RE: [Asterisk-Users] NewBie to Ast Server, help need for the configuration
http://sourceforge.net/projects/asteriskathome/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ramSent: Friday, November 25, 2005 11:33 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] NewBie to Ast Server,help need for the configuration Hi can some one point me where can i download this ?? ram On 11/25/05, Juan Janczuk [EMAIL PROTECTED] wrote: It seems like [EMAIL PROTECTED] could be your best solution. It has a nice user interface (AMP, you can try to install it in your actual asterisk box), that lets you do all you say. Regards. Juan. -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ]En nombre de ramEnviado el: Viernes, 25 de Noviembre de 2005 02:54 a.m.Para: asterisk-users@lists.digium.comAsunto: [Asterisk-Users] NewBie to Ast Server,help need for the configuration Hi all iam setting PBX for outgoing calls at this moment once iam success this , iam planning to do config inbound to So iam start configuring with Outbound calls Ring now my config looks like follow Lan Users-- Astrix--- VoIP provider I have one account with VoIP provider, i can make multiple calls using that accounts i have 20 Lan users, who start making called to out going all of the them connected to Lan Swtich where astrix connected I have downloaded Asterisk+addons+sounds and comipled with any errors now iam looking what are the files need to configured to achieve the following setup. here my question about the config 1. where should i config this Account of VoIP to register, so i can make calls out 2. how do i create 20 users and register them and start making calls 3. where can i see which user called where, and duration 4. how do i configure 20 users can talk each other using extensions. 5. the user side can be Soft Phone using PC or Any cisco ATA Box. what are the config files i need to look any suggestions will be appriciated. ram ___--Bandwidth and Colocation sponsored by Easynews.com --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 ___ --Bandwidth and Colocation sponsored by Easynews.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 Extension behind NAT, Asterisk on a public domain
I'm pasting something from another user on this list from 14/11/05 I would recommend that you do a little research on google, voip- info.org, and the list archives. To connect to an Asterisk box that sits behind NAT, you need to forward ports 5060 and 1-2 too the asterisk box, and you need to configure the externip, localnet, and nat variables in sip.conf. audio problems are almost always due to the RTP stream (ports 1-2) not being forwarded properly, either due to the port forwarding setup or the sip.conf settings. Tom -- Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bharath KhambadkoneSent: Wednesday, November 23, 2005 9:29 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain By default AMP had NAT=yes in sip.conf, I read in some posts to change it to one, i was just trying my luck if that works. I have tried NAT=yes, The Phone gets registered, I can also make recieve calls but as soon as the call is picked I dont hear anything at both ends. Does this have anything to do with codecs?Thanks On 11/22/05, C F [EMAIL PROTECTED] wrote: On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote: Hello All,I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution. My Asterisk is on a public domain, there is no NAT or firewall in front ofIf no nat then why do you have nat=1 in sip.conf? the asteris box. I have sucessfully connected iax2 softphones was able to recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2)with [EMAIL PROTECTED] installed.my Sip.conf :[2008] ;(Sipura2002)username=2008type=friendsecret=2008record_out=Adhocrecord_in=Adhocqualify=noport=5060nat=1[EMAIL PROTECTED] host=dynamicdtmfmode=rfc2833context=from-internalcanreinvite=nocallerid=device 2008[2009] ;X-Lite Soft Phoneusername=2009type=friend secret=2009record_out=Adhocrecord_in=Adhocqualify=noport=5060nat=1[EMAIL PROTECTED]host=dynamicdtmfmode=rfc2833context=from-internal canreinvite=nocallerid=device 2009Thanks in advance.. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___--Bandwidth and Colocation sponsored by Easynews.com --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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Zaptel cards on SuSE?
