server such as the winagents product allows you to limit the range of secondary ports that are used allowing you to somewhat tighten firewall publishing rules. Does anyone know how to do this using the linux tftp server? Thanks, Chad ---End of Original Message- ___ 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] Is it possible to use CHAN_CAPI with ZAPHFC enabled card ?
I have an ISDN card, Billion ISDN PCI Card I tried to use the ZAPHFC, I patched the kernel, I did anything (also followed reccomandation on use on Suse Linux Professional 9.2 --my box is) using bristuff last version. In the end I succesfully compile zaphfc, but I am not able to use the card (a lot of problem running zapcfg, a loto of problem starting asterisk saying about wrong anything (from signalling to any other parameter specified in zapata.conf) I don't want to spend any more time trying to make this run, I have a PRI e1 well configured on same machine. but the question is : is it possibile to do not use zaphfc and configure in some way a CHAN_CAPI channel pointing to Billion card ?? I succesfully iìnstalled (on another box) 3 fritzcard as chan_capi channel without any problem. But here ? I don't know what to write down in /etc/capi.conf !! In the box with the 3 fritzcard, i wrote: fcpci - - - - - - f2pci - - - - - - f3pci - - - - - - but I don't know what to write here ! Any help will be greatly appreciated Thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ 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] Generic Question: Why should I use Asterisk over SIPxchange?
For those of you who have been working with asterisk for a while and who have experience with SIPxchange, why have you chosen Asterisk over the latter? What are some significant differences between the two that those of you familiar with both have discovered? Brent ___ 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] Is this maillist down?
It's not just him. The list was majorly down from sometime on the 29th until the 1st. MARK. Derek Whitten wrote: must be just you.. get messages all day every day here.. :-) On Mon, 2005-08-01 at 05:49, Howard Leadmon wrote: This is usually a very active list, but looking at my procmail log the last message I have received arrived on: From [EMAIL PROTECTED] Fri Jul 29 03:04:17 2005 Subject: Re: [Asterisk-Users] How can I use MySQL in the dialplan? Since that message there has been a gaping silence, any idea what is up, as I am sure seeing mail from everything else. Actually I don't think I have seen any mail from any of the asterisk lists, since that time so guessing this list is having some kind of problem... --- Howard Leadmon - [EMAIL PROTECTED] http://www.leadmon.net ___ 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] 7970 SIP
There is DEFINITELY 7970 SIP firmware out there...maybe Betabut it's out there. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, August 03, 2005 7:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 7970 SIP Nkm [EMAIL PROTECTED] : On 8/2/05, Darren Wright wrote: Can anyone point me to the location of the 7970 SIP image? I'm logged There's no SIP firmware for 7970, only SCCP firmware. Am I right? Sergio ___ 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] TFTP Secondary Ports
Rich Adamson wrote: Just a data point... tftp works just fine in RHv9 and FC3 with remote 7960's. Images, config files, etc, get transferred correctly every time, and the 7960's are between elcheapo firewall boxes. If you really want to restrict who can access the tftp server, run one of the firewall app's on the linux server. The Linux firewall knows about tftp also. You just load ip_nat_tftp and it will handle the data ports in a secure manner for you - you just need to open port 69 so that the 7960's can initiate the request. Tony ___ 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] SIP phone procedural question
A lot of my customers have people who are in the office most of the time but occasionally wish to work from home. So they may have a sip phone which is extension 208 in the office. When they work from home they can of course plug in a sip phone into their broadband connection and work with that. But it would be ideal if they could be same extension as phone in office. If they try to register as same sip user - eg extn 208 - will it work. Then problem is phone on their desk will still ring p***ing all their colleagues off. How do people deal with this sort of thing. Ideally, would want person to be able to easily switch from office to home but use same extension. Or does sip somehow deal with this? Is there a standard sip way of dealing with this? You should be able to find multiple ways to do that on the wiki. Use keywords such as call forwarding in the search. One way to do it for x1234 (as a high level example only) is to: - assign 9234 as a call forward control extension - employee dials x9234 - asterisk writes a value into the db (see show application dbput) - when a call is sent to x1234, dialplan code checks for value using dbget. - if value is set, ring at-home extension; if not set, ring at-office extension (eg, x1234). If you want to make the above a little more sophisticated, when the user dials x9234 prompt the user for which extension to forward his calls to and write that to the db. When a call arrives, the call is call-forwarded to whatever extension the user entered. Want to complicate that more, write a macro to do that for all extens. ___ 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] AstLinux - Anyone running on a Soekris Engineering net4826
