Re: [Asterisk-Users] AMP - recording call
last I heard that feature wasn't supported how are you getting it to work? - Original Message - From: Alexis F. [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, June 30, 2005 12:29 AM Subject: [Asterisk-Users] AMP - recording call Hi, I'm using the new AMP which provides a call recording. The options of recording call Always and Never are well working. But how to use the On-Demand option ? Should I press a pad ? Is this configured in the featuremap of features.conf ? Why my modifications in that features.conf have no effects ? Please advice me. Alexis. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?
he must have just added it all I saw last I looked was the forums thanks for this ifnormation take care hank - Original Message - From: Dean Collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 30, 2005 4:12 AM Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new? Hank, There is, look again on the [EMAIL PROTECTED] sourceforge site. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of hank Sent: Wednesday, 29 June 2005 11:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new? there is no list on sourceforge I checked whitch is why I offered to start one up sense it isn't aloud on this list. once again folks let me know thanks hank - Original Message - From: Chris Mason [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 8:18 PM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new? hank wrote: there is no list for [EMAIL PROTECTED] and why aint it aloud on this list its asterisk related aint it? chill out guys AAH is an abstraction layer for Asterisk, and the issues that relate to it and not Asterisk belong on it's own list. If there is no list, which would surprise me, use the forums on sourceforge. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP - recording call
okay so this is something new any one using this feature in amp? if so how are you using it meaning how do you get this to work? - Original Message - From: Dean Collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 30, 2005 4:14 AM Subject: RE: [Asterisk-Users] AMP - recording call Always and never work. The on demand hasn't been implemented yet. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of hank Sent: Thursday, 30 June 2005 3:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AMP - recording call last I heard that feature wasn't supported how are you getting it to work? - Original Message - From: Alexis F. [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, June 30, 2005 12:29 AM Subject: [Asterisk-Users] AMP - recording call Hi, I'm using the new AMP which provides a call recording. The options of recording call Always and Never are well working. But how to use the On-Demand option ? Should I press a pad ? Is this configured in the featuremap of features.conf ? Why my modifications in that features.conf have no effects ? Please advice me. Alexis. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?
he must have just added it all I saw last I looked was the forums thanks for this ifnormation take care hank - Original Message - From: Dean Collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 30, 2005 4:12 AM Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new? Hank, There is, look again on the [EMAIL PROTECTED] sourceforge site. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of hank Sent: Wednesday, 29 June 2005 11:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new? there is no list on sourceforge I checked whitch is why I offered to start one up sense it isn't aloud on this list. once again folks let me know thanks hank - Original Message - From: Chris Mason [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 8:18 PM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new? hank wrote: there is no list for [EMAIL PROTECTED] and why aint it aloud on this list its asterisk related aint it? chill out guys AAH is an abstraction layer for Asterisk, and the issues that relate to it and not Asterisk belong on it's own list. If there is no list, which would surprise me, use the forums on sourceforge. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP - recording call
okay so this is something new any one using this feature in amp? if so how are you using it meaning how do you get this to work? - Original Message - From: Dean Collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 30, 2005 4:14 AM Subject: RE: [Asterisk-Users] AMP - recording call Always and never work. The on demand hasn't been implemented yet. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of hank Sent: Thursday, 30 June 2005 3:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AMP - recording call last I heard that feature wasn't supported how are you getting it to work? - Original Message - From: Alexis F. [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, June 30, 2005 12:29 AM Subject: [Asterisk-Users] AMP - recording call Hi, I'm using the new AMP which provides a call recording. The options of recording call Always and Never are well working. But how to use the On-Demand option ? Should I press a pad ? Is this configured in the featuremap of features.conf ? Why my modifications in that features.conf have no effects ? Please advice me. Alexis. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gizmo: Skype done right?
they claim to have a windows download but I can't get the program. also they give no instructions on how to get it connected to asterisk - Original Message - From: Jerry Glomph Black [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 30, 2005 1:31 PM Subject: [Asterisk-Users] Gizmo: Skype done right? I've just submitted this as a Slashdot story, too. I have absolutely no connection with any of the principals, I just think they are doing the right thing. This could have a major impact on the Asterisk community, and VoIP usage in general. Michael Robertson, of mp3.com fame, has been battling for open standards in the IP telephony world, in addition to his better-known Lindows (now Linspire, at http://www.linspire.com) venture to promote Linux on the desktop. His sipphone.com VoIP operation works great for me, but Michael has been long concerned about the totally closed and proprietary nature of Skype (as well as a lot of the misleading hype surrounding it). Today his crew released Gizmo (at http://www.gizmoproject.com) (a tentative name until a better one is found) which has the main benefits of Skype, PLUS it is layered upon SIP, DUNDI, and the existing sipphone.com infrastructure, meaning it is fully interconnectable to the world by obvious and nonobvious techniques, Asterisk being on the top of the obvious charts... This is certainly what I've been waiting for, being totally cheesed by the smarminess of Skype and its founders. Open Standards is one of the most abused concepts this side of Lake Washington, but this comes pretty damn close! ___ 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] Gizmo: Skype done right?
win isn't out yet - Original Message - From: Erik Espinoza [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 30, 2005 3:30 PM Subject: Re: [Asterisk-Users] Gizmo: Skype done right? Agreed. IAX2 would have been a much better way to go. Regardless i don't see how an open source, standards based softphone will compete with Skype. Skype has a few things going for it: 1) Hype, lots of it. It's no coincidence that that the two rhyme 2) Built in traversal of firewalls - p2p style (have I mentioned I hate sip + nat) 3) Encryption, Encryption, Encryption An open source, standards based free implementation does not win over users. There needs to be more, just ask the Ogg folks how MP3's doing. Also it's worth noting that both are free, however Skype has a Linux version! Skype = Win, Mac, Lin x86, PocketPC Gizmo Beta = Mac, Win (Coming Soon: Linux?) Erik On 6/30/05, Matt Fredrickson [EMAIL PROTECTED] wrote: On Thu, Jun 30, 2005 at 01:31:55PM -0700, Jerry Glomph Black wrote: I've just submitted this as a Slashdot story, too. I have absolutely no connection with any of the principals, I just think they are doing the right thing. This could have a major impact on the Asterisk community, and VoIP usage in general. Michael Robertson, of mp3.com fame, has been battling for open standards in the IP telephony world, in addition to his better-known Lindows (now Linspire, at http://www.linspire.com) venture to promote Linux on the desktop. His sipphone.com VoIP operation works great for me, but Michael has been long concerned about the totally closed and proprietary nature of Skype (as well as a lot of the misleading hype surrounding it). Today his crew released Gizmo (at http://www.gizmoproject.com) (a tentative name until a better one is found) which has the main benefits of Skype, PLUS it is layered upon SIP, DUNDI, and the existing sipphone.com infrastructure, ^^^ Looks like they already messed up... If they're going to redo all of this anyway, they might as well use a protocol like IAX where you don't have NAT problems. Matthew Fredrickson ___ 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] Shoutcast Music On Hold problems?
um do I paste the below info in to a file and name it something? this looks really odd. from what my screen reader is reading to me it looks like to be some sort of script file or something - Original Message - From: Huddleston, Robert [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 5:55 AM Subject: RE: [Asterisk-Users] Shoutcast Music On Hold problems? bash-3.00# cat musiconhold.conf | more ; ; Music on hold class definitions ; [classes] ; Christian Rock.NET ;default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/ ;loud = mp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/ ; Cleft in the Rock Radio (TESTING) default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.97.198.50:30518/ loud = mp3:/var/lib/asterisk/mohmp3-empty,http://209.97.198.50:30518/ bash-3.00# pwd /var/lib/asterisk/mohmp3-empty bash-3.00# ls -la total 8 drwxr-xr-x 2 root root 4096 Jun 15 15:21 . drwxr-xr-x 9 root root 4096 Jun 15 15:18 .. -rw-r--r-- 1 root root0 Jun 15 15:21 empty.mp3 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: Wednesday, June 29, 2005 1:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Fw: [Asterisk-Users] Shoutcast Music On Hold problems? - Original Message - From: hank [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 10:52 PM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? I am using [EMAIL PROTECTED] 1.0 my mp3 is called mp3 it has nothing before it it is 0 bytes does my mp3 of 0 bytes need to have a .mp3 or does it need to be called anything? thanks hank - Original Message - From: Huddleston, Robert [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 11:52 AM Subject: RE: [Asterisk-Users] Shoutcast Music On Hold problems? Worked for me with a different stream... I ran into this same problem before - but it was my own fault for not RTM... Both the manual and ast install advised of verifying correct version of mpg123... I had wrong version and thus got no noise... If you follow the directions explicitly laid out on the wiki you should have no problems. I use christianrock.net's shoutcast stream Like this in musiconhold.conf default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: Tuesday, June 28, 2005 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? I tried that the stream i tried to use orriginally was http://209.97.198.50:30518 all I get is silence when I put the person on hold thanks hank - Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 2:50 AM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? On Mon, 2005-06-27 at 22:51 -0700, hank wrote: mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/ I haven't tried this myself but if I put www.waixwave.com:8000 in Firefox I get connection refused. Try another site that actually streams music. Shoutcast.org should have a nice list. Regards, Patrick ___ 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 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
Re: [Asterisk-Users] Teliax Problems
I use them and I have another friend with them so far they are okay, support is awesome, not any outages thus far and have been with them for about 3 weeks, not sure if they support iax or not, they do allow biod, prices are good. hth - Original Message - From: Chris Coulthurst [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 4:48 AM Subject: RE: [Asterisk-Users] Teliax Problems Does anyone have anything +/- to say about TeleSIP? They appear to have local DIDs where I live and all comments on the wiki indicate they are reputable.. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Wednesday, June 29, 2005 5:22 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Teliax Problems | | | I'm currently unable to register with Teliax's server via IAX2 and | can't reach them via either of their phone numbers. Their |website is | up and I have logged a support incident. | | Is anyone else experiencing the same problems? Having been |caught up | in the Broadvoice fiasco a couple of months back, I'm hoping that | Teliax is not going through the same sort of thing. | |An ethereal trace indicates the IP address is active, but it |is not responding to iax packets (registration). So, either |their asterisk app has failed or they have folded their tent as well. | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-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
Fw: [Asterisk-Users] Shoutcast Music On Hold problems?
