[asterisk-users] 4 Port FXO interface
I am looking to build a small PBX for an office that has 3 incoming analog lines and less than 10 extensions. For the Asterisk server I am going to use a small form factor PC with no-PCI slots so the FXO interface needs to be either FXO-SIP or USB. Can anyone make suggestions? I am looking at an AudioCodes MP114 FXO or possibly two Sangoma U100's but don't have experience with either. = Eric Merkel ejmerkel.li...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pri CLI command not available
I am in the process of trying to terminate a PRI into a new * server. The server has an old T100P T1/PRI card in it. I have compiled the following on Centos 5.4. dahdi-linux-complete-2.2.1+2.2.1 libpri-1.4.10.2 asterisk-1.4.29 Everything seems to have compiled fine. DAHDI reports Found a Wildcard: Digium Wildcard T100P T1/PRI on bootup. Dahdi_tool shows that the span is up and active with no alarms however the phone company is not seeing the trunkgroup going into service. I was wanting to take a look at the PRI debugs but for some reason the CLI pri option is not available. I libpri compiled without any issues prior to compiling asterisk. What would cause the pri debug commands to not be available in the CLI? = Eric Merkel ejmerkel.li...@gmail.com -- _ -- 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] pri CLI command not available
Thanks you were exactly right. I had a problem in my chan_dadhi.conf file. Basically, I had the channels defined before the signaling and it wouldn't load. It did not show any errors that I could see on startup and there were no messages in the /var/log/asterisk/messages but when doing a load chan_dahdi.so from the command line showed me the problem. Thanks again! = Eric Merkel ejmerkel.li...@gmail.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony Francis - Handy Networks LLC Sent: 2010-01-21 15:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pri CLI command not available This is often caused by the dahdi module not loading, check /var/log/asterisk/messages for the reason, or better yet, from the cli load the module manually and see the error in real time. If I had to guess I would say it is a configuration error. Thank you and have a nice day, Anthony Francis From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Merkel (Mail Lists) Sent: Thursday, January 21, 2010 1:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] pri CLI command not available I am in the process of trying to terminate a PRI into a new * server. The server has an old T100P T1/PRI card in it. I have compiled the following on Centos 5.4. dahdi-linux-complete-2.2.1+2.2.1 libpri-1.4.10.2 asterisk-1.4.29 Everything seems to have compiled fine. DAHDI reports Found a Wildcard: Digium Wildcard T100P T1/PRI on bootup. Dahdi_tool shows that the span is up and active with no alarms however the phone company is not seeing the trunkgroup going into service. I was wanting to take a look at the PRI debugs but for some reason the CLI pri option is not available. I libpri compiled without any issues prior to compiling asterisk. What would cause the pri debug commands to not be available in the CLI? = Eric Merkel ejmerkel.li...@gmail.com -- _ -- 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] CallerID ANI issues
Does anyone else have any ideas about this short of changing the voicemail service (can't do that). Most voicemail/answering service dont' care about callerid or ani, they instead use the DID that the call comes in on to decide how to answer the call. Get a different voicemail/answering service. On Tue, Jan 20, 2009 at 9:32 AM, mail-lists mail-li...@peachnet.com wrote: Hello, We're having some issues with CallerID and I thought someone here might be able to shed some light as none of our carriers seem to know what I'm talking about. The issues is this: A client of ours uses an after-hours voicemail service as mandated by their corporate office. We have a Day/Night setting that lets them turn this on and off. A call comes in from one of their customers and gets forwarded back out to the voicemail service with the CallerID set to the clients DID. The voicemail service checks the callerID of the incoming call to determine which agency is calling. The problem is this: The voicemail service (who uses verizon), looks at the ANI field in the CallerID which shows up as something other than our clients DID (notably our BILLING number) We've called two of our carriers (one a SIP provider, the other our PRI provider) and they both say they make no distinction between 'regular' CallerID and the ANI field. The PRI provider said if we have 'station-level' callerID (which we do) the number should show up fine. I've contacted the voicemail service and they say there's nothing they can do on their end. I've played around with setting the ANI fields on our asterisk servers and as far as I can tell the ANI is correctly set to the same as the callerID (Tried Set(CALLERID(all) and Set(CALLERID(ANI) ). Does anyone have any idea what we might look at next to get this resolved? I'm pretty eager to figure this out as we potentially have a dozen clients that are interested in signing with us, provided we have this working. Thanks a lot! Steve ___ -- 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] CallerID ANI issues
Hello, We're having some issues with CallerID and I thought someone here might be able to shed some light as none of our carriers seem to know what I'm talking about. The issues is this: A client of ours uses an after-hours voicemail service as mandated by their corporate office. We have a Day/Night setting that lets them turn this on and off. A call comes in from one of their customers and gets forwarded back out to the voicemail service with the CallerID set to the clients DID. The voicemail service checks the callerID of the incoming call to determine which agency is calling. The problem is this: The voicemail service (who uses verizon), looks at the ANI field in the CallerID which shows up as something other than our clients DID (notably our BILLING number) We've called two of our carriers (one a SIP provider, the other our PRI provider) and they both say they make no distinction between 'regular' CallerID and the ANI field. The PRI provider said if we have 'station-level' callerID (which we do) the number should show up fine. I've contacted the voicemail service and they say there's nothing they can do on their end. I've played around with setting the ANI fields on our asterisk servers and as far as I can tell the ANI is correctly set to the same as the callerID (Tried Set(CALLERID(all) and Set(CALLERID(ANI) ). Does anyone have any idea what we might look at next to get this resolved? I'm pretty eager to figure this out as we potentially have a dozen clients that are interested in signing with us, provided we have this working. Thanks a lot! Steve ___ -- 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] CallerID ANI issues
Not a possibility I'm afraid. Our client is an insurance agent and the voicemail/answering service is mandated by corporate. There also are not various DID's to call in on. All voicemail calls go to an 800 number Thanks for your advice though. Most voicemail/answering service dont' care about callerid or ani, they instead use the DID that the call comes in on to decide how to answer the call. Get a different voicemail/answering service. On Tue, Jan 20, 2009 at 9:32 AM, mail-lists mail-li...@peachnet.com wrote: Hello, We're having some issues with CallerID and I thought someone here might be able to shed some light as none of our carriers seem to know what I'm talking about. The issues is this: A client of ours uses an after-hours voicemail service as mandated by their corporate office. We have a Day/Night setting that lets them turn this on and off. A call comes in from one of their customers and gets forwarded back out to the voicemail service with the CallerID set to the clients DID. The voicemail service checks the callerID of the incoming call to determine which agency is calling. The problem is this: The voicemail service (who uses verizon), looks at the ANI field in the CallerID which shows up as something other than our clients DID (notably our BILLING number) We've called two of our carriers (one a SIP provider, the other our PRI provider) and they both say they make no distinction between 'regular' CallerID and the ANI field. The PRI provider said if we have 'station-level' callerID (which we do) the number should show up fine. I've contacted the voicemail service and they say there's nothing they can do on their end. I've played around with setting the ANI fields on our asterisk servers and as far as I can tell the ANI is correctly set to the same as the callerID (Tried Set(CALLERID(all) and Set(CALLERID(ANI) ). Does anyone have any idea what we might look at next to get this resolved? I'm pretty eager to figure this out as we potentially have a dozen clients that are interested in signing with us, provided we have this working. Thanks a lot! Steve ___ -- 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
