[asterisk-users] Quintum configuration ASM200 Analog 2 tenor port
Hi, just wondering if there is anyone that can help me configure my quintum box to operate with asterisk. i have tried and made numerous attemtps configuring the tenor to work with [EMAIL PROTECTED] but have been unlucky. anyone out there has a cheat sheet to configure this device. thanks.. for some reason i cannot get it to work. your help is appreciated. _ Dont miss your chance to WIN 10 hours of private jet travel from Microsoft® Office Live http://clk.atdmt.com/MRT/go/mcrssaub0540002499mrt/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] QUINTUM CONFIGURATION.-
Has anyone been succesful at configuring tenor with asterisk. I have a few analog fxs-fxo boxes and would like to put them to use With asterisk, but I need help Thanks. On 9/10/06 11:54 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: Zaptel-1.2.9 compile error (Rich Adamson) 2. Re: How to send correct Caller ID on PRI (George Pajari) 3. Re: Zaptel-1.2.9 compile error (Bill Maidment) 4. Re: Submenus (Mir) 5. Re: sip peer question (Rushowr) 6. call notification for queues? (Rajkumar S) 7. Satellite link-IAX Jitter Buffer. (ANANGWE Nelson) 8. How could i get bridged channel partner (Mohammad Shokuie) 9. Re: Satellite link-IAX Jitter Buffer. (Yusuf) 10. Re: How to send correct Caller ID on PRI (Doug Lytle) 11. Re: Whcih phones are better for mass deployment (Thomas Kenyon) 12. Re: Whcih phones are better for mass deployment (Alberto Sagredo) 13. Re: Whcih phones are better for mass deployment (Thomas Kenyon) 14. Re: Whcih phones are better for mass deployment (Michael Graves) 15. Re: Whcih phones are better for mass deployment (Dave Cotton) 16. Re: Quintum tenor configuration with asterisk help ([EMAIL PROTECTED]) 17. Take 3 -- Trying to get SIP firmware on a 7970G (Jason Lixfeld) -- Message: 1 Date: Sun, 10 Sep 2006 02:49:11 -0500 From: Rich Adamson [EMAIL PROTECTED] Subject: Re: [asterisk-users] Zaptel-1.2.9 compile error To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Samy Antoun wrote: --- Bill Maidment [EMAIL PROTECTED] wrote: Hi I've just tried to compile the zaptel-1.2.9 release and I get the following error: Same here, using CentOS 4.4 kernel 2.6.9-42.0.2.ELsmp, got these errors when compiling zap: make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found make[3]: *** No rule to make target `/usr/src/zaptel/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h', needed by `/usr/src/zaptel/wct4xxp/vpm450m.o'. Stop. make[2]: *** [/usr/src/zaptel/wct4xxp] Error 2 make[1]: *** [_module_/usr/src/zaptel] Error 2 make: *** [linux26] Error 2 Hope someone has a workaround for this problem Have you tried: cd /usr/src/zaptel make update make install -- Message: 2 Date: Sun, 10 Sep 2006 01:17:06 -0700 From: George Pajari [EMAIL PROTECTED] Subject: Re: [asterisk-users] How to send correct Caller ID on PRI To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed There are several reasons why your attempts to set CNAM might fail: (1) your upstream PRI provider does not allow you to set it (although normally in this case you will find that you cannot set the CID either); (2) someone between you and the called party will set the CNAM using a directory lookup; (3) your upstream PRI provider is expecting the CNAM in a facility frame and not the default used by Asterisk (fix this by setting facilityenable=yes in zapata.conf) -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) Hosted IP PBX Services for SOHO Small Businesses - www.ip-centrex.ca VoIP Service, Equipment, Systems, and Consulting - www.netvoice.ca -- Message: 3 Date: Sun, 10 Sep 2006 18:36:03 +1000 From: Bill Maidment [EMAIL PROTECTED] Subject: Re: [asterisk-users] Zaptel-1.2.9 compile error To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Rich Adamson wrote: Have you tried: cd /usr/src/zaptel make update make install From a tarball? I don't think so! That would only work for SVN and as we know the SVN version is OK. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] QUINTUM TENOR ASM200 Configuration
Hi, this message is for Steve. Sorry for replying to the digest. It wasn't my intention. I would appreciate if you can guide as to how make the tenor asm200 work with asterisk. I am using asterisk at home. I guess my problem is configuring the tenor so that it is recognized and can take calls from asterisk (both ways). If you can help me out and send me a sample config i would be very thankful. My config is an asterisk at home box, and i wish to be able to have my quintum register to the asterisk. Both devices will be in different lans. My intention is to be able to call my asterisk box (in my home), using my quintum box (in my office) and vice versa. thanks, Francisco _ Get the new Windows Live Messenger! http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: asterisk-users Digest, Vol 26, Issue 54