- Original Message - From: hank [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 8:52 AM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? let me know what happens with the Cleft in the Rock Radio weather that works or not also the way my screen reader was reading that.conf file was really odd it took me a while to figure out what that was supposed to be I am getting silence but that may be to the version of mpg123 that [EMAIL PROTECTED] is using in regards to the results of the conf file is [EMAIL PROTECTED] using the correct version? thanks hank - Original Message - From: Huddleston, Robert [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 5:55 AM Subject: RE: [Asterisk-Users] Shoutcast Music On Hold problems? bash-3.00# cat musiconhold.conf | more ; ; Music on hold class definitions ; [classes] ; Christian Rock.NET ;default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/ ;loud = mp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/ ; Cleft in the Rock Radio (TESTING) default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.97.198.50:30518/ loud = mp3:/var/lib/asterisk/mohmp3-empty,http://209.97.198.50:30518/ bash-3.00# pwd /var/lib/asterisk/mohmp3-empty bash-3.00# ls -la total 8 drwxr-xr-x 2 root root 4096 Jun 15 15:21 . drwxr-xr-x 9 root root 4096 Jun 15 15:18 .. -rw-r--r-- 1 root root0 Jun 15 15:21 empty.mp3 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: Wednesday, June 29, 2005 1:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Fw: [Asterisk-Users] Shoutcast Music On Hold problems? - Original Message - From: hank [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 10:52 PM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? I am using [EMAIL PROTECTED] 1.0 my mp3 is called mp3 it has nothing before it it is 0 bytes does my mp3 of 0 bytes need to have a .mp3 or does it need to be called anything? thanks hank - Original Message - From: Huddleston, Robert [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 11:52 AM Subject: RE: [Asterisk-Users] Shoutcast Music On Hold problems? Worked for me with a different stream... I ran into this same problem before - but it was my own fault for not RTM... Both the manual and ast install advised of verifying correct version of mpg123... I had wrong version and thus got no noise... If you follow the directions explicitly laid out on the wiki you should have no problems. I use christianrock.net's shoutcast stream Like this in musiconhold.conf default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: Tuesday, June 28, 2005 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? I tried that the stream i tried to use orriginally was http://209.97.198.50:30518 all I get is silence when I put the person on hold thanks hank - Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 2:50 AM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? On Mon, 2005-06-27 at 22:51 -0700, hank wrote: mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/ I haven't tried this myself but if I put www.waixwave.com:8000 in Firefox I get connection refused. Try another site that actually streams music. Shoutcast.org should have a nice list. Regards, Patrick ___ 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
Re: [Asterisk-Users] Multiple Timezones with Asterisk
for that matter how do you set it up for pst? mine is set to est and its really anoying - Original Message - From: Max Clark [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 12:53 PM Subject: [Asterisk-Users] Multiple Timezones with Asterisk Hi all, I am curious if it is possible to have multiple timezones registered on an Asterisk server for Voicemail (i.e. so that PST users get PST time, and EST users get EST time)? Ideally I would like to set my Asterisk box to GMT and have a switch depending on where the user was registered from. Is this possible? Thanks, Max ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?
what is the dhcp server used for? - Original Message - From: JD Austin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 5:49 PM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new? New features include: CentOS 3.5 Asterisk 1.0.8 New Zaptel Driver from CVS Built-in DHCP server David Shaw wrote: Hello I saw Ver1.2 is out. Whats new? Thanks for the hard work, David ___ 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 -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?
there is no list for [EMAIL PROTECTED] and why aint it aloud on this list its asterisk related aint it? chill out guys - Original Message - From: Mike [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 3:04 PM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new? This is NOT the AAH list, please check sf.net for information On Wed, 29 Jun 2005, David Shaw wrote: Hello I saw Ver1.2 is out. Whats new? Thanks for the hard work, David ___ 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] Asterisk@Home Ver 1.2 Whats new?
perhaps I should create a list for [EMAIL PROTECTED] any one want me to do this? I can create one on yahoogroups I find email list easier to use then the forums and I am sure I am not alone in this. let me know off list [EMAIL PROTECTED] if you want a [EMAIL PROTECTED] list started laters hank - Original Message - From: Chris Mason [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 8:18 PM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new? hank wrote: there is no list for [EMAIL PROTECTED] and why aint it aloud on this list its asterisk related aint it? chill out guys AAH is an abstraction layer for Asterisk, and the issues that relate to it and not Asterisk belong on it's own list. If there is no list, which would surprise me, use the forums on sourceforge. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?
there is no list on sourceforge I checked whitch is why I offered to start one up sense it isn't aloud on this list. once again folks let me know thanks hank - Original Message - From: Chris Mason [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 8:18 PM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new? hank wrote: there is no list for [EMAIL PROTECTED] and why aint it aloud on this list its asterisk related aint it? chill out guys AAH is an abstraction layer for Asterisk, and the issues that relate to it and not Asterisk belong on it's own list. If there is no list, which would surprise me, use the forums on sourceforge. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?
okay my apologies thanks hank - Original Message - From: Dean Collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 8:21 PM Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new? Hank, yes there is. Go to the sourceforge site, there is a forum just for aah there. Any specific aah questions should be posted there. Any specific amp questions should be posted on the AMP sourceforge page. Any general asterisk questions should be posted here. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of hank Sent: Wednesday, 29 June 2005 10:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new? there is no list for [EMAIL PROTECTED] and why aint it aloud on this list its asterisk related aint it? chill out guys - Original Message - From: Mike [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 3:04 PM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] Ver 1.2 Whats new? This is NOT the AAH list, please check sf.net for information On Wed, 29 Jun 2005, David Shaw wrote: Hello I saw Ver1.2 is out. Whats new? Thanks for the hard work, David ___ 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 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] Shoutcast Music On Hold problems?
I tried that the stream i tried to use orriginally was http://209.97.198.50:30518 all I get is silence when I put the person on hold thanks hank - Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 2:50 AM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? On Mon, 2005-06-27 at 22:51 -0700, hank wrote: mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/ I haven't tried this myself but if I put www.waixwave.com:8000 in Firefox I get connection refused. Try another site that actually streams music. Shoutcast.org should have a nice list. Regards, Patrick ___ 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] LiveVoip is Bankrupt - Why this thread
a man loose his mind? what was the archive posting on that one? I want to read that :) laters hank - Original Message - From: Michael Di Martino [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 6:44 AM Subject: RE: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Monday, June 27, 2005 5:26 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread On Monday 27 June 2005 15:46, steve szmidt wrote: One could probably argue effectively for an Asterisk-Basic list. Or an Asterisk-Advanced user list. Something that makes it easier to get started without being overwhelmed by 10,000-15,000 users posts. A place that frequently posted links to the beginner pages on the wiki. We've effectively argued it to death many many times over the course of the last few years. Check the archives -- it's been thought up and re-thought up and dismissed each and every time. Basic issue: nobody will want to sit on the newbie list because they'll end up answering all the same questions over and over since nobody really seems to want to read for themselves. It's the same argument that comes around for forums, except that last time I think we actually witnessed a man lose his mind on the mailing list. That was entertaining. :-) ~ Their would not be so many newbie questions if their was 1. A fully indexed searchable archive list and 2. Good solid documentation. ___ 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] How do you handle NAT?
how easy is it to set up a stun server? with asterisk amd will this fix part of the nat problem? - Original Message - From: Ray Van Dolson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 8:14 AM Subject: Re: [Asterisk-Users] How do you handle NAT? We've been feeling our way along with the NAT stuff (using SIP) as well. At this point we are fairly small, so the keep-alive packets are not too bad. What type of user load are you at and what are the specs on your Asterisk box? I'm concerned we may run into this as well. We do have the luxury that each Sipura device we use is sitting behind its own NAT (a customer CPE). So we can do port-forwarding and in combination with a STUN server (MyStun), things work quite well. The only issues left to deal with are a lingering problem with ip_conntrack entries staying cached because of the keep alive packets due to qualify=yes after the CPE's IP address changes. Curious to hear other's setups as well. I would *love* to start using the IAXy instead, but it has a couple shortcomings over the Sipura 2002's we're using now: - About $10/more - Only has one line (apparently two lines is a bit more of a selling point). Still trying to figure out a good way to make a case for the IAXy though. Ray On Tue, Jun 28, 2005 at 09:59:49AM -0500, Matthew Boehm wrote: We are interested in how other people are handling NAT problems. We have several customers all of which have some sort of firewall/NAT device at their location. For simplicity sake, all customers' internal networks are 192.168.*.*. Our asterisk box is on public IP not blocked by any FW/NAT. I use QUALIFY=yes on all our customers' phones and I feel that sending out 80-something keep-alive packets is causing our box to crawl and cause bad calls. Would SER be better in this case? Should I have phones register with SER instead of with Asterisk? Thanks, Matthew P.S. Yes, I have read stuff on NAT on the wiki. I'm more interested in other real world, working, solutions. ___ 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] Re: teliax [Was: LiveVoip is Bankrupt]
how do they know if your calling your tax dude or something? what do they do monitor the calls or something? - Original Message - From: John Goerzen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 6:46 AM Subject: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt] On 2005-06-28, r00t [EMAIL PROTECTED] wrote: I'll second voipjet for outbound only. While many reported problems to VoipJet bothers me for two reasons. First, their terms of service are absolutely insane. Users are specifically forbidden to place calls regarding medical or financial matters over VoipJet. So I couldn't call my tax preparer or schedule a doctor appointment under their contract. There are many other insane things about it; see the thread at http://lists.digium.com/pipermail/asterisk-users/2005-March/094251.html. One of the little gems is that if you tell anyone you use VoipJet, you violate your contract. So all of you that have been praising your VoipJet service here: prepare to be disconnected! :-) Secondly, they are not honest about what they are doing. They clearly are aiming some services at small self-sufficient end-users, yet they claim to provide services to commercial carriers only (and their ToS tries to enforce that.) Got to love little statements like Emerging VoIP service providers can make payments through PayPal. -- John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do you handle NAT?
I am fighting this as we speak I have a friend who can't connect to me cause of a damn nat frankly its irritating me so any recommendations are welcome - Original Message - From: Matthew Boehm [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 7:59 AM Subject: [Asterisk-Users] How do you handle NAT? We are interested in how other people are handling NAT problems. We have several customers all of which have some sort of firewall/NAT device at their location. For simplicity sake, all customers' internal networks are 192.168.*.*. Our asterisk box is on public IP not blocked by any FW/NAT. I use QUALIFY=yes on all our customers' phones and I feel that sending out 80-something keep-alive packets is causing our box to crawl and cause bad calls. Would SER be better in this case? Should I have phones register with SER instead of with Asterisk? Thanks, Matthew P.S. Yes, I have read stuff on NAT on the wiki. I'm more interested in other real world, working, solutions. ___ 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] enabling stun on asterisk?
hello I am going to be setting up a stun server on windows how do I enable it to work withasterisk? thanks hank ___ 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] Shoutcast Music On Hold problems?
how is your mp3 named? in the empty directory? I think I may have mine screwed up can you email me the mp3 that you have so I can just drop in to the directory and then add the line to my musiconhold.conf? thanks hank - Original Message - From: Huddleston, Robert [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 11:52 AM Subject: RE: [Asterisk-Users] Shoutcast Music On Hold problems? Worked for me with a different stream... I ran into this same problem before - but it was my own fault for not RTM... Both the manual and ast install advised of verifying correct version of mpg123... I had wrong version and thus got no noise... If you follow the directions explicitly laid out on the wiki you should have no problems. I use christianrock.net's shoutcast stream Like this in musiconhold.conf default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: Tuesday, June 28, 2005 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? I tried that the stream i tried to use orriginally was http://209.97.198.50:30518 all I get is silence when I put the person on hold thanks hank - Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 2:50 AM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? On Mon, 2005-06-27 at 22:51 -0700, hank wrote: mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/ I haven't tried this myself but if I put www.waixwave.com:8000 in Firefox I get connection refused. Try another site that actually streams music. Shoutcast.org should have a nice list. Regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do you handle NAT?