Re: [asterisk-users] Asterisk with or without OpenSER
Steve, Hijacking this post here - How 'good' is freeswitch currently. I'm looking for some sort of SIP proxy and have looked into openser and ser. Freeswitch seems to have more functionality than these and it seems a lot easier to configure. I particularly like the xml config files, etc. Our long term goal is to use some sort of SBC for sip registrations, call routing, maybe even basic applications like voicemail and use Asterisk for media gateways, maybe transcoding, etc. Am I completely missing the mark as to whether freeswitch can do this sort of thing or is there a 'better' way to do it. Thanks! Look into FreeSwitch. http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ On Tue, Nov 18, 2008 at 7:29 AM, Yehavi Bourvine [EMAIL PROTECTED] wrote: Hello, I am running a small installation of asterisk and looking for future expansion of it to handle thousands of users. From what I read I see that usually large installation place OpenSER (or similar solution) in front of Asterisk in order to provide high call rate because OpenSER does only signalling while Asterisk does all. My question is: If Asterisk also does only signalling (i.e. the voice traffic goes directly between the phones and not via asterisk) is it still that slow? I preffer to have one software package rather than dealing with two. Thanks! __Yehavi: ___ -- 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] Asterisk with or without OpenSER
Alex Balashov wrote: mail-lists wrote: Steve, Hijacking this post here - How 'good' is freeswitch currently. I'm looking for some sort of SIP proxy and have looked into openser and ser. Freeswitch seems to have more functionality than these and it seems a lot easier to configure. I particularly like the xml config files, etc. What do you mean by functionality? Are you looking for low-level or high-level functionality? I guess I mean FS has more high level functionality like conference rooms and voicemail modules which might allow us to offload some of this from *. OpenSER has some of this as well I think. I'm not sure FS lets you interact directly with the SIP stack like OpenSER/SER does though (I might be completely wrong about this) Also, XML is not a reasonable format for config files. I don't know what sipping-the-property-file-Kool-Aid J2EE droids decided that, but it's made me like UNIX a lot less than I did before now that they're proliferating. I like XML. I know there's a lot of extra grammar but it keeps things straight in my head. I don't have a a great deal of experience with various config options but in the past I've much preferred XML based phone configs to others. To each their own I suppose. ___ -- 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] OT: Asterisk on fitPC
I can't see why not. You should easily have enough power for asterisk. You can probably also run it as your firewall in a home environment thanks to the dual RJ45's I don't know whether or not you can use the built in RJ11 to interface with your POTS line though - maybe someone else could speak to this? Hi, I’d like to install Asterisk at home. But don’t want to use a full blown PC to host it. I was thinking of using fitPC www.fit-pc.com http://www.fit-pc.com to do all the Asterisk work, interfacing with the local Bell Canada line, and using a SIP VoIP line as well. What do you experts think of it? Thanks, Mark. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Asterisk on fitPC
Hm. $300 in the US and the UK disty is selling them for just short of £240, so they can go stuff themselves, low-power or not. (I buy 1GHz systems with 1GB of RAM, running at 15W for half that. No drive though) Gordon, If you don't mind my asking: What do you get for $150.00 ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switch recommendation?
Woah, How weird. I JUST bought this off of ebay 2 minutes ago. The exact one. This will be my first time playing with PoE. I have all cisco phones here but I'll let you know how it goes. This will be my first major asterisk experiment and I'm trying to choose a PoE switch for 15-24 phones. I was going to spend $400 on this: http://www.newegg.com/product/product.asp?item=N82E16833124053 but then I see this on ebay: http://cgi.ebay.com/WS-C3524-PWR-XL-EN-Cisco-3524-24-FE-Switch-W-PoE-VoIP_W0QQitemZ370043264927QQihZ024QQcategoryZ51268QQssPageNameZWDVWQQrdZ1QQcmdZViewItem and I'm thinking, hey, thats a lot cheaper and it is PoE. Will the Cisco IP phone's proprietary wizardry be a problem for my flock on Linksys IP phones? Because as long as it can do vlan qos and poe I think I can scrape by for half the price, right? Thanks for reading! ___ -- 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] VoIP service providers/PSTN termination points
Same here - Gafachi has been great. Decent rates, very stable and great voice quality. I use Gafachi.com http://Gafachi.com and have good quality with no minimum requirements. Try them at www.gafachi.com http://www.gafachi.com On 12/16/07, *Benjamin Jacob* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere http://www.iconnecthere.com http://www.iconnecthere.com Vonage http://www.vonage.com Teliax http://www.teliax.com I found something known as Inphonex http://www.inphonex.com. These had the cheapest rates and quite a good coverage too. Anyone with experience on this one? I am looking at a combination of decent prices and good quality. Any other suggestions or ideas welcome too. TiA - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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
Re: [asterisk-users] Asterisk B2BUA and Site to Site transfers
Chris Bennett wrote: Hi All, I am seeking input from anyone who may have seen a similar configuration and dealt with similar issues to what I'm experiencing. Configuration: - 2 sites (site A and B) - Asterisk 1.2.23 on each site (Trixbox) - Internet 512/512 symmetric at each site, dedicated to VOIP calls only. - IAX trunk between the sites, with data travelling across the 512/512 Symmetric link - PSTN inbound/outbound via a Sangoma PCI FXO card. The required configuration is inbound calls at either site need to be answered by a reception at Site A. Calls coming in via PSTN to Site B, will result in a SIP extension at Site A to be dialled and answered. This will result in an active channel between site B's asterisk server, and the user at Site A. If Site A transfers that call *back* to site B, this will result in another call leg being established to the user at site B. Every RTP packet will travel: - in via PSTN @ Site B - across 512/512 DSL link to Site A's asterisk server - back across 512/512 DSL link to user at Site B We are noticing jitter and voice quality problems. A call can degrade in quality over time. We are using G729 for the voice codec. Can anyone suggest further debugging I can do to determine the cause of voice quality degradation? Is there a way I can configure the asterisk servers to not communicate the RTP traffic across the DSL links and back again? Any suggestions will be much appreciated. I'm don't think setting reinvites on will fix your problem. The only thing I can think of is that you use some sort of call parking to park the call on SiteB's asterisk server and then have the person at siteB pick up the call from the parking lot Anyone else know a better way to do this? ___ --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] Asterisk SIP Microsoft Outlook Integration
Does anyone know how I could integrate my Asterisk setup with Outlook so that when I click on a phone number is my outlook address book it will dial the number and ring my SIP phone so that I can just pick it up? I am interested in this integration for WinXP with Outlook 2003 and WInVista with Outlook 2007. On a related note... Is there anything out there that integrates with thunderbird??? I've looked many times but the only thing I've found are this: http://www.maxcole.com/tbird-asterisk.html and this: http://cockatoo.mozdev.org/ Both of which don't work with current versions of Tbird. There's also this: http://labs.abbeyphone.com/tools/Thunderbird_Voip.html But that only works through THEIR SIP proxy - which is useless. I wonder if anyone has anything that actually works - Seems like this would be a killer plugin for Thunderbird. Thanks ___ --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] Cisco 7911/7941/7970/7971 Softkey XML Files
Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn’t find much up there. As far as I know (and I might be very wrong), you can't change the soft key configuration of Cisco phones with the SIP Firmware. Maybe you can with Cisco's CallManager - I don't know. Someone PLEASE correct me if I'm wrong because I've been wanting to do this for a year ___ --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] Outgoing PRI CID?