I ran across AstLinux today, and noticed they had a build for Soekris Engineering net4801. Is anyone running this board with AstLinux in a production environment? If so, what type of load have you been able to put on it? Any luck getting Digium hardware to run on it? Any other thoughts/opinions, etc? Thanks. Doug Logan ___ 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] Best way to connect asterisk to an traditional PBX
Sounds to me like your phone vendor is talking out of his arse. You should be able to place a crossover cable between your * box and your pbx. They both think the other is a phone company. I've done this with Avaya Definity G3's a few times now and it works fine. Mark Administrator TOOTAI wrote: Hi list, we want to connect asterisk to an traditionnal PBX (EADS 6550/Matra). People from telco told that they can't connect two PBX's using E1/T1 or only with QSig signaling. I wanted to use EuroISDN. In this case, it was me told that VN6-VN7 would be used. The PBX has a spare ADQ card installed on which we would connect. Has someone a such working setup? Is it working well? More generaly, could you please tell me how you're connecting * to an traditional PBX, what you think is the best solution, which signaling you're using and which card(s). At the moment, the two PBX's are connected through 2 TDM cards, one 4 FXS the other 4 FXO. Our goal is to have a max of 30 lines available and available in the same time. Location is in France. Thanks for your feedback. -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] SIP WEB Phone (Wanna implement Click to Call)
Hi Tim, I would like to test it as well. Thanks, Derek On Wed, 2005-08-03 at 00:37, Boris Zolotarev - Pamet wrote: Hello Tim, I am definitely interested in testing it. Please contact me off the list. Best Regards, Boris. If anyone is interested I'm (slowly) developing a GPL'd Java applet that works as an IAX softphone. I should have a test version out at the end of the week for a limited number of testers. Tim. __ ___ 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 -- -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- signature.asc Description: This is a digitally signed message part ___ 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] Re: Polycom Soundpoint 600
Hi Eric - I am having trouble with one of our IP600. Every five days or so, the phone locks up. This is the third 600 I have put in place. I am running asterisk 1.0.9. Has anyone had this problem with the IP600? What version of the bootrom and sip firmware are you using? Can we see your phone's config files, and maybe the asterisk sip.conf file? I'll knock on wood, but I have many 600's that run indefinitely without any incidents. - Noah ___ 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] what phones support this when running with asterisk
Tim Litwiller wrote: I've been using * at home at my house for while and like it but for work I didn't know the answers to these questions. But now my new employer is wanting to upgrade a very old phone system and wants to make sure our new system has some features I've talked to him about using asterisk and he put this on the required options list. a button with a light for each incoming physical line and a button and light for each user and the ability to transfer a call by pressing hold and then that users button. Better look into other systems. Asterisk and SIP can't yet handle this COMMON requirement. See other postings regarding this very subject. Asterisk is not yet ready for prime time in this arena. Asterisk IS NOT a Hybrid key/pbx, which is what you really are asking for Users want buttons and lights, access and status of lines, and most want handsfree intercom as well. Many good and affordable choices, depending on your country, number of lines and number of stations. For small requirements, under 12 lines and 32 stations, Panasonic, and NEC have affordable systems available in the US through supply houses, or at higher cost off of eBay , new. If you are replacing an existing wired system, you may even be able to install yourself and on the wants but not required list a way for each user to automatically have a call log and tapi (click to dial) on windows xp desktops a way to push a record button on the phone to record on demand. Check your laws on recording. In the US this varies from state to state, and could result in a serious problem. John Novack ___ 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] Is it possible to use CHAN_CAPI with ZAPHFC enabled card ?
[EMAIL PROTECTED] ha scritto: In the end I succesfully compile zaphfc, but I am not able to use the card (a lot of problem running zapcfg, a loto of problem starting asterisk saying about wrong anything (from signalling to any other parameter specified in zapata.conf) You may want to post both the configuration files AND the error messages here... is it possibile to do not use zaphfc and configure in some way a CHAN_CAPI channel pointing to Billion card ?? I don't think so, unless someone has written a CAPI layer for HFC-S PCI A cards! Bye, -- Emanuele ___ 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 TDM card connected to phone lines AND fax line
Hello I want to setup an Asterisk with three analog lines. Two of the analog lines are the main office number. The other line is the fax number. The fax machine plugs into the line 3 but also will be a connection to the third port on a Digium analog card. Reason for the third line into Asterisk is so if two lines in use someone can still dial out over third (fax) line. Is this going to cause a problem? How would I stop the Idiom card answering on line 3? Angus ___ 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] Is this maillist down?