I think my problem is numbrer 3 cause basicly my friend who is not on my router is trying to get connected to me but can't and I am the 1 that is behind a nat. thanks hank - Original Message - From: Sebastian Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 12:45 PM Subject: Re: [Asterisk-Users] How do you handle NAT? Hi everyone. 1. Asterisk as a SIP client behind nat, connecting to outside SIP Proxies: #1 works with a NAT-supporting proxy as SIP Express router as the outside proxy. (Get an account at IPtel.org and try!). Fails with Free World Dialup. 2. Asterisk as a SIP client behind nat, connecting to inside SIP proxies: #2 Works- no NAT in between 3. Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk: #3 Works with port forwarding and some header mangling magic 4. Asterisk as a SIP server behind nat, clients on the inside connecting to Asterisk: #4 Works - no NAT in between 5. Asterisk as a SIP client outside nat, connecting to outside SIP proxies: #5 is no problem. No NAT in the middle 6. Asterisk as a SIP client outside nat, connecting to inside SIP proxies: #6 is a problem if no port forwarding is done, similar to 3 above. 7. Asterisk as a SIP server outside nat, clients on the outside connecting to Asterisk: #7 is no problem. No NAT in the middle 8. Asterisk as a SIP server outside nat, clients on the inside connecting to Asterisk: #8 is solved with nat=yes and qualify=xxx in sip.conf for the client in most cases. Some clients (X-lite) assist themselves by using STUN and sending UDP keep-alive packets. Qualify sends keep-alive packets from Asterisk to the client on the inside. from wiki Now, if you net to define a NAT, you have to set asterisk to canreinvite=no, qualify=yes and nat=1. Also, INSTEAD of NAT, you can use a STUN server. To use a STUN server you should set asterisk to canreinvite=no, qualify=no and nat=0 (the STUN configuration is in your agents). Sebas hank wrote: how easy is it to set up a stun server? with asterisk amd will this fix part of the nat problem? - Original Message - From: Ray Van Dolson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 8:14 AM Subject: Re: [Asterisk-Users] How do you handle NAT? We've been feeling our way along with the NAT stuff (using SIP) as well. At this point we are fairly small, so the keep-alive packets are not too bad. What type of user load are you at and what are the specs on your Asterisk box? I'm concerned we may run into this as well. We do have the luxury that each Sipura device we use is sitting behind its own NAT (a customer CPE). So we can do port-forwarding and in combination with a STUN server (MyStun), things work quite well. The only issues left to deal with are a lingering problem with ip_conntrack entries staying cached because of the keep alive packets due to qualify=yes after the CPE's IP address changes. Curious to hear other's setups as well. I would *love* to start using the IAXy instead, but it has a couple shortcomings over the Sipura 2002's we're using now: - About $10/more - Only has one line (apparently two lines is a bit more of a selling point). Still trying to figure out a good way to make a case for the IAXy though. Ray On Tue, Jun 28, 2005 at 09:59:49AM -0500, Matthew Boehm wrote: We are interested in how other people are handling NAT problems. We have several customers all of which have some sort of firewall/NAT device at their location. For simplicity sake, all customers' internal networks are 192.168.*.*. Our asterisk box is on public IP not blocked by any FW/NAT. I use QUALIFY=yes on all our customers' phones and I feel that sending out 80-something keep-alive packets is causing our box to crawl and cause bad calls. Would SER be better in this case? Should I have phones register with SER instead of with Asterisk? Thanks, Matthew P.S. Yes, I have read stuff on NAT on the wiki. I'm more interested in other real world, working, solutions. ___ 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 -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250 4 piso Tl.: 4 706- (int. 121) [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http
Fw: [Asterisk-Users] Shoutcast Music On Hold problems?
- Original Message - From: hank [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 10:52 PM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? I am using [EMAIL PROTECTED] 1.0 my mp3 is called mp3 it has nothing before it it is 0 bytes does my mp3 of 0 bytes need to have a .mp3 or does it need to be called anything? thanks hank - Original Message - From: Huddleston, Robert [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 11:52 AM Subject: RE: [Asterisk-Users] Shoutcast Music On Hold problems? Worked for me with a different stream... I ran into this same problem before - but it was my own fault for not RTM... Both the manual and ast install advised of verifying correct version of mpg123... I had wrong version and thus got no noise... If you follow the directions explicitly laid out on the wiki you should have no problems. I use christianrock.net's shoutcast stream Like this in musiconhold.conf default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: Tuesday, June 28, 2005 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? I tried that the stream i tried to use orriginally was http://209.97.198.50:30518 all I get is silence when I put the person on hold thanks hank - Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 28, 2005 2:50 AM Subject: Re: [Asterisk-Users] Shoutcast Music On Hold problems? On Mon, 2005-06-27 at 22:51 -0700, hank wrote: mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/ I haven't tried this myself but if I put www.waixwave.com:8000 in Firefox I get connection refused. Try another site that actually streams music. Shoutcast.org should have a nice list. Regards, Patrick ___ 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] Fw: shoutcast mp3 music onhold with amp portal?
- Original Message - From: hank To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, June 27, 2005 9:13 PM Subject: shoutcast mp3 music onhold with amp portal? hello I use amp the asterisk management portal I went to www.voip-info.org I followed the instructions that were given on http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf it isn't working what do I need to do to get this to work? I have a empty mp3 file like it said. thanks hank ___ 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] Shoutcast Music On Hold problems?
hello I followed the info given and I can't seem to get this to work has any one sucessfully done this? if so can you help me out? I am trying to use a 128 kbps mp3 feed to stream to people while there on hold the info I am using is below. Shoutcast Music On Hold You can have asterisk use a streaming source for on-hold music. Make a directory and put a 0 size file ending in .mp3.I called my directory: /var/lib/asterisk/mohmp3-empty in musiconhold.conf, add a line such as:default = mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/ thanks hank ___ 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 problem
go in to the [EMAIL PROTECTED] set up http://yourip/maint inter in your user name and password go to config edit sip.conf go to the general section Note: if you are behind a NAT Firewall, you will probably need to add thefollowing lines to hte [General] section of your sip.conf file. Adjust thenumbers as needed to match your configuration: externip=66.5.21.6 replace the above with your public ip address localnet=192.168.5.0/255.255.255.0 the 192 adddress mentioned above needs to be replaced with the ip address of your asterisk server eg mine is on 192.168.15.101 so it would look like localnet=192.168.15.101/255.255.255.0 hth hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 - Original Message - From: Betl Gzlkolu To: asterisk-users@lists.digium.com Sent: Tuesday, May 24, 2005 3:57 AM Subject: [Asterisk-Users] nat problem Hi; Using [EMAIL PROTECTED] and it working well in network but when can not logged in over internet although the server is reachable Does anybody has any idea? Thanks Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mmkn olmayabilir. Mesaji alan kisi, mesajin gnderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkrler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___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] Asterisk@home
and man does it kick ass to!!! :) email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 - Original Message - From: Ariel Batista To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, May 24, 2005 4:19 AM Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] Asterisk @ Home This CD will install everything you need to get your Software PBX going. Its a complete ISO CD that brings together the OS (CentOS 3.4) Asterisk Software version 1.0.7 stable AMP Asterisk Management Portal Web GUI FTP TFTP Plus many more items. Every pre-configured to install and run out of the box. Just put it into your CD drive it will format and setup asterisk for you. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of QuintinSent: Tuesday, May 24, 2005 6:41 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] [EMAIL PROTECTED] Hi Can any one tel me what is [EMAIL PROTECTED] Thx Q ___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] BudgeTone 101 doesn't register with FirmWare1.5.23
they ever going to fix it? email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 - Original Message - From: Bob Goddard [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 24, 2005 7:05 AM Subject: Re: [Asterisk-Users] BudgeTone 101 doesn't register with FirmWare1.5.23 On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote: Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0.3.81 version, the phone register properly. Is ther any know bug with the SW Version? There is a registration bug with the BT101s which has been around for years which is still in 1.0.6.3. If the phone is is rung but not answered, allowed to reregister as normal, then rung again and not answered, then the phone will never regiseter again until rebooted. Grandstream have known about this bug since 3rd March but have only acknowledged it since 15th May. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice delivers CID even when restricted?
I missed the numbers can some one repost? thanks hank email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 - Original Message - From: Darren Nickerson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 24, 2005 12:58 PM Subject: Re: [Asterisk-Users] Broadvoice delivers CID even when restricted? Matthew Crocker [EMAIL PROTECTED]: I know David Epstein and Dan Geopp personally, they are good guys, posting their direct office numbers on the mailing list is extremely bad form. As someone who has been given unacceptably vague, meaningless and often blatantly dishonest replies from badly trained support staff over the past three weeks, I'm glad to see these numbers released. Broadvoice has no escalation procedure ..., and they simply don't have any information for the people who eventually answer the phone to give to customers. I mean if you're going to pick up the phone, you'd think someone would at least tell them what to say!? They may be nice guys but their handling of this recent crisis means that they should anticipate taking a few calls. After three weeks of we're working on it - call us in a few days answers, we've been told that Broadvoice doesn't have inbound 800 service any more, and that they haven't actually been looking into our issue at all, since it's a network-wide failure whose reasons they understand, but just don't generally divulge. Apparently they're working with a new carrier to bring 800 back, but at this time it's completely out of service (can anyone confirm this?). And yet I can still buy an instant-activation 800 number online, and have been able to for this entire period. I've been a loyal customer, been patient and stuck with them for weeks. All I want now is the truth, and it's good to have new numbers at which to seek that truth. If inbound 800's not coming back, I'll look elsewhere for inbound 800 service and pick through the pros/cons of the various other ITSPs who have been reported to interoperate (with varying degrees of success) with Asterisk. If it really is coming back in any meaningful way, I can wait a few more weeks. -d ___ 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] BudgeTone 101 doesn't register with FirmWare1.5.23
hey let me know when its fixed so I can upgrade mine to :) take care hank email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 - Original Message - From: Bob Goddard [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 24, 2005 2:49 PM Subject: Re: [Asterisk-Users] BudgeTone 101 doesn't register with FirmWare1.5.23 On Tuesday 24 May 2005 17:07, Daniel ANDRE wrote: Bob Goddard a écrit : On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote: Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0.3.81 version, the phone register properly. Is ther any know bug with the SW Version? There is a registration bug with the BT101s which has been around for years which is still in 1.0.6.3. If the phone is is rung but not answered, allowed to reregister as normal, then rung again and not answered, then the phone will never regiseter again until rebooted. Grandstream have known about this bug since 3rd March but have only acknowledged it since 15th May. Thank you Bob but I have just found what was wrong: two dhcp servers on the same network. Another question, Is the 1.0.6.3 stable enough for production use? I don't know. I only check for this single bug and until it's fixed no phone gets upgraded. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spa-1001 with asterisk?