Turbo Fredriksson wrote: We have now got our new PRI line (10 channels, 100 numbers) connected and everything is working except the outgoing caller ID. Whatever SIP phone I'm using, the CID that's shown is the very first number... I don't know if the same is true for you but we had to call our telco and have them set our callerid settings to 'station level'. Not sure if your telco offers this but they should. ___ --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] shared system - how to monitor channels
I was wondering how everyone here is giving users (say via the BLF on a Polycom, or the sidecart/buddies) the ability to see how many channels they have in their group and how many are in use. Since so many users are used to seeing Line 1, 2, 3 etc on a key system I have been trying to think about how to show channels as a buddy (ie hint). Any suggestions? Bill ___ --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] Strange Call problems on some numbers
I am having some really strange problems calling from 2 asterisk boxes of mine. One is version 1.2.22 the other 1.2.18. The problem is identical on both boxes. When I try to call certain numbers (8006375410, for instance) the call rings and rings and rings. Eventually the receiving end will pick up in the middle of an IVR as if I had been connected for some time already. When I call this number from a cell phone it connects normally. When I call this number from a sip phone connected to an asterisk box running 1.4 and out through a pri it connects normally. When I call it from these other two asterisk boxes out through whatever provider(my pri box, voip providers, whatever) it exhibits this behaviour.. Does anyone know what might be causing this? It's turning int a significant problem. Thanks Steve Glaus Peachnet Communications. ___ --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] Which Asterisk version to use?
Razza wrote: On 27/09/2007, Eric B. [EMAIL PROTECTED] wrote: For starters, what is the difference btwn the 1.2 and 1.4 branches of Asterisk? I can't seem to find a document that describes the changes. Anyone? Not much/Lots Depends what you're looking for. Important considerations for us in moving to 1.4 were: jabber/gtalk support t.38 passthrough support shared line appearance support You can probably have a look at the Changelogs for more details. If you don't need the extra features 1.2.Current is still the most stable solution IMO. ___ --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] asterisk cli - vi keybindings ?
Tzafrir Cohen wrote: On Mon, Sep 24, 2007 at 02:04:05AM +0100, Ex Vito wrote: This might sound lika a small issu, but here it goes: I'm a long time unix user and my shell history usage and editing is configured to use vi keybindings; it's something that's already built into my fingers and using different bindings, like the arrow keys to fetch previous lines, really blows me !... :-( Is there any way to setup the asterisk cli to use such keybindings ? I took a quick glance at 1.4.11 source and found readline.[ch] files, but asterisk is not behaving to my inputrc configuration... Googled for a while to no effect. Set in your environment: AST_EDITOR=vi AWESOME! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] canreinvite
How can I know that the traffic went directly between the endpoints and did not go via the asterisk? I'm sure there are many ways to do this one way would be to do rtp debug on the cli and watch for media packets another would be to do tcpdump on the command line and watch for packets there. Regards Bilal Ghayad Mobile: 009659849460 - By default assuming you have no global setting otherwise, if asterisk doesnt see a need to stay in the path then it wont. hence if it has to transcode between different codecs, capture DTMF or different protocols it will stay in the path. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send traffic to each other directly or it will be via Asterisk? Regards Bilal Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] PRI Card
Hello, We're in the process of moving to a PRI circuit for our asterisk switch. Can anyone point me in the right direction as far as PRI Cards are concerned? Thanks! ___ --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] PRI Card
Jared Smith wrote: On Thu, 2007-07-19 at 11:28 -0400, mail-lists wrote: We're in the process of moving to a PRI circuit for our asterisk switch. Can anyone point me in the right direction as far as PRI Cards are concerned? You haven't given us many details on your setup, but I'll take a stab at answering your question anyway. For a single-port PRI card, I recommend the Digium TE120P card[1]. It can be configured for either T1 (United States and Canada), E1 (Europe, South America, and most of the rest of the world), or J1 (Japan). It will work with both channelized T1s as well as PRI circuits. This is a PCI card, and will work in either a 3.3 volt or 5 volt PCI slot. [1] http://www.digium.com/en/products/hardware/te120p.php Jared, thanks for your reply - Our setup isn't complicated at all - just a PRI coming into an asterisk box. Maybe you could answer another question for me - what disadvantages does a PRI have from a channelized T1? or vice versa. is the TE205 the double port version of the TE120P? Also, what's required as far as echo cancellation goes? Is that built into these cards or do you have to move up to a TE207P? What is the difference between a TE205 and a TE210? Sorry about all the qeustions - the info on digiums web site doesn't really make this clear. ___ --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] PRI Card
Jared Smith wrote: On Thu, 2007-07-19 at 12:11 -0400, mail-lists wrote: thanks for your reply - Our setup isn't complicated at all - just a PRI coming into an asterisk box. Maybe you could answer another question for me - what disadvantages does a PRI have from a channelized T1? or vice versa. A channelized T1 is 24 channels over a T1 -- basically little more than 24 POTS lines that happen to come across a 2-pair digital interface rather than 24 pair analog interface. PRI over T1, on the other hand, only gives you 23 voice channels (often call B or Bearer channels), and uses the 24th channel for call signalling between you and the upstream switch. (This channel is called a D or Delta channel and I've even heard it called a Data channel, although my friends in big telco say that the D doesn't stand for Data.) PRI gives you much more advanced call control, allows you to more easily do things like DIDs, and gives much quicker call setup and dialing. Depending on your location, the only major downside to PRI might be price... they're often more expensive than channelized T1s. That's what I pretty much thought - We're about to sign a two year agreement so I wanted to make double sure a PRI was the best route. is the TE205 the double port version of the TE120P? Yes, the TE205P is the 2-port version of the TE120P. Once you get to the 2-port versions and 4-port versions of the Digium cards, they come in three flavors: a card for 5-volt PCI slots (the TE205P, for example), a card for 3.3-volt PCI slots (the TE210P, for example), and a card for a PCI-Express slot (the TE220P). Also, what's required as far as echo cancellation goes? Is that built into these cards or do you have to move up to a TE207P? It depends... if you're OK with software echo cancellation, you don't need anything special. If you want the hardware echo cancellation (which many people do), you could move to the TE207P or TE212P cards (for PCI slots in 5v and 3.3v, respectively), or the TE220P plus a VPMOCT064 echo cancellation module. I'm going to look into what sort of overhead software echo canceling incurs and go from there. Is it possible to get a TE205 and then add the echo cancellation separately or are they sold as a unit? I don't see any info about the TE220 and a google search for vpmocto64 didn't turn up much. What is the difference between a TE205 and a TE210? Explained above... difference between the different types of PCI slots. Sorry about all the qeustions - the info on digiums web site doesn't really make this clear. I agree. Luckily, I know Digium's marketing department is working to improve the information on the website so that it's clearer which hardware is appropriate for different situations. Thanks for all the advice/answers. ___ --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] PRI and Local numbers