Yep, I second (or third) that observation. Ryan It's not just him. The list was majorly down from sometime on the 29th until the 1st. MARK. Derek Whitten wrote: must be just you.. get messages all day every day here.. :-) On Mon, 2005-08-01 at 05:49, Howard Leadmon wrote: This is usually a very active list, but looking at my procmail log the last message I have received arrived on: From [EMAIL PROTECTED] Fri Jul 29 03:04:17 2005 Subject: Re: [Asterisk-Users] How can I use MySQL in the dialplan? Since that message there has been a gaping silence, any idea what is up, as I am sure seeing mail from everything else. Actually I don't think I have seen any mail from any of the asterisk lists, since that time so guessing this list is having some kind of problem... --- Howard Leadmon - [EMAIL PROTECTED] http://www.leadmon.net ___ 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
[Asterisk-Users] Astcc Start up
Hello.. I am new to the asterisk/astcc domain and have to do some maintenance work on an existing system. As far as I know astcc has been installed and has worked previously. All of a sudden it has stopped working. Since I am not aware of how the interfacing between astcc and asterisk, I need some quick pointers as to where I need to look at to see what is broken. Thanks Marios -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 2/8/2005 ___ 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] Problems with zaptel and voice prompts/voicemail
This seems to be due to a driver conflict. If I unload Zaptel, the sound returns. I'm having the same issue with a 2.4 kernel on whitebox 3 using HEAD. Still investigating... let me know if you find anything new. Jack On 6/29/05, Jeremy McDermond [EMAIL PROTECTED] wrote: I've looked all around, and I can't find an answer to this. I apologize if this has been discussed already or is buried somewhere in voip-info.org. I have an asterisk setup on linux 2.6.11.11 kernel, a revision E/F TDM400P, and Polycom IP501 phones. As soon as I load the zaptel module into the kernel, the voice prompts and voicemail system ceases to work. The asterisk logs say that the gsm files are being played, but nothing comes out on the other end. This is for both calls coming in via our VoicePulse Connect lines, or when dialing locally from our SIP phones. As soon as I rmmod the zaptel driver, asterisk acts just fine. Thanks for any assistance the list may be able to provide. -- Jeremy McDermond Xenotropic Systems ___ 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] Voicemail and musiconhold sound stopped working
I'm having the same issue. If I unload Zaptel, and restart asterisk... the sound does return. On 7/25/05, Arnd Vehling [EMAIL PROTECTED] wrote: Hi, i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07 and everything worked fine sofar when suddenly the voicemail and musiconhold sound output stopped working. The voicemailmenu still works though. I can see the voiceprompts etc in the debug messages on the asterisk CLI but i cant hear anything. Everything else works fine though. I can call out fine etc. I did some network sniffing using ngrep and verified that the voicemail app is indeed not sending _any_ udp/rtp packets towards my sip fones. I did restore old, working configs back but still no change. I reinstalled asterisk from the cvs and even rebootet my linux box (kernel 2.4.27) still no change. This stuff is now bugging me for 5 hours and i am slowly going nuts. I am using an installation with several different sip-fones, zaptel+zaprtc as well as fcpci+capi on a teles isdn card. Any ideas where to look for? thx, Arnd ___ 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] Transfer to outside line.
Finally got everything up and run with the help of Manny Wise last night. So I am setting up my digital assistant and getting down to the task I need this box to perform the most. I need to have a custom app that I can call that will take me pressing 2 at the menu and have it transfer the call to a offsite phone number utilizing my Zap Trunk. Im sure someone has done this already. Anyone want to point me in the right direction? Tim King ___ 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] Re: two UA with the same usr/pwd
Rich Adamson wrote: All of these postings about ringing two (or more) phones is well known and fairly well understood by everyone. The issue that everyone seems to want to ignore in the postings is the busy lamp field functionality of key systems (not pbx's). I'm not the OP and I've been around * and sip phones for over two years. The issue that the OP was asking about (as have many many others over the same two years) is that associated with a lamp (led) indicating when a specific extension is in use, AND, being able to press the button associated with that lamp and truly pick up that specific extension. Over my 20+ years of being a technical engineer for a very large telco, the best example that I've seen over and over again is that of an executive and secretary where neither one can see the other. The executive will typically yell for the secretary to pick up the line that is on hold and finish handling a call. The secretary can't tell which of the executive's six lines are on hold, can't use call pickup (cause the phone ain't ringing), can't use directed call pickup (since the secretary doesn't know which of the six lines is on hold), and suggestions to train the executive will likely involve seeking employment. Multiple devices registering as the same user is not the solution to wanting a Busy Lamp Field. What you want to do is search the Wiki and mailing list archive for the hint priority. In SIP Busy Lamp Fields are done using PUBLISH/SUBSCRIBE, not using REGISTER. MANY people have gotten Busy Lamp Fields with Asterisk without needing to register the same username to multiple devices. Of course your problem will become the fact that virtually no SIP devices support more than 5 line buttons. The devices that do support more than 5 line buttons don't run SIP. The only device that I know of is the SNOM, but I've never used it. The Cisco sidecar does not run SIP. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Call Interception
Hi all, I'm thinking of setting up an Asterisk based VoIP system between two offices and I wanted to know if it is possible to intercept calls with Asterisk if so how does one set it up? 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] Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary D-channel of span 1