hello my friend is trying to get his spa-1001 (sipura) 1001 connected to my asterisk box. he reset his spa-1001 to factory defaults I emailed him the voip-info page I found on google and yes I did look on google anyways he isn't able to get the thing to connect to it eg getting a dial tone, he did install x-lite and it worked fine with that am running [EMAIL PROTECTED] 1.0 can some one please tell me the steps to get the spa-1001 working for my friend, I will be passing the instructions to him. thanks for any help you can give. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 ___ 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] play gsm files in windows
there is also a winamp plugin for playing gsm files go to www.winamp.com/plugins do a search for gsm you will find it hth hank email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 - Original Message - From: El Flynn [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 23, 2005 8:07 AM Subject: Re: [Asterisk-Users] play gsm files in windows Brett, Gary wrote: Does anybody know of a WINDOWS application (preferably freeware) that will simply playback asterisk GSM sound files, I don't want to record them, just playback the ones that are currently there. Any help would be greatly appreciated You could try Audacity (http://audacity.sourceforge.net). You have to use the Import Raw Data feature and open it as a GSM file. Flynn ___ 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] spa-1001 not getting a dial tone on my pbx
hello my friend has the proxy set up his extention set up his password set up but he isn't getting a dial tone is there a second setting we need to put the address in? he is going to advenced settings line1 and in the proxy address box he is putting the info in below is the way he has it set up Sipura SPA Configuration Sipura Technology IncInfoSystemSIPProvisioningRegionalLine 1User 1User Loginbasic |advanced Table with 4 columns and 75 rows Line Enable:yes Streaming Audio Server (SAS)SAS Enable:noSAS DLG Refresh Intvl:30SAS Inbound RTP Sink: NAT SettingsNAT Mapping Enable:yesNAT Keep Alive Enable:yesNAT Keep Alive Msg:$NOTIFYNAT Keep Alive Dest:$PROXY Network SettingsSIP TOS/DiffServ Value:0x68Network Jitter Level:highRTP TOS/DiffServ Value:0xb8 SIP SettingsSIP Port:5060SIP 100REL Enable:noEXT SIP Port: Auth Resync-Reboot:yesSIP Debug Option:none Call Feature SettingsBlind Attn-Xfer Enable:noMOH Server: Xfer When Hangup Conf:yes Proxy and RegistrationProxy:67.183.118.6Use Outbound Proxy:noOutbound Proxy: Use OB Proxy In Dialog:noRegister:yesMake Call Without Reg:noRegister Expires:60Ans Call Without Reg:noUse DNS SRV:noDNS SRV Auto Prefix:noProxy Fallback Intvl:3600 Subscriber InformationDisplay Name:Herbie AllenUser ID:202Password:*Use Auth ID:yesAuth ID:202 Mini Certificate: SRTP Private Key: Supplementary Service SubscriptionCall Waiting Serv:yesBlock CID Serv:yesBlock ANC Serv:yesDist Ring Serv:yesCfwd All Serv:yesCfwd Busy Serv:yesCfwd No Ans Serv:yesCfwd Sel Serv:yesCfwd Last Serv:yesBlock Last Serv:yesAccept Last Serv:yesDND Serv:yesCID Serv:yesCWCID Serv:yesCall Return Serv:yesCall Back Serv:yesThree Way Call Serv:yesThree Way Conf Serv:yesAttn Transfer Serv:yesUnattn Transfer Serv:yesMWI Serv:yesVMWI Serv:yesSpeed Dial Serv:yesSecure Call Serv:yesReferral Serv:yesFeature Dial Serv:yes Audio ConfigurationPreferred Codec:G711uSilence Supp Enable:noUse Pref Codec Only:noSilence Threshold:mediumG729a Enable:yesEcho Canc Enable:yesG723 Enable:yesEcho Canc Adapt Enable:yesG726-16 Enable:yesEcho Supp Enable:yesG726-24 Enable:yesFAX CED Detect Enable:yesG726-32 Enable:yesFAX CNG Detect Enable:yesG726-40 Enable:yesFAX Passthru Codec:G711uDTMF Tx Method:AutoFAX Codec Symmetric:yesHook Flash Tx Method:NoneFAX Passthru Method:NSERelease Unused Codec:yesFAX Process NSE:yes Dial PlanDial Plan:(xx.|*xx.|**xx.|#xx.)Enable IP Dialing:no FXS Port Polarity ConfigurationIdle Polarity:ForwardCaller Conn Polarity:ForwardCallee Conn Polarity:Forward table end Undo All Changes Submit All Changes User Loginbasic |advanced Copyright © 2003 Sipura Technology. All Rights Reserved. --No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 5/22/2005 email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 ___ 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 with vonage linksys adapter?
hello do you know if vonage unlocks there linksys adapter to use with other providers? I want to use my ixisting vonage adapter with asterisk and cancil my vonage service. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 ___ 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] having asterisk play music on hold to callers while phone rings?
hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 ___ 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] having asterisk play music on hold to callerswhile phone rings?
what config is this found in? email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 - Original Message - From: Jon Gabrielson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 21, 2005 2:05 PM Subject: Re: [Asterisk-Users] having asterisk play music on hold to callerswhile phone rings? use option m in the cmd dial. Cheers, Jon. On Saturday 21 May 2005 03:26 pm, hank smith wrote: hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 ___ 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] having asterisk play music on hold tocallerswhile phone rings?
yep I have hold music other wise looks like I am going to have to go in to the [EMAIL PROTECTED] and configure it via that method can you give me pointers on what the dial line lookslike so I dont screw this thing up?? they dont recommend editing this stuff bye hand unless you know what you are doing. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 - Original Message - From: Gary Lawrence To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Saturday, May 21, 2005 2:09 PM Subject: RE: [Asterisk-Users] having asterisk play music on hold tocallerswhile phone rings? Edit the extensions.conf and put an m at the end of the dial line. Do you have hold music otherwise? Sincerely; Gary Lawrence ITcom.Net 866.4ITcom1 866.448.2661 -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank smithSent: Saturday, May 21, 2005 4:26 PMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] having asterisk play music on hold to callerswhile phone rings? hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith5 ___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] Re: Asterisk-1.0.7 Build - Serious issues
isn't [EMAIL PROTECTED] included in 1.07? of asterisk? also I checked the asterisk.org site and saw 1.06 but not the latest when was it put up on asterisk.org? - Original Message - From: Tony Mountifield [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, March 31, 2005 8:52 AM Subject: [Asterisk-Users] Re: Asterisk-1.0.7 Build - Serious issues In article [EMAIL PROTECTED], Kanuri, Seshu (Company IT) [EMAIL PROTECTED] wrote: Folks! I want to let everyone know that I have been trying to migrate from 1.0.6 to 1.0.7 last few days and I have come across serious issues in the build 1.0.7. What I found are listed below. I would recommend everyone to hold off any upgrade till the next build. But many people have successfully used 1.0.7, so it's possible the problem is at your end. 1)Voicemail - No Audio. Asterisk is not able to stream the voice to the Uas. 0-9 Digit files seem to be missing and Asterisk does not try to say extension numbers for the called user. My guess is all these .gsm files are corrupt and hence you don't hear anything. They are fine in both the tar.gz file and from CVS stable. 2)Music on hold - .MP3 files in the ../mohmp3 and other folders are corrupt. When we tried to play these files using a media player, all we hear is gibberish. So are these. 3)DTMF is screwed up. Whatever worked in 1.06 does not work now when we configure this for RFC2833. Has anyone upgraded to 1.0.7 from 1.0.6 and had these issues and been able to find a fix? Try downloading again. If using FTP, ensure you have BINARY mode enabled. Seshu NOTICE: If received in error, please destroy and notify sender. But why destroy the sender as well as notifying them? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] using amp with asterisk?
hello I have asterisk 1.0 running on fedora core3 and amp version 1.06 I think is the version its the version down below the current release, I have fwd working threw iax on outbound calls fine but I can't get inbound to work, has any one successfully gotten this to work? if so can you tell me what you did to get it working? there isn't alot of documentation on amp so I am kind of lost. thanks hank ___ 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] Xten-lite for linux
do you know if it is gtk2? - Original Message - From: Bruno Hertz [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, March 30, 2005 10:31 PM Subject: Re: [Asterisk-Users] Xten-lite for linux Kris Edwards [EMAIL PROTECTED] writes: This is the best linux sip phone I've used so far. Audio quality has been perfect and it seems really stable, so hopefully it will be out of beta soon. I might actually pay for the full version! (not counting console games, that would be the second piece of software I've purchaced since 1987). Sounds rather like you want to sell the full version. Myself, I don't know about recent betas since, frankly, I didn't care anymore after initial experiences being pretty much disappointing. The first beta I got produced no audio at all, and we had a tough time to convince the developer that it wasn't a driver issue. The next releases then had huge latencies, primarily due to the Xlite audio setup. Now, I admit that setting up audio for interactive/'realtime' apps on linux is a mess, but various open source projects have already done much better. So no, in contrast to your plug I'm not as enthusiastic myself, especially since audio quality resp. latency is the one major trouble I had with linux softphones. E.g. iaxcomm would be great and totally satisfying for me if latency were (significantly) less than 1 second. Regards, Bruno. ___ 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] Setting Up @Home 0.8 Guide
is there a way you can write those screen shots in to text format on the user guide? I am a blind computer user and am unable to see the examples that are shown on the site. thanks hank - Original Message - From: Kerry Garrison [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, March 30, 2005 11:12 PM Subject: [Asterisk-Users] Setting Up @Home 0.8 Guide Because of all of the changes to AMP, we have written up a completely new How-To Guide for [EMAIL PROTECTED] v0.8. Our first example uses BroadVoice for the trunk. http://www.geekgazette.com -Kerry ___ 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] phpconfig
Hello, I recently downloaded phpconfig from http://asterisk.espia-net.net/horde/chora/cvs.php/phpconfig?login=2 but on installing it, my interface does not look like the one at http://rd.it.utah.edu/phpconfig/. The main differences are: 1)On opening a file for editing, on the left menu mine has ony two links i.e Header and the filename.conf as opposed to the deffrent sections on the demo site. Even when i click Header, the page just refreshes and doesn't pick out only the header. 2) I also lack the other links on the right which are in most cases numbers Questions: 1) Am i using an older version? If so, where can i get a newr version? 2) Am i missing some configuration, which one? Thanks in advance, Allan ___ 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] phpconfig
Hello, That's the document i read and got all the relevant links. I also tried to follow all the predures . More help is appreciated, Thanks very much Allan On Mon, 28 Feb 2005 10:13:02 -0500, Time Bandit [EMAIL PROTECTED] wrote: Questions: 1) Am i using an older version? If so, where can i get a newr version? 2) Am i missing some configuration, which one? See this newly created document, it explains everything you need to make it work. http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig It's been written with the help of peoples on this list. hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@Home software?
hello I was looking around sourceforge.net and came accraust this application for asterisk pbx [EMAIL PROTECTED] was wondering if any one has tried it and how good it works? what all can you do with this thing as far as feature set goes? the url for this project is http://sourceforge.net/projects/asteriskathome/ thanks hank business site:http://awsomesavings.hanksmith.netskype:hanksmith5aim:hanksmith5msn messenger:[EMAIL PROTECTED]personal email:[EMAIL PROTECTED]business email:[EMAIL PROTECTED]business phone:18663677484home phone:15092321855cell phone:15093890569 No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.289 / Virus Database: 265.5.0 - Release Date: 12/9/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No ring signal when calling internal extensions ?
can you make asteirsk do a fast ring as well?
business site:
http://awsomesavings.hanksmith.net
skype:
hanksmith5
aim:
hanksmith5
msn messenger:
[EMAIL PROTECTED]
personal email:
[EMAIL PROTECTED]
business email:
[EMAIL PROTECTED]
business phone:
18663677484
home phone:
15092321855
cell phone:
15093890569
- Original Message -
From: Eric Wieling aka ManxPower [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, December 09, 2004 3:41 PM
Subject: Re: [Asterisk-Users] No ring signal when calling internal
extensions ?
Robert Rozman wrote:
Sorry, wasn't specific enough.
Caller is not hearing any ringing tone. That means just plain silence til
local extension picks up or goes to voicemail.
I have this one in macro:
exten = s,4,Dial(${ARG1},30,tr) ; 20sec timeout
but obviously something else is missing. 30 secs of plain silence and
then
further action without any ring (for caller)...
What version of Asterisk are you using? Versions before 1.0.x did not
always provide ringing sounds to the caller.
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--
No virus found in this incoming message.
Checked by AVG Anti-Virus.