Hello, We're running into a problem I thought some of the enlightened people on this list might be able to help with. Our VoIP stuff has grown to the point where it makes sense to get a PRI (we've been doing things purely voip till now). The problem we're running into is this: We have several offices scattered across the SE states. We would like to be able to get local numbers for each of these offices that ring into the PRI we're about to get installed. I've spoken with several carriers and this is something that most of them do one way or another but it costs an arm and a leg - 5.00 per number plus 2.5 cents a minute. Does anyone know of a product offering that encompasses this? Can anyone recommend a way to get this accomplished cheaply and cleanly? What do VoIP providers do to get phone numbers in every rate center across the US? Thanks for any advice, Steve. ___ --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] Polycom multiple registrations
Noah Miller wrote: The 430's have two line appearances. I'm trying to get the second line registered to a different extension but for some reason it's not allowing me to do this. The first line will register fine but the second line never seems to register no matter how I swap the device ID's and permissions around. I've played with the lineKeys and callsperlinekey settings to no avail. For what you want to do, you'll have to set lineKeys to 1 for both of your registrations. callsPerLineKey can be anything from 1 to up to (I think) 6, your preference. Can you share the reg ... / statement from your phone.cfg file? Also your sip.conf? - Noah ___ --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 Noah, Thanks for your reply. It was just me not being careful. When I looked at the reg.x.server.y.address setting I noticed I had server.2 instead of server.1. Thanks for your help. ___ --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] Polycom multiple registrations
Hello all, I have some polycom 430's which I'm trying to get to work with asterisk. I have them working for the most port other than one little issue. The 430's have two line appearances. I'm trying to get the second line registered to a different extension but for some reason it's not allowing me to do this. The first line will register fine but the second line never seems to register no matter how I swap the device ID's and permissions around. I've played with the lineKeys and callsperlinekey settings to no avail. Has anyone run into this problem before? Thanks ___ --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] Re: Slightly OT:CSU on Digium cards, and it's requirement
I probably shouldn't be hijacking this thread but it seems that there's some people paying attention here that know what they're talking about. We've recently acquired a cisco IAD 2400 router with 2MFT-T1 VWIC card in it. Doing some cursory reading It seems that this card can be interfaced with a PRI. (I really DON'T know what I'm talking about here so my terminology might be all wrong). My question is this: If we want to get an analog trunk into the building and interface that trunk with the 2MFT card, can we then use asterisk to receive/send calls over this cisco router? How would this be accomplished? Do the cisco routers take calls via SIP or is there some other mechanism to pass calls off through this card. The reason I ask is that my boss is a cheap bastard and wants to avoid spending the $'s on a digium card if possible. Thanks, Steve. ___ --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] How to tell what codec is used for each end of a call MD110-H323-SIP
[EMAIL PROTECTED] wrote: Hi. Calling from Ericsson MD110 via H.323 trunk to an asterisk 1.4.4 I get the call established but no sound heard on either end. What is the best/correct way to try and see what codecs Asterisk is using on each end of the call as it passes through Asterisk for SIP I use 'sip show channels' I'm not sure what the equivilent h323 command is. And is there any way to see that voice is in fact being passed through Asterisk during the call (some counters etc.)? Try 'rtp debug' and the rtp packets should scroll by. Thank you for your time and effort to respond. ___ --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] click to call
Anton Krall wrote: I have been looking around for examples or code on making a click to call application for web sites... has anybody had any luck on this topic? Is there any open source code out ther that could do this? What we have done in the past is created url's like this : sip:4044565941. Xlite will register itself as the sip handler on your system. If you want a generic click to call (ability to call numbers on any given website) check out moziax ___ --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] Bottom line on fax reception
shadowym wrote: So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. One simple question - VOIP or PSTN? ___ --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 Problems continue...
Ken, I have similar problems every now and then on one of my asterisk boxes. I'm also running CentOS4 on that box. I've found that doing a sip reload when in that state results in something along : Last reload not yet finished (can't remember the exact wording) We're using cisco 7960's here. The ONLY time I've seen this happening is when I reload everything VIA freepbx. It used to do it every time I reloaded. I read somewhere that this was a result of DNS queries not being done in a timely fashion - So I went and replaced all the host statement in my trunk with IP addresses and now it doesn't do it very often at all. I don't know if this is your problem at all but it might be worth a shot. Replace any host names with IP addresses in sip.conf and anywhere else. Failing that and if you're still pulling your hair out at the end of the week ( I know how it is), I would really consider re-installing the box (I'm using centos5 now on this server I'm configuring currently) and starting from scratch. I know it sounds like a cop out but that's what I would do. ___ --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] app_txfax, app_rxfax
ax. The downside of rx_fax is that you need to compile it into asterisk. The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. The last fax setup I did was for a small 2-person office where they had an existing fax machine that answered, listened for the remote fax squawk, if it didn't get it, then it rung the phones daisy-chained to it, and if they didn't answer it went to answering machine. I implemented this in asterisk fairly easilly with rx_fax. I'm not sure if you can do that with iaxmodem. Another question along these lines : How does everyone one fax detection on a sip channel? The only thing I've found is NvFaxDetect - anyone know of anything else? thanks ___ --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] app_txfax, app_rxfax
The downside of rx_fax is that you need to compile it into asterisk. The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. The last fax setup I did was for a small 2-person office where they had an existing fax machine that answered, listened for the remote fax squawk, if it didn't get it, then it rung the phones daisy-chained to it, and if they didn't answer it went to answering machine. I implemented this in asterisk fairly easilly with rx_fax. I'm not sure if you can do that with iaxmodem. Another question along these lines : How does everyone one fax detection on a sip channel? The only thing I've found is NvFaxDetect - anyone know of anything else? thanks ___ --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] zaptel compile error
Forrest Beck wrote: The problem is that your kernel is newer than the xbus-core.c file is looking for. See: http://forums.digium.com/viewtopic.php?t=15317sid=7beaf6bfed1550f4a8676427283800c5 I just did a make menuselect and eliminated the xpp module. It is for USB Astribank, something I will never use. On 5/4/07, mail-lists [EMAIL PROTECTED] wrote: I get the following error when trying to compile zaptel on CentOS 5 kernel 2.6.18-8.1.3.el5 CC [M] /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c: In function â /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c:171: error: â has no member named â make[3]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o] Error 1 make[2]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp] Error 2 make[1]: *** [_module_/root/asterisk-src/zaptel-1.2.17.1] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-8.1.3.el5-i686' make: *** [all] Error 2 I'm kind of at my wits end with this - been trying for several hours.. Thanks! ___ --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 That's what I was going to do Do you know how to eliminate the xpp model on a non 1.4 zaptel build? ___ --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] zaptel compile error