Yep, another list posting on this topic :) All the messages I've read on this are from people experiencing these errors in quiet times - I get them as soon as I plug a port on our TE410P to an Inter-Tel AXXESS PBX.. and I get them continuously... I'm just sticking an * box in between ISDN30e (we're in the UK so euroisdn) and the PBX.. and whilst the telco ISDN30e side works like a charm [1] I simply can't get a reliable link to the PBX.. I've tried two different T1 crossovers (1-4, 2-5) with identical results and zapata.conf is indeed using signalling=pri_cpe for the telco ISDN30e and pri_net for the PBX Digium support have taken me through loopback testing which came out perfect, and the card is not sharing any IRQ, yet this error renders the card useless :( Digium are reluctant to accept a return and replace the card since they don't believe it to be at fault - and neither do I. I see the same behaviour with 1.0.9 asterisk / libpri and 1.0.9.1 zaptel... and CVS-HEAD versions of everything. Any ideas/advice would be warmly received right now! Cheers, Gavin. [1] http://www.voip-info.org/tiki-index.php?page=UK+Asterisk+Details ___ 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] Full T38 sip Faxing now Available
Chris Mason (Lists) [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Carlos [EMAIL PROTECTED] lazily top-posted: Has anyone got a response from this? It was just spam. Forget it. I have an account with them, just waiting for a suitable ATA to arrive. Good for you. Personally, I never buy anything from spammers. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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] fax -- grandstream 286 -- asterisk -- pstn
Hi all, Im having problems using a fax machine conected trough a grandstream 286 sip ATA, it must be able to send and recive fax from pstn, but fax always ends with communication errors 252/244/232 and others. Im using alaw/ulaw codes on pass trough mode, also have tried asterisk faxdetection, nvfaxdetect, disable echo cancellation by hand always with same results. Grandstream ATA is using firmware version 1.0.6.7 Im using efax with a serial modem, to send faxes to asterisk, hopping it forwards them to the fax machine, here it the log efax -d /dev/ttyS0 -o f -t 913664813 sun480-boot efax: Wed Aug 3 16:37:35 2005 efax v 0.9a-001114 Copyright 1999 Ed Casas efax: Wed Aug 3 16:37:35 2005 efax v 0.9a-001114 Copyright 1999 Ed Casas efax: 37:35 compiled Dec 2 2004 14:28:56 efax: 37:35 opened /dev/ttyS0 efax: 37:36 using V1.002.C04-K56_DLS in class 1 efax: 37:36 dialing 913664813 efax: 38:00 connected efax: 38:01 received UNKNOWN efax: 38:01 Warning: bit-reversed HDLC frame, reversing bit order efax: 38:01 received CSI - answering ID efax: 38:01 remote ID -913664813 efax: 38:02 received DIS - answering capabilities efax: 38:02 remote has no document(s) to send, and can receive efax: 38:02 local 196lpi 14.4kbps 8.5/215mm any 1D- - 0ms efax: 38:02 remote 196lpi 14.4kbps 8.5/215mm any 2D ECM-64 - 0ms efax: 38:02 session 196lpi 14.4kbps 8.5/215mm any 1D- - 0ms efax: 38:02 sent TSI - caller ID efax: 38:04 sent DCS - session format efax: 38:07 sent TCF - channel check of 2700 bytes efax: 38:11 received DCS - session format efax: 38:11 local 196lpi 14.4kbps 8.5/215mm any 1D- - 0ms efax: 38:11 remote 196lpi 14.4kbps 8.5/215mm any 2D ECM-64 - 0ms efax: 38:11 session 196lpi 14.4kbps 8.5/215mm any 1D- - 0ms efax: 38:11 sent TSI - caller ID efax: 38:12 sent DCS - session format efax: 38:16 sent TCF - channel check of 2700 bytes efax: 38:19 received DCS - session format efax: 38:19 local 196lpi 14.4kbps 8.5/215mm any 1D- - 0ms efax: 38:19 remote 196lpi 14.4kbps 8.5/215mm any 2D ECM-64 - 0ms efax: 38:19 session 196lpi 14.4kbps 8.5/215mm any 1D- - 0ms efax: 38:19 sent TSI - caller ID efax: 38:21 sent DCS - session format efax: 38:25 sent TCF - channel check of 2700 bytes efax: 38:28 received DCS - session format efax: 38:28 Error: no command/response from remote efax: 38:28 sent DCN - disconnect efax: 38:32 failed - sun480-boot efax: 38:32 done, returning 3 (invalid modem response) and the asterisk log -- Starting simple switch on 'Zap/4-1' Aug 3 16:37:14 NOTICE[16135]: chan_zap.c:5377 ss_thread: Got event 2 (Ring/Answered)... -- Executing Answer(Zap/4-1, ) in new stack -- Executing NVFaxDetect(Zap/4-1, 4|dt) in new stack Aug 3 16:37:16 NOTICE[16135]: app_nv_faxdetect.c:215 nv_detectfax_exec: Redirecting Zap/4-1 to fax extension -- Executing Dial(Zap/4-1, SIP/gw3) in new stack -- Called gw3 -- SIP/gw3-4ffd is ringing -- SIP/gw3-4ffd answered Zap/4-1 == Spawn extension (linea-fax, fax, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' Does anyone know what could be happening? any suggestions? Tell me if more info is needed. Thanks, Jaime. ___ 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] invalid extension dilemma
On Wed, 2005-08-03 at 07:52 +0200, Wilson Pickett wrote: In the example below if I dial valid extension 1000, the Invalid context plays pbx-invalid as it is included with _7 context. Include voicemail in the main context. Thanks, I new it must be something simple. Simply reposition the context voicemail before goto-dialout did the trick. -- #Joseph ___ 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] IDSN 30 PRI UK
Hi I am ordering a ISDN 30 line in from BT to use with digium hardware. Was wondering if there was anything specific I should ask for when getting the service in place. Thanks Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ 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] app_dbodbc for asterisk stable 1.09
app_dbodbc has been publically deprecated by the author and he isn't updating it. Functionality provided by ast_data is provided by RealTime. You will need CVS-HEAD to use RealTime. Or wait a month for 1.2 to come out. -Matthew Quoting Umar Sear [EMAIL PROTECTED]: Hi, Has anyone manage to comile app_dbodbc or ast_data with the latest stable release (1.09). If so can you give some guidence on howto do it as I have trouble getting either working. Umar ___ 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 This message was sent using IMP, the Internet Messaging Program. ___ 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] Mozphone
Has anyone tried this? I got in to download but now I can not get back into mozdev.org. It did not come with any directions or help. If anyone has it working where did you get instructions? TIA Bob ___ 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 TDM card connected to phone lines AND fax line
Buy a 3 porst fxo card and 1port fxs (green) card from digium. Plug your fax the the fxs port. Assign an extension to the fax at extension.conf Create a menu. Since the call will be bridged from fxo to fxs natively, there is very few loss and the fax works ok. Anyway, the diferrence between having the tdp400 with 3 or 4 porst isn´t much... Angus Comber wrote: Hello I want to setup an Asterisk with three analog lines. Two of the analog lines are the main office number. The other line is the fax number. The fax machine plugs into the line 3 but also will be a connection to the third port on a Digium analog card. Reason for the third line into Asterisk is so if two lines in use someone can still dial out over third (fax) line. Is this going to cause a problem? How would I stop the Idiom card answering on line 3? Angus ___ 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] Is it possible to use CHAN_CAPI with ZAPHFC enabled card ?