Version: 7.0.289 / Virus Database: 265.5.0 - Release Date: 12/9/2004
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.289 / Virus Database: 265.5.0 - Release Date: 12/9/2004
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[Asterisk-Users] asterisk gui?
hello is there a gui that would allow me to configure everything from phones, to extentions, to voice mail to basicly everything that asterisk can do? I did go to www.voip-info.org and none of the guis I saw there do the trick and the ones that come close aren't downloadable just wanted to see status on this thanks hank My Inbox is protected by SPAMfighter893 spam mails have been blocked so far.Download free SPAMfighter today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice
I am hooked up with broadvoice and have been having no problems that are major there voice mail system went on the blits for about 30 minutes yesterday but that was about it. what kind of problems you expierencing? - Original Message - From: Joel Gathercole [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 11, 2004 9:19 PM Subject: [Asterisk-Users] Broadvoice Hello, I am just curious how many people are hooked up with BroadVoice and have recently been experiencing a lot of dificulty. Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1117 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Telephony with Asterisk by Paul Mahler
is it in ebook format at all? I am a blind computer user and have no way of getting it scanned in to my computer even if I were to purchase it. thanks hank - Original Message - From: Sys. Concept Inc. [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 11, 2004 10:08 PM Subject: [Asterisk-Users] VoIP Telephony with Asterisk by Paul Mahler Does anybody have the book: VoIP Telephony with Asterisk by Paul Mahler. Is it for beginners or advanced users? -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1117 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD
the user you are calling is currently offline is what I get when calling fwd number hth hank - Original Message - From: Steve Maroney [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 11, 2004 9:29 AM Subject: [Asterisk-Users] FWD Im trying to get IAX to work between my * and FWD. I activated my iax2 account on iax.fwdnet.net and I get the output: Registered to '65.39.205.121', who sees us as 68.14.203.254:4569 when I start asterisk. I tried used the Call Me tool on fwdnet.net but I dont get any calls even though the Call Me tool says everything looks ok. Can someone call my FWD number and just leave me a message if i dont answer. FWD Number is 474538. My * box is configured to ring one of my extentions. Thank you, Steve Maroney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1104 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
yasr is text based but the interesting part is going to see if it works running on a windows platform with this version of linux with out that I can't do anything with this so I will have to see. take care. hank - Original Message - From: Greg Boehnlein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, September 10, 2004 12:00 AM Subject: Re: [Asterisk-Users] astwind has any one got this thing to work? On Thu, 9 Sep 2004, hank smith wrote: is there going to be a gui for co linux and astwind? No. AstWind is just a Debian GNU Linux distribution with a precompiled Asterisk installation running under a CoLinux kernel. I will have to see if either there is going to be a gui or if yasr a screen reader for the blind will work with this thing. I do not know. I would assume that a blind user would probably prefer a text based interface, but I have no clue. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1050 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
it works it works it works! sorry it took it so long for the info to click thanks for the help guys!!! take care hank - Original Message - From: Greg Boehnlein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 11:57 PM Subject: Re: [Asterisk-Users] astwind has any one got this thing to work? On Thu, 9 Sep 2004, hank smith wrote: I have a SiS 900 PCI Fast Ethernet Adapter what do I put in there or is that what I put in the xml file? Go read: http://www.colinux.org/wiki/index.php/coLinuxNetworking Specifically: If in doubt, the name of the card can be found in colinux-daemon startup log as follows: bridged-net-daemon: Checking adapter: NDIS 5.0 driver bridged-net-daemon: Checking adapter: TAP VPN Adapter. bridged-net-daemon: No matching adapter Error initializing winPCap The correct name here is NDIS 5.0 driver and not Karta Realtek RTL8139(A) PCI Fast Ethernet Adapter. It may help to use the default console, rather than the NT-Native (as the initial window has scrollback). I tried it with winpcap v 3.0 and 3.1beta. Currently works well with 3.1 beta Deja Vu.. Is there an echo in here? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1053 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference Phone
what phone did you purchase and how much - Original Message - From: Deon Rodden [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, September 10, 2004 5:59 AM Subject: Re: [Asterisk-Users] Conference Phone We use a nice Polycom conference phone and plugged it into the Sipura and it works crystal clear. Was cheaper than Polycom's conference phone w/ built in VOIP capabilities. Joe Dennick wrote: If it were me; I'd opt for one of the Polycom Conference phones (they are just regular analog phones), and use an FXS card to connect it to Asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Brown Sent: Thursday, September 09, 2004 4:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Conference Phone Any advice on a good conference phone that works with Asterisk? I like the Cisco line and was wondering if anyone has used the 7935 or 7936 phones. From what I can tell they dont have a sip load. Has anyone verified this or gotten an ETA from Cisco? Chad --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.745 / Virus Database: 497 - Release Date: 8/27/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1062 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxy vs sipura
how much ram you got on the pc running the vm? also will microsoft Virtual PC run on xp home? thanks hank - Original Message - From: Bill Seddon [EMAIL PROTECTED] To: 'Benjamin on Asterisk Mailing Lists' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Friday, September 10, 2004 6:34 AM Subject: RE: [Asterisk-Users] iaxy vs sipura I run Asterisk on Redhat 8.0 with a VM hosted by Microsoft's Virtual PC which, in turn, runs on Windows 2000 Server. Works like a charm. Can't use Zaptel cards but that's OK for me. I can put it into standby any time and it takes only a few seconds to start up the VM from its saved state and at that time the Linux session (and Asterisk) is available once again. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists Sent: September 10, 2004 2:03 PM To: Andy Powell Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iaxy vs sipura On Fri, 10 Sep 2004 14:05:09 +0200, Andy Powell [EMAIL PROTECTED] wrote: At the risk of stating the obvious if you have a laptop not running MacOSX (ie perhaps running windows) download my asterisk live! cd ( http://www.automated.it/asterisk/ ), burn it and test it on your laptop and bung it in your laptop case along with your iaxy/sipura/whatever and errm... problem solved.. :D Certainly an option, but most business folks will want to have their Outlook contacts and Excel spreadsheets in front of them when they are on the phone. Dual boot environments are not ideal in those situations. Imagine you're talking to some guy on the phone about prices and he tells you I cant' tell you what the discounts are right now because I would have to shut down the phone system to open Excel. However, you could use VMware on an Intel notebook to run both Windoze and Linux concurrently. This wouldn't be ideal for a real PBX for performance reasons, but since all you are going to use Asterisk for is to be a gateway for one single user, it's probably ok in this particular scenario. I remember there was a guy in Romania who reported he had VMware with Windoze and Asterisk on Linux running as a home PBX on his PC and it seemed to be alright. If you'd combine such a setup with a Windoze GUI tool that will start and stop the Linux environment and Asterisk at the push of a button, then you'd have a fairly convenient and workable SIP/IAX gateway solution for travelling biz folks. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1068 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxy vs sipura
I have to have access to sound on linux to use the screen reader for linux and from what I under stand colinux don't support sound. otherwise this would be the perfict sullution. - Original Message - From: Greg Boehnlein [EMAIL PROTECTED] To: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Cc: Andy Powell [EMAIL PROTECTED] Sent: Friday, September 10, 2004 9:13 PM Subject: Re: [Asterisk-Users] iaxy vs sipura On Fri, 10 Sep 2004, Benjamin on Asterisk Mailing Lists wrote: However, you could use VMware on an Intel notebook to run both Windoze and Linux concurrently. This wouldn't be ideal for a real PBX for performance reasons, but since all you are going to use Asterisk for is to be a gateway for one single user, it's probably ok in this particular scenario. Or you could use AstWind, which runs concurrently with Windows and is built entirely on Open Source software (CoLinux Kernel, Debian, Asterisk) and avoid paying for Vmware! ;) Plus, installation is a snap. See Digium's press release: http://www.digium.com/index.php?menu=astwind You can find more information on AstWind at: http://www.voip-info.org/wiki-AstWind -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1084 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
can you post the information on how you got that thing working? thanks hank - Original Message - From: Chris HARIGA [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Wednesday, September 08, 2004 8:55 PM Subject: RE: [Asterisk-Users] astwind has any one got this thing to work? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1030 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940
what is the price range in us dollars? - Original Message - From: Jody N. Rudolph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 11:32 AM Subject: RE: [Asterisk-Users] Polycom IP500 vs Cisco 7940 The Polycom IP500s do support customized ringtones and can use a customized ALERT_INFO for all of them. One thing that is worth noting in this comparison is that the IP500 doesn't support the XHTML microbrowser that the IP600 does. Since they both use the same SIP application I am hoping they enable this in future but as of now it doesn't work. I actually had 30 of these before I found this out but would still recommend these over any phone in the price range. Jody N. Rudolph Heartland Communications Internet Services, Inc 1301 Boadway Paducah, KY 42001 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Laird Sent: Thursday, September 09, 2004 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940 On Sep 9, 2004, at 9:53 AM, Matt G wrote: I've been asked to determine which phones our organization should go with. And I've narrowed it down to the Polycom IP500 or the Cisco 7940. From my travels through google, it's hard to find a definitive comparison of the two phones. So I thought I would ask the people that have probably used both. From what I can tell, the only major benefit the Cisco has over the Polycom is * 24 ring tones * XML support * Help Button * Larger Screen (is this true? 2x24 vs the 160x80 on the polycom) The screen on the Cisco isn't very big, either--192x96 or so, if I remember correctly. I'm running 6.3 on my 7940, and I haven't seen the ability to do anything interesting with ringtones. In theory, you can feed new tones to it, but you can't use them for ALERT_INFO-driven distinctive ringing. The XML support is okay, but rumors suggest that the newest Polycom firmware supports something very close to XHTML, which would be a lot more powerful then Cisco's sparsely-documented XML dialect. Another question that came up while discussing the Cisco phones was if the 24 ring tones are 'assignable' (ie, user calls in with callerid saying 'sales' and it rings a certain way, if they call in with callerid saying 'tech support' it rings something else). I couldn't find any information on this on google, so if anyone has the answer to this that would be great. I don't think it can do that. You can set ALERT_INFO in Asterisk to Bellcore-drX, where X is 1..5, and the phone will ring slightly differently, but that may or may not be good enough for your purposes. Other than that, the polycom seems to have all the features we want, and according to the wiki works quite well with asterisk and has many features enabled that seem pretty interesting (MWI, etc). The Cisco's on the other hand seem less straightforward to configure and not as much talk on the wiki, nor support. MWI works just fine on the 7940, so I'm not sure that I'd count that as an advantage for the Polycom. I haven't seen a Polycom in person, but I haven't heard anything bad about them. My 7940 works well, and I wouldn't hesitate to recommend it, but for the money, the Polycom is quite likely a better phone. I didn't find the 7940 to be particularly difficult to configure, *EXCEPT* for the initial installation of the SIP firmware. It's a multi-step upgrade, because you can't directly upgrade from the SCCP image that it ships with to a modern SIP image. Once you get past that, it isn't too bad, particularly if you have multiple phones. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1033 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival Speech Synthesis 1.95:beta July 2004 Eval
do you got a number I can call to take a listen? - Original Message - From: Steve Murphy To: [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 12:06 PM Subject: [Asterisk-Users] Festival Speech Synthesis 1.95:beta July 2004 Eval Hello--In the interests of playing around and wasting time, I've installed the latest version of theFestival stuff, 1.95beta.And, in the interests of future Asterisk-Festival connectivity, I applied the 1.4.3 patch to put in theasterisk related routines. I did it by hand, but, it looks like the patch will apply with no comment.Asterisk works with the new server...BUTthe speed of what's played over the speaker vs. what you hear over the phone is off by maybe 2x.The voice isn't shifted in frequency at all. On Asterisk, the voice just speaks twice as fast as it does coming from festival over the speakers. And, if you are having it say jokes at double speed, well, it reminds me of speed reading.There must be some lever or pulley or switch or something to modify the speed.ANDHoo boy, try putting this in your siteinit.scm file:(set! voice_default 'voice_cstr_us_awb_arctic_multisyn)and listen to this:(SayText "Hello there, kyootee pie.")(SayText "Don't you just love the sound of my voice.")(SayText "My wife, Sonya, Makes the best bread there ever was")Best synthetic voice I've ever heard.murf -- Steve Murphy Electronic Tools Company ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter1037 spam mails have been blocked so far.Download free SPAMfighter today!___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
I have a SiS 900 PCI Fast Ethernet Adapter what do I put in there or is that what I put in the xml file? - Original Message - From: Greg Boehnlein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 9:03 PM Subject: RE: [Asterisk-Users] astwind has any one got this thing to work? On Wed, 8 Sep 2004, Chris HARIGA wrote: I make it work!! My Astwind is up and running! Now is 11:53 PM and I'm going to bed. Tomorrow morning I will post how I fix the Ethernet connection. I bet you followed the following directions! ;) From: http://www.colinux.org/wiki/index.php/coLinuxNetworking If in doubt, the name of the card can be found in colinux-daemon startup log as follows: bridged-net-daemon: Checking adapter: NDIS 5.0 driver bridged-net-daemon: Checking adapter: TAP VPN Adapter. bridged-net-daemon: No matching adapter Error initializing winPCap The correct name here is NDIS 5.0 driver and not Karta Realtek RTL8139(A) PCI Fast Ethernet Adapter. It may help to use the default console, rather than the NT-Native (as the initial window has scrollback). I tried it with winpcap v 3.0 and 3.1beta. Currently works well with 3.1 beta -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1047 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
is there going to be a gui for co linux and astwind? I will have to see if either there is going to be a gui or if yasr a screen reader for the blind will work with this thing. thanks hank - Original Message - From: Greg Boehnlein [EMAIL PROTECTED] To: arsal siddiqui [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 9:11 PM Subject: Re: [Asterisk-Users] astwind has any one got this thing to work? On Thu, 9 Sep 2004, arsal siddiqui wrote: dear khurram, i need to know the price for x100p. i've emailed convergence.com.pk and never get a reply. If you could help me in this regards, i'll be greatful. I need to know the price. send me an email off the list. if you can help me in getting * hardware. Waiting for your reply Just as a side note... CoLinux CANNOT YET interface with any Digium hardware! So if you plan to run an X100P under AstWind you may be waiting a long time before it works! ;) Someone needs to port Zaptel to CoLinux! ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1047 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with length of voicemail
I had that problem when I was running asterisk on my linux box before it went down so you aren't the only one having that problem - Original Message - From: Marty Mastera To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, September 07, 2004 10:50 PM Subject: RE: [Asterisk-Users] Problems with length of voicemail I wonder if anyone else's Asterisk box drops the connection to voicemail after 30 secs even when the maxmessage parameter is set to 180 (3 mins). Here is the general section of my voicemail: Roger, There has been very recent discussion regarding this topic exactly...specifically when using BroadVoice as a sip provider. Calls toyour BroadVoice DID that end up in VM terminate after 30 seconds The current theory is that during VM recording, * doesn't send any audio packets back to BroadVoice...after 30 seconds BroadVoice thinks that the connection has been lost and terminates the call...(I'm paraphrasing the thread that recently appeared on this topic, forgive me if this isn't completely accurate) Assuming that this is correct, you could be using BroadVoice, or another provider who disconnects after not receiving audio for some period of time... Hope that helps, Marty ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter983 spam mails have been blocked so far.Download free SPAMfighter today!___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astwind has any one got this thing to work?