Forrest Beck wrote: The problem is that your kernel is newer than the xbus-core.c file is looking for. See: http://forums.digium.com/viewtopic.php?t=15317sid=7beaf6bfed1550f4a8676427283800c5 thank you for pointing me in the right direction with this - the answer is write there in xbus-core.c: -CODE /* * As part of the inode diet the private data member of struct inode * has changed in 2.6.19. However, Fedore Core 6 adopted this change * a bit earlier (2.6.18). If you use such a kernel, Change the * following test from 2,6,19 to 2,6,18. */ #if LINUX_VERSION_CODE KERNEL_VERSION(2,6,19) #define I_PRIVATE(inode)((inode)-u.generic_ip) #else #define I_PRIVATE(inode)((inode)-i_private) #endif END CODE--- So change KERNEL_VERSION(2,6,19) to KERNEL_VERSION(2,6,18), And away it goes!! Thanks! ___ --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] zaptel compile error
Forrest Beck wrote: The problem is that your kernel is newer than the xbus-core.c file is looking for. See: http://forums.digium.com/viewtopic.php?t=15317sid=7beaf6bfed1550f4a8676427283800c5 thank you for pointing me in the right direction with this - the answer is write there in xbus-core.c: -CODE /* * As part of the inode diet the private data member of struct inode * has changed in 2.6.19. However, Fedore Core 6 adopted this change * a bit earlier (2.6.18). If you use such a kernel, Change the * following test from 2,6,19 to 2,6,18. */ #if LINUX_VERSION_CODE KERNEL_VERSION(2,6,19) #define I_PRIVATE(inode)((inode)-u.generic_ip) #else #define I_PRIVATE(inode)((inode)-i_private) #endif END CODE--- So change KERNEL_VERSION(2,6,19) to KERNEL_VERSION(2,6,18), And away it goes!! Thanks! ___ --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] app_txfax, app_rxfax
Hello, I'm trying to compile asterisk from source (1.2.18). Faxing is fairly critical for us, so in the past we've used spandsps app_rxfax and app_txfax to support faxing in asterisk. Unfortunately I can't find these applications on soft-switch.org anymore and even so I have a feeling they wouldn't patch correctly into 1.2.18. Does anyone know how to best handle faxing in 1.2.18? Is it even necessary to compile these two apps into asterisk? what about spandsp? Any advice would be appreciated, Thanks! Steve ___ --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] app_txfax, app_rxfax
Stefan Wintermeyer wrote: Steve, Am 04.05.2007 um 14:44 schrieb mail-lists: I'm trying to compile asterisk from source (1.2.18). Faxing is fairly critical for us, so in the past we've used spandsps app_rxfax and app_txfax to support faxing in asterisk. Unfortunately I can't find these applications on soft-switch.org anymore Forget them! Use Hylafax and iaxmodem instead. Does anyone know how to best handle faxing in 1.2.18? Is it even necessary to compile these two apps into asterisk? what about spandsp? Any advice would be appreciated, I only have a German howto available. But you should get the idea: http://www.das-asterisk-buch.de/stable/installation-iaxmodem.html http://www.das-asterisk-buch.de/stable/installation-hylafax.html Stefan -- Stefan, My name is spelled Stefan too :) I AM using hylafax/iaxmodem on my production boxes. I guess I'm not entirely clear on WHAT app_rxfax, app_txfax do. Are iaxmodem/hylafax essentially a replacement for these asterisk internal applications? Also, In the past I've installed Trixbox/FreePbx. It uses NvFaxDetect to detect incoming faxes - is this an application I need to build for asterisk? Thanks! ___ --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] zaptel compile error
I get the following error when trying to compile zaptel on CentOS 5 kernel 2.6.18-8.1.3.el5 CC [M] /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c: In function â /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c:171: error: â has no member named â make[3]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o] Error 1 make[2]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp] Error 2 make[1]: *** [_module_/root/asterisk-src/zaptel-1.2.17.1] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-8.1.3.el5-i686' make: *** [all] Error 2 I'm kind of at my wits end with this - been trying for several hours.. Thanks! ___ --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] Re: Problem converting a Cisco 7960 to SIP
Salvatore Giudice wrote: You can go directly to 5.2 and then move on to 7.x and 8.6. 5.2 allows you to upgrade to the newer firmware releases that have an app loader, which Cisco added in later releases. Beware that some cisco non-sip loads can not generate the proper firmware filename to download from tftp when they read the version numbers from the version text. I always go directly to 7.2 and it works fine for me. If anyone needs the 7.2 firmware, let me know off list. ___ --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] asterisk on mini-itx
Boot it off flash and have it load an initrd.gz into RAM. Everything will run entirely from RAM - no writes to the flash at all! I can get everything inside a 48MB flash drive, but I use 64MB ones which gives me space to store configs, etc.. (of-course, I make it sound so simple ;-) but I'd already worked this out some years back for a diskless router project) I'm guessing you don't have any sort of graphical UI? I was hoping to run freepbx in some way - probably have the mysql database stuff stored somewhere else.. In the 64MB flash devices I use, I can squeeze in a fairly full featured Linux system (and I've not even bothered with uClib or busybox yet) including apache and php. I don't use mysql, and I removed PERL. My web based GUI stores everything in flat text files, and generates the relevant /etc/asterisk files from these text files. It's very fast and without MySQL is one less thing to go wrong. (And I suspect my aplication is actually lot faster than going through a MySQL layer too!) I did test the speed of this by generating config files with 1000 users and there wasn't an issue handling it. (other than having a list of 1000 names on screen which wasn't helpful!) Is your GUI something you wrote yourself or something that's commercially available? I'm using freePBX on all of my installs and while it lets you do almost everything from the interface I've come to find it's not very user-friendly for novices not to mention having to have mysql as a back-end. I've been looking for a leaner - more simplistic GUI but haven't really come across anything. Maybe Digiums own GUI will meet the needs for this at some point.. ___ --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] asterisk on mini-itx
Hello, I'm trying to put together a low cost - low powers PBX appliance for several customers. I have purchased a couple of the soekris net4801 boards and have asterisk up and running on them fine but they just don't quite cut it in the processing power department. I've been able to get about 10 simultaneous SIP calls with simple ulaw (no encoding decoding). While this might be OK for a very small business or home I just don't think it leaves a lot of overhead to do anything else. I've had a look around and I think I have settled on one of the VIA EPIA fanless boards. Does anyone have any experience with these running asterisk as far as performance and reliability is concerned? Has anyone run asterisk with any compressed codecs on this setup? I am going to TRY to run the system from flash memory one way or another - I realize the hoops I might have to jump through to prevent a large number of read/write cycles but I'd really like to have the whole thing solid state... Maybe someone has a better idea regarding program storage? Also, I would really like to run this as a router/firewall appliance as well so that that the box can sit on a public IP if the client only has one. For this reason I kind of have my heart set on openbsd. The routing and firewall utilities on openbsd are very simple to configure and easy to use. Does anyone know what limitations asterisk might have on openbsd (besides lack of zaptel.. ) ? I have run asterisk 1.2.? on openbsd before and found it worked pretty well. Failing that I suppose I would settle for running the routing/firewalling on linux. I've just found the linux networking tools very awkward up until now - perhaps someone know of a linux distribution - or tool - that makes routing/firewall/NAT as painless as on openbsd? Maybe I just need to sit down for a day and learn the tool properly ;) Anyways, I know there are a lot of questions in here but perhaps someone has done one or all of these things? Thanks for any advice or warnings! Steve Glaus ___ --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] asterisk on mini-itx
Hey these look pretty good - I was going to build my own but if they're ~$350 like you say it's probably not worth it. I haven't played around with Astlinux at all. I'm assuming it doesn't install freepbx does it? I don't really need (or want) the gui - but clients will. I'm assuming all your calls are pure VOIP? ___ --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] asterisk on mini-itx