Hi Ramon, I have used Asterisk 1.09, SuSE 9.3 with a TDM400M with 4 FXOs. I'm planning on trying 10, but haven't found the time. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, November 13, 2005 9:19 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Zaptel cards on SuSE? Hello: So far I have been using Asterisk with SIP and VoIP only. I just received a couple a Zaptel cards from Digium (one analog 2 FXS + 2 FXO, one T1), but I am hesitant to install them because I am afraid I may break the kernel or something. Since Asterisk is not tested under SuSE, I prefer to proceed with caution. So, is there anyone out there using the Zaptel cards under SuSE? TIA, -Ramon F Herrera ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Intel Desktop MotherBoards Unsuitable for DigiumBoards
Hi George, I run an Intel D865GBF Desktop board with Digium's TDM400P with 4 FXOs just fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Pajari Sent: Tuesday, November 08, 2005 6:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for DigiumBoards FYI: We're trying to standardise on a tier one motherboard for the Asterisk boxes we build for customers and thought we'd try to use a low-end Intel Desktop Board since even a low-end Celeron has more than enough horsepower to handle a typical 8x32 PBX. To make a long story short, according to Intel Dealer Technical Support (we became Intel dealers in order to get answers to our questions) there is no Intel motherboard that permits the IRQs to be configured uniquely. They are all hardwired and shared. This information applies to both the Intel Desktop Board and Server Board product lines. Please let me know if your experience differs from what I've been told by Intel. Otherwise, you've been warned -- Intel mobos appear to be unsuitable for use with Digium hardware. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Cisco 7960 Can't be unlocked
Are you trying to go right to 7.4? I had to install 6.3 first and then I could install 7.4. Others have had to upgrade in version increments. (ie from 3 to 4 to 5 to 6 to 7) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Mensel Sent: Friday, May 13, 2005 6:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 Can't be unlocked OK...now that I've been able to get in to the phones, I have (yet another) strange problem: I've configured the phone with a Static IP for testing. TFTP server settings are pointed at a known good TFTP server that other machines on the network are able to access and GET files from. When the phone boots up, it will appear on the network for about 5 pings worth of time, and then dissappear, only to reappear about 20 pings later. The phone's status message indicates a tftp server timeout. My tftp server's logs do not indicate any TFTP activity. The phones are Fw ver 3.1 MF.G2. (SCCP) I'm trying to convert them to SIP (obviously.) Any help will be greatly appreciated. John Mensel On Thursday 12 May 2005 13:06, John Mensel wrote: Tim, Thank you, that took care of the problem -- I'm much obliged. John On Thursday 12 May 2005 11:51, Timothy R. McKee wrote: Those are SCCP based phones. move the cursor to option 3, but do not press select. press **#, then press select. You should see the padlock icon with an unlocked appearance. press 32 and see if you have a YES option (alternate TFTP). If so press yes, then go to option 8 and edit the ip address. The phone sometimes locks itself in the middle and I have to start over. tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Mensel Sent: Thursday, May 12, 2005 12:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 Can't be unlocked Odd problem here--I just got a couple of Cisco 7960s from Ebay that are not functioning as expected.. These 7960s can't seem to be unlocked for manual configuration via any mechanism that I can find. If you go to settings, there is no option 9 (unlock). Available options stop at 4 (Status). **# has no effect. The Phones report that thier current firmware version is 3.1 MF.G2. When plugged into a known good DHCP/TFTP server, the phones will *sometimes* get a DHCP lease that is reflected in SettingsNetwork Configuration, but at no point will they grab new firmware via TFTP. DHCP server logs show the phones trying acquire a lease and then immediately requesting a new one. If anyone has encountered a similar situation, please advise. Thanks, John Mensel ___ 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] cisco7940 upgrading problem