On Wed, 3 Aug 2005, Emanuele Pucciarelli wrote: is it possibile to do not use zaphfc and configure in some way a CHAN_CAPI channel pointing to Billion card ?? I don't think so, unless someone has written a CAPI layer for HFC-S PCI A cards! Isn't mISDN providing this? Armin ___ 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] what phones support this when running with asterisk
John Novack wrote: Tim Litwiller wrote: I've been using * at home at my house for while and like it but for work I didn't know the answers to these questions. But now my new employer is wanting to upgrade a very old phone system and wants to make sure our new system has some features I've talked to him about using asterisk and he put this on the required options list. a button with a light for each incoming physical line and a button and light for each user and the ability to transfer a call by pressing hold and then that users button. Better look into other systems. Asterisk and SIP can't yet handle this COMMON requirement. See other postings regarding this very subject. Asterisk is not yet ready for prime time in this arena. Asterisk IS NOT a Hybrid key/pbx, which is what you really are asking for Users want buttons and lights, access and status of lines, and most want handsfree intercom as well. Many good and affordable choices, depending on your country, number of lines and number of stations. For small requirements, under 12 lines and 32 stations, Panasonic, and NEC have affordable systems available in the US through supply houses, or at higher cost off of eBay , new. If you are replacing an existing wired system, you may even be able to install yourself and on the wants but not required list a way for each user to automatically have a call log and tapi (click to dial) on windows xp desktops a way to push a record button on the phone to record on demand. Check your laws on recording. In the US this varies from state to state, and could result in a serious problem. We have - it is legal in Kansas - You don't even have to tell the other party. But we would tell the customer that we are recording thier phone order for accuracy. John Novack ___ 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] Is it possible to use CHAN_CAPI with ZAPHFC enabled card ?
Thank you for your answer. anyway I just destryed my linux box, and I am installing it again. The problem was, I think, that the driver was not loaded, sayng something about pci card not found. Really funny, becouse the Yast detected it and let you configure it. In the end all the modules in my box were taitenig the kernel, after I tried to install mISDN as an alternative to zaphfc So I decided to scratch anything, reinstall all and throw away the billion card; I think the coexistence with a E1 Primary adapter is not good for it. Unfortunately the PCI fritz card I have (which can be configured as chan_capi) is not pluggable in my dell (pci bus not compatible) thanks, Andrea Emanuele Pucciarelli [EMAIL PROTECTED] To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 03/08/2005 15.48 Re: [Asterisk-Users] Is it possible to use CHAN_CAPI with ZAPHFC enabled card ? Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com [EMAIL PROTECTED] ha scritto: In the end I succesfully compile zaphfc, but I am not able to use the card (a lot of problem running zapcfg, a loto of problem starting asterisk saying about wrong anything (from signalling to any other parameter specified in zapata.conf) You may want to post both the configuration files AND the error messages here... is it possibile to do not use zaphfc and configure in some way a CHAN_CAPI channel pointing to Billion card ?? I don't think so, unless someone has written a CAPI layer for HFC-S PCI A cards! Bye, -- Emanuele ___ 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] Chan_bluetooth and AudioGateway phone [long]