On 3/10/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Sat, 10 Mar 2007, Mail Lists wrote: I've had a look around and I think I have settled on one of the VIA EPIA fanless boards. Does anyone have any experience with these running asterisk as far as performance and reliability is concerned? Has anyone run asterisk with any compressed codecs on this setup? I've built several systems based on this motherboard (the 1GHz fanless one) Compressed codecs are fine - as long as you aren't transcoding ;-) I figured I could push 30 non transcoded calls through one, but I've never had the ability to fully test it out. The max. I had going on one system was 20 calls. I probably will be doing transcoding phone(ulaw)-PBX(gsm)-VTSP At least in some circumstances. Boot it off flash and have it load an initrd.gz into RAM. Everything will run entirely from RAM - no writes to the flash at all! I can get everything inside a 48MB flash drive, but I use 64MB ones which gives me space to store configs, etc.. (of-course, I make it sound so simple ;-) but I'd already worked this out some years back for a diskless router project) I'm guessing you don't have any sort of graphical UI? I was hoping to run freepbx in some way - probably have the mysql database stuff stored somewhere else.. I keep voicemail on a 2nd flash IDE device mounted as ext2 (not 3 as ext3 writes regularly!) and force the fsck at boot time if it's dirty - I'd rather lose all voicemail than have it dump itself into single user mode waiting for keyboard input... (your thoughts here might be different :) Have you ever burnt out a flash drive from voicemail usage alone? Also, I would really like to run this as a router/firewall appliance as well so that that the box can sit on a public IP if the client only has one. For this reason I kind of have my heart set on openbsd. The routing and firewall utilities on openbsd are very simple to configure and easy to use. Does anyone know what limitations asterisk might have on openbsd (besides lack of zaptel.. ) ? I have run asterisk 1.2.? on openbsd before and found it worked pretty well. I run similar motherboards as routers, booting off flash too. Also running Linux, but then I find the Linux firewall an easy thing to work with for most simple cases. Watch your interrupts - especially if you're plugging in a 2nd Ethernet card and a TDM card. The VIA motherboard which has 2 Ethernet ports has a processor with only 64MB of cache ram. The ones I'm using have 128KB cache. I don't think TDM is even a consideration - at least not right now. Do the boards you use have 2 PCI slots?? Drop me an email and I'll send you a simple shell script to setup a basic firewall, do nat, etc. I'd probably not recomend running the router/firewall on the same box as asterisk though... That'd be great thanks! Why would you not do that? security? resources? Single point of failure? Thanks a lot for all your advice - its nice to know that this sort of setup is working for people. Up till now I've only run asterisk on IBM eservers with redundant everything - which works well - but for most small-medium size clients it's definitely overkill and not very elegant. ___ --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] Newbie Question
[test] disallow=all allow=gsm ;GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw Are you sure that the xlite phone can handle gsm?? ___ --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] Newbie Planning Help
a) to what extent Asterisk can manage everything necessary to allow machines A and B to communicate if they were SIP phones. Is it possible to go for a setup with the firewalls/NAT devices as shown If the asterisk machine isn't NATed you shouldn't have a problem at all. If you're using SIP clients just make sure nat=yes is set in each of the client definitions in sip.conf b) if I go with IAX softphones, does communication between A and B have to go through S, or can Asterisk hand-off the IAX conversation so that A and B talk directly. I'm not sure in this case since both clients are going to be NATed. I'm pretty sure that this wouldn't work with SIP clients. Since IAX has less problems with NAT traversal it might work fine - try setting canreinvite=yes in your iax.conf and monitor rtp traffic at the asterisk CLI c) the example documentation shows seperate entries in iax.conf for incoming and outgoing calls. In my case (assuming IAX softphones) would I just have entries for A and B of type friend? Can someone give me some advice about how to proceed. type=friend works for me... If you decide to use iax check out moziax - a firefox plugin iax client that's simple to set up. ___ --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] Kernel and zaptel versions
If your connections are VoIP, the first area to look at for quality is network jitter/congestion/drops. I'm mostly worried about drops. A little bit of garbling I can deal with but a dropped call is just VERY bad. Especially when it happens again and again. Does anyone know any methods for tracing dropped calls? All I see is a normal hangup in the logs. The dropped calls seem VERY random and happen regardless of VSP. All I can determine is that it's asterisk that's at fault but I really have no justification for it. All of our calls are VOIP only. ___ --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] Kernel and zaptel versions
Hello, Can anyone recommend the 'best' kernel and zaptel versions to use with asterisk? we're currently running trixbox and are having numerous call quality issues(disconnects, echo, garbled speech) and I'm considering wiping the asterisk box and installing a virgin copy of centos, compiling asterisk myself and installing freepbx on it's own.. Is there anyone who can recommend specific software versions that have been proven to be stable and reliable? We're running trixbox 1.2.3 in case anyone has similar issues that they might know how to solve Thanks! ___ --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] Long call setup times on SIP to zaptel calls
Yes, But try calling your cell from one of your phones - Does the cell start ringing the moment you hear ringing on the SIP phone? or does it ring a half ring/full ring later? I'm just curious because I do the same thing. (Very strict pattern match) but I don NOT have the 'r' option set in the Dial command. This way, I don't here a ring on the SIP phone side until the phone at the other end actually starts to ring. It's confused some people who sit there waiting and wondering - but I like it so too bad :) All of our SIP phones dial instantly when the users finished dialing. We can do this because we have no ambiguous extension lengths. i.e. no _XXX and _ and we don't use the . pattern match. Shane Spencer wrote: I only say this because nobody in our office knew how to use the checkmark on snom phones to initiate a call, they always just waited for the phone to initiate the call for them :) On 2/15/07, Shane Spencer [EMAIL PROTECTED] wrote: do your sip phones dial after a timeout? If the timeout is set to around 5 seconds you may have a dialplan issue on your sip hardware. ___ --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
[asterisk-users] Dropped calls
I've been getting a number of dropped calls today with the following visible in the logs: Feb 5 12:06:29 DEBUG[8340] channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/peachnet-213243-b7830fd8, c1=SIP/2142-08f4b6a0, flags: No,No,No,Yes Feb 5 12:06:29 DEBUG[8340] channel.c: Bridge stops bridging channels SIP/peachnet-213243-b7830fd8 and SIP/2142-08f4b6a0 Feb 5 12:06:29 DEBUG[8340] res_features.c: Timed out for feature! Feb 5 12:06:29 WARNING[8340] res_features.c: Bridge failed on channels SIP/peachnet-213243-b7830fd8 and SIP/2142-08f4b6a0 Feb 5 12:06:29 DEBUG[8340] chan_sip.c: update_call_counter(2142) - decrement call limit counter Can anyone point out what might be calling this sort of behaviour? ___ --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] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)