hi i need helpl configuring a quintum tenor analog gateway using sip with asterisk. anyone, help is appreciated the model of the gteway is asm200 i need the settings to configure it with asterisk. for some reason it registers with asterisk but when try to call the extension from the quintum it is not recognized. help help help thanks From: [EMAIL PROTECTED] Reply-To: asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 26, Issue 54 Date: Sat, 9 Sep 2006 12:00:25 -0700 (MST) MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc2-f18.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Sat, 9 Sep 2006 12:03:59 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 3C2CA41D5;Sat, 9 Sep 2006 12:00:25 -0700 (MST) X-Message-Info: LsUYwwHHNt1Qrly5/IdcOLxnJ5Hdz4bhYGyQtYHi6jU= X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 09 Sep 2006 19:04:00.0431 (UTC) FILETIME=[B57B13F0:01C6D442] Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: Call Forwarding in SIP.conf ([EMAIL PROTECTED]) 2. RE: Call Processing Slow 11 seconds (G.Jacobsen) 3. Re: Intel Based G.729 and SVN-trunk-r42453 (Daniel Pocock) 4. RE: Call Processing Slow 11 seconds ([EMAIL PROTECTED]) 5. Re: Call Processing Slow 11 seconds (Alberto Sagredo) 6. Re: Intel Based G.729 and SVN-trunk-r42453 (Jason Lee) 7. Re: What don't I get about SIP? (John Marvin) 8. Re: Intel Based G.729 and SVN-trunk-r42453 (Daniel Pocock) 9. Re: Intel Based G.729 and SVN-trunk-r42453 (Jason Lee) 10. RE: What don't I get about SIP? (Mike) -- Message: 1 Date: Sat, 09 Sep 2006 17:12:54 + From: [EMAIL PROTECTED] Subject: Re: [asterisk-users] Call Forwarding in SIP.conf To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Skipped content of type multipart/alternative-- next part -- An embedded message was scrubbed... From: Tim St. Pierre [EMAIL PROTECTED] Subject: Re: [asterisk-users] Call Forwarding in SIP.conf Date: Sat, 9 Sep 2006 16:52:40 + Size: 2109 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/828bebdd/attachment-0001.eml -- Message: 2 Date: Sat, 9 Sep 2006 19:17:23 +0300 From: G.Jacobsen [EMAIL PROTECTED] Subject: RE: [asterisk-users] Call Processing Slow 11 seconds To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii In case you use an adapter or voip phone: Did you try to press hash # after the number ? - then the adapter/voip phone dials immediately and doesnt wait for the next digit timeout. Cheers Gerry -Original Message From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Samstag, 9. September 2006 15:15 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call Processing Slow 11 seconds I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3 -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/632afcb4/attachment-0001.htm -- Message: 3 Date: Sat, 09 Sep 2006 18:23:37 +0100 From:
[asterisk-users] Quintum tenor configuration with asterisk help
Hi I need help configuring a quintum box with asterisk. Anyone has it working ? Thanks, Please let me know what I should do. I want to be able to register the asm200 with an extension, and be able to hopoff calls when calling from my asterisk, Thanks, On 9/9/06 6:47 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: Another (quick) Polycom 501 question (Kevin Smith) 2. RE: asterisk-users Digest, Vol 26, Issue 54 (FRANCISCO PEREZ-LANDAETA) 3. Re: Call Processing Slow 11 seconds ([EMAIL PROTECTED]) 4. Re: Zaptel-1.2.9 compile error (Samy Antoun) 5. Problems configuring Polycom 301 (Jim Freeze) 6. Re: Zaptel-1.2.9 compile error (Nigel Godfrey) 7. ztdummy installed but choppy audio warning on load (Nigel Godfrey) 8. Re: ztdummy installed but choppy audio warning on load (Daniel Pocock) 9. Re: Zaptel-1.2.9 compile error (Samy Antoun) 10. Scope of contexts (Rene) 11. Re: What don't I get about SIP? (John Marvin) 12. Re: Scope of contexts (Doug Lytle) 13. Re: Scope of contexts (Moises Silva) 14. Re: Grandstream GX-2000, doesn't send calls to free lines (Zeeshan Zakaria) 15. Re: How to send correct Caller ID on PRI (Zeeshan Zakaria) 16. Re: How to use Grandstream GX-2000 phones for paging (Zeeshan Zakaria) 17. Re: Grandstream, how to use the configuration tool (Zeeshan Zakaria) 18. Re: Roundrobin not working on PRI (Zeeshan Zakaria) 19. Using option 'r' in queue doesn't announce frequeny etc. (Zeeshan Zakaria) -- Message: 1 Date: Sat, 09 Sep 2006 15:24:44 -0400 From: Kevin Smith [EMAIL PROTECTED] Subject: Re: [asterisk-users] Another (quick) Polycom 501 question To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi Mike, As far as I know, you need to at least start the dialing (ie New call, speaker, etc) for the digitmap to even come into play. The only settings that I am aware of that you can try to change are dialplan.impossibleMatch-Handling and dialplan.digitmap from sip.conf. Kevin Mike wrote: Hi all, That's my last one for a while (I hope). How can I (if at all possible) make the 501 turn on the speaker phone as soon as a digit is dialed (if the handset is not lifted)? Sort of like what a normal speakerphone does. The reason I want this is I want the 501 digitmap to be taken into consideration even if the handset isnt lifted and the speakerphone button isn't consciously pressed. For all those users who don't want to press send, but like dialing without lifting the handset (and can't be bothered to press the speakerphone button). Yes I know it's capricious, but we have the users we have... Yes, I have read the admin manual, but couldn't find the info. I am assuming I just don't know what to look for, but that this functionality exists. