Re: [asterisk-users] Asterisk and Faxing
For what it's worth I'm also using HylaFAX 6.0.3 with IAXmodem to talk to my 1.6.0.17 box to send faxes across my SIP provider (they only support SIP at this time). I can't really speak to the reliability for high-volume loads, but I haven't had any problems with the dozen or so that I've sent out over the last couple of months. Travis - Original Message - From: "Chris Hillman" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, December 23, 2009 7:39:30 AM Subject: Re: [asterisk-users] Asterisk and Faxing I'm using hylafax/aixmodem for a fax solution. If your SIP DID provider has a T.38 path to you, faxing should be pretty reliable. I'm on * 1.4 and I'm not too familiar with current fax offerings in 1.6. Hylafax can be configured to email incoming faxes as a PDF, or to accept outbound faxes from a print driver on your client machines Chris Hillman Systems Administrator Clearwater Research, Inc. chill...@clearwater-research.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Fawthrop Sent: Wednesday, December 23, 2009 7:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk and Faxing Hi All I have been looking around and haven not been able to find a working example I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri 1.4.10.2 I use a sangoma A200 card so I am using wanpipe 3.4.7 If I use zaptel which I read I need for app_rxfax then asterisk crashes with segfaults on startup asterisk[2624]: segfault at 30353466 ip b7eb538b sp bffda26c error 4 in libc-2.7.so[b7e3f000+155000] asterisk[2647]: segfault at 30353466 ip b7e6338b sp bfb7708c error 4 in libc-2.7.so[b7ded000+155000] asterisk[2666]: segfault at 30353466 ip b7dfe38b sp bfa6e4ec error 4 in libc-2.7.so[b7ded000+155000] When I use dahdi at least asterisk will start but app_rxfax will fail with unknown symbol in ast_register_application I could only find a precompiled-linux-spandsp-app-fax is there anyway to get the source and compile for myself? Where I can compile for dahdi and not zaptel. Or can someone explain why the segfaulting?? my machine O/S is: Debian kernel : 2.6.26-2-686 i686 My goal is to connect a fax machine to the the sangoma card so I can send paper based faxes. I have a teliax provided SIP phone number which will be the fax number to receive all faxes and have them emailed to a central email address, hopefully in PDF format. where they can be printed and/or forwarded. It would be nice to have the incoming fax emailed to a specific address based on either subject or senders phone number. If this is possible I would like to know how. Thanks in advance Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip show channels display
Hi All, I'm running 1.6.0.17 and wanted to adjust the output from the command sip show channels. When there is a current call in progress the User/ANR field shows 10 numbers. The problem I'm having is that it is including a 1 such as 1949555121 which is truncating the last number. Is there a way to increase the number of characters displayed to 11 or to have the 1 dropped so I can see the full phone number? Thanks, Travis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on queues
Hi Jerry, I use the built-in function queue_member http://www.asterisk.org/docs/asterisk/trunk/functions/queue_member?type=functions&value=QUEUE_MEMBER and check with a GotoIf statement to check if the number is equal to zero. If it is not I send the call to the queue, if it is I pass the call to dial a cell-phone number or go directly to voicemail depending on which queue the call was originally destined for. Travis - Original Message - From: "Jerry Geis" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, December 13, 2009 4:20:40 PM Subject: [asterisk-users] question on queues I have been looking for a way from the dialplan to inquire if there are any members in a queue. So what I want to do is if no users are members of a queue then I can send the call to a given extention. I have the queue setup all that is working. Just need to be able to send the call to a certain user if no-one is logged into the queue. How do I do that? Thanks Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many lines do you use.