Hello, I start trying to use a USB dongle and a Bluetooth GSM phone to make GSM call with asterisk using the BLT channel provided by the GSM phone. Unfortunately I get a Everyone is busy/congested at this time whenever I try to Dial(IAX2/[EMAIL PROTECTED]/2, BLT/MotorolaLara/3474501***) For sure I make some mistake in the configuration. Unfortunately I don't find any step-by-step guide to configure USB-Dongle + Asterisk + chan_bluetooth What channel I have to use? If I try the following command as specified on the configuration file /etc/asterisk# sdptool search --bdaddr 00:0a:28:83:a9:cf 0x111F Class 0x111F Searching for 0x111F on 00:0A:28:83:A9:CF ... Service Name: Hands-Free voice gateway Service Description: Hands-Free voice gateway Service Provider: Motorola Service RecHandle: 0x10007 Service Class ID List: Handfree Audio Gateway (0x111f) Generic Audio (0x1203) Protocol Descriptor List: L2CAP (0x0100) RFCOMM (0x0003) Channel: 7 Language Base Attr List: ... Profile Descriptor List: Handsfree (0x111e) Version: 0x0101 Maybe the channel 7? I have also the following channels available /etc/asterisk# sdptool browse 00:0a:28:83:a9:cf Browsing 00:0A:28:83:A9:CF ... Service RecHandle: 0x0 Service Class ID List: SDP Server (0x1000) Protocol Descriptor List: L2CAP (0x0100) SDP (0x0001) Profile Descriptor List: SDP Server (0x1000) Version: 0x0100 Service Name: Dial-up networking Gateway Service Description: Dial-up networking Gateway Service Provider: Motorola Service RecHandle: 0x10001 Service Class ID List: Dialup Networking (0x1103) Protocol Descriptor List: L2CAP (0x0100) RFCOMM (0x0003) Channel: 1 Language Base Attr List: ... Profile Descriptor List: Dialup Networking (0x1103) Version: 0x0100 Service Name: Voice Gateway Service Description: Headset Audio Gateway Service Provider: Motorola Service RecHandle: 0x10003 Service Class ID List: Headset Audio Gateway (0x1112) Generic Audio (0x1203) Protocol Descriptor List: L2CAP (0x0100) RFCOMM (0x0003) Channel: 3 Language Base Attr List: ... Profile Descriptor List: Headset (0x1108) Version: 0x0100 Service Name: Hands-Free voice gateway Service Description: Hands-Free voice gateway Service Provider: Motorola Service RecHandle: 0x10007 Service Class ID List: Handfree Audio Gateway (0x111f) Generic Audio (0x1203) Protocol Descriptor List: L2CAP (0x0100) RFCOMM (0x0003) Channel: 7 Language Base Attr List: ... Profile Descriptor List: Handsfree (0x111e) Version: 0x0101 Service Name: OBEX Object Push Service Description: OBEX Object Push Service Provider: Motorola Service RecHandle: 0x10008 Service Class ID List: OBEX Object Push (0x1105) Protocol Descriptor List: L2CAP (0x0100) RFCOMM (0x0003) Channel: 8 OBEX (0x0008) Language Base Attr List: ... Profile Descriptor List: OBEX Object Push (0x1105) Version: 0x0100 Service Name: OBEX file transfer Service Description: OBEX file transfer Service Provider: Motorola Service RecHandle: 0x10009 Service Class ID List: OBEX File Transfer (0x1106) Protocol Descriptor List: L2CAP (0x0100) RFCOMM (0x0003) Channel: 9 OBEX (0x0008) Language Base Attr List: ... Profile Descriptor List: OBEX File Transfer (0x1106) Version: 0x0100 On asterisk I have the following result: *CLI bluetooth show information --- Version : $Rev: 38 $ Monitor PID : 8487 RFCOMM AG : Channel 1, FD 12 RFCOMM HS : Channel 2, FD 13 Device : hci0, MAC Address 00:10:60:A9:99:CA --- I try to use either channels, 3 and 7, but the result is the same, Everyone is busy/congested at this time. Here it is the result of the connection using channel 7 *CLI [AG] MotorolaLara AT+BRSF=23 [AG] MotorolaLara +MBAN: Copyright 2000-2002 Motorola, Inc. [AG] MotorolaLara +BRSF: 63 [AG] MotorolaLara OK [AG] MotorolaLara AT+CIND=? [AG] MotorolaLara +CIND: (Voice Mail,(0,1)),(service,(0,1)),(call,(0,1)),(Roam,(0-2)),(signal,(0-5)),(callsetup,(0-3)),(smsfull,(0,1)) [AG] MotorolaLara OK [AG] MotorolaLara AT+CIND? [AG] MotorolaLara +CIND: 0,1,0,0,3,0,1 [AG] MotorolaLara OK [AG] MotorolaLara AT+CMER=3,0,0,1 [AG] MotorolaLara OK [AG] MotorolaLara AT+CLIP=1 [AG] MotorolaLara OK [AG] MotorolaLara AT+CGMI=? [AG] MotorolaLara ERROR [AG] MotorolaLara +CIEV: 5,4 note the ERROR during AT+CGMI=? command. However I patch the chan_bluetooth.c and get the correct answer: ... [AG] MotorolaLara AT+CGMI [AG] MotorolaLara +CGMI: Motorola CE, Copyright 2000 [AG] MotorolaLara OK [AG] MotorolaLara AT+CGMI [AG] MotorolaLara +CGMI: Motorola CE, Copyright 2000 [AG] MotorolaLara OK [AG] MotorolaLara +CIEV: 5,3 Command output: *CLI bluetooth show peers BDAddrName Role Status A/C SCOCon/Fd/Th Sig - -- --- --- --- 00:0A:28:83:A9:CF MotorolaLara AG Ready Yes -1/-1/0 Yes Using the channel number 3 I get
RE: [Asterisk-Users] 7970 SCCP configs?
