Re: [asterisk-users] UK Colocation services
Hi,We can provide UK toll-free inbound and can recommend a co-lo provider where you'll have single-hop access to our network. Feel free to contact me off list.Kind regards,Simon On 9/27/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Can anyone direct me to a colo provider in the UKwhere I can park an asterisk server and buy UK toll free inbound services over SIP? Thanks Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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] DUNDi Servers
Hi Doug,On 9/25/06, Douglas Garstang [EMAIL PROTECTED] wrote: Lets say you have a cluster of, say, 10 Asterisk servers. After doing a local lookup to see if a number is available locally, in order to find out if the number is available on one of the other 9 servers, this peer has to query all 9 remaining peers. Is that true?Yes, or it could send one query to a server which in turned queried the other 9. Either way though, all 9 get queried unless the answer was cached. Caching is tricky with registrations as you don't want to cache a registration which hasn't been renewed. Is there a way to have 'registration servers' that accept registrations from phones, and which somehow notify 'DUNDI servers' (two for redundancy) that the registration servers query? To terminate a call, a peer would only have to query the DUNDi servers, not every other peer. After looking at the config files, I can't imagine how this could work, or if it's even possible with DUNDi. Yes, it is possible to push peer information as well as pull it. You could also, as you say, limit the number of registration servers (i.e. servers doing both the registration and DUNDi) and then only query to them. I'm sure the hybrid model you suggest would work as well although it'd need testing to see whether you got more performance out of splitting the DUNDI and registration roles or just adding more dual-purpose machines. SimonDoug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Accounting and re-invite
Hi Ronald,On 9/19/06, Ronald Wiplinger [EMAIL PROTECTED] wrote: I am thinking if re-invite will interfere accounting.No it won't Please help me to figure it out:Phone A is registered at asterisk and calls a gateway. If the gatewayallows re-invite than the rtp would go directly from phone A to thegateway, while the sip messages are still going through Asterisk. Asterisk will be informed when the call ended.If it is a postpaid accounting, just bill the customer, however, how isit for a pre-paid (calling card user)?I think Asterisk will have no power to turn off the call from A to the gateway.Even more, if the gateway would allow to end a call and continue with anew call, the new call would not be billed (or would it)?The SIP messages control the call, irrespective of where the RTP goes so Asterisk can terminate/set-up calls exactly as if the RTP was being handled. I guess the solution must be re-invite=noHowever, re-invite=no means that each call is going with rtp also through my server, what means for a remote phone, I have to provide forboth legs the bandwidth.Personally, I would always handle the RTP on quality/accountability/consitency grounds but, yes, that will incur a bandwidth overhead. ___ --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] macros in Realtime
Hi Ben,Yes it is but you need to remember to still include[macro-stdpbx1exten]switch = Realtime/in your extensions.confSimonOn 9/6/06,
Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello all,Another question related to Realtime.Is it possible to call macros using Realtime arch?I have a macro definition in table extensions_conf in my MySql db as: 30 | macro-stdpbx1exten | s |
1 | SET| fwdedNum=${DB(CFWD/${ARG1})}And calling the Macro using another entry in table extensions_conf inMysql db: 40 | pbx1 | _[345]. |
1 | Macro| stdpbx1exten|${EXTEN}I get errors like :Sep6 11:18:53 WARNING[14493]: app_macro.c:154 macro_exec: No suchcontext 'macro-stdpbx1exten' for macro 'stdpbx1exten'
Are there issues with the same??TiA-Ben.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list
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Re: [asterisk-users] iax vs. sip?
Hi Steven,The provider's implementation will have a bigger affect than any differences within Asterisk, e.g. how they are load-balancing and whether in fact SIP is serviced by Asterisk at all. Compared like-for-like within Asterisk we find there is not a lot in it, with each having their own pros and cons. We support both and whilst we have more customers on SIP than IAX, currently favour IAX for new customers where they are undecided given lower support overhead and simplified load-balancing. I'd recommend you try both with the provider you're considering. Simonwww.esms.comOn 8/31/06, BerkHolz, Steven [EMAIL PROTECTED] wrote: I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am asking about SIP vs. IAX if the provider does either. (we are looking at testing Teliax next) I have seen posts about jitter in IAX, so I am not sure if SIPmight bebetter to use right now. Also, since IAX uses the same port for all of the calls, the call separation has to be done higher in the OSI stack. I do not know if this is better or worse or neither. Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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: Nokia E60/61/70 and SIP
Hi Benny,The E61 handles this just fine. With SIP as the default channel to dial and no WiFi coverage, you get a message asking if you'd like to dial by cellular. Works nicely other than a few stability issues. SimonOn 24 Aug 2006 11:23:39 +0200, Benny Amorsen [EMAIL PROTECTED] wrote: H == Haspers[EMAIL PROTECTED] writes:H We are using some E61 and E70's with asterisk. Only problem we haveH at this moment is that we are unable to use a password for the H authentication. I haven't found out yet why this isn't working.H They are working good, but I would like to see some small thingsH changed in future firmware versions (like being able to select H multiple WLAN points (Access groups) instead of just one.E70 works with passwords here. No trouble.The main issue is that the E70 can't automatically switch to cellularwhen the phone is out of WLAN coverage. It is a bit silly to have to click option-s___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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: Nokia E60/61/70 and SIP
Hi Haspers,Makes sure you have created an 'Internet tel' profile. It doesn't appear to do anything but was vital in getting it working for me. The other settings in the how to look sensible.Simon On 8/24/06, Haspers [EMAIL PROTECTED] wrote: Strange,What settings do you use? I followed this linkhttp://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html but without anyluck. -Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] ] On Behalf Of Benny AmorsenSent: donderdag 24 augustus 2006 11:24To: asterisk-users@lists.digium.comSubject: [asterisk-users] Re: Nokia E60/61/70 and SIP H == Haspers[EMAIL PROTECTED] writes:H We are using some E61 and E70's with asterisk. Only problem we haveH at this moment is that we are unable to use a password for the H authentication. I haven't found out yet why this isn't working.H They are working good, but I would like to see some small thingsH changed in future firmware versions (like being able to select H multiple WLAN points (Access groups) instead of just one.E70 works with passwords here. No trouble.The main issue is that the E70 can't automatically switch to cellular whenthe phone is out of WLAN coverage. It is a bit silly to have to click option-s___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] E61