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Message: 2 Date: Sat, 09 Sep 2006 19:48:27 + From: FRANCISCO PEREZ-LANDAETA [EMAIL PROTECTED] Subject: [asterisk-users] RE: asterisk-users Digest, Vol 26, Issue 54 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; format=flowed hi i need helpl configuring a quintum tenor analog gateway using sip with asterisk. anyone, help is appreciated the model of the gteway is asm200 i need the settings to configure it with asterisk. for some reason it registers with asterisk but when try to call the extension from the quintum it is not recognized. help help help thanks From: [EMAIL PROTECTED] Reply-To: asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 26, Issue 54 Date: Sat, 9 Sep 2006 12:00:25 -0700 (MST) MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc2-f18.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Sat, 9 Sep 2006 12:03:59 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost
[Asterisk-Users] Broadvoice Problems.-
I am a user of broadvoice. I have recently acquired another account to use it with my asterisk box. However, there seems to be a problem with the registration. I keep on getting the following message : Nov 8 18:30:27 WARNING[2306]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call Nov 8 18:30:41 NOTICE[2306]: chan_sip.c:4053 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again -- parse_srv: SRV mapped to host proxy.dca.broadvoice.com, port 5060 Any ideas, how to make it work. I read there is a patch, but I am not sure if I should install it. Can someone send me a working configuration file just in case to [EMAIL PROTECTED] Thanks, On 11/10/04 4:04 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Problem adding zaprtc to Asterisk CVS on debiansarge (Sebastian Mauer) 2. Broadvoice asterisk patch ([EMAIL PROTECTED]) 3. Broadvoice Patch (TELUX) 4. Re: Broadvoice asterisk patch (Ryan Wilkins) 5. Re: Re: Is this good or bad (Rich Allen) 6. RE: Broadvoice asterisk patch (Tim Jackson) -- Message: 1 Date: Wed, 10 Nov 2004 20:51:39 +0100 From: Sebastian Mauer [EMAIL PROTECTED] Subject: [Asterisk-Users] Problem adding zaprtc to Asterisk CVS on debiansarge To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello there, I'm still new to Asterisk and until today i did not even know that Asterisk needs a timing source for correct use of MeetMe and MOH. So I looked after ztdummy (but I found out that I have USB-OHCI instead of UHCI) and came finally to zaprtc But there the trouble starts I fetched all zaptel, and asterisk from CVS, and put zaptrtc in the right directory: Making zaprtc: trinity:/usr/local/src/zaptelrtc# make cc -c zaprtc.c -D__KERNEL__ -DMODULE -DEXPORT_SYMTAB -fomit-frame-pointer -O2 -Wall -I/usr/src/linux/include -I../zaptel -Wall -DMODVERSIONS -include /usr/src/linux/include/linux/modversions.h In file included from /usr/src/linux/include/linux/spinlock.h:6, from /usr/src/linux/include/linux/module.h:12, from zaprtc.c:60: /usr/src/linux/include/asm/system.h: In Funktion ?__set_64bit_var?: /usr/src/linux/include/asm/system.h:192: Warnung: dereferencing type-punned pointer will break strict-aliasing rules /usr/src/linux/include/asm/system.h:192: Warnung: dereferencing type-punned pointer will break strict-aliasing rules gcc -s -Wall -Wstrict-prototypes rtctest.c -o rtctest rtctest.c: In Funktion ?main?: rtctest.c:30: Warnung: implicit declaration of function `exit' gcc -s -Wall -Wstrict-prototypes rtcsetup.c -o rtcsetup rtcsetup.c: In Funktion ?main?: rtcsetup.c:31: Warnung: implicit declaration of function `exit' rtcsetup.c:23: Warnung: unused variable `i' rtcsetup.c:23: Warnung: unused variable `irqcount' rtcsetup.c:24: Warnung: unused variable `tmp' rtcsetup.c:24: Warnung: unused variable `data' rtcsetup.c:25: Warnung: unused variable `rtc_tm' sync And now I try to activate it finally with make load: trinity:/usr/local/src/zaptelrtc# make load sync modprobe zaptel /lib/modules/2.4.27-1-386/misc/zaptel.o: /lib/modules/2.4.27-1-386/misc/zaptel.o: unresolved symbol create_proc_entry_R6f790657 /lib/modules/2.4.27-1-386/misc/zaptel.o: /lib/modules/2.4.27-1-386/misc/zaptel.o: unresolved symbol remove_proc_entry_R96e689cb /lib/modules/2.4.27-1-386/misc/zaptel.o: /lib/modules/2.4.27-1-386/misc/zaptel.o: unresolved symbol register_chrdev_Rbe900e0b /lib/modules/2.4.27-1-386/misc/zaptel.o: /lib/modules/2.4.27-1-386/misc/zaptel.o: unresolved symbol proc_mkdir_R8be6b1b2 /lib/modules/2.4.27-1-386/misc/zaptel.o: /lib/modules/2.4.27-1-386/misc/zaptel.o: unresolved symbol add_wait_queue_Rd4158ead /lib/modules/2.4.27-1-386/misc/zaptel.o: /lib/modules/2.4.27-1-386/misc/zaptel.o: unresolved symbol remove_wait_queue_R93e76632 /lib/modules/2.4.27-1-386/misc/zaptel.o: /lib/modules/2.4.27-1-386/misc/zaptel.o: unresolved symbol __pollwait_R8701678a /lib/modules/2.4.27-1-386/misc/zaptel.o: insmod /lib/modules/2.4.27-1-386/misc/zaptel.o failed /lib/modules/2.4.27-1-386/misc/zaptel.o: insmod zaptel failed make: *** [load] Fehler 255 I don't know what to do, but I hope someone here on this maillist can help me. By the way, YES rtc is disabled as told in the zaptrtc README, it's a 2.4.27 baked by myself Thanks in Advance, and
[Asterisk-Users] Digum board TDM to Phonejack --Quicknet --Trandsfering calls.