Mark Coccimiglio wrote: Marty, Where are you paying $1000 for a 1600 series Cisco? I can get you 20% off that price on any quantity (note: Sarcasam). Its not the 1990's anymore. You can get them on eBay ($50-150) for only slightly more then the Linksys. The performance is rock solid. Three-quarters of the world have used them for decades. I know of units running 2 and 3 YEARS between reboots. The power company reboots my equipment more then I do. Ok it is true that Cisco does not support the models anymore, but you can't buy a services contract for a linksys router either. It can sometimes be a little difficult to configure without any technical knowledge but that is what most of us get paid for. It does impress the customer when you bring in the grey box labled Cisco. As for performance just try to put 50 people behind a linksys/netgear/dlink. I've used 1605R supporting +100 users. Not even a blink. Finally, untill everyone is using 10Mps FTTH the broad band link is still the slowest part of the connection. Not to shabby for antiquated technology. Mark C Martin Joseph wrote: On 2007-01-06 00:48:11 -0800, Mark Coccimiglio [EMAIL PROTECTED] said: Mike I'm using a Cisco 1605R [running IOS 12.3(5a)] small office router with Fair-Weight queueing enabled. Works great. The nice thing about Fair-Weight queueing is that it dynamically adapts to lower the priority of higher demand traffic (e.g. large downloads). If you want quality stick with quality stuff. Mark C Reread the subject line please. $1000 (US) isn't inexpensive by any stretch. Marty ___ --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 Mark, Do these 1600 series Cisco routers you mention that you find on eBay for $50-$150 support Layer3 routing? I have a managed switch setup on my home network with several VLANs defined. (work subnet, home subnet, VOIP subnet) I currently have to use a Linux box to route between the VLANs. I'd like to move to Gigabit routing, but I'd need to replace the Linux box(more processor power and new NICs) and that gets expensive. I'd much rather have a router or smart switch for that matter that does Gigabit Layer3 routing all in one unit. Do you have any recommendationsthat wouldn't break the bank? Thanks, Ed ___ --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] (OT) HylaFAX, IAXModem, Asterisk
Steve Totaro wrote: Doug Lytle wrote: Steve Totaro wrote: Steve, You neglet to mention: Distro Version of HylaFAX Version of iaxmodem Version of Asterisk How you're connecting to the PSTN (From previous conversations, I'm guessing PRI) I can't say that I'm not experiencing the same issue as you, 99% of our faxes are incoming. I'm running 23 iaxmodems along with HylaFAX 4.3.0.12 on a PRI and Asterisk 1.2.12.1, running on Mandriva 2006. My setting below: [iaxmodem] device /dev/ttyIAX01 port4599 refresh 60 server 127.0.0.1 record peernameiaxmodem.com01 secret 12345 cidname WhereIWork cidnumber 269xxx codec slinear [Asterisk] [iaxmodem.com01] ; Software modem COM01 type=friend host=dynamic trunk=no allowcallerid=yes disallow=all allow=slinear secret=12345 qualify=no trunk=no context=sip [HylaFAX] ModemType: Class1 # use this to supply a hint ModemSetOriginCmd: AT+VSID=%s,%d Class1AdaptRecvCmd: AT+FAR=1 Class1TMConnectDelay: 400 # counteract quick CONNECT response Class1RMQueryCmd: !24,48,72,96 # enable this to disable V.17 ModemResetCmds: AT+VCID=1 # enables CallID display I am running on CentOS 4.4, Asterisk 1.2.10, hylafax-4.3.0-2, iaxmodem-0.1.10. Your config.tty files are much shorter than mine. I think I used the addfax script instead of copying the sample from iaxmodem. I guess it is time to upgrade a few components and try again. Thanks, Steve ___ --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 The place to start would be upgrading your iaxmodem to at least 0.1.14. There are significant improvements. ___ --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] How to get CDR to show answered calls only
shadowym wrote: Is there anyway to get CDR to show just the answered calls. Not by exporting to a spreadsheet and editing. We have ring groups and queues and CDR shows everything as calls received. Even if it's multiple extensions ringing it shows them as multiple calls received. This seems kind of goofy. ___ --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 What I've come to realize is that CDR in asterisk is awful. We're looking at doing our own 'CDR' via the userfield ___ --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] DTMF Corruption Problem
Also, I am not using a zaptel timer. Could this possibly be causing problems with DTMF?? I really don't know for certain but here's what I experienced: When calling out asterisk gives the option to allow called numbers to transfer by hitting the '#' by putting 'T' (or 't'?) as an option in th dial string. This would very seldom work. I could hit the '#' on the called phone it would say 'extension' but would always reply with 'not valid extension' I recently upgraded to 1.2.12 and noticed that there was no ztdummy running! I compiled my own zaptel installed it, loaded the modules on boot and now the transfer works perfectly. Also: my moh wasn't working for some reason. After I installed the ztdummy module it works too.. I'm not sure whether the transfer issue was fixed by using the ztdummy module or by the asterisk issue but my point is that you should always have the ztdummy module installed if possible. Just my .02. Hope it helps ___ --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] DTMF Corruption Problem
Erick Perez wrote: I can report that with asterisk 1.2.13, internal SIP calls work perfectly but (in my particular case) my asterisk box cannot recognize DTMF digits when it receives a call via our SIP provider. we are both using rfc2833 and I have tried relaxdtmf=yes/no when i use an internal sip extension and call somebody outside via my sip provider, dtmf is recognized. On 11/9/06, mail-lists [EMAIL PROTECTED] wrote: Also, I am not using a zaptel timer. Could this possibly be causing problems with DTMF?? I really don't know for certain but here's what I experienced: When calling out asterisk gives the option to allow called numbers to transfer by hitting the '#' by putting 'T' (or 't'?) as an option in th dial string. This would very seldom work. I could hit the '#' on the called phone it would say 'extension' but would always reply with 'not valid extension' I recently upgraded to 1.2.12 and noticed that there was no ztdummy running! I compiled my own zaptel installed it, loaded the modules on boot and now the transfer works perfectly. Also: my moh wasn't working for some reason. After I installed the ztdummy module it works too.. I'm not sure whether the transfer issue was fixed by using the ztdummy module or by the asterisk issue but my point is that you should always have the ztdummy module installed if possible. Just my .02. Hope it helps ___ --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 Erick, Do you have ztdummy running? What SIP provider are you using. Incoming calls work fine for me (and always have as far as I know). ___ --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] Voxee lag problems ?
Vicky wrote: Anyone having problems with voxee since last few days or is it just me ? In peek hours i get LAGGED when i do a iax2 show peers or even 1000 ms latency . Most of time it is 20 ms or so but when i start sending traffic to them latency increases to 1000 ms or even LAGGED ( also shows high in peak time even when no high latency ). No problems with any other provider . Anyone else having same problem ? ___ --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 Yes.. not just the last few days either... the last few weeks I havent' bothered to write voxee about it as their support sucks horribly and it takes about a week most times for them to get back to you. I have a voxee trunks on 2 seperate boxes and both do the same ___ --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] VOIP Bandwidth questions
Hello everyone, This probably isn't the correct place to ask this but I thought I'd check here first. We're getting ready to roll out a hosted pbx solution on a very limited trial basis (some company employees are going to get voip service at home). Our main issue is of course bandwidth. We have enough bandwidth (spread across two locations) to accommodate the few employees (around 10) for the near future but we're worried about how this is going to scale. Obviously at some point we'll need to consider 'real' bandwidth. My question is this: How do huge voip companies like vonage handle bandwidth. I'm pretty sure that they have to have sufficient bandwidth available for X numbers of simultaneous calls, in other words ALL VOIP traffic runs through their servers, right? My boss is of the mind that there is no way that this is a viable business model and his insistence has me doubting myself. So, to clarify - Vonage has to have the necessary bandwidth to handle whatever amount of simultaneous calls. I can imagine that one vonage user calling another vonage user would use some sort of sip re-invite and perhaps even calls to other huge providers (packet8) are direct client to client. (Last time I read about this it seems that even calls to other large voip providers go through the PSTN though). Barring voip to voip calls, everything must run through their bandwidth right? If I'm right on this, I guess we need to come up with some sort of viable business model to do sell our own service. I want to concentrate on smb clients to whom we can then provide an asterisk box which would leave our bandwidth free, but my boss isn't particularly keen on this route. Anyways, Thanks for any insight and advice on this question, sorry if I'm asking this in the wrong place Thanks, Steve ___ --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: AW: [asterisk-users] Snom or Cisco Phones?