Hi Andreas,That is incorrect. It works just fine through NAT providing:- The server is proxying RTP as it has no support for STUN etc.- The NAT is the basic domestic router style, not a full blown firewall requiring port mappings SimonOn 8/24/06, Andreas Sikkema [EMAIL PROTECTED] wrote: Anyone here use the Nokia E61 ? I am looking to invest in a wifi phone and I want to get the best. Is it good as far as reception ? That is of most importance to me. Thanks.I've tried it in the last couple of days. The biggest issue for me ist that it HAS to be on the same side of a NAT as theserver it talks to (asterisk, ser, etc). If it is on theprivate side of a NAT and the server is on the public side, itdoesn't work. I've read something on the Nokia forums that Nokia is aware of the problem and it will be solved.My problem is that they want to solve this using STUN etc,while I would prefer they also wouldn't have the softwarecare if it is on the inside of a NAT like most other CPE's so our platform can take care of things.--Andreas SikkemaBBeyondSoftware EngineerPlaneetbaan 4+31 (0)23 70743422132 HZ Hoofddorp___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] E61
Hi Andreas,I'm on 1.0610.04.04 19-04-06 RM-89WOn 8/24/06, Andreas Sikkema [EMAIL PROTECTED] wrote:Simon, That is incorrect. It works just fine through NAT providing: - The server is proxying RTP as it has no support for STUN etc. - The NAT is the basic domestic router style, not a full blown firewall requiring port mappingsStrange, then you must have some other firmware, because I just can't get it registered at all, let alone make calls.We do have proxies for RTP ;-)--Andreas SikkemaBBeyondSoftware EngineerPlaneetbaan 4+31 (0)23 70743422132 HZ Hoofddorp ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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: Nokia E60/61/70 and SIP
Hi,Yes, the proxy and registrar settings are identical in my set-up. Getting registered is the hard part but you've done that. I'd be looking at the Asterisk end of things rather than the E61 to see why that authentication issue is arising. WOn 8/24/06, Haspers [EMAIL PROTECTED] wrote: I've got them all. It registers correctly with Asterisk, and get incoming calls, but it complaints about outgoing calls (Connection Error). SIP Debug is giving me: SIP/2.0 407 Proxy Authentication Required But those settings are the same (Proxy Server/Registrar Server). So what could be the problem? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Simon WoodheadSent: donderdag 24 augustus 2006 12:51To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Re: Nokia E60/61/70 and SIP Hi Haspers,Makes sure you have created an 'Internet tel' profile. It doesn't appear to do anything but was vital in getting it working for me. The other settings in the how to look sensible.Simon On 8/24/06, Haspers [EMAIL PROTECTED] wrote: Strange,What settings do you use? I followed this linkhttp://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html but without anyluck. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Benny AmorsenSent: donderdag 24 augustus 2006 11:24To: asterisk-users@lists.digium.comSubject: [asterisk-users] Re: Nokia E60/61/70 and SIP H == Haspers[EMAIL PROTECTED] writes:H We are using some E61 and E70's with asterisk. Only problem we haveH at this moment is that we are unable to use a password for the H authentication. I haven't found out yet why this isn't working.H They are working good, but I would like to see some small thingsH changed in future firmware versions (like being able to selectH multiple WLAN points (Access groups) instead of just one.E70 works with passwords here. No trouble.The main issue is that the E70 can't automatically switch to cellular whenthe phone is out of WLAN coverage. It is a bit silly to have to click option-s___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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] Apache for FastAGI
Hi Doug,We run with SOAP on both the client/server side, i.e. conventional AGI making SOAP request to SOAP server running on Apache. I'm sure FastAGI would be quicker but wasn't around and other than the overhead of making the SOAP request is very lightweight on the Asterisk side. SimonOn 8/22/06, Douglas Garstang [EMAIL PROTECTED] wrote: I'm not sure how one would build a HTTP header on the client side, given that all you have to work with is a single line entry in extensions.conf. -Original Message- From: Tielin Xu [mailto: [EMAIL PROTECTED]] Sent: Friday, August 18, 2006 12:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Apache for FastAGI It is an valid option, but you have to build a HTTP header in your request to your web server, which CGI programs or Java servlets on web server could interpret your request from Asterisk. Tielin [EMAIL PROTECTED] 08/18/06 11:28 AM Here's an idea... Rather than writing your own multi-thread socket server for use with FastAGI, has anyone tried to use an Apache web server instead? After all, it does all that for you. I just gave it a shot, but Asterisk tries to send all the agi params to the web server, which it doesn't like it... [Fri Aug 18 12:25:28 2006] [error] [client xxx.yyy.141.162] Invalid URI in request agi_network: yes Doug ___ --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 listTo 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] Calls over VPN
We've done this with OpenVPN and it works fine. I'd recommend that the VPN server is not on the same box as Asterisk. Stick it on a firewall/gateway box giving access to the network containing the Asterisk boxes behind it. This way the Asterisk box(es) is seeing normal unencrypted traffic and the VPN server(s) can be specified to meet the VPN requirement. On 8/23/06, Joseph [EMAIL PROTECTED] wrote: Is anybody making calls over VPN?If so what is the penalty asencryption is involved.I was planning to use VPN to register Sipura units to my local asteriskthis way I don't have to deal with NAT issues. --#Joseph___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] IAX unstable with large number of calls?
Hi Curt,That probably suggests that with SIP they're handing off the RTP to their upstream provider and just dealing with the signalling which is very low overhead. With IAX they have to transport both unless they're interconnecting upstream by IAX and can transfer. In my experience the load is about the same for IAX or SIP+RTP and limited [on a single box] by the specification of the box itself. Whilst I risk being shot down, I'd be wary of any provider who isn't themselves handling the RTP for a multitude of quality reasons (and just because that is what you're paying them for), as well as one who quotes capacities in single box terms. SimonOn 8/15/06, Curt Shaffer [EMAIL PROTECTED] wrote: I was just talking with an unnamed provider and the guy told me that they recommend their users not to use IAX because it is unstable at 50 concurrent calls and unusable at 100 or more calls. Now I have personally worked on an asterisk box that was pushing more than 50 and there were no problems. Anyone else out there have any data either for or against this suggestion? Thanks Curt ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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] regexten / Realtime WAS DUNDI / regcontext