hello I am fitteling with the astwind-installer-0.1.1.exe asterisk for windows and am having trouble getting the thing to connect to the meers to download the updates and stuff. I looked at the wiki and set up networking and stuff with no success, has any one got this thing to work successfully? my windows box is the faster of the 2 machines and my main linux box is down at the moment. I am running a netgear rp614 router behind nat if this helps but I have tried and tried and tried to get this sucker up with no luck any help would be greatly greatly appreciated. thanks hank My Inbox is protected by SPAMfighter983 spam mails have been blocked so far.Download free SPAMfighter today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I get DIDs for remote areas in Canada
are you serious? that it is elegal to watch hbo? if so what is the logic behind that one? that is so stupid email me off list on this one [EMAIL PROTECTED] - Original Message - From: Brandon Patterson (peering) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, September 08, 2004 1:25 PM Subject: Re: [Asterisk-Users] How do I get DIDs for remote areas in Canada Good luck. If you did you will pay through the nose. Did you know that the CRTC in Canada is holding hearings late Sept on VOIP? Decision due in Feb 2005. Can we say why waste time? 10 people decide your entire future from radio to phone to tv. Hey, its against the law to watch HBO in Canada! Don't invest any money in small towns unless you want to go broke. Contact me off the list and I will be happy to go further. Our Motto Canada Owned and Operated by the Very Few I want the ability to setup DIDs in a variety of different remote locations in Canada. There are various providers that have DIDs in major cities, but none that focus on the smaller cities. The question is how do I actually setup these DIDs? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1002 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astwind has any one got this thing to work?
please do I can't get mine to work thanks hank - Original Message - From: arsal siddiqui [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, September 08, 2004 5:47 PM Subject: Re: [Asterisk-Users] astwind has any one got this thing to work? I am download astwind-installer-0.1.1.exe, I'll post an update if I manage to make this thing work. Regards Arsal - Original Message - From: hank smith [EMAIL PROTECTED] Date: Wed, 8 Sep 2004 00:14:37 -0700 Subject: [Asterisk-Users] astwind has any one got this thing to work? To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] hello I am fitteling with the astwind-installer-0.1.1.exe asterisk for windows and am having trouble getting the thing to connect to the meers to download the updates and stuff. I looked at the wiki and set up networking and stuff with no success, has any one got this thing to work successfully? my windows box is the faster of the 2 machines and my main linux box is down at the moment. I am running a netgear rp614 router behind nat if this helps but I have tried and tried and tried to get this sucker up with no luck any help would be greatly greatly appreciated. thanks hank My Inbox is protected by SPAMfighter 983 spam mails have been blocked so far. Download free SPAMfighter today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1010 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ## transfer into CVS when? Plus Suggestion. Attendant Transfer possible..
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 how can I get this feature inplimented? - - Original Message - From: James Gardiner [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 16, 2004 1:17 AM Subject: [Asterisk-Users] ## transfer into CVS when? Plus Suggestion. Attendant Transfer possible.. Hi, I have just had a look at the double ## transfer option, Very nice I like it a lot. I was wondering when this will be rolled into the main CVS? I also have a suggestion, can you make it do ATTENDANT transfer, like ## - Blind Transfer What it does now. (Apart from parking, hybrid really.) #0 - ATTENDANT TRANFER, leading with ## - complete transfer or #0 to Cancel and go back to the call.. Then again, I suppose using Parking kinda archives this. Ie park call, Call other person telling where to pick up parked call. Just a thought.. Really after getting it into the main CVS. Thanks, James Gardiner --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.737 / Virus Database: 491 - Release Date: 11/08/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: PGP 8.1 iQA/AwUBQSDctNPHBmkC+PAZEQKIjgCg827Py24yWVgZcB1G4jsrkyB/s0gAoMHi Je07IrZmWPYMgobu6ct4fEyJ =Gyd6 -END PGP SIGNATURE- My Inbox is protected by SPAMfighter 431 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free MOH MP3
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 any recommendations for games I could use? - - Original Message - From: Holger Schurig [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 16, 2004 3:02 AM Subject: Re: [Asterisk-Users] Free MOH MP3 Some GPLed open-source games have great music. Convert them from OGG to MP3 and you have a GPLed music file. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: PGP 8.1 iQA/AwUBQSEFJNPHBmkC+PAZEQKByACeK+2JMdRg1nScgLGrIc5uyEt7z8IAoLT6 FHNjM9nhQeCbLKT24EdHG1Ya =YGv8 -END PGP SIGNATURE- -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 any recommendations for games I could use? - - Original Message - From: Holger Schurig [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 16, 2004 3:02 AM Subject: Re: [Asterisk-Users] Free MOH MP3 Some GPLed open-source games have great music. Convert them from OGG to MP3 and you have a GPLed music file. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: PGP 8.1 iQA/AwUBQSEFJNPHBmkC+PAZEQKByACeK+2JMdRg1nScgLGrIc5uyEt7z8IAoLT6 FHNjM9nhQeCbLKT24EdHG1Ya =YGv8 -END PGP SIGNATURE- My Inbox is protected by SPAMfighter 433 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] taking asterisk out of nat?
-BEGIN PGP SIGNED MESSAGE-Hash: SHA1 - -BEGIN PGP SIGNED MESSAGE-Hash: SHA1 hello I have a router that is behind a nat, I want to take asteriskout of nat so I can use it with sip. what would be the best way togo about doing this? I have cable internet and everything is hookedup to a router currently.thankshank - -BEGIN PGP SIGNATURE-Version: PGP 8.1 iQA/AwUBQSFpx9PHBmkC+PAZEQIVDgCePnga44T9RkvXNI7BPWYrEUBY5mcAoLr3a7ueKvfNmw4rBXF4iXT62OWb=t3Px- -END PGP SIGNATURE- -BEGIN PGP SIGNATURE-Version: PGP 8.1 iQA/AwUBQSFp1tPHBmkC+PAZEQIXUQCfRr7cExlgQ8KQcqi1yN1Cx23eo1MAoKk+U8qZ5PG6410+oy21ep+fkb2q=vZte-END PGP SIGNATURE- My Inbox is protected by SPAMfighter437 spam mails have been blocked so far.Download free SPAMfighter today!
Re: [Asterisk-Users] taking asterisk out of nat?
-BEGIN PGP SIGNED MESSAGE-Hash: SHA1 add a second nic to the linux box? can I then use it as if itweren't behind a nat? I want to use services sip services to be exactthat won't work behind a nat with asterisk.thankshank- - Original Message - From: Steve Totaro To: [EMAIL PROTECTED] Sent: Monday, August 16, 2004 7:34 PMSubject: Re: [Asterisk-Users] taking asterisk out of nat? make your asterisk box the router and add a second nic.- - Original Message - From: hank To: [EMAIL PROTECTED] Sent: Monday, August 16, 2004 10:13 PMSubject: [Asterisk-Users] taking asterisk out of nat? - -BEGIN PGP SIGNED MESSAGE-Hash: SHA1 - - -BEGIN PGP SIGNED MESSAGE-Hash: SHA1 hello I have a router that is behind a nat, I want to take asteriskout of nat so I can use it with sip. what would be the best way togo about doing this? I have cable internet and everything is hookedup to a router currently.thankshank - - -BEGIN PGP SIGNATURE-Version: PGP 8.1 iQA/AwUBQSFpx9PHBmkC+PAZEQIVDgCePnga44T9RkvXNI7BPWYrEUBY5mcAoLr3a7ueKvfNmw4rBXF4iXT62OWb=t3Px- - -END PGP SIGNATURE- - -BEGIN PGP SIGNATURE-Version: PGP 8.1 iQA/AwUBQSFp1tPHBmkC+PAZEQIXUQCfRr7cExlgQ8KQcqi1yN1Cx23eo1MAoKk+U8qZ5PG6410+oy21ep+fkb2q=vZte- -END PGP SIGNATURE- - --- --My Inbox is protected by SPAMfighter437 spam mails have been blocked so far.Download free SPAMfighter today! -BEGIN PGP SIGNATURE-Version: PGP 8.1 iQA/AwUBQSFyLtPHBmkC+PAZEQKsVACeIdQvS/vD1rwjrgvm2H85ZkkXrFkAnRnu553QjimVAMVkT334C8ofMynJ=zf5e-END PGP SIGNATURE- My Inbox is protected by SPAMfighter437 spam mails have been blocked so far.Download free SPAMfighter today!
[Asterisk-Users] asterisk with InPhonex?
hello has any one got asterisk to work with InPhonex? if so can you send me your conf information? we are having some problems getting ours up and running. my friend is helping me get it set up. thanks hank My Inbox is protected by SPAMfighter415 spam mails have been blocked so far.Download free SPAMfighter today!