I have succesfully integrated some phonejacks with Zaptels. I am able to transfer calls from my tdm board to my phonejack (from quicknet) using the hangup button (pressing it once). But I am unable to do this the other way around with the quicknet board. This works : Phonejack --Digium TDM -Phonejack But, THIS DOES NOT WORK Digium TDM to ---Phonejack -- X ( does not work ) --another extensions with a phonejack ANY HELP IS APPRECIATED. Thanks Francisco ___ 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] Quintum ASM400, ASM200 and ASTERISK
I have some ASM200 and ASM400, these are analog gatewyas, The ASM 200 - 2 fxs 2 fxo ports (only two simulatenous calls) The ASM 400 - 4 fxs 4 fxo Ports ( only 4 simulatenous calls ) My intention is to integrate them with Asterisk, so that I can use their FXS channels as internal extensions in conjunctions with my ZAP boards and their fxo ports as outgoing to the pstn line. The only issue is I'm not use how to integrate or use the (ports) in the quintum box to asterisk, so asterisk can see them as extensions. ?? Any clues on this. Currently, there is a firmware that supports SIP and H323.. Hope to get some help from u guys !!! Cheers I am sure this will be helpful for some people out there... Thanks, Francisco ___ 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] Personal Phone Gateway PCI and USB Phone.-
Hi, I was looking at a board that I have from tjnet.com and noticed that I looks almost the same as digiums X100p. I was wondering if anyone has tried using this board with Asterisk ? I also have a pair of USB phones form tjnet and I believe they use the tiger 560B and / or Tiger320. Their website is www.tjnet.com Has anyone played around with this hardware and made it work. Are there any linux drivers for them ? Greetings, Francisco ___ 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] . Re: Quicknet Linejack Asterisk PBX (Jeremy McNamara)
Thanks Jeremy !! We met at the conference. I will do as you say ! ;-) Regards. Francisco ___ 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] 4. Re: Quicknet Linejack Asterisk PBX (Lubomir Christov)
Thanks for your advice. I will consider moving on. Regards, Frank ___ 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] Quicknet Linejack Asterisk PBX
Hi, I am in the process of setting up an Asterisk PBX with some Quicknet Linejacks that I have. Has anyone been successful with this setup ? I have a PC with 7 Linejacks and would like to set it up as a PBX with two incoming lines and 7 extensions. These two lines will be to dial out and to receive incoming calls. For this setup I don¹t need voicemail or any of the fancy features, just a phone that rings so that one can pick up the call and transfer it. I am just wondering if the linejacks (7 of them) will work ok with Asterisk I am 100% sure that digium will work but I am not sure about the linejacks. I need to do this project and have it ready this week so your help and cooperation is appreciated. I know that many people don¹t like the Linejacks but this is a project and I must make it work. Are there any tricks to transfer the calls to other phones ? Thanks guys ! Frank ___ 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] Re: Asterisk-Users Digest, Vol 1, Issue 5082
Anyone using the recently MAC OS X ? Version of asterisk ? Thanks, Francisco Perez-Landaeta From: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Fri, 27 Aug 2004 13:08:24 -0500 (CDT) To: [EMAIL PROTECTED] Subject: Asterisk-Users Digest, Vol 1, Issue 5082 Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: how to fetch a call? (Tony Mountifield) (Sudhir Kumar) 2. Asterisk Assistants Custom Icon (Sunrise Ltd) 3. RE: Overhead Paging (Jay Milk) 4. libr2 (Vikram Rangnekar) 5. Speech Recognition and Asterisk (Mike Meyer) 6. Re: FXOs (Marcelo Mercio Dandrea) 7. RE: sip change? (Jerry Roy) 8. RE: sip change? (Chad Brown) 9. RE: Overhead Paging (Rich Adamson) 10. Re: sip change? (Doug Shubert) 11. RE: Faxing with SPANDSP or any other mean ? Isitpossible ? Am I dreaming ? (Jean-Fran?ois Rousseau) 12. RE: Faxing with SPANDSP or any other mean ? Isitpossible ? Am I dreaming ? (Jean-Fran?ois Rousseau) 13. RE: Faxing with SPANDSP or any other mean ? Isitpossible ? Am I dreaming ? (Jean-Fran?ois Rousseau) -- Message: 1 Date: 27 Aug 2004 13:07:10 -0400 From: Sudhir Kumar [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: how to fetch a call? (Tony Mountifield) To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED], [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain Remote Call Pick up feature is very much implemented in asterisk. You can pick up a call for another extension by dialing *8# To be able to do that, you need to have the extensions in the same pickup group, configurable through sip.conf and zapata.conf. -- sudhir -- Message: 14 Date: Fri, 27 Aug 2004 14:17:26 + (UTC) From: [EMAIL PROTECTED] (Tony Mountifield) Subject: [Asterisk-Users] Re: how to fetch a call? To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] In article [EMAIL PROTECTED], Roger Schreiter [EMAIL PROTECTED] wrote: Hi, there is a feature, which I would like to use with asterisk, and I assume it exists. Unfortunately I don't know how to say it in english. In german it's einen Ruf heranholen. It means: The phone set of my collegue is ringing, and I'm hearing the ringing. I know, that my collegue is not at his desk, and now I want to answer the call at my phone (instead of running to my collegue's desc to answer at his phone). I don't know whether it is implemented or not in Asterisk, but the feature is known in English as call pickup. mfg, Tony -- Message: 2 Date: Sat, 28 Aug 2004 02:14:45 +0900 (JST) From: Sunrise Ltd [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Assistants Custom Icon To: astusr [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-2022-jp I think I need to clarify what I meant by custom icon for the Asterisk Assistants in my earlier posting. On the Mac an Assistant is what the Windoze world calls a Wizard and there is a generic icon for it - the front of a dinner suit with bow tie, the one you can see on the Wiki. However, many of Apple's own assistants have a little mark in the lower right corner of the generic icon which further hints at what the respective assistant is for. An example for that is the Airport Assistant which has a little Airport base station in the lower right corner ... http://www.sunrise-tel.com/screenshots/AirportAssistantIcon.png (Aiport is what Apple calls their WiFi gear) What I had in mind for the Asterisk Assistants is an icon just like the one at the above link, but with an Asterisk replacing the Airport base station in the lower right corner. This would fit in and still project Asterisk's brand into the Mac world. So, please don't get too fancy with this, it's more of a cut and paste kind of job which I had in mind. For those who are interested in making more fancy icons, we'll have other tools outside of the Assistant series later on ;-) thanks rgds benjk -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __ GANBARE! NIPPON! Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE http://mail.ganbare-nippon.yahoo.co.jp/ -- Message: 3 Date: Fri, 27 Aug 2004 12:27:22 -0500 From: Jay Milk [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Overhead Paging
[Asterisk-Users] Deltathree and Go2call registration ? Anyone ?