Hello all, Do you know if it IS possible to use multiple lines/extensions on SIP with a Cisco 7960 or other phone models? My boss wanted to have 1 physical phone but have it register to a couple of different extensions, then use different ringtones to identify which line was ringing when a call came in. Thanks, Travis - Original Message - From: "Michiel van Baak" To: asterisk-users@lists.digium.com Sent: Wednesday, November 25, 2009 2:40:07 AM Subject: Re: [asterisk-users] How many lines do you use. On 10:18, Wed 25 Nov 09, Julian Lyndon-Smith wrote: > Just for some information really : How many of you use multiple sip lines on > a phone ?. > > I'm sitting here looking at my 7960, with it's 6 lines. I've every only used > one line, and I was wondering if I was a weirdo ;) > > The only time I've ever found a use was when I had two systems (production > and test) and it caused so much grief (could have been asterisk or cisco) I > simply use a softphone for testing now. > > Curious minds are wanting to know ... I use three lines on my cisco 7960 (not sip, but that's not really relevant here) 1 - Private home number 2 - Daytime job number I got from work and is redirected to my home asterisk box from the office pbx 3 - number for my private business. The other three buttons are speeddial. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD "Why is it drug addicts and computer aficionados are both called users?" ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues
Hi Michael, Your web interface for the "on-call roster" is pretty close to what we're trying to trying to achieve. I would like to have people signing into the on-call queue be the method that determined whose cell phone to call. I was hoping there was a way to pass the call exiting the queue to a variable or two that was composed of the extensions currently logged into the queue. I set up an extension number for people to call into and enter a forwarding number which writes an entry into the ASTDB. I have my dialplan check to see if there is an ASTDB entry for that extension before it tries to dial their deskphone, and if there is an entry it dials the forwarded number stored in the database instead. The closest thing so far I have found to what I am trying to achieve is to hard code a couple of spare extensions into the dialplan, and then have whoever is on-call set one of those extensions to their cell phone number. I'll definitely take another look at followme to see if I can adapt that what I'm trying to achieve. Thanks Danny & Michael, Travis - Original Message - From: "Michael Wyres" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, November 16, 2009 2:23:49 PM Subject: Re: [asterisk-users] Queues Hi Travis, There’s lots of different ways to attack “on-call” roster solutions in Asterisk – as Danny suggested, FollowMe() is definitely an option (and normally the best), but it doesn’t always suit the “business need”. However, also as Danny suggested, in most cases using ASTDB in some way to simplify dialling plans is the way to go - then you just have to decide how you want to update the information as to the number to call, in ASTDB. For example, I had a customer a couple of years back who desperately wanted to manage his “on-call roster” routing using a web interface. I dollied up a simple PHP/MySQL web interface with a list of all the people (and their mobile/cell numbers) in a drop down list – they could simply select the right person, and click a “First Call” button to make that person the first in the roster, select another person and click a “Second Call” button to make that person the second in the roster, and so on. Using the Asterisk manager interface – (or even “asterisk –rx ” if you’re not comfortable using the AMI) – you get the numbers selected into ASTDB. The dialplan just comes along then and reads the appropriate numbers from ASTDB as it steps through, and dials the people in order. As with many things in Asterisk – there is more than one way to “hump the leg”. Cheers Michael From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, 17 November 2009 08:57 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Queues Since followme is “extension-based”, you have at least two options. Option 1 is to have a few extensions designated for “following” where you punch in the cell numbers as you wish. Option 2 is to use “day logic” to point to the “following” guys based on days. If I were doing option 2, I’d try to use ASTDB to control this instead of having to code a lot of dialplan, but that’s just me… From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues I had looked at followme as a solution but ran into the same stumbling block of having to hard code the cell phone list. I didn't see a dynamic way of the list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on Tuesday without editing the extensions.conf file manually each day. Did I overlook something in how followme works? - Original Message - From: "Danny Nicholas" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, November 16, 2009 1:37:04 PM Subject: Re: [asterisk-users] Queues It should be realistic, but have you considered just using followme to add the cell phones to the queue list? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an "on-call" queue. A call comes in and it rings the "on-call" extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the &quo
Re: [asterisk-users] Queues
I had looked at followme as a solution but ran into the same stumbling block of having to hard code the cell phone list. I didn't see a dynamic way of the list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on Tuesday without editing the extensions.conf file manually each day. Did I overlook something in how followme works? - Original Message - From: "Danny Nicholas" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, November 16, 2009 1:37:04 PM Subject: Re: [asterisk-users] Queues It should be realistic, but have you considered just using followme to add the cell phones to the queue list? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an "on-call" queue. A call comes in and it rings the "on-call" extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the "on-call" queue. I've got the queue setup and the people log into and out of it by dialing extensions that use AddQueueMember() and RemoveQueueMember() respectively. I tried using QUEUE_MEMBER_LIST to write to a database list when the call comes in however it keeps adding duplicates each time the call goes into the queue. I'm just not seeing how to pass the call that goes into the queue to a dynamic list on the way out. Is attempting something like this even realistic? Thanks in advance, Travis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queues
Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an "on-call" queue. A call comes in and it rings the "on-call" extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the "on-call" queue. I've got the queue setup and the people log into and out of it by dialing extensions that use AddQueueMember() and RemoveQueueMember() respectively. I tried using QUEUE_MEMBER_LIST to write to a database list when the call comes in however it keeps adding duplicates each time the call goes into the queue. I'm just not seeing how to pass the call that goes into the queue to a dynamic list on the way out. Is attempting something like this even realistic? Thanks in advance, Travis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