Folks,Just as an update on this, DUNDI is working prefectly and regexten is working fine for peers defined in sip/iax.conf. However, for peers defined in Realtime the regexten does not appear to be created although the console reports that it is. If anyone knows of any gotchas with regexten and Realtime, I'd be grateful for any pointers.Many thanks,SimonOn 7/17/06, Simon Woodhead [EMAIL PROTECTED] wrote:I'm expecting regcontext to create a context of regcontext and an priority 1 extension for either the value of regexten or the peer name. The context is created, the extension says it is created but isn't. It works fine with a staticaly defined extension of the same name as defined for regexten or the peer name. SimonOn 7/17/06, Watkins, Bradley [EMAIL PROTECTED] wrote: So you are relying on the behavior of regexten to default to peer name? Is that what you are expecting? And if so, could you test with a statically defined extension for the per-peer regexten parameter? Regards, - Brad From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] ] On Behalf Of Simon WoodheadSent: Monday, July 17, 2006 5:53 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] DUNDI / regcontext Thanks for the reply Brad.The relevant section of sip.conf was posted:[general]regcontext=sipregistrationIf you mean extensions.conf, I wasn't creating the extension in there other than for testing. RegContext correctly creates the context on registration but does not create the extension. If I create the extension manually, the DUNDi lookup works just fine. Simon On 7/16/06, Watkins, Bradley [EMAIL PROTECTED] wrote: Could you possibly put up the relevant section(s) of your sip.conf?It sounds like the DUNDi portion is set up properly, and obviously it's not going to find an extension that doesn't exist.Regards,- BradFrom: [EMAIL PROTECTED] on behalf of Simon WoodheadSent: Sat 7/15/2006 5:59 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] DUNDI / regcontextHi folks,I've been having a go at getting DUNDI working this evening to enableusers to register to any Asterisk box and to look them up from another. The DUNDI part works just great (very impressed), as does the subsequentjoining of calls between the two servers but I'm struggling withregcontext and would be grateful for any input.sip.conf includes: [general]regcontext=sipregistrationWhen a user registers, I get the Added extension 'XX' priority 1 tosipregistration message. However, 'show dialplan' does not show theextension and a DUNDI lookup does not return it. The sipregistration context has been auto-created but is empty. If I manually create thesipregistration context and add the NoOp extension, then everythingworks as expected.I've tried this across multiple boxes, each running different versions right up to the latest stable but the behaviour is the same. It is alsothe same with both SIP and IAX registrations and doesn't make adifference if the peer is defined in the .conf file or Realtime. They doall have identical configurations though so I suspect there might besomething in our setup which is conflicting.Any input gratefully received.All the best,SimonThe contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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] Redundant Ethernet
Have a google for 'interface bonding'. You bond your two cards together to appear as a single one and then bind Asterisk to an IP address on it. The cards work in loadbalance or failover mode as you specify. On 7/20/06, shadowym [EMAIL PROTECTED] wrote: Has anyone had any success creating a redundant ethernet connection fromtheir Asterisk server?What I would like it to do is use both ethernetcontrollers on my motherboard so that if one fails the other one takes over. I don't see anyway to make it work seamlessly with 2 IP addresses it wouldprobably have to be a hot standby in software type of thing.Preferrably with Debian Sarge but CentOS 4.3 is an option.___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] DUNDI / regcontext
Thanks for the reply Brad.The relevant section of sip.conf was posted:[general]regcontext=sipregistrationIf you mean extensions.conf, I wasn't creating the extension in there other than for testing. RegContext correctly creates the context on registration but does not create the extension. If I create the extension manually, the DUNDi lookup works just fine. SimonOn 7/16/06, Watkins, Bradley [EMAIL PROTECTED] wrote: Could you possibly put up the relevant section(s) of your sip.conf?It sounds like the DUNDi portion is set up properly, and obviously it's not going to find an extension that doesn't exist.Regards,- Brad From: [EMAIL PROTECTED] on behalf of Simon WoodheadSent: Sat 7/15/2006 5:59 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] DUNDI / regcontextHi folks,I've been having a go at getting DUNDI working this evening to enableusers to register to any Asterisk box and to look them up from another. The DUNDI part works just great (very impressed), as does the subsequentjoining of calls between the two servers but I'm struggling withregcontext and would be grateful for any input.sip.conf includes: [general]regcontext=sipregistrationWhen a user registers, I get the Added extension 'XX' priority 1 tosipregistration message. However, 'show dialplan' does not show theextension and a DUNDI lookup does not return it. The sipregistration context has been auto-created but is empty. If I manually create thesipregistration context and add the NoOp extension, then everythingworks as expected.I've tried this across multiple boxes, each running different versions right up to the latest stable but the behaviour is the same. It is alsothe same with both SIP and IAX registrations and doesn't make adifference if the peer is defined in the .conf file or Realtime. They do all have identical configurations though so I suspect there might besomething in our setup which is conflicting.Any input gratefully received.All the best,SimonThe contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] DUNDI / regcontext
I'm expecting regcontext to create a context of regcontext and an priority 1 extension for either the value of regexten or the peer name. The context is created, the extension says it is created but isn't. It works fine with a staticaly defined extension of the same name as defined for regexten or the peer name. SimonOn 7/17/06, Watkins, Bradley [EMAIL PROTECTED] wrote: So you are relying on the behavior of regexten to default to peer name? Is that what you are expecting? And if so, could you test with a statically defined extension for the per-peer regexten parameter? Regards, - Brad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Simon WoodheadSent: Monday, July 17, 2006 5:53 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] DUNDI / regcontext Thanks for the reply Brad.The relevant section of sip.conf was posted:[general]regcontext=sipregistrationIf you mean extensions.conf, I wasn't creating the extension in there other than for testing. RegContext correctly creates the context on registration but does not create the extension. If I create the extension manually, the DUNDi lookup works just fine. Simon On 7/16/06, Watkins, Bradley [EMAIL PROTECTED] wrote: Could you possibly put up the relevant section(s) of your sip.conf?It sounds like the DUNDi portion is set up properly, and obviously it's not going to find an extension that doesn't exist.Regards,- BradFrom: [EMAIL PROTECTED] on behalf of Simon WoodheadSent: Sat 7/15/2006 5:59 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] DUNDI / regcontextHi folks,I've been having a go at getting DUNDI working this evening to enableusers to register to any Asterisk box and to look them up from another. The DUNDI part works just great (very impressed), as does the subsequentjoining of calls between the two servers but I'm struggling withregcontext and would be grateful for any input.sip.conf includes: [general]regcontext=sipregistrationWhen a user registers, I get the Added extension 'XX' priority 1 tosipregistration message. However, 'show dialplan' does not show theextension and a DUNDI lookup does not return it. The sipregistration context has been auto-created but is empty. If I manually create thesipregistration context and add the NoOp extension, then everythingworks as expected.I've tried this across multiple boxes, each running different versions right up to the latest stable but the behaviour is the same. It is alsothe same with both SIP and IAX registrations and doesn't make adifference if the peer is defined in the .conf file or Realtime. They doall have identical configurations though so I suspect there might besomething in our setup which is conflicting.Any input gratefully received.All the best,SimonThe contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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] DUNDI / regcontext