Ok I've got SCCP running I have my 7970 firmware files. Can anyone send an XMLdefault config and an SEP config file? There are a bunch of sbn files in the package...not sure what needs to be loaded. -Darren ___ 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] Transfer to outside line.
I think what you want is called DISA http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA DISA (Direct Inward System Access) Allows someone from outside the telephone switch (PBX) to obtain an internal system dialtone and to place calls from it as if they were placing a call from within the switch. A user calls a number that connects to the DISA application and is given dialtone and context. Doug At 09:12 AM 8/3/2005, you wrote: Finally got everything up and run with the help of Manny Wise last night. So I am setting up my digital assistant and getting down to the task I need this box to perform the most. I need to have a custom app that I can call that will take me pressing 2 at the menu and have it transfer the call to a offsite phone number utilizing my Zap Trunk. Im sure someone has done this already. Anyone want to point me in the right direction? Tim King ___ 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] Call Interception
I assume you mean to save on tolls between the two offices.
If so, the simplest way is to set up asterisk on both ends
and specify in your dialplan which numbers you want to
go out over IP and which you want to go out over landline.
Asterisk makes this easy as it uses the most specific first,
so for instance:
exten = _1NXXNXX,1,Macro(dialout1,${EXTEN:0},tT)
exten = _1573NXX,1,Macro(dialout2,${EXTEN:0},tT)
exten = 1573555,1,Macro(dialout3,${EXTEN:0},tT)
Would mean that 573555 would use dialout3, all other 573
numbers would user dialout2 and all other numbers would user
dialout1
Hope this helps,
Jon.
On Wednesday 03 August 2005 09:29 am, [EMAIL PROTECTED] wrote:
Hi all,
I'm thinking of setting up an Asterisk based VoIP system between two
offices and I wanted to know if it is possible to intercept calls with
Asterisk if so how does one set it up?
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] Dell Servers
http://www.digium.com/index.php?menu=compatibility What servers does one recommend though using ? Our company hates using HP junk, dell used to be a good choice for most of our stuff. IBM is way overpriced. Anyone have any suggestions? Sascha From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Callum McGillivray Sent: August 2, 2005 9:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell Servers Sascha, Where did you see the information about the Dell machines that Digium do not recommend ? Do you have a link ? Thanks, Callum William Boehlke wrote: 1850s work fine with T1 cards but not with TDM. If you need to use an 1850 use an external gateway. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sascha Ferley Sent: Tuesday, August 02, 2005 12:56 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Dell Servers Hi, I was wondering if anyone had any problems with the Dell 1800 series servers, with TDM400 cards? I saw that digium seems to recommend against a lot of the dells. Please let me know Thanks Sascha -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005 ___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
Re: [Asterisk-Users] Anyone know of an open source sip video phone like eyebeam available?
Angus Comber wrote: I just wondered - might save me some development effort! Angus http://www.gnomemeeting.org/ ? Jorge ___ 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] Dell Servers
Supermicro's can be nice. Problem is that Supermicro's aren't sold in Canada and as per our specification is it needs to be a tower based server. Anyone know any other decent fully manufactured systems? .. Systems with support on them like dell, that would work well with asterisk. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: August 2, 2005 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell Servers Sascha Ferley wrote: I was wondering if anyone had any problems with the Dell 1800 series servers, with TDM400 cards? I saw that digium seems to recommend against a lot of the dells. I would recommend Super Micro racks. We're using them with no problems. I haven't tried the dells, although we had some problems with the Fujitsu Siemens machines we had. -- 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
Re: [Asterisk-Users] Mozphone
Take a look at http://moziax.mozdev.org/ Take care. On Wed, 2005-08-03 at 11:11 -0400, Robert A. Rawlinson wrote: Has anyone tried this? I got in to download but now I can not get back into mozdev.org. It did not come with any directions or help. If anyone has it working where did you get instructions? TIA Bob ___ 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 -- Arnaldo M. Pereira egghunt at gmail dot com http://ansi-c.org/~arnaldo ___ 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 TDM card connected to phone lines AND fax line
On Wed, 2005-08-03 at 10:23 +, Andres Tello Abrego wrote: Assign an extension to the fax at extension.conf Create a menu. Why even bother to do that much? Just put the 3rd port/line into its own extension where s automatically dials the fax machine on 4. You can still use 1, 2, and 3 for outbound if you group them and dial with one of zaptel's grouping options. Other idea being to make sure the fax machine picks up first, but this issue's been discussed on the list before. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ 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] Mozphone
Robert A. Rawlinson a écrit : Has anyone tried this? I got in to download but now I can not get back into mozdev.org. It did not come with any directions or help. If anyone has it working where did you get instructions? The project home page is: http://moziax.mozdev.org/ (unfortunately mozphone.mozdev.org had already been registered but nothing there). If you have specific question, go ahead I'll try to help as much as I can. I'm also very interested in feedback. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ 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] Inter-Tel AXXESS failure: HDLC Bad FCS (8) onPrimary D-channel of span 1
On Wednesday 03 August 2005 17:33, Jens von Bülow wrote: Gavin, Any ideas/advice would be warmly received right now! You are not going to like my response... Erk :) The only way I could get this to work (luckily I had 2 identical sites and was busy with the upgrade to the gen2 card) was to downgrade to zaptel 1.0.7. Alas no - just moved down to zaptel, libpri and asterisk 1.0.7 with identical behaviour, both with span=1,0,0,ccs,hdb3,crc4 and span=1,1,0,ccs,hdb3,crc4 - I don't have any other active spans in the system :/ Tim Panton: As above, I've already tried timing source twiddles (and even changing the build-out length values, even though the cable is 2 metres :)) My whole zaptel.conf is span = 1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk defaultzone=uk With zapata.conf snippet: switchtype=euroisdn immediate=no overlapdial=yes pridialplan=unknown prilocaldialplan=unknown group = 1 signalling=pri_net context = fromaxxess channel = 1-15 Cheers, Gavin. ___ 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] AstriCon 2005 - Early Bird Registration Open (Free IAXy To First 50!)