Hi Guys, I am really having a hard time with this configuration. I will like to test Asterisk using Deltathree or Go2Call. I am totally clueless. I have check the wiki and followed the instructions but have been unlucky. It would be nice if someone could tell me how to configure this for asterisk. I have asked this questions a few times now but nobody has answered me yet. Well. I look forward to receiving some help. Thanks ! Francisco ___ 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] SIP Providers Setup.- Go2call, deltathree, etc...setup files ? any
Hi guys, . i have been playing around with it for a while but havenot figure out how to configure SIP Providers like go2call. They provided me a test account but i have been unable to set it up. I have a test account with no password and an ip address, i am not connecting behind a firewall. I have some x100p and TDM boards from digium as well as phonejacks, linejacks,etc. I am not sure if this will work since go2call requires g729 and i am not sure asterisk will do the conversions (although the codec is present in the phonejack). thanks for your help. Best Regards, Francisco Perez-Landaeta This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this for the addressee, you must not use, copy, disclose or take any action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply e-mail and delete this message. Thank you for your cooperation.
[Asterisk-Users] go2call setup ?
Hi guys, Anyone running go2call setup ? can anyone send me the configuratio sip.conf lines ? I am planning on using asterisk with a linejack and phonejack. I am not sure if this will work. These cards use g729 and g723.1. I also have some x100p and tdm cards from digium but without the codecs. I am not sure if this will work since asterisk acts as a passthrough.. What is really the meaning for pass through ? iin the case of asterisk ? your help is greatly appreciated ! Francisco _ Discover the best of the best at MSN Luxury Living. http://lexus.msn.com/ ___ 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] Setup for Go2call ? Or any SIP provider using phonejack or linejack g729 g723
Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723 Thanks, From: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Mon, 19 Jul 2004 19:48:02 -0500 To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #4610 - 12 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: Re: Cisco 7960 SIP V6 and distinctive ring. (Sam Tilders) 2. Re: Asterisk + NEC Electra Elite IPK Integration (Jason Kawakami) 3. RE: Polycom IP 500 Voicemail (Wiley E. Siler) 4. Re: uip200 clips audio prompts (Ryan Courtnage) 5. MWI - Config Stupidity or Notify Issues? (Robert Jackson) 6. RE: RE:RE: [Asterisk-Users] Codecs - Advantages (Wiley E. Siler) 7. RE: Polycom IP 500 Voicemail (Wiley E. Siler) 8. RE: Polycom IP 500 Voicemail (Chris A. Icide) 9. Echo on a PRI (David Goldfein) 10. Suscription (Carlos Clemares) 11. RE: Echo on a PRI (Wiley E. Siler) 12. Re: SIP to H323 call timeout (administrator tootai) --__--__-- Message: 1 Date: Tue, 20 Jul 2004 09:25:11 +1000 From: Sam Tilders [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Cisco 7960 SIP V6 and distinctive ring. Reply-To: [EMAIL PROTECTED] On Mon, Jul 19, 2004 at 02:09:34PM -0700, asteriskstuff @ ziplip. com wrote: Thanks..it's a numeric value!! in the wiki it refers to a text field!! The wiki is also correct... I have: exten = 101,1,SetVar(ALERT_INFO=Bellcore-dr1) And that works fine. What was the error message you were getting? -- -- Sam Tilders [EMAIL PROTECTED] (Move to Jupiter) --__--__-- Message: 2 From: Jason Kawakami [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration Date: Mon, 19 Jul 2004 17:28:09 -0600 Reply-To: [EMAIL PROTECTED] Date: Mon, 19 Jul 2004 14:54:44 -0500 From: Christopher L. Wade [EMAIL PROTECTED] Organization: Unistar-Sparco Computers, Inc. To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration Reply-To: [EMAIL PROTECTED] Would the TLI(2)-U10 ETU work as well? That is a 2 port analog tie line card, I don't think that Digium has a card that can be set up as an analog 4W EM trunk. bad idea anyway, the t-1 will be a much better interface and if you ever press the eject on the IPK you could use the t-1 as a PSTN interface. --__--__-- Message: 3 Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail Date: Mon, 19 Jul 2004 16:28:25 -0700 From: Wiley E. Siler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Mine does the same. Once in Message center I can choose selection 1.Message Center and then soft key Select.Then I select the registered line that I want to check voice mail on. That is no less than 4 key strokes just to get into your voice mail. Not many to me but tons to an unskilled user. However, in the documentation regarding the bypassInstantMessage value, supposedly, setting bypassInstantMessage to 1 is supposed to allow you to go right into voice mail without navigating the Message Center. That is the big question on my mind at this point. I have yet to get this to work and I also don't think I am receiving any SIMPLE messages ti show me that I have messages waiting. Do you get a message waiting indicator? W -Original Message- From: Chris A. Icide [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 3:03 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail On 12:40 PM 7/19/2004, Wiley E. Siler wrote: My Polycom is on loan as a demo and I assume it is one of the first revision models. In fact it shows as Rev A on the back of the phone. I have all the same buttons you listed save for the Messages button. The 3rd from the bottom on the right column of buttons sayd Voice Mail on my version. That corresponds to the location of your button that says Messages. I assume this was changed by Polycom since their phone has other messaging capability (isntant message for instance) and it was easier to use Messages and unify the meaning instead of Voice Mail and lock it into one type of messaging. Does your Messages button dump you right into voice mail
[Asterisk-Users] MAC OS X Panther :?