Hi folks, I've been having a go at getting DUNDI working this evening to enable users to register to any Asterisk box and to look them up from another. The DUNDI part works just great (very impressed), as does the subsequent joining of calls between the two servers but I'm struggling with regcontext and would be grateful for any input. sip.conf includes: [general] regcontext=sipregistration When a user registers, I get the Added extension 'XX' priority 1 to sipregistration message. However, 'show dialplan' does not show the extension and a DUNDI lookup does not return it. The sipregistration context has been auto-created but is empty. If I manually create the sipregistration context and add the NoOp extension, then everything works as expected. I've tried this across multiple boxes, each running different versions right up to the latest stable but the behaviour is the same. It is also the same with both SIP and IAX registrations and doesn't make a difference if the peer is defined in the .conf file or Realtime. They do all have identical configurations though so I suspect there might be something in our setup which is conflicting. Any input gratefully received. All the best, Simon smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Add Country to CDR's
Yep, cmd setCDRUserField will do this for you assuming you have the field set up. I'd be keen to hear if anyone has a way of achieving the same thing across multiple user fields to save having to explode multiple values out of a single user field seperately. SimonOn 6/20/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Tue, 2006-06-20 at 11:14 -0400, William Piper wrote: Thanks Bret, but how about an example or webpage? I'm not finding anything on google about this command for asterisk. What about AppendCDRUserField()... would this work? that seems to be the same thing.the userfield lets you stick arbitrarydata into your cdr records.--Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402Utrecht NL +31 306 553058US WA +1 360 207 0479US NY +1 516 687 5200FreeWorldDialup: 635378http://www.trxtel.com the VoIP provider that pays you!-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.3 (GNU/Linux)iD8DBQBEmBee+1olxlzQw5cRAnnjAKCFf4NDjUlCDlf1Pb//LyeauifNbwCfUB+5cMObnxnTQYcuP3VlTYHsxZg==g7ro -END PGP SIGNATURE-___--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] asterisk load balance
That sounds fine except where registrations are involved although I'd suggest you look into SRV as well as RR for the DNS to more finely balance the load for clients which support it. Doug's mail says it all where registrations are involved - not all state information is stored in the database so you need to ensure that incoming and outgoing traffic for a given user is hitting the asterisk box with which they've registered. SimonOn 6/17/06, unplug [EMAIL PROTECTED] wrote: Hi,I am designing a asterisk load balancing model as follow.There are3 asterisks connected to a single DB and a single server storing allthe configuration file and voicemail.Round Robin DNS will distribute the request to asterisks.DNS round robin ---+ asterisk1--+ DB and file server +---asterisk2---+ +---asterisk3---+ Does anyone has load balancing experience implemented in asterisk thatcan share?Does my design work?Does any conflict will happen in mydesign?Any comment?Thanks!___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Voicemail with NFS
We use Unison Doug and it works just fine. It isn't perfect in theory but we've had no issues in practice. Your concerns over sacalbility are resolved by implementation - do you need it on every single Asterisk box, or maybe local to just two with routing to them and failover in the dial-plan? Unison is like two way rsync and consequently extremely efficient. SimonOn 6/17/06, Douglas Garstang [EMAIL PROTECTED] wrote: Mike,I don't think unison is a workable solution. It doesn't scale. The network and system load would increase exponentially as we added asterisk servers to our cluster.Doug.-Original Message- From: Mike Diehl [mailto:[EMAIL PROTECTED]]Sent: Fri 6/16/2006 9:40 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionCc: Subject: Re: [Asterisk-Users] Voicemail with NFSI don't know how big your voicemail system is, but have you considered usingUnison to syncronize the vm accross all your servers?I'm deploying multiple servers with two vm servers, each sync'ed every 5? minutes.If one fails,the other one should be good enough.Just a though,MikeOn Friday 16 June 2006 16:14, Brian Capouch wrote: Douglas Garstang wrote: Douglas Garstang wrote: I hope someone isn't going to tell me that the voicemail directory going away is going to cause Asterisk to fall in a heap on the floor. Brian Capouch wrote: You never give up on dissing Asterisk, do you, Pococurante? This would be acceptable behaviour for you? An NFS-mounted volume isn't ever going to be as reliable as one mounted on the local filesystem.You are introducing additional points of failure both with respect to there now being two hard drives involved, as well as an interposed network that can fail in a variety of ways. So by definition this arrangement isn't going to be as reliable as one based on a native filesystem. And you never have answered the direct question: what do you expect the logical thing would be to happen if all the sudden an important system resource has just gone away? Regardless of the answer (because a rejoinder to that would then be, So add that behavior into Asterisk, or help the developers do so . . ) my point isn't that you are finding--actually looking for--places where catastrophic behavior makes Asterisk suffer. The problem is that you don't ever say, So what are some reasonable things that might be done in this situation; instead you emit a scathing remark (fall in a heap on the floor) that would indicate you've discovered some glaring design flaw that any idiot would have known to design around ahead of your finding it. It is not automatically the case that if Asterisk doesn't do something you think it should do it means that Asterisk is horribly and glaringly flawed.But that's what you *always* assume, and you always--ALWAYS--do so snidely. Pococurante. B.___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo 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] Realtime SIP
Hi Doug,We use Realtime SIP via a central MySQL database (2 actually in Master Master config) but registration is only available on the box to which the client has registered. Clients can register with any database and the table does get updated with some registration information (ip address, expiry time etc.) but they are not reachable by any of the other boxes sharing the config. I've been following the cluster thread with great interest for a workable solution to this. All the best,SimonOn 3/14/06, Douglas Garstang [EMAIL PROTECTED] wrote: Is anyone using realtime sip for friends (ie phones) with multiple Asterisk boxes all pointing to the one central MySQL database? Does it work? Are phones that are registered to the database from Asterisk box able to reach phones registered to the database from another Asterisk box? Doug.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] * point to point t1 solution?