Re: [Asterisk-Users] 831 Santa Cruz/Watsoncille, Calif. DIDs
odd my broadvoice has been working fine over here. - Original Message - From: lists-jmhunter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 09, 2004 10:37 PM Subject: [Asterisk-Users] 831 Santa Cruz/Watsoncille, Calif. DIDs Hey there, I don't know who else has suffered broadvoices terrible service, but I am about to my end with them. The lack of a LBR codec, the outages, the changing of servers without notifying subscribers haspushed me to my end. Now most incoming calls are abbruptly cut off within a minute of the call starting. Anyone know of any other * friendly providers that have DID, besides Voicepulse, Nufone, broadvoice. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 303 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: VoIP SPAM, what's next ?
voip spam? I have never gotten any yet. - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 10, 2004 11:13 AM Subject: [Asterisk-Users] Re: VoIP SPAM, what's next ? At 7:14 PM +0200 on 8/10/04, Soren Rathje wrote: Gang, Do anyone have a clue on how they do this ?? QOVIA FILES PATENTS FOR VOICE SPAM BLOCKING TECHNOLOGY http://www.qovia.com/company/news/06.28.2004_voip_spam_patent_app_final.htm Qovia ready to take on VoIP spam http://www.nwfusion.com/news/2004/071204qovia.html Next thing will probably be a sbl.e164.org service to block spammers like we do with email... :-) Hmm.. Imagine a built-in reporting tool in Asterisk. Hit **666**# and Asterisk will report the IP address of the caller (and possibly also the CID but it can be forged as we all know) on-line and in real-time to a SBL list for immediate blocking and further processing... Any takers ?? /Soren It is the mark of an educated mind to be able to entertain a thought without accepting it. - Aristotle VOIP Spam is actually pretty trivial to take care of, if only the manufacturers would wise up. We're in the same place we were with SMTP about twelve years ago. I'm sure we'll see a slew of patents and chest-pounding by people with obvious or trivial solutions - welcome to the New WIPO World. The solution is simple: End devices should have the option to only accept authenticated requests. That's pretty simple, but that is the key to the whole solution. However, most end devices will blindly accept any call that they're given, so long as the destination number is correct. I've seen a few phones (Polycom is the only one that comes to mind) which will challenge INVITEs. SIP devices are pretty smart, but I don't think they're capable of being totally smart. The proxy in the middle will have to retain some intelligence and reference some type of permissions model or database to allow calls through or not. I trust that industry (and quasi-industry, like Asterisk) programmers will come up with dozens of ways of intercepting and thrashing unsolicited phone call, so long as there is no back door that the spammer can sleaze through to get right to the desktop. TLS SIP is also a nice concept, since it would require some sort of root authentication that could be revoked or at least recognized if a spam origin was adequately recognized. This is all starting to sound a lot like an anti-spam thread, so I'll stop here. Most intelligent people on the list should be able to figure out a bunch of ways to prevent spam, but the primary one is accountability of origin. Anything that allows that accountability to be compromised from the perspective of the destination means that spam will inevitably slide in, so it is our job to enforce sane authentication/authorization mechanisms NOW on the vendors from whom we buy equipment/firmware. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 321 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: VoIP SPAM, what's next ?
hello wolfgang. I am curious did the packit have any sound to it when your friend tried it out? I am assuming voip spam would have audio. thanks hank - Original Message - From: Wolfgang S. Rupprecht [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 10, 2004 2:10 PM Subject: Re: [Asterisk-Users] Re: VoIP SPAM, what's next ? [EMAIL PROTECTED] (Loek Gijben) writes: hank [EMAIL PROTECTED] wrote: voip spam? I have never gotten any yet. It's is just waiting for the first one to arrive.. The mechanics are just too appealing for spam-like businesses. I got one the other day, but it turns out it was a buddy trying out his skills at generating UDP from a shell script. I figure if voice spam gets to be a problem I'll simply use a whitelist arrangement where some aspect of the caller is looked up in an asterisk DB. Callers in good standing get to ring the phone. Others go to a voice-menu tree that asks them to press a certain key if they are a telemarketer, or calling for a political party, or collecting for a charity. They will then get a canned message to please put us in their do not call list. All other callers are encouraged to press a different key to ring through to me. Unlike email, phone calls are interactive and sorting the robo-caller from the real people shouldn't be hard. The only thing bugging me is, is there a law that would prevent a telemarketer from lying and pressing the key for I am not a telemarketer. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 327 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound file quality
can you use .wav files or does it have to be gsm? - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 09, 2004 4:23 AM Subject: Re: [Asterisk-Users] Sound file quality On Mon, 2004-08-09 at 06:07, David Gurr wrote: I'm building a phone-in demo system to use for introducing Asterisk to prospective clients. One of the things I'm wary of is their likely preconceptions that VoIP systems will have poor audio quality. As a result, I'd like to ensure that the voice prompts I'm using have the best possible audio quality. Is it possible to use sound files at higher than 8kHz sampling? My callers will be coming in over PSTN to a VoIP gateway and then to me by uLaw/aLaw ... would higher sampling rates gain me anything in this configuration? PSTN is 8khz sample rate. So obviously a higher sample rate will not get you any where. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 290 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Generic X100P setup issues
I got mine from the same place and the only way I got mine working was to upgrade kirnil to 2.6.5 you may want to try dibian sarge unstable distro with kirnil 2.6.5. hth hank - Original Message - From: Lyle Giese To: [EMAIL PROTECTED] Sent: Sunday, August 08, 2004 6:59 AM Subject: Re: [Asterisk-Users] Generic X100P setup issues I am running SuSE v9.0 with kernal 2.4.21-99 and my clone cards run just fine. But I did purchase them from Digit Networks, so I knew they were good clone cards. I think this guy does not have true X100P clone cards. Lyle - Original Message - From: hank To: [EMAIL PROTECTED] Sent: Saturday, August 07, 2004 11:49 PM Subject: Re: [Asterisk-Users] Generic X100P setup issues you tried kirnil 2.6.5? that sounds simular to what I experenced with my clone card on my dibean box until I upgraded my kirnil. give that a shot. hth hank - Original Message - From: Graham W. Mitchell To: [EMAIL PROTECTED] Sent: Saturday, August 07, 2004 5:41 PM Subject: [Asterisk-Users] Generic X100P setup issues I am starting to dip my toes into the asterisk world, and to that end Ive scavenged an old PC (this is a home project, and I have basically $0 to spend on it), and installed FC1. Ive purchased a clone X100P (or at least I was told it was), and I am trying to get it to work. However, when I try and load the wcfxo module, I get the following errors [EMAIL PROTECTED] root]# insmod wcfxo Using /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Unfortunately, there is no other info in /var/log/messages or from dmesg The lspci vv for the card shows the following 00:08.0 Modem: Intel Corp.: Unknown device 1080 (rev 04) (prog-if 00 [Generic]) Subsystem: Intel Corp.: Unknown device 1000 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping+ SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32 (250ns min, 15500ns max), cache line size 08 Interrupt: pin A routed to IRQ 10 Region 0: Memory at db10 (32-bit, non-prefetchable) [size=4K] Region 1: I/O ports at e800 [size=256] Capabilities: [80] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Is this indeed a generic X100P? If so, does anyone have any suggestions what to look at next? Thanks Graham My Inbox is protected by SPAMfighter272 spam mails have been blocked so far.Download free SPAMfighter today! My Inbox is protected by SPAMfighter283 spam mails have been blocked so far.Download free SPAMfighter today!
Re: [Asterisk-Users] System Reqirements HELP
um where did you get that system for 100.00 at? wow what a deal. can I get url? thanks hank - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 08, 2004 8:37 AM Subject: RE: [Asterisk-Users] System Reqirements HELP You overpaid. Whether it's a P4 OR a Celeron (which one is it?), a 2.2Ghz machine with 256MB RAM and two small drives shouldn't have cost you more than $400-$500. I got a 2.7GHz Celeron/MB combo for $120 (less $40 rebate), 256MB RAM for $40 and 40GB drives shouldn't run you more than $50/each. $100 more of it's a P4 instead of a Celeron. Add a case+PS for $40-$50. -Original Message- From: Steven P. Donegan [mailto:[EMAIL PROTECTED] Sent: Sunday, August 08, 2004 7:45 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] System Reqirements HELP I bought a 5013G-i barebones 1U from Supermicro (www.supermicro.com). Nice chassis :-) Added 256M ram and a 2.2 Ghz p4 celeron and 2 40g ata 133 ide drives - complete 1U rack mountable system for 1k$. Installed RedHat 9 plus all updates/errata, my Asterisk CVS get/make/install scripts and voila - instant Asterisk box :-) This makes Asterisk #3 in the home network :-) The SIP stuff you reference is dead easy. The ISDN - well, ISDN is pretty much dead here in the US (except PRI) so on that I'm sure someone else will assist. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 283 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse problems?
what provider is this that is in beta? also have you looked in to broadvoice? thanks hank - Original Message - From: Ken Wiesner [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 08, 2004 9:17 AM Subject: RE: [Asterisk-Users] Voicepulse problems? Bruce, Yeah I'm having the same problems with VoicePulse. It's getting to be ridiculous because this happens all the time now. There's a new voip provider coming out that is working with some of the larger telcos. It will be offering similar quick turn up of services like voicepulse but much better service. In fact, there is even a phone number that you can call if you have problems where a person actually answers! AMAZING CONCEPT for a phone company! :-) Soon as it's up and out of beta I'm giving voicepulse the boot! ~Ken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Komito Sent: Sunday, August 08, 2004 10:29 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicepulse problems? Is any one else having problems with Voicepulse today? Suddenly, I can't register and calls to my Voicepulse numbers get a fast busy. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.734 / Virus Database: 488 - Release Date: 8/4/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.734 / Virus Database: 488 - Release Date: 8/4/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 283 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-update script
where can we get the script at? - Original Message - From: Steve Szmidt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 08, 2004 12:04 PM Subject: [Asterisk-Users] asterisk-update script -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Here's a version I modified which grabs either a development or stable verision, and does a backup before updating from CVS. It also asks for addon's and cc. Leif Madsen did the original development and Mark released it. My changes does the minimum changes to previous version, to get what I need. It does the same version checking as the Make script uses should .version file not exist. It runs well for me. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBFnlOljK16xgETzkRAiy1AKDAK/4E6yHWkA+eNcJVdWIOmaOVEgCgqySe TnsmIgkMMVhUSfIFun2OKtE= =eb68 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 283 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interesting catalog: Viking Electronics
how did you get a door phone set up? sounds pritty cool. - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 06, 2004 10:58 PM Subject: RE: [Asterisk-Users] Interesting catalog: Viking Electronics Make a sign -- I've been trying train my mail-carriers to use the DOORBELL and not just knock. Geez, people, what does it take?? -Original Message- From: David Hickman [mailto:[EMAIL PROTECTED] Sent: Friday, August 06, 2004 11:03 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Interesting catalog: Viking Electronics Just make sure the device will handle external power if needed. I have a door phone and run its external power on an extra copper pair. My only problem is getting people to use it. It seems that a door phone is a foreign concept in St. Louis. FedEx guys are the worst about it. dhh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 268 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk : No Sound Issues
nat and sip won't work as far as I know unless things have changed. - Original Message - From: niko singh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 07, 2004 12:42 AM Subject: [Asterisk-Users] Asterisk : No Sound Issues Hi , Thanks greg , for pointing out the valuable resources for reference. I tried SJphone in a windows environment to connect to fwd and it worked fine(including (audio). Now have to do the same thing for linux(red hat 9 ) and hope the nat issue is resolved. Now i would like to connect asterisk to fwd and instead of the SJ phone connecting to fwd directly i would wish to connect through asterisk, writing the extensions to transfer all dailled numbers from my SJphone to fwd. At a later stage make asterisk accept calls dialled to my fwd number and operate thm through the SJ phone How can nat issues be resolved with asterisk. I am a newbie in the area of firewall and security issues hence a bit detailed replies would be obliging. Thanks niko _ On the road to retirement? Check out MSN Life Events for advice on how to get there! http://lifeevents.msn.com/category.aspx?cid=Retirement ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 268 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voice mail greeting not updating?
hello when I try to change my voice mail greeting over the phone it says voice mail greeting saved etc but it is still playing the greeting I had on there before. is there something in my voicemail.conf that needs to be changed? thanks hank My Inbox is protected by SPAMfighter272 spam mails have been blocked so far.Download free SPAMfighter today!