I had to upgrade to version 6.3 first. I then was able to install 7.4 on afterwards. -Original Message- From: [EMAIL PROTECTED] on behalf of Betül Gözlükoglu Sent: Tue 5/3/2005 8:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cisco7940 upgrading problem Hi; I have Cisco 7940 with version 3.2 sccp and want to upgrade it to sip...I got the firmware P0S3-07-4-00...When the phone tries to reach the tftp server gives conf. error and in status messages It says CFG File Not found...Is it because of version problem or something else? Does anybody have any idea? Thanks in advance Betul winmail.dat___ 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] cisco 7960 SIP setup
Mike, Under status, what is the firmware version? You're looking for Appl Load ID, Boot Load ID and Version. Most likely you'll have to get a version 6 SIP image and then you'll be able to install the current 7.4 SIP image after that. In order to get these image files, you have to be a contract paying Cisco client to download them from Cisco's site. I just went through this on my 3 7940s, but I have them all converted over to SIP. You also need to run a TFTP server. I used Cisco's old TFTP program on my Windows XP Pro box. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mk111 Sent: Thursday, April 14, 2005 9:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cisco 7960 SIP setup I can't get the 7960 to reconfigure and work. I am a newbie to voip. I went through the list and read some other comments about the 7960 and unlocking it. It is a used 7960 that came with CallManager. I need to have SIP. I first reset the phone to factory defaults then I changed the TFTP server address in the settings. I have unlocked the phone with **# and it shows the lock as unlocked in the upper right hand corner. I was told that the phone should be able to download the SIP... file once the TFTP address was changed. So far nothing though. Any ideas? Mike ___ 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] Cisco 7940G SIP Conversion
Boris, Thanks for sharing your results. I had to upgrade to image P0S3-06-3-00 first. Once I was at this level, I could then upgrade to image P0S3-07-4-00. I WAS aware of the differences in spelling in the OS79XX.TXT file and SIPDefault.cnf and KEPT the file names DIFFERENT according to a document on voip.info.org. Thanks Again, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Boris Bakchiev Sent: Wednesday, April 13, 2005 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Cisco 7940G SIP Conversion I made the same mistake with my 7960 The content of 'OS79XX.TXT' should be P0S3-07-4-00 and not P003-07-4-00 Same goes for SIPdefault.cnf. After the change everything worked like magic -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michael West Sent: Wednesday, 13 April 2005 22:27 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7940G SIP Conversion Hi, I have three Cisco 7940G phones that I'm trying to convert to SIP Image P0S3-07-3-00 or P0S3-07-4-00. The phone I'm attempting right now has App Load ID P00305000500. I'm running Cisco's TFTP (v1.1) on a Windows XP platform. I have configured my DHCP server to hand out the correct TFTP address as the phone confirms it knows where to find a TFTP server. In the Cisco TFTP status window, I'm receiving the following message continuously: Wed Apr 13 08:21:44 2005: Sending 'OS79XX.TXT' file to 192.168.1.30 in binary mode# I would expect it to attempt to load the image file next that is listed in the OS79XX.txt file (P003-07-4-00, but it continues to load JUST the OS79XX.TCT file continuously. Any ideas? Michael J. West [EMAIL PROTECTED] WESTMark Consulting, Inc. 34 Wasilla Drive Worcester, MA 01604-2411 ___ 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 7940G SIP Conversion
Hi, I have three Cisco 7940G phones that I'm trying to convert to SIP Image P0S3-07-3-00 or P0S3-07-4-00. The phone I'm attempting right now has App Load ID P00305000500. I'm running Cisco's TFTP (v1.1) on a Windows XP platform. I have configured my DHCP server to hand out the correct TFTP address as the phone confirms it knows where to find a TFTP server. In the Cisco TFTP status window, I'm receiving the following message continuously: Wed Apr 13 08:21:44 2005: Sending 'OS79XX.TXT' file to 192.168.1.30 in binary mode# I would expect it to attempt to load the image file next that is listed in the OS79XX.txt file (P003-07-4-00, but it continues to load JUST the OS79XX.TCT file continuously. Any ideas? Michael J. West [EMAIL PROTECTED] WESTMark Consulting, Inc. 34 Wasilla Drive Worcester, MA 01604-2411 ___ 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] Modprobe: FATAL: error running install command for wctdm
Hi Group, Asterisk newbie here- New install using Digium TDM400P with 4 FXO modules using wctdm driver. I was thinking an IRQ conflict, but was able to move from IRQ5 to IRQ3 and still have error. ___ 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