I concur, We run about 50 7960's and they work quite well. Sound quality is pretty good. Like Aaron said, unless you're running call manager you can't program soft keys, etc. We're looking at going to a different phone that would give us some more customization options. Also.. Cisco's come with skinny firmware. You'd have to acquire the SIP firmware from somewhere. And NO cisco will NOT 'give' it to you If you need some, I have some for sale! :) :) That and Cisco won't give you the time of day if you don't use their stuff ;) We have about 1600 of the Cisco's on campus, and unless you run them on the call manager, you're not gonna have nearly as many features as any other phone that's designed with SIP in mind. That said, if you need a phone with dialtone, a pretty screen, and limited xml services, then I will say that the cisco's are extremely easy to provision once you figure out the upgrade paths. (Oh, and we're running 7940's and 7960's... if you're looking at the 7912's, etc, good luck, they're a _complete_ pain to work with) Aaron On Tue, 2006-10-31 at 20:41 +0100, Christian Stredicke wrote: I think one of the differences is: We do pay attention to Asterisk and this mailing list ;-) CS -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Joao Pereira Gesendet: Dienstag, 31. Oktober 2006 13:47 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] Snom or Cisco Phones? Hello I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom and Cisco Phones. Can you gurus, please, give me your impression of these 2 brands? I need to focus more in SIP and Asterisk compatibility and less in pricing (yes, I know the Cisco are more expensive). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao Pereira ___ --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 ___ --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] Re: VOIP Bandwidth questions
Martin Joseph wrote: On 2006-11-02 07:34:15 -0800, mail-lists [EMAIL PROTECTED] said: snip My question is this: How do huge voip companies like vonage handle bandwidth. I'm pretty sure that they have to have sufficient bandwidth available for X numbers of simultaneous calls, in other words ALL VOIP traffic runs through their servers, right? My boss is of the mind that there is no way that this is a viable business model and his insistence has me doubting myself.snip Also, it's not true that all the traffic need to flow through there servers. Once the connections are setup in a well designed system, the data could flow directly. I'm not sure what you mean here. What connections? If they're terminating to the PSTN Either they're paying someone to do it or they're doing it themselves right? If they're doing it themselves they have to handle the bandwidth requirements. As I said before - if both endpoints are on vonage the data might go from device to device ___ --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] AstFax Sending a Fax
Barry Fawthrop wrote: Thanks Andrew I have no plans to VoIP my Faxes to a VoIP provider I just would like to send them from my desktop (which is windows) to my PBX (which is AstLinux inside a net 4801) The PBX connects to PSTN lines via a FXO Gateway (CG-410 in my case) So really it's trying to get Windows to detect the modem or phone line one the 410. Can this still be done ? Thanks again Barry Andrew Joakimsen wrote: You can use the fax server Hylafax ( http://www.voip-info.org/wiki/index.php?page=Asterisk+IAXmodem ) with IAXmodem ( http://iaxmodem.sourceforge.net/howto.php ) You really don't want to be sending faxes over the internet via VoIP providers, not yet because there is no t.38 support for that. As long as the connection to the PSTN is on a card on the same machine or possibly over a network connection perhaps over a private line maybe using TDMoE then it should work fine On 10/24/06, *Barry Fawthrop* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi All I'm trying to understand how I would send my fax ? If I use Word or what ever word processor or even an email client to create what I want faxed. I have *asterisk setup with and FXO Gateway that will make the call to the fax number I dial SIP extension 320 is the FXO gateway. How do I now get my email or word document to TIFF to then fax to the FXO gateway or SIP/320 ? I don't understand that part. They all talk about an email with a TIF attachment and the TIF attachment is sent to the number in the subject line. Thanks all Barry As far as I know you can't send anything BUT a TIFF. This is how we do it: We have winprint hylafax on client machines. This allows you to 'print' to the fax client from any program (word, adobe, webbrowser,etc). This communicates with hylafax on the asterisk server which in turn talks to asterisk via IAXModem (we ARE using VOIP to fax - with about 95% success). If you're not using voip to fax, I suppose hylafax would talk to whatever modem you're using which in turn would call out through asterisk. There are other options I think, such as email to fax gateways in which you could attach a document to an email. From what I understand this would require you to have conversion software somewhere in the loop to convert from pdf to tiff or from word to tiff, etc. This is just too painful for me at the moment. One other option that I'm considering is having clients install some sort of conversion software on the client machine. I don't know if such is available or not The point of this is that I think you need somesort of fax server software on your asterisk box. Hylafax works great for us. ___ --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] CID Issues
Hello, I've posted this at the trixbox and freepbx forums and haven't been able to get an answer. I thought perhaps the guru's here might be able to help me out :) I'm having some issues with setting caller IDs. There are 2 problems that I would like to solve. 1. I have a DID pointing to a ring group. The only 'extension' in that ring group is an external number (cell phone). So essentially the DID fwds to a cell phone. The problem is that the CID that shows up on the Cell phone is the number that's set on the outgoing trunk the the CALLERS #. Is there a simple way to override this? Or better yet, is there a prefered method for forwarding calls out with freePBX? 2. I have sevaral trixbox installs connected through DUNDI. The DUNDI works very well.. I can call local extensions from every PBX. The PBX's are connected via an IAX trunk. In freePBX I've created a custom trunk that accepts a 4 digit extension and puts the call into a 'trydundi' context. The problem I'm having is that whenever someone calls from an extension at one location to an extension at another location the CallerID that shows up at the other location is the one set either in #1 The custom trunk, or #2 in the 'Outbound CID' field in the users screen. What I WANT this to be set to is the Name of the extension ie. just like local calls are. Is there a way to do this painlessly. Is it possible to hook dundi into a different context so that it would think all calls are local.. I'm kinda guessing here. Sorry about the length of these descriptions and thanks for any advice! ___ --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] Virtualise asterisk on Xen
Rene wrote: Hi Arik, I have Asterisk running as guest on my Debain Xen system and it works fine. I used to work with an AVM Fritz!PCI ISDN card as well by compiling the ISDN driver (Hisax) in the XEN-kernel. You have to be aware that the host system does not use the ISDN PCI card by putting physdev_dom0_hide=pci_id in your grub config. I never used chan_misdn myself, but if you compile the xen kernel with misdn support I don't see any reason it can't work. Rene Hi, has anybody experience running asterisk on a (i.e. fedora-based) Xen system? What about mISDN support etc.? For a low-load system I thought about using: 1. Sempron 2800+ 2. some memory, in your opinion how much should I attribute to the asterisk guest system? 3. A AVM Fritz!PCI card for PSTN access 4. HFCPCI-S card in nt-mode for internal ISDN bus provision 5. Asterisk 1.2 with chan_misdn for the ISDN-card support It would be great to hear some of your thoughts on this set-up? Regards, Arik NB: I have the impression that virtualisation is not a big issue on this mailing list... Is that due to a show-stopper I overlooked, just because everything goes so smoothly that nobody even bothers to mention it ;-), or because everybody has plenty of hardware they can dedicate to their PBXs? ___ --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 I have attempted to use Xen on FC5 for my Asterisk setup as well. This is for my home office, so very few extensions, so I thought a Dell Optiplex PIII 550Mhz should handle this nicely. In my setup I have a single Digium X101P PCI card to provide FXO access. The problem I've run into is the FC5 kernels for Xen do NOT allow the PCI hiding capability, it is not compiled into the Dom0 and DomU Fedora provided kernels. Because of this, I cannot get access to my Digium card in the DomU...so I'm stuck. I don't really want to compile my own kernel...just for this. If I could get past this issue, I'd be running Asterisk in a Xen DomU domain. Ed ___ --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] New here...
I am trying to get an initial setup up and going which I assume is a very common question here. My basic questionsare the following: Can I get Asterisk up and going without voice cards using it with SoftPhones internally as a proof of concept. (just calling extensions and leaving voice mail) Is there a jump start config that would accomplish this? What is the recommended SoftPhone that is "Open Source" Thanks!