Bad day Damon? I think your comments are a little harsh towards someone who is an active and informed contributor to the list. Jean-Michel could have ignored you but he chose to share what he could. Maybe someone else will have the complete answer to your question. On 1/26/06, Damon Estep [EMAIL PROTECTED] wrote: Jean-Michel,You missed the entire point - the question is IS ASTERISK CAPABLE OF EMULATING A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY WEB BASED HINTS I MIGHT LOOK AT?Not WILL YOU DO IT FOR ME? Your response to this post was un-informative and quite frankly it is the type of useless response that most mailing lists and newsgroups could do without.Damon -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] ] On Behalf Of Jean-Michel Hiver Sent: Thursday, January 26, 2006 1:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * point to point t1 solution? Damon Estep a écrit : Can anyone point me to a reference or sample config for bypassing a nailed up (point to point) t1 between two PBXs with asterisk and a pair of t1 cards? Right now I have 2 Nortel norstars connected to each other via a leased line t1. I also have a solid 10mbps low latency microwave link between the 2 sites. You probably need a couple of T1 cards, and some paid consulting to get it working (I've never done it myself but that's how I would do it if I was in a hurry) My goal is to run an asterisk box at each end with a t1 card and Ethernet card to act as a TDMSIP gateway to bypass the nailed T1 in a relatively dumb configuration, with the goal of migrating off of the norstars eventually. If it's a point to point Asterisk - Asterisk configuration, why use SIP rather than IAX? IAX configuration is very easy, so once you get the norstar - asterisk link up it'll be a piece of cake. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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 listTo 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] UK Provider
We will Scott. http://www.esms.com or drop me a mail off-list. Kind regards, SimonOn 1/24/06, scott [EMAIL PROTECTED] wrote: HiDoes anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account.www.SipGate.co.uk are great but they only allow 1 telephone number per user, you can register another telephone number by registering as another user but Asterisk doesn't allow multiple registrations.Many ThanksScott___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo 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] Asterisk Hosting
We've been trying Unison (http://www.cis.upenn.edu/~bcpierce/unison/) on a 1 minute cron job. There are some theoretical issues but it has been great so far. We use it to synch prompts as well as messages. SimonOn 12/27/05, BILL GITONGA [EMAIL PROTECTED] wrote: What is the best method of storing voice main messagesso that they are accessible to different asteriskservers in a hosted environment? I have consideredAsterisk real time but I don't think it stores theactual voice mail folder in the database. I'm thinking of using NFS for this and put my voice mail folders onthe NFS so that it is accessible by the differentservers. Is this a good way to do it or is there abetter way of doing this?__ Yahoo! for Good - Make a difference this year.http://brand.yahoo.com/cybergivingweek2005/___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Dynamic IAX2 hosting in the UK
Hi Bails, We'll help. Drop me a mail off-list. Simon http://www.esms.comOn 12/8/05, bails [EMAIL PROTECTED] wrote: Hi all, just got an iaxy box for a customer and its great, but!I really dont want to host and bill this customer myself and i cannot find a voip to pstn breakout that will let him have a dynamic IP.Gradwell require a static ipVoiptalk wont support itAny Ideas where else to try?Thanks in advanceBails___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] IAX test service
Our free UK numbers can forward to IAX: http://www.esms.com/services_numbers_pure_free.php Simon On 11/3/05, Gabor Horvath [EMAIL PROTECTED] wrote: Dear Asterisk users, can you suggest me a free service where I can test my IAX trunks? Thank you. Gabor ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound fax solution
Fax over VoIP is just not reliable in my opinion. I'd run with doing it directly to PSTN as the other poster suggested or via Hylafax. We've used Hylafax behind Asterisk very succesfully in the past. SimonOn 10/29/05, KARIM MOUSLI [EMAIL PROTECTED] wrote: my problem is to triger the transfer to sip provideri always get worng number error*** REPLY SEPARATOR***On 28/10/2005 at 20:27 Chris Mason (Lists) wrote:Teliax works for me, generally. I don't know why but no other provider does. I suspect the other translate to G729 and send SIP.--Chris MasonNetConcepts(264) 497-5670 Fax: (264) 497-8463Int:(305) 704-7249 Fax: (815)301-9759Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED]___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Prevent transcoding
Hi folks, Is anyone aware of a way to prevent transcoding or better still apply some kind of weighting to codec selection based on other channels in the call? Let's say we support g729 and gsm, a peer supports both and a client supports one of them. We're seeing calls frequently coming in on one and being transcoded to the other whilst it would be much more efficient to pass it straight through for the negligible bandwidth saved. Is there any way of achieving this? Thanks, Simon PS - before anyone suggests re-inviting, not doing so is intentional! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hangs
Hi Rene, Yes, I've seen that but our version from CVS is a month or so old os it may well have been rectified now. On our version reloads cause the process to die about 50% of the time, work fine about 45% and cause it to hang in the way your describe probably 5%. Simon On 19/10/05, René Enskat [Teamware GmbH] [EMAIL PROTECTED] wrote: Since some CVS Updates the asterisk hangs after command: reload orrestart now.Then i have to kill -9 th eprocess.Nothing will be outout inside the CLI but i can type commands.Somebody know this problem? And the CallerID bug still seems to be in there too.Regards rene___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cmd SIPRedirect for loadbalancing
Hi folks, I've just been reading about the above command and wonder if anyone has made use of it for load-balancing or if doing so would be completely inappropriate!? I'm thinking of the scenario where there are a number of Asterisk gateways and incoming SIP traffic. From what I've read, with a box in front receiving all incoming traffic the SIPRedirect command could be used to redirect traffic to one of the gateways, perhaps with an AGI to manage the load balancing and registration to handle failover. Conventional wisdom suggests using SER for this but I wonder if a pure Asterisk deployment is now possible/viable or sensible? Secondly, with the gateways themselves sharing a Realtime database could a client registered with one, deliver calls to another or is this not yet fully supported? Thanks in advance, Simon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_queue, fewestcalls and leastrecent logic
Hi Brian, Thanks for the explanation. In my second scenario, agent 1002 wouldn't be sat idle as the call would rotate around all them (like round robin) but in the order they stand in the league of fewestcalls / least recent calls. They'd all ring but the order would change according to activity. This'd need to be coupled with your auto-logout logic though as the fewestcalls / least recent has potential to always ring first and go unanswered which could delay calls in going to live and active agents. W - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 09, 2003 8:48 PM Subject: Re: [Asterisk-Users] app_queue, fewestcalls and leastrecent logic Well their are a few diffrent ways to do a roundrobin type setup. One I didn't mention is circular. Circular is like this. You have Agents 1001,1002 and 1003 First calls hits 1001... second call hits 1002 even if 1001 isn't busy. Third call hits 1003 even if 1001 and 1002 are not busy. Now it will start roundrobin(ie or hunting as you might see it) if the agent doesn't answer. Fourth call would start back at 1001. I think I seen app_circular somewhere. And yes I agree it should be a fallback options or just create diffrent variations of the current fewestcalls and leastrecent like fewestcallsrr and leastrecentrr, but that doesn't address the fact that if agent 1001 and 1002 are logged in you don't want agent 1001 ringing forever over and over while agent 1002 sits idle. If no agents answer it should try something else or have the option to try something else. bkw On Sat, 9 Aug 2003, Simon Woodhead wrote: Hiya, First off, thanks to everyone involved in app_queue. Its a great addition to an already great system. For my two penneth (or cents!), I think the following would be good: - the fallback method should be optional if at all possible, so it can be set up for, say, fewest calls with ring all as the fallback rather than roundrobin. - it would also be good to have no fall-back at all but to stay in fewestcalls / leastrecent mode such that if the no. 1 in the league of fewestcalls / leastrecent calls doesn't answer, it goes on to no. 2, then 3 etc. etc. The second one may actually be how you see roundrobin working as I'm a little confused as to how it actually works in queue scenarios. I've only ever used for passing unanswered calls to a specific extensions to other members of the workgroup. In a queue scenario, what order does it rotate? Will the same person always be the first and the order thereafter always the same? W - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 09, 2003 8:06 PM Subject: [Asterisk-Users] app_queue, fewestcalls and leastrecent logic First of all I would like to thank Mark for getting roundrobin to go roundrobin. Good job. Now we have some options here for leastrecent and