Just wondering if anyone has tried MAC OS X and panther. I will like to do SIP to H323, not sure if this will be possible on the MAC because of the Libraries PWlib and OPenh32 for Linux.. Just curious.. Anyway, anyone has an easy guide (step by step) to setup oh323 with asterisk. I saw a guide but i am not very savy on linux. thanks, Francisco - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 19, 2004 12:25 PM Subject: Asterisk-Users digest, Vol 1 #4598 - 14 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: STILL NO AUDIO (Michael Manousos) 2. Re: TDM400P Internal Extenion Config (Nick Cobley) 3. Re: ZyXEL 2000W (Jason Williams) 4. Channel banks, voicemail, and immediate=no (Chris A. Icide) 5. RE: STILL NO AUDIO (Eric Wieling) 6. Re: STILL NO AUDIO (Holger Schurig) 7. RE: Mac OS X installer for Asterisk (Wallingford, Ted) 8. Re: PhoneGaim? ([EMAIL PROTECTED]) 9. Re: BroadVoice problems? (Chris Shaw) 10. RE: STILL NO AUDIO (Sebastian Nocetti) 11. Re: TDM400P Internal Extenion Config (Jason Williams) 12. IP Phone recommendation (Yiannis Costopoulos) 13. Re: Cheap PoE switches/injectors? ([EMAIL PROTECTED]) 14. RE: STILL NO AUDIO (Sebastian Nocetti) --__--__-- Message: 1 Date: Mon, 19 Jul 2004 18:24:39 +0300 From: Michael Manousos [EMAIL PROTECTED] Organization: inAccess Networks To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] STILL NO AUDIO Reply-To: [EMAIL PROTECTED] Why don't you use asterisk-oh323? Michael. Sebastian Nocetti wrote: I WANT TO USE G729, I HAVE TO USE IT... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Eric Wieling Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] STILL NO AUDIO I suspect it will be solved when you put disallow=all and allow=ulaw in sip.conf and h323.conf (and NO OTHER ALLOW= LINES) On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote: I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING It happened in both: SIP - CHAN_H323 and CHAN_H323 - SIP... when it will be solved? --__--__-- Message: 2 Date: Mon, 19 Jul 2004 23:26:06 +0800 From: Nick Cobley [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] TDM400P Internal Extenion Config Reply-To: [EMAIL PROTECTED] Thanks Steve, The SIP handsets are working find as I can make calls to other handsets as well as receive incoming calls via the FXO module. So all is good there. Cheers Nick Steven Critchfield wrote: On Mon, 2004-07-19 at 07:13, Nick Cobley wrote: If I dial the extension I just get a 404 error on the phone (Grandstream), but no errors at all on the console. I am using CVS-HEAD-07/14/04. Here is a snippet of what I have in the various config files. Welcome to SIP. Dialtone is local to your phone and is not dependent on proper config. Hope that helps put you on the correct step to fix that problem. --__--__-- Message: 3 Date: Mon, 19 Jul 2004 16:26:26 +0100 From: Jason Williams [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ZyXEL 2000W Reply-To: [EMAIL PROTECTED] On Fri, 16 Jul 2004 01:23:54 +1000, Andrew Yager [EMAIL PROTECTED] wrote: Does anyone have the call hold feature working? If you do... how did you make it work? The instructions say to press the left button to place the call on hold, and the right button to take it off - except when I am in a call, these keys have no effect. I've tried teh 000c firmware, the 000e firmware and the Pulver 0011 firmware - but none work, so I'm wondering if this feature just simply isn't implemented, or if there is likely to be something wrong with my asterisk config. No it does not work, you need to use # transfer which will mean you will not be able to dial # into ivr's. Search on wiki for # transfer Regards Jason --__--__-- Message: 4 Date: Mon, 19 Jul 2004 08:26:32 -0700 To: [EMAIL PROTECTED] From: Chris A. Icide [EMAIL PROTECTED] Subject: [Asterisk-Users] Channel banks, voicemail, and immediate=no Reply-To: [EMAIL PROTECTED] When using a channel bank for analog handsets, you have a couple options in the way you handle transactions involving the analog handsets and origination. With immediate set to no, it appears to me that soon as a digit is pressed after going off-hook, the single digit is taken and processed against
[Asterisk-Users] Doubt about IP address setting for Asterisk
Hi, I have a doubt with the installation of asterisk and redhat 9 when i tried setting up the redhat, and said something about the HOST FILE. I had to modify it and put my address. XXX.XXX.XXX.XXX. is this correct or will this affect the configuration of asterisk in another way. Then, since i dont have a domain like pulver.com or iax. dot something. what will i have my domain or server be ? thanks, - Original Message - From: Greg Boehnlein [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 11, 2004 10:51 AM Subject: Re: [Asterisk-Users] Radius On Wed, 10 Mar 2004, Anton Tinchev wrote: Just make a wrapper. 100 lines in perl. Do you have an example that you can share? Derek Samford wrote: I know this has been hashed, and rehashed, but I saw that a few people had said they were going to release their code soon. Is there a working implementation of RADIUS for Asterisk out there? Not looking to start a debate on how bad it is for billing purposes, that's a given, but I need it for legacy systems. Thanks, Derek ### This message has been scanned by F-Secure Anti-Virus ___ 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 -- 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 ___ 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] newbie
Hi guys, I am kind of overwelmed with all the information in the asterisk site and have no clue where to start. I have review some files but i am not certain how to assemble all this. I got a dev kit with one fxo, and one fxs port. I would like to setup my server to take incoming calls and hop them off to the local pstn line. I also intend to have another pc equiped with the fxs port to be able to use my psntn server (asterisk).. has anyone come accross with the same situation ? What files in order should i modify ? thanks, your help is appreciated. Please reply to [EMAIL PROTECTED] have a great day. Francisco _ Frustrated with dial-up? Lightning-fast Internet access for as low as $29.95/month. http://click.atdmt.com/AVE/go/onm00200360ave/direct/01/ ___ 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] newbie