Re: [Asterisk-Users] Generic X100P setup issues
you tried kirnil 2.6.5? that sounds simular to what I experenced with my clone card on my dibean box until I upgraded my kirnil. give that a shot. hth hank - Original Message - From: Graham W. Mitchell To: [EMAIL PROTECTED] Sent: Saturday, August 07, 2004 5:41 PM Subject: [Asterisk-Users] Generic X100P setup issues I am starting to dip my toes into the asterisk world, and to that end Ive scavenged an old PC (this is a home project, and I have basically $0 to spend on it), and installed FC1. Ive purchased a clone X100P (or at least I was told it was), and I am trying to get it to work. However, when I try and load the wcfxo module, I get the following errors [EMAIL PROTECTED] root]# insmod wcfxo Using /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Unfortunately, there is no other info in /var/log/messages or from dmesg The lspci vv for the card shows the following 00:08.0 Modem: Intel Corp.: Unknown device 1080 (rev 04) (prog-if 00 [Generic]) Subsystem: Intel Corp.: Unknown device 1000 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping+ SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32 (250ns min, 15500ns max), cache line size 08 Interrupt: pin A routed to IRQ 10 Region 0: Memory at db10 (32-bit, non-prefetchable) [size=4K] Region 1: I/O ports at e800 [size=256] Capabilities: [80] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Is this indeed a generic X100P? If so, does anyone have any suggestions what to look at next? Thanks Graham My Inbox is protected by SPAMfighter272 spam mails have been blocked so far.Download free SPAMfighter today!
[Asterisk-Users] iaxtel, asterisk, and sipura 1000 am having trouble with codecs
hello I am trying to set up iaxtel with asterisk and am using a sipura 1000 when my friend calls me he is sounding like he is in a metal tank that is the best way I can describe it, how ever when he calls me on my grand stream budjet phone 101 it sounds fine. is there a fix for this really anoying problem? thanks hank My Inbox is protected by SPAMfighter264 spam mails have been blocked so far.Download free SPAMfighter today!
Re: [Asterisk-Users] Vonage working with asterisk
did vonage finally allow there service to work with asterisk? when I was with them they wouldn't give out there server info. thanks hank - Original Message - From: Assaf Benharoosh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 06, 2004 9:43 PM Subject: RE: [Asterisk-Users] Vonage working with asterisk I still didn't get it to work. When calling the number- it goes to voicemail. No indication on the CLI. The 'sip show peers' shows: vonage/16464855 216.115.25.199 N 255.255.255.255 5061 Unmonitored 'sip show registry': HostUsername Refresh State sphone.vopr.vonage.net:5061 16464855183 15 Registered Help anyone? Assaf Benharoosh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, July 14, 2004 6:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Vonage working with asterisk atlast after working of 7 hours i got voange soft account working on asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 264 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk, fwd, and grandstream?
I have all ready been there the only refference I saw was the tips and tricks for asterisk and grandstream is there some info I am missing? thanks hank - Original Message - From: Holger Schurig [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 06, 2004 12:06 AM Subject: Re: [Asterisk-Users] asterisk, fwd, and grandstream? can this be accomplished? Yes. You should start reading documentation before asking. A good starting place is http://www.voip-info.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback over Console
do you got your speakers in the 2 floors of your house hooked up to the computer? am just curious. how do you got your sound system set up? email me off list. this may be off topic. email [EMAIL PROTECTED] - Original Message - From: Chris Foster [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 05, 2004 5:48 PM Subject: [Asterisk-Users] Playback over Console I'm trying to setup a primitive announcement-paging system in my house using the line-out from my * box to a cheap amplifier that runs to speakers on our first and second floors from the basement. I have a extension that connects to Console, and console is set to auto-pickup. I'm using alsa drivers. This all works great, except for one thing. I want to play a tone over the console after the console picks up. What i'm doing right now is calling Playback after the Dial. However, No playback sound or background sound is being heard over the console speakers or are any error messages appearing in the command line. The extensions.conf entry looks like this: [access-internal] include = parkedcalls exten = 31,1,Dial(SIP/line1,30,t) exten = 31,2,Voicemail(u1) exten = 32,1,Dial(SIP/line2,30,t) exten = 32,2,Voicemail(u1) exten = 33,1,Dial(SIP/grand,30,t) exten = 33,2,Voicemail(u1) exten = 310,1,Dial(Console/dsp) ;; intercom exten = 310,2,Playback(tt-weasels) Thanks to anybody who can help! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling an outside phone number as part of a hunt
how would I do this but do it with broadvoice? I want to give people the oppsion to call my cell phone but I use a voip carier - Original Message - From: Hall, Eric M. [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 05, 2004 7:51 PM Subject: [Asterisk-Users] Calling an outside phone number as part of a hunt I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has expired. My hardware is 2 X100P card exten = 2000,1,Dial(SIP/2000,20) exten = 2000,2,Dial(Zap/1/5551212,10) exten = 2000,3,Voicemail(u2000) exten = 2000,102,Voicemail(b2000) exten = 2000,103,Hangup Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk, fwd, and grandstream?
hello I want to use my grandstream witch is currently configured for fwd to use asterisk, my asterisk is configured with fwd threw iax, but I want to still recieve calls on my grandstream threw fwd threw asterisk if this makes any sense is this possible? I basicly want all of my phones to use asterisk but be able to use them with all the networks, my fwd account, my broadvoice account, etc etc. can this be accomplished? thanks hank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling an outside phone number as part of a hunt
will do. thanks - Original Message - From: Greg Hill [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 05, 2004 9:26 PM Subject: Re: [Asterisk-Users] Calling an outside phone number as part of a hunt On Mon, 5 Jul 2004, hank smith wrote: how would I do this but do it with broadvoice? I want to give people the oppsion to call my cell phone but I use a voip carier stay tuned to see how he gets the thing figured out, then change exten = 2000,2,Dial(Zap/1/5551212,10) to exten = 2000,2,Dial(SIP/[EMAIL PROTECTED],10) or similar. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback over Console
what was the problem? - Original Message - From: Chris Foster [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 05, 2004 10:02 PM Subject: Re: [Asterisk-Users] Playback over Console Thanks for responding. I figured it out. On Mon, 5 Jul 2004 22:22:37 -0600 (MDT), Greg Hill [EMAIL PROTECTED] wrote: I suspect that after Dial has happened the auto-answer connects you to the console, and the call doesn't reach Playback until after the console hangs up. Which is exactly right. As for how to do what you're after.. I dunno! Maybe you can find a way to pick up the console as if to dial from the console out to somewhere and issue the Playback then. Sort of. It turns out that Dial has option that does exactly what I want, namely, play a tone over the speakers (console) to alert people that somebody is about to speak/ A(x): Play an announcement (x.gsm) to the called party. Put that in Dial's options and Asterisk sends it out to the caller, which is exactly what I wanted. You can find more in the Tiki The only bad part about all of this is that my roommate found a air-raid siren sound, and so now when you call you don't get a tone but a 2 minute long warning that a tornado is approaching. I'll have to change that. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] music on hold question with asterisk
hello I'm trying to figure out if anyone's accomplished putting someone on hold with a hardphone that doesn't have a hold button or multiple lines. I'm thinking transferring the caller to a specific extension or something...is this possible? Has it been done? thanks hank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] looking for newbie resources
hello andy is your user guide updated? - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 04, 2004 5:24 PM Subject: Re: [Asterisk-Users] looking for newbie resources On 04/07/2004 at 14:53 Steven M. Sawczyn wrote: Hi, I am very interested in VOIP and telephony in general, although admittedly, I don't know much about the theories and protocols behind it. Having also an interest in Linux, I was really excited to come upon Asterisk. I would really like to learn more about Asterisk and VOIP in general and am wondering if anyone could suggest some beginner resources? Of course I've found that the best way to learn something is to just dive in and try it, but I don't think I'm ready to tackle installing Asterisk yet. In which case, http://www.automated.it/asterisk/ You'll find a link there for my Asterisk Live! CD (it's a test version, but feedback so far has been favourable) I'm running Slackware Linux on a machine which at the moment, is just hosting mail. In addition, I have accounts with both Vonage and Broadvoice. My idea is to set up a mini PBX here at home using both VOIP providers as my main lines and using my LAN to connect a few extensions. Might this be a good way to start learning, or am I way off track? Again, I am very new to this, so any info/resources/suggestions greatly appreciated. You could also try http://www.automated.it/guidetoasterisk.htm to get you going... The wiki has useful info too http://www.voip-info.org Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream ringtones - makering.pl usage for 1.0.50
how can you create your own ring tone? - Original Message - From: Maron Kristófersson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 08, 2004 6:57 AM Subject: [Asterisk-Users] grandstream ringtones - makering.pl usage for 1.0.50 If you wan't to create a ringtone with makering.pl for firmware 1.0.50, be sure to create it as ring.bin and then rename it to ring1.bin / ring2.bin or ring3.bin. This seems to be the only change between the format from 1.0.4.68. Regards, Maron ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Sniffing Calls for recording
how can you record calls with asterisk? I didn't even know this was possible can some one point me to a url for info on this? - Original Message - From: lists [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 07, 2004 1:30 PM Subject: [Asterisk-Users] Network Sniffing Calls for recording Ok assuming I don't want to record calls using * but instead want a dedicated server that listens to a mirror port and records calls. Is there a cheap software package out there for doing this for mgcp/sccp? I know if evern cut over to * there is a way but I doubt I will even cut 100% over to * so I was wonder what the list has heard of for call recording via sniffing my gates. I know there are some out there but $100k for 40 users is to high for my blood. Offlist is fine for all flames and answers since this is a bit off topic [EMAIL PROTECTED] OK it's a Monday when it takes 5 tries to get a email to the right list from the right account. Either that or someone switched the coffee pot to decaf again. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DNS SRV records
can enum be used with asterisk? if so how? - Original Message - From: Duane [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 07, 2004 6:09 PM Subject: Re: [Asterisk-Users] Re: DNS SRV records Adam Goryachev wrote: In fact, I think it would be nice if all modules/apps/chans/etc were marked noload by default (except the bare minimum required to get asterisk to start with no channels...). Then this goes back to my original point, I will not suggest people use SRV records if they want to receive calls as a large majority of Asterisk users won't be able to call them. If they want a simple method of allowing calls they should use enum, least then it's obvious that it isn't a email address and that they would possibly need to enable a few things to make it work. -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk to broadvoice?
hello is there any info on connecting asterisk to broadvoice? thanks hank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zapata?
what is hadrware? - Original Message - From: Richard Neese [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 06, 2004 7:37 PM Subject: Re: [Asterisk-Users] Zapata? as for hadrware digitalnetworks has made a clone card . but only digium has made any majoor card changes. there have been 2 ne rev to the cards i kow of and you can rea d on the digium site about them. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD network from Asterisk through NAT
Hi there, I'm trying to dial into the FWD network using Asterisk, though a NAT. The sources I've read say that it's unconfirmed to work through a NAT, but I'm wondering if anyone's done it anyway. So, anyone got a clue how to do this? Hank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: setting the number of rings befor asterisk picks up?
- - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. - Original Message - From: hank [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 23, 2004 6:19 PM Subject: setting the number of rings befor asterisk picks up? hello how do I set the number of rings picks up on? I am using a single port fxo card and currently asterisk is answering after 1 or 2 rings and I want it answering after 4 5 or 6 rings thanks hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: creating a single user voice mail box on asterisk?
- - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. - Original Message - From: hank [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 23, 2004 3:56 PM Subject: creating a single user voice mail box on asterisk? hello how do I go create a single boice mail box on asterisk? thanks hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