fewestcalls strategy. It needs some work on the logic and Mark recommend that I ask the list and get some input before he makes any changes to it. fewestcalls from what I have seen would always ring the agent with the fewestcalls first then go into roundrobin if that agent didn't answer. Next new caller would ring fewestcalls agent first then start roundrobin. What do you think should happen in fewestcalls? Right now it just rings the agent with the fewestcalls over and over with current app_queue logic. leastrecent from what I have been looking at will ring the agent that has least recently take a call first then if they don't answer go into roundrobin. Then the next new call coming from queue would first go to the leastrecent first then try every agent in roundrobin till answered then starting over again. New caller from queue hits leastrecent agent first. Same thing happens in leastrecent strategy. The leastrecent agent will ring over and over with current app_queue logic. Now some of you might recommend autologoff options. But that also might need some work. I don't want to log off an agent for not answering the phone only once. So here is how I would like to see autologoff work. Example: queue timeout = 20 agent autologoff = 60 The agent would have to not answer their phone 3 times in a row to get logged off. As it stands now they did not answer just once and get logged off. Thus allow for an employee to use the excuse for not working when they should be logged in and taking calls. Unless i'm wrong here. Please post your input on these options and how you would like them to see them function function. Thanks, Brian CWIS Internet Services ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GSM file format
Hi folks, We're getting some IVR/busy messages recorded for Asterisk which I understand have to be in GSM format but we've been given a list of options which don't mean a lot: RAW GSM 6.10 audio stream RAW 'byte aligned' GSM 6.10audio stream US ROBOTICS VOICE MODEM W.O. HEADER US ROBOTICS VOICE MODEM W. HEADER I've looked in the archive but can't see anything. Can anyone help? Thanks, Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_queue, fewestcalls and leastrecent logic
Hiya, First off, thanks to everyone involved in app_queue. Its a great addition to an already great system. For my two penneth (or cents!), I think the following would be good: - the fallback method should be optional if at all possible, so it can be set up for, say, fewest calls with ring all as the fallback rather than roundrobin. - it would also be good to have no fall-back at all but to stay in fewestcalls / leastrecent mode such that if the no. 1 in the league of fewestcalls / leastrecent calls doesn't answer, it goes on to no. 2, then 3 etc. etc. The second one may actually be how you see roundrobin working as I'm a little confused as to how it actually works in queue scenarios. I've only ever used for passing unanswered calls to a specific extensions to other members of the workgroup. In a queue scenario, what order does it rotate? Will the same person always be the first and the order thereafter always the same? W - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 09, 2003 8:06 PM Subject: [Asterisk-Users] app_queue, fewestcalls and leastrecent logic First of all I would like to thank Mark for getting roundrobin to go roundrobin. Good job. Now we have some options here for leastrecent and fewestcalls strategy. It needs some work on the logic and Mark recommend that I ask the list and get some input before he makes any changes to it. fewestcalls from what I have seen would always ring the agent with the fewestcalls first then go into roundrobin if that agent didn't answer. Next new caller would ring fewestcalls agent first then start roundrobin. What do you think should happen in fewestcalls? Right now it just rings the agent with the fewestcalls over and over with current app_queue logic. leastrecent from what I have been looking at will ring the agent that has least recently take a call first then if they don't answer go into roundrobin. Then the next new call coming from queue would first go to the leastrecent first then try every agent in roundrobin till answered then starting over again. New caller from queue hits leastrecent agent first. Same thing happens in leastrecent strategy. The leastrecent agent will ring over and over with current app_queue logic. Now some of you might recommend autologoff options. But that also might need some work. I don't want to log off an agent for not answering the phone only once. So here is how I would like to see autologoff work. Example: queue timeout = 20 agent autologoff = 60 The agent would have to not answer their phone 3 times in a row to get logged off. As it stands now they did not answer just once and get logged off. Thus allow for an employee to use the excuse for not working when they should be logged in and taking calls. Unless i'm wrong here. Please post your input on these options and how you would like them to see them function function. Thanks, Brian CWIS Internet Services ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E-mail (still version 1) is not being Delivered
Hi Uriel, Forgive me if you've already done this, but have you checked disk space on the mailserver? Its caught me before and might save you hours debugging something that isn't broke. W - Original Message - From: Uriel Carrasquilla [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 09, 2003 9:40 PM Subject: RE: [Asterisk-Users] E-mail (still version 1) is not being Delivered For some reason my Voice-mail is not sending E-mails with the voice attachment anymore. It just stopped working. any suggestions on how to debug? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some questions about a potential usage scenario for asterisk
Hi Dave, I think it will. We have a very similar requirement... My aim is simply to have incoming calls identified (using CID) and logged, then initiating a request - from the relevant client pc - to the webserver for a specific page (the customer history/details page). We're working towards exactly the same. Connect to telephone line - can I use a simple PCI modem on a standard POTS line for testing purposes?; No. An ISDN card is your quickest way to get started. Beyond that you'll need a card from Digium. Configure for routing incoming calls to the client machines using VOIP - this means the client machines (on a lan) need a headset with mic and can be used instead of a telephone?; No problem. You'll need a softphone on each PC such as X-Lite or may prefer to get a POTS card from Digium and keep standard handsets. Configure for CID value to cross-reference customer database, then (somehow) instruct the client machine to request the relevant customer details page. As I'm unable to implement http-push due to client machines running mainly MSIE, I was thinking, even a basic messenger pop-up displaying customer name and number might suffice; We're doing this with a simple web-service providing tray-app on the client PC. Asterisk can call Perl scripts so we're having one which will call the appropriate client PC and pass the caller ID. The tray app then fires up IE with the CLID in the URL. I'd also like the ability to perform http-get and post operations directly from the asterisk box as the web server is in a remote location (via VPN); Shouldn't be a problem. I realise I could simply use a smaller module to pull out the CID but, frankly I'm interested in the PBX functionality too for future development. Is asterisk the right tool for me and am I thinking along the right lines with the above? Very much. Can't recommend it highly enough. W ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones
Hi Nick, You'll probably run into quality problems making calls over the ISDN from Xten via *. We did which led us to try several other softphones which were better and worse, e.g. Pingtel was great from a quality point of view but the interface wasn't. We're using snom 100s at the moment which are working great (apart from a headset issue which may or may not be relevant). We also have one of the 4 port analogue cards from Digium with traditional POTS phones/faxes connected in to it. They work fine as well. For the future, we'll be getting snoms for all the new phones and keeping the existing analogues to the capacity of the card. We'll only be (only are) using softphones for out of office laptop use although have also set up an 0800 dial-in on * to enable authorised users to make free calls in from their home/mobile and then dial internal extensions, or external numbers on the company which really works very well. We started out with the ideal of a softphone and have retraced sharply from it. A hardware sip phone is more expensive yet provides many benefits in usability and quality. Using analogue phones is great for users (as little changes) and the quality is fine although per port it is a more expensive route than a hardware sip phone, certainly at the lower end. Maybe I'll be shot down in flames but that is certainly how it has panned out for us. All the best, Simon - Original Message - From: Nick Knight [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 21, 2003 10:21 AM Subject: [Asterisk-Users] Phones Hello all, I am a newbie to this list - and so far very impressed with the functionality of Asterisk. So far I have setup a simple soft phone running on a windows PC making calls to other SIP soft phones. Later this week I hope to get UK ISDN2e up and running with it! My question is I would like the experience and feedback from users about what equipment/software you are all using for phones to connect to Asterisk, so fat I have been playing with xten soft phone which works very well but before I make my decision on which phones to use would like the feedback of the group. Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Again Asterisk and VMWare - it works now!