thanks, i will see if i can start...this. - Original Message - From: Andrew McRory [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 12:05 PM Subject: Re: [Asterisk-Users] newbie On Thu, 4 Mar 2004, FRANCISCO PEREZ-LANDAETA wrote: Hi guys, I am kind of overwelmed with all the information in the asterisk site and have no clue where to start. I have review some files but i am not certain how to assemble all this. I got a dev kit with one fxo, and one fxs port. I would like to setup my server to take incoming calls and hop them off to the local pstn line. I also intend to have another pc equiped with the fxs port to be able to use my psntn server (asterisk).. has anyone come accross with the same situation ? I can offer some links that helped me... http://www.codepipe.com/id25.htm http://www.jaredsmith.net/misc/hgta/ http://www.wwworks-inc.com/asterisk/ http://www.fnords.org/~eric/asterisk/ http://bcwireless.net/moin.cgi/VoIPHowTo http://www.automated.it/guidetoasterisk.htm http://www.asterisk.org/index.php?menu=support http://www.voip-info.org/wiki-Asterisk+config+files http://www.voip-info.org/tiki-index.php?page=Asterisk http://asterisk.sohoskyway.net/Asterisk_Doc/current/docs-html/c526.html If anyone has other links I'd appreciate them! -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. PO BOX 3791 Tallahassee, FL 32315 (850)224-5737 (850)294-7567 ___ 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 setup.-
Hi, I recently received my development kit with 1 x100p and one tdm400p (1) fxs port. I installed everything from the digium disk that i received with my kit, however, i dont; know what to do next. I would like to be able to call through the internet using xten (pc2phone) and terminate the call in my gateway. anyone has a standard setup ? thanks, Francisco - Original Message - From: Steven E. Frazier [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 06, 2004 12:30 AM Subject: [Asterisk-Users] Adding another X100P after X100P and TDM400P is already configured History: 1. Added X100P to my system 2. Added TDM400P (2 port) Worked fine so far 3. Now I want to add an additional X100P Is the following configs files ok and is there any issue with adding the X100P (channel 4) after my 2 analog FXS channels? Thanks. Steve Here is my /etc/zaptel.conf fxsks=1,4 fxols=2-3 loadzone = us defaultzone = us Here is my /etc/asterisk/zapata.conf ; Zapata telephony interface sample configuration file ; [channels] ; ; X100P plugged into PSTN ; X100P # 1 context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel = 1 ; ; ; ; TDM200B Port #1 plugged into analog Phone ; ; context=toll-access signalling=fxo_ls callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes callerid=Livingroom 2201 mailbox=2201 channel = 2 ; ; TDM200B Port #2 ; ; context=toll-access signalling=fxo_ls callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes callerid=Kitchen 2202 mailbox=2202 channel = 3 ; X100P # 2 context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel = 4 ___ 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 settings
Hi, i finally was able to getdialtone on my fxs board. !! however, i think i am missing something in the fwd setting to make work my account. i am getting an error authenticating my account could someone send me the exact settings to put on sip.conf ? to make it work ? i have my own account, password but i am getting it wrong. thanks, Francisco
[Asterisk-Users] help *** newbie
can anyone help me on this ? i am having problems configuring the asterisk. i have included an attachment because for some reason i could not cut and past from the terminal to my hotmail account. your help is appreciated. thanks, *** please look at the errors francisco _ Check out the new MSN 9 Dial-up fast reliable Internet access with prime features! http://join.msn.com/?pgmarket=en-uspage=dialup/homeST=1 My config.sxw Description: OpenOffice Writer document
[Asterisk-Users] setting up ---- newbie
hi guys, i am getting today my dev kit with fxo and fxs boards. i intend to do the following : 1) be able to route an incoming call from the pstn fxo port to an ip (answering with netmeeting or anyother sip softphone) 2) be able to call from netmeeting to my pstn fxo port to place calls. questions : how can i do this ? what are the commands for this simple setup ? how can i place calls using a webbrowser (explorer, etc ?) can i use messenger to call to call my pstn port ? can i translate h323 to sip and viceversa ? thanks, francisco
[Asterisk-Users] Phonejack
Sorry, if this question has been asked before. I have redhat 9 and asterisk installed. I would like to know how can i make work the phonejack with asterisk. From what i have read not too many people are happy with it, but i will like to give it a try before i receive my digium. Any instructions or guideline isappreciated. Thanks, Francisco
Re: [Asterisk-Users] Phonejack
so, i have two questions : 1) how soon will the this be ready 2) Is there an alternate method ? thanks, Francisco - Original Message - From: Bruce Ferrell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 08, 2004 9:54 AM Subject: Re: [Asterisk-Users] Phonejack the most difficult thing about the phonejack is getting the driver installed. We (Quicknet) ate working on getting the latest driver inserted into the kernel so that the outside installation isn't needed anymore. Francisco Perez-Landaeta wrote: Sorry, if this question has been asked before. I have redhat 9 and asterisk installed. I would like to know how can i make work the phonejack with asterisk. From what i have read not too many people are happy with it, but i will like to give it a try before i receive my digium. Any instructions or guideline is appreciated. Thanks, Francisco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lindows ?