// AstriCon 2005 - Oct 11 - 14, 2005 - Anaheim, California USA // [ REGISTRATION NOW OPEN] -- Digium and Ipsando are pleased to announce that AstriCon 2005 Early Bird Registration is now open. Early Bird registration can save you 20% ($110.00 USD) off the full conference admission. The first 50 to qualify for Early Bird by purchasing an AstriCon All Access Pass will also RECEIVE A FREE IAXy from Digium*. Register Now: https://www.astricon.net/2005/register/ [ WHAT IS ASTRICON? ] -- The only conference dedicated exclusively to Asterisk. AstriCon includes: * Two Pre-Conference Events: - The Asterisk Developer Summit - Meet Asterisk! - An Introductory Seminar * A full day of Asterisk Tutorials: - Beginner: Learn to install and implement Asterisk - Intermediate: Learn tips and tricks for enhancing your PBX - Advanced: Scale and cluster Asterisk, improve security * Two Full Days of Conference: - Keynote from Asterisk creator Mark Spencer - Presentations from lead developers - Asterisk Industry Perspectives - Panel discussions round tables - BOF Sessions * The Asterisk Exposition Trade Show: - Service Providers from around the globe - IP Phone manufacturers and distributors - VARs and Integrators - Training Support Organizations [ WHEN WHERE IS IT? ] -- AstriCon 2005 will be held from October 11 through October 14 at the Hyatt Regency Orange County in Anaheim California. [ WHO WILL BE THERE? ] -- Last year’s AstriCon drew nearly FIVE HUNDRED attendees. The goal for this year is nothing short of doubling the previous attendance. Attendees include: enterprise users, Internet telephony service providers, competitive local exchange carriers, interconnect vendors, consultants, systems integrators, VARs, developers, ISPs, and hobbyists. [ EXHIBIT or SPEAK at ASTRICON ] -- For information on speaking opportunities or for exhibition information, contact us. Email: [EMAIL PROTECTED] Phone: +1 816 256 8916 IAX2: IAX2/[EMAIL PROTECTED] See you at AstriCon! * Winners will be able to pick up their IAXys from Digium at the Digium booth at AstriCon. begin:vcard fn:Steven Sokol n:Sokol;Steven email;internet:[EMAIL PROTECTED] tel;work:816.822.1807 x-mozilla-html:FALSE url:http://www.sokol-associates.com version:2.1 end:vcard ___ 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] Full T38 sip Faxing now Available
Why do you put me down? I have not done a thing to you and I'm not a spammer. Please stop this activity It's not professional. If I were to give you bad service please feel free to comment negatively but I've never dealt with you nor do you have an account with us. Sincerely Michael D. Schelin ___ 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] Dell Servers
Hi, Sorry for the top post, but the precedent has already been set. :-( We're using a Dell 1850 with a TDM04B without any problem so the previous post is incorrect about TDM cards not working in this machine. We're using it with Fedora Core 1. The only problem was that we had to add a PCI Ethernet card since FC1 didn't recognize the on-board Ethernet. Michael Swan Neon Software, Inc. At 10:25 AM 8/3/2005 -0600, you wrote: http://www.digium.com/index.php?menu=compatibility What servers does one recommend though using ? Our company hates using HP junk, dell used to be a good choice for most of our stuff. IBM is way overpriced. Anyone have any suggestions? Sascha From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Callum McGillivray Sent: August 2, 2005 9:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell Servers Sascha, Where did you see the information about the Dell machines that Digium do not recommend ? Do you have a link ? Thanks, Callum William Boehlke wrote: 1850s work fine with T1 cards but not with TDM. If you need to use an 1850 use an external gateway. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Sascha Ferley Sent: Tuesday, August 02, 2005 12:56 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Dell Servers Hi, I was wondering if anyone had any problems with the Dell 1800 series servers, with TDM400 cards? I saw that digium seems to recommend against a lot of the dells. Please let me know Thanks Sascha -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005 ___ 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