Hey Dan, Get a http://www.mini-itx.com/ and disguise it as a fruit bowl. She'll never know! I can't wait until someone builds one looking like a shoe or a handbag and then I can have them all over the house and the more I have, the happier the other half will be!! W - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 11:12 AM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! Roy, Please do not give me such a solution. I know how and where to buy or how to build a very cheap PC (I work in this field). I now that this is a cheaper option (to buy or build a new pc), but I don't want another computer running 24/7 in my house. It is so difficult to understand that? I have a small flat with two rooms. I want to be able to sleep too in the same house. My wife for sure will not accept another one... I feel that it can fully work on my config (allmost it does it now). It is more challenging to make it work under those circumstances...;-) Why to choose everytime the easiest solution available? I want to do it for my ..soul...;-) Best regards, Dan P.S. I have several PCs available for this, but.. I DON'T WANT TO USE THEM! - Original Message - From: Roy Sigurd Karlsbakk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 12:33 PM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! IIRC you were given URLs for all sorts of cheapo PCs. Perhaps you've got an old P90 lying around? Or perhaps someone else has? Use that! Not vmware! If you're to use vmware, do it the other way around - linux host with vmware windoze guest. This works fine for me on my PC. On Sat, 2003-07-19 at 08:49, Dan wrote: Hi, I have succeed using Asterisk on VMWare on an [EMAIL PROTECTED] with 128 MB allocated for the Linux virtual machine. I have connected this PBX with another one using IAX/GSM. I can call the other part and the sound is great, without any interruption. The phone used is a Cisco7960 with G.711, so still a codec conversion is in place (GSM/G.711) and Asterisk/VMWare Wkst performs very well. The problem is only when I try to call local services, like echo test or Digium Demo. Then, the sound of the informative message for the Digium Demo is choppy, but the sound from the Digium server (after connection) is very good. So.. the problem is only to play local files when in virtual machine (menus, informative messages, etc.). Why? It is clear that this is not a computer performance issue and/or a timing problem during the codec conversion. More, the inband DTMF works like a charm under the virtual machine. Even the known problem with double digits for Cisco phones dissapear. BR, Dan P.S. Please do not answer again that this setup cannot work. In this moment I cannot accept such an answer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX G729 Codec
You would still use PRI if you need bulk lines. You could use channelized T1, but you get a lot more options with PRI. Currently our phone server is in our colo rack and our phone lines are sent down to us via our data T1 line. Thanks Steven. I'll go investiagte that. Cheers, Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX G729 Codec
Hi Steven, BTW, my problems where on our private T1 line that sees round trips in the 4ms range. Our semi educated guess was that we had a problem with the jitter buffer causing echo cancel to go nutty when our ping times would occasionally jump to 20ms. When I turned off the jitter buffer, the call quality became so clear that people don't believe we are VoIP. Would you mind elaborating on what you mean by going 'nutty'? Long distance * to * G.729 over IAX works perfectly but we're also using G.729 over H.323 to our telco's Pace Vega Stream. The ping is only 10-12ms, occasionally jumping to 20ms. Latency is non-existent and the quality is excellent but periodically, mid-way through a call it goes wrong with the other party sounding literally like they are underwater. This only happens on incoming calls though strangely. Any ideas? Cheers, Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Virtual fax on the Asterisk box
Hylafax Dan. It isn't that elegant though as you'll need to wire an analogue port to each fax/modem. AFAIK there isn't a virtual fax/modem provided by * that another programme can use. W - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 05, 2003 7:20 AM Subject: [Asterisk-Users] Virtual fax on the Asterisk box Hi all, I want to get the following functionality: define one extension as a virtual fax machine. Every fax redirected to that extension to be converted in a picture file (bmp/jpg/gif or something else) and then attached to an email and send to an e-mail address. Are you aware of a linux based application who does something like this and can be installed on the same computer as Asterisk? Another possibility is a Windows based software SIP fax which can be registered with Asterisk. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Virtual fax on the Asterisk box
I don't think such a thing exists Dan but please let me know if you find one as we need the same. The only additional hardware you'd need is a fax/modem or two for Hylafax (assuming your * already has analogue extensions set up). One appealing thing about this solution is that the fax can be on a separate machine to * which is worth considering given the possible cpu implications of a lot of pstiff/pdf conversions. W - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 07, 2003 10:13 AM Subject: Re: [Asterisk-Users] Virtual fax on the Asterisk box Hi, What I need is a pure software solution, to avoid any other hardware to get that functionality. Thanks, Dan - Original Message - From: Simon Woodhead [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 07, 2003 11:50 AM Subject: Re: [Asterisk-Users] Virtual fax on the Asterisk box Hylafax Dan. It isn't that elegant though as you'll need to wire an analogue port to each fax/modem. AFAIK there isn't a virtual fax/modem provided by * that another programme can use. W - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 05, 2003 7:20 AM Subject: [Asterisk-Users] Virtual fax on the Asterisk box Hi all, I want to get the following functionality: define one extension as a virtual fax machine. Every fax redirected to that extension to be converted in a picture file (bmp/jpg/gif or something else) and then attached to an email and send to an e-mail address. Are you aware of a linux based application who does something like this and can be installed on the same computer as Asterisk? Another possibility is a Windows based software SIP fax which can be registered with Asterisk. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Bandwidth Question
Your uplink is pretty limited at both ends. I'd be using g.729 over IAX in your situation giving enough uplink for several calls, or a call and normal use at least. GSM is a bit too hungry for that kind of connection. - Original Message - From: Jay Tyndall To: [EMAIL PROTECTED] Sent: Monday, July 07, 2003 10:58 AM Subject: [Asterisk-Users] IAX Bandwidth Question Hi, I am using IAX to communicate between 2 sites, each site is using a 256k/64k ADSL Connection. I have noticed that when I connect my ping time to my 1st hop jumps from 30~ms to over 12,000ms in over a period of about 10 minutes, it just keeps climbing until the link is saturated. Naturally, there is a very long delay when speaking. What bandwidth would be adequate for IAX? or how can I tune my config to work better with my current bandwidth situation. I am using GSM codec and bandwidth=low in iax.conf Thanks in advance. Jay.
Re: [Asterisk-Users] Asterisk and Hot Desks??
The snom phones (and I assume others) allow you to have multiple SIP accounts on a single phone. The user logs in to the phone which logs in *. The downside is that you can only log in to accounts set up on the phone rather than any account set up on * but is useful for shared desks etc.. W - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 11:21 AM Subject: [Asterisk-Users] Asterisk and Hot Desks?? Hi, Has anyone worked out a way to use Asterisk in a Hot Desk environment?? I have not been able to think of a way for the user to have control over which IP phone will ring when that users extension is dialed without the user needing to reconfigure the phone.. Something like this would be cool.. User dials *8555 (or similar) and is prompted to enter their extension and then password, after successfully validating the user is then prompted for phone number (being some IP phone ID number or an external Mobile or Home phone number).. All calls made to that users DID number or extension are now routed to the registered destination device.. Any calls the user makes from any IP Phones carry the correct caller ID information as well.. Anyone got something like this or any form of user manageable extension control or hot desk type solution working?? Or any suggestions how it could be archived?? Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP phone with asterisk
For the Snom100 is a IAX Image available at asterisk's ftp site. Can you tell me more? Is it a patch to enable IAX or replacement firmware? Many thanks, Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 Licencing..