Has anyone tested asterisk with Lindows ? just curious ? thanks, Francisco
[Asterisk-Users] Setting up asterisk on Rh 9
Hi Friends, I am new to linux and new to asterisk. I need some help setting up asterisk in my linux box. Does anyone have a step by step guide ? On my PC i have installed a phonejack (from Quicknet) as well. Your help is appreciated.. I kind lost.. thanks, _ Expand your wine savvy and get some great new recipes at MSN Wine. http://wine.msn.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting up asterisk on Rh 9
thanks.. i will try it out.. I really appreciated... Francisco From: WipeOut [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Setting up asterisk on Rh 9 Date: Sat, 27 Dec 2003 09:39:33 + FRANCISCO PEREZ-LANDAETA wrote: Hi Friends, I am new to linux and new to asterisk. I need some help setting up asterisk in my linux box. Does anyone have a step by step guide ? On my PC i have installed a phonejack (from Quicknet) as well. Your help is appreciated.. I kind lost.. thanks, My install guide may help.. http://members.lycos.co.uk/wipe_out/asterisk Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ Working moms: Find helpful tips here on managing kids, home, work and yourself. http://special.msn.com/msnbc/workingmom.armx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unsubscribe
unsubscribe - Original Message - From: jerk face [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 26, 2003 9:57 AM Subject: Re: [Asterisk-Users] RE: T100P/ TSU 600 installation problem Oops ... I found out my problem span= --- jerk face [EMAIL PROTECTED] wrote: I am using a crossover cable. My channel definitions are: fxoks=1-22 fxsks=23-24 in zaptel.conf --- Alex Lopez [EMAIL PROTECTED] wrote: What cable are you using, The SU600 to Digium cards need a crossover cable. 1 to 4 2 to 5 4 to 1 5 to 2 That would stop it from not working, also make sure that you have a span definition on the zaptel.con file. Message: 7 Date: Mon, 25 Aug 2003 12:52:12 -0700 (PDT) From: jerk face [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T100P/ TSU 600 installation problem To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Each port is set to the proper signalling type (FXO, FXS). I can't find any other options for the individual ports. As for the timing and configuration of NI, I have tried NI: Timing Mode as both DTE and NI (my only choices) Where else should I be checking? (Before this morning, I hadn't even seen a channel bank before, so I'm a little lost at the moment). Thanks for your time. --- Wade Weppler [EMAIL PROTECTED] wrote: Have you configured the TSU600 properly? You have to allocate each FXO/FXS channel to a timeslot before it will work. This is not automatically done (like the Adtran Total Access series). Mind you, you should still have a sync light on the T1 card... -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of jerk face Sent: Monday, August 25, 2003 3:01 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T100P/ TSU 600 installation problem My zapata.conf is located in /etc/asterisk and my zaptel.conf is located in the /etc directory. --- Adams, Gavin [EMAIL PROTECTED] wrote: -Original Message- From: jerk face [mailto:[EMAIL PROTECTED] I seem to be having a problem with the T100P card. So far I have done the following: vi zaptel.conf fxoks=1-22 fxsks=23-24 ... vi zapata.conf ... signalling=fxo_ks ... channel = 1-22 ... signalling=fxs_ks ... channel = 23-24 I then run modprobe zaptel modprobe wct1xxp ztcfg -vv There are no errors to report. In /proc/zaptel/1 all of the configured channels are listed. Crazy question, is zaptel.conf is /etc or /etc/asterisk? If the latter, try: ztcfg -c /etc/asterisk/zaptel.conf -vvv --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speex
does anyone know what compression uses Speex ? and which gateways used it so they can work with Asterisk. Thanks, - Original Message - From: Emanuele Pucciarelli [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 17, 2003 8:49 AM Subject: Re: [Asterisk-Users] Speex On Tue, Jun 17, 2003 at 02:27:38PM +0200, Tjardick van der Kraan wrote: It seems * is not loading speex. When i did a make in the codecs sub dir, the following error pops up when making speex: codec_speex.c:34:19: speex.h: No such file or directory is this file missing in the cvs as i just removed the whole * dir and did a new checkout and still seem to get this error, or do i need to get/install something before speex works ? Yes, you ought to install libspeex! It is probably installable by your distro (libspeex-dev in Debian, for example). Bye, -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] InternetPhoneWizard
I also have the tigerjet usb phone. I was wondering if it would work too. does the tiger work fine with 723.1 ? do you know if the pci board they sell is compatible with asterisk ? thanks, - Original Message - From: Humberto Atristain V. [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 15, 2003 12:39 PM Subject: Re: [Asterisk-Users] InternetPhoneWizard I have used Tiger jet usb phones and works with asterisk but with Open h323 softphone in h.323 or eyepmedia sip softphone , it`s only a USB speaker and mic , what wee need to ahve is the dialpad working into this device and the best of all this Hardware is $20.00 bucks regards Humberto Mensaje citado por: Kim C. Callis [EMAIL PROTECTED]: I recently had a Active InternetPhoneWizard USB sent to me from iconnecthere. Although they push the software that iconnecthere runs, I figured there is a way to make use of it exclusive of iconnecthere. Has anyone played with this device? It would make for a cheap way to get a connection to a * box. Any information would be appreciated! Kim Callis Humberto Atristain V. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quicknet.-
Hi, I am sure this question has been asked a hundred times. I am planning on purchasing the Dev Kit to experience with * and do some voip configurations. However, I see that Quicknet phonejack and linejacks are compatible. IS it worth the time to try Quicknet hardware or is it a waste ? My plans are to setup a gateway. Thanks, Francisco - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, June 13, 2003 6:11 AM Subject: [Asterisk-Users] Is it possible? Dear Folks, Before using my precious time trying to configure what I'm imagining, I would like to confirm if it's possible with asterisk. I'm planning to buy several E100 cards to plug in my PSTN network here in Japan and configure another H323 device in Brazil. I would receive all calls here in Japan through IVR, get the user ID and its password. So, I authenticate that in a database(postgres or mysql) and play a message or a beep in the IVR allowing the user to press the destination number. After that, the call is completed through H323. I need H323 because the other side is running H323. For inbound calls testing here in Japan, in the beginning I'd use a Linejack because it's already in my drawer. Questions: 1) Using the current * code, can this mission be accomplished ? 2) A scheme of billing would be possible as well? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
