Re: [Asterisk-Users] Testing asterisk faxing functionality
You could always use System() to copy a call spool file to launch the outbound fax call. I don't really think a 3rd party app is necessary. -Corey On Mon, 27 Mar 2006, Gary Richardson wrote: I was playing with the fax stuff over IP on Friday. Unless you're receiving faxes from a PSTN circuit, it doesn't work so well. Also, I don't think you can chain txfax and rxfax like that. When you hit the s,2 part, it's going to play the fax out to the handset you dialed from. You'll need something like hylafax to send the fax. And you probably want to call Dial(Local/[EMAIL PROTECTED]) to call a local extension.. On 3/27/06, patryk [EMAIL PROTECTED] wrote: I have asterisk with rxfax txfax modules.I want to test fax sendig and reciving in one asterisk instance, in extensions.conf I have : exten = 1234567,1,rxfax(/home/patryk/fax-new.tif|debug) exten = s,1,Dial(1234567) exten = s,2,txfax(/home/patryk/fax.tif|caller|debug) but I doesn't seem to work.But when I'm calling on this number I can hear fax tones. I can't find sip client with fax fuctionality for linux I think it would help with testing. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * This message has been scanned for viruses and dangerous content, and is believed to be clean. * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ --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] OT: XML Content Manager for Cisco 79XX Phones
For anyone interested, our company released a PHP/MySQL based content manager for the Cisco 79XX series IP Phones compatible with the SIP load yesterday. It's available via: http://www.sourceforge.net/projects/open79xxdir Best wishes, -Corey * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP 400 Bad Request from Cisco 7960/7940
Olle, Thanks for looking into it. In doing some ngrep work I figured out where my problem is. Acutal error from the 79xx inside the SIP header is: Warning: 399 Bad Request - 'Malformed/Missing FROM: field' From looks like this: From: Sales Queue sip:12345... Those double-quotes looked bad, so I assumed that the problem was related to this: Set(CALLERID(name)=Sales Queue) that executes before the offending queue. I changed to: Set(CALLERID(name)=Sales) and no success. then to: Set(CALLERID(name)=Sales) and it's OK. Am I just using the Set() command wrong? It seems pretty counter-intuitive not to enclose multi-word strings in quotes but if that's the problem let me know. FYI, we're testing with (right now) CVS-Nv1-2-0-beta1-10/01/05-20:43:03 SIP Firmware on the phone is 7.4. -Corey On Sun, 2 Oct 2005, Olle E. Johansson wrote: Doug Lytle wrote: Olle E. Johansson wrote: Corey S. McFadden wrote: Here's the CLI output: -- Got SIP response 400 Bad Request back from 192.168.249.94 -- SIP/502-9a58 is circuit-busy I've tried a few different Asterisk versions CVS-HEAD, stable, even 1.2 beta. I've also bounced between SIP firmware 7.4 and 7.5 on the 7960/7940 phones. As of Friday evening, we've been seeing this on our system as well. Olle, do you want debugs from other people as well, or will the one you've requested be enough? Just make sure I get one. If I can't figure that one out, I might need more. Thank you for asking. The first one in my mailbox tomorrow morning (it's late in Sweden) will get my attention :-) I need to know version of Asterisk as well. As we are getting very close to release, it's important for us to track down and resolve all outstanding bugs as quickly as possible. The SIP channel has been changing quite a lot during the last two months, so there are a lot of new code in there right now. Thank you for your assistance! /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * This message has been scanned for viruses and dangerous content, and is believed to be clean. * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP 400 Bad Request from Cisco 7960/7940
Steve, I'm glad to know what the problem is. We're back to normal now. FWIW, this was working up until about a week and a half ago and didn't affect our non-Cisco phones... I'm not sure what component (Asterisk, chan_sip, 79xx firmware, etc.) became less tolerant of the error between then and now but I hope it's not indicative of a larger issue. Thanks again, -Corey Am I just using the Set() command wrong? It seems pretty counter-intuitive not to enclose multi-word strings in quotes but if that's the problem let me know. Yeah, that's the problem. Steve * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with VM Distribution Groups