Hiya, Yes it does. The only thing to be careful of, as we learnt to our mistake, was that a single purchase gives you a single key for all and thus you cannot buy 10 licenses intending to use some on one server and some on another. I guess this would be possible by special request though. Simon - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 1:44 PM Subject: [Asterisk-Users] G.729 Licencing.. Hi, Does the G.729 module support adding more licences?? From what I understand the module generates a code that unlocks it for a given number of licences.. I would probably want to buy 2 or 3 licences to test with and then later as I needed more add then on as needed one or two at a time.. Is this possible?? Thanks -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 Licencing..
No, you can reinstall up to 3 times I believe. - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 2:11 PM Subject: Re: [Asterisk-Users] G.729 Licencing.. What if you change the hardware? The licenses are lost? Thanks, Dan - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 3:44 PM Subject: [Asterisk-Users] G.729 Licencing.. Hi, Does the G.729 module support adding more licences?? From what I understand the module generates a code that unlocks it for a given number of licences.. I would probably want to buy 2 or 3 licences to test with and then later as I needed more add then on as needed one or two at a time.. Is this possible?? Thanks -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 Licencing..
We've just moved servers and it went fine. - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 2:57 PM Subject: Re: [Asterisk-Users] G.729 Licencing.. As I understand, the key you get depend on the software hardware installation you have. If you change Asterisk to another computer (different hardware), then you still can use that codec? I have installed Asterisk on a Compaq Armada 1700 notebook (celeron/300MHz) and it works like a charm with 6 IP phones and 2 analog phones through a Cisco ATA186. I need now to add a FXO interface and for this purpose I need a system with a PCI bus. I can try the codec now on this installation (notebook) and then move it to the new system when it will be available and still keep working? Thanks, Dan - Original Message - From: Simon Woodhead [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 4:38 PM Subject: Re: [Asterisk-Users] G.729 Licencing.. No, you can reinstall up to 3 times I believe. - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 2:11 PM Subject: Re: [Asterisk-Users] G.729 Licencing.. What if you change the hardware? The licenses are lost? Thanks, Dan - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 3:44 PM Subject: [Asterisk-Users] G.729 Licencing.. Hi, Does the G.729 module support adding more licences?? From what I understand the module generates a code that unlocks it for a given number of licences.. I would probably want to buy 2 or 3 licences to test with and then later as I needed more add then on as needed one or two at a time.. Is this possible?? Thanks -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk IAX over VSAT satellite.
Hey Jim, All sounds good. We tried a satellite system here a few months ago but couldn't get on with it. Glad you've had more success. In theory, it shouldn't matter whether the TCP/IP link between your sites is going over satellite, modem or any other medium but the issues we found with satellite that would be particularly damaging for VoIP were as follows: - Latency. You're onto this one already by the sounds of it. We were seeing 750ms pings so you're looking at delays of around 1 second; 1.5-2 seconds for someone to hear what you've said and reply. That doesn't prevent a conversation but might make it sound a little strange to the other party who doesn't know what is going on. - Upstream. We had a system with 2Mbps downstream but since the upstream is the expensive part for providers to provide it is usually much much smaller - ours was only 128k. That is one call for many codecs without allowing for any other use you'll be making of the line. G.729 would improve this a lot as you've spotted. - Drop-outs. A satellite system should theoretically provide continuous service like a leased line or modem connection so you shouldn't get call dropouts. However, we found that we'd lose all connectivity from our provider for several seconds at a time. It could have been a peculiarity of the way they were prioritising traffic, routing, excessive contention or even the non-TCP/IP method for the dishsatdish part of the link but it seems whenever other customers were making heavy downloads others would slow down to just a few bps or drop out completely. That wouldn't be good for the quality of any calls in progress even if the connection was maintained. I'm not meaning to be negative or dash your enthusiasm but if I had a choice of links to do VoIP over, satellite would be at the bottom, even below modems. Our experience could be unique of course and if you own both ends of the link then you have far more control over the issues I've mentioned, other than latency of course. All the best, Simon - Original Message - From: Jim Ockers [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 7:47 PM Subject: [Asterisk-Users] Asterisk IAX over VSAT satellite. Hi all, For some reason VSAT or Satellite Internet services are not mentioned (or searchable) in this list's archives. I thought I'd let you know that I tested Asterisk using IAX (not IAX2) to make a phone call from an analog phone hooked up to an Asterisk system behind a Linksys router connected to a Gilat VSAT satmodem, and it worked. The other end (gateway) is a P200MMX with a X100P FXO card. I have bi-directional calling set up so that the VSAT-phone can make outbound calls using the X100P in the gateway, and if the X100P gets a ring it answers and transfers the call to the analog phone on the other side of the VSAT. There is about a 1-2 second propagation delay in voice from the VSAT phone, as expected. The echo is not bad at all, and the voice quality is quite good. I don't think the VSAT network was very busy so I don't know how well this will work if the available bandwidth is less. We are not using the G.729 codec - just gsm. I have tos=reliable set in iax.conf. I didn't get disconnected during my test calls, but they weren't very long in duration. I haven't tried a fax but maybe I will. Anyway congratulations Mark et al on your fine work making such a robust VoIP system. Thanks! -- Jim Ockers, P.Eng. ([EMAIL PROTECTED]) Contact info: please see http://www.ockers.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the going rate for the Snom 100 in the UK?
That's about right, inlcuding postage. - Original Message - From: nathan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 12:44 PM Subject: [Asterisk-Users] What is the going rate for the Snom 100 in the UK? Hi All, What is the going rate for the Snom 100 in the UK? I've found a couple of suppliers with prices around the £170 (exc vat) mark. Regards, Nathan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P BT
Hi Richard, So is a PBX style the equivalent of a straightforward adapter or is there more to it than that? I noticed last night that our analogue phones actually have RJ11 sockets that the BTRJ11 lead plugs into so I hooked one of them up with an RJ11RJ11 modem lead but got nothing. Am I doing something eternally dumb? Cheers, Simon - Original Message - From: Richard Alexander [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 27, 2003 11:37 PM Subject: RE: [Asterisk-Users] TDM400P BT My question, is do they need to be Master, Slave or PBX type ? The PBX style is just a Master without the test resistor surge protector - it is meant for internal cabling. A PBX style is probably fine for the first connection. If you daisy chain other connectors they should be slaves. Be aware that US and UK line levels are different. Be prepared to adjust gain settings accordingly if you connect UK style handsets. CallerID will not be displayed on BT style caller id devices. Email me for a UK phone number if you want a chat. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