Hi, We're experiencing a problem with basic VM distribution groups where messages won't be delivered. VM is called with a command like: Voicemail(203039... CLI Shows: Oct 1 20:54:33 NOTICE[26943]: app_voicemail.c:1990 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Oct 1 20:54:38 WARNING[26943]: app.c:1125 ast_lock_path: Failed to lock path '': File exists for each attempt and nothing is delivered. I've read a number of threads where the VM distribution list exceeds 256 characters and this breaks things, but that isn't the case here. Delivery fails with even a small number of mailboxes. Voicemail works normally otherwise. Can anyone advise? Thanks, -Corey * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP 400 Bad Request from Cisco 7960/7940
We've been experiencing an odd issue lately. I'm not sure when it started because it's not happening on most calls--it seems confined to a couple of our queues. It's consistent though. Here's the CLI output: -- Got SIP response 400 Bad Request back from 192.168.249.94 -- SIP/502-9a58 is circuit-busy I've tried a few different Asterisk versions CVS-HEAD, stable, even 1.2 beta. I've also bounced between SIP firmware 7.4 and 7.5 on the 7960/7940 phones. Anyone else seeing anything like this? -Corey * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Orinoco Injectors
Has anyone gotten the Lucent / Orinoco injectors (AE-1, AE-6, AE-12) to work with the Cisco 79* series phones? I'm not sure if the are the statndard POE or not Cisco's phones are not standard POE. They reversed the polarity, and I think they run the power hot all the time. Can't remember specifically. Cisco phones will work with any 802.3af standard 48V PoE midspan injector or PoE switch. You just need a patch cable made to the correct spec. * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Good Polycom Dealer?
Kenny, We use T2 Supply. They're one of Polycom's biggest dealers and only deal with resellers. Good service and prices. http://www.t2supply.com/ -Corey On Tue, 6 Sep 2005, Kenny Kant wrote: Could any of you provide me information on a good Polycom phone dealers to utilize. One who provides firmwares ..etc Thank you! Kenny __ Click here to donate to the Hurricane Katrina relief effort. http://store.yahoo.com/redcross-donate3/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * This message has been scanned for viruses and dangerous content, and is believed to be clean. * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 'multi-line' configuration
Pat, To my knowledge the only way to turn on and off the Call Waiting function is on-screen with the phone itself. There are quite a few of these 'little' features I wish would be configurable via the config file but don't seem to be... Best wishes, -Corey Great info! The only question I would have is on the call waiting setting. What should it be set to, and is the setting the one in the SIPX.conf file? Pat -- Corey S. McFadden ([EMAIL PROTECTED]) McFadden Associates - Technology Consultants phone 215-825-2121 x510 - web.csma.biz * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ 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] 7960 'multi-line' configuration
Guys, I added some content to the Wiki on this feature. I don't think it's well documented anywhere. Please expand upon what I put in there if you have more details. http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx -Corey -- Corey S. McFadden ([EMAIL PROTECTED]) McFadden Associates - Technology Consultants phone 215-825-2121 x510 - web.csma.biz * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ 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] Quick Question on Wildcard T100P
Guys, This is probably a stupid question, but I've got a client ordering service from a CLEC and they're going with a fractional T1. Only 6 channels are going to be voice. Is this a problem with the Wildcard T100P? We've only worked with a full PRI before. Thanks for any insight. -Corey * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ 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] AgentCallbackLogin by other means
Hi, Were looking at options for logging agents into the system programmatically via Perl/PHP and I was wondering if anyone else is doing this and if so, how. We're using AgentCallbackLogin now but would like to set up a web interface instead. I've been looking at Asterisk::Manager and didn't see anything relevant and wanted to ask the group before we dove into the Asterisk source. Any input would be immensely appreciated... -Corey -- Corey S. McFadden ([EMAIL PROTECTED]) McFadden Associates - Technology Consultants phone 215-825-2121 x510 - web.csma.biz * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ 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
