Re: [asterisk-users] Mountain ahead of me!
Hello, I want to set up a Voip Farm (c) (tm) (patent pending) but don't know how to do it. Please help. Oh, the irony :) Cheers Jean-Michel. 2009/4/2 Gabriel - IP Guys gabr...@impactteachers.com: Dear All, Thanks for taking the time to read this. I have been presented with a massive task. I'm not an asterisk expert, but I do know my way around a linux server and infrastructure, and I know when things are not done correctly. A large number of minutes are routed every month, (1m+) and I wish to do this in the most efficient way possible. I've been presented with three linux servers, all in varying states of upkeep, and I've decided, instead of attempting to clean the systems I'm presented with, it is better for me to build a stable platform for asterisk to be migrated onto. This makes my question two fold. 1 What steps should I take, or consider, if I wish to migrate an existing asterisk installation, without it being offline for too long 2 What steps should I look out for, if I wish to move to a MySQL backed for the configuration files, so that I can remove the systems dependence on local configuration. My long term plan is to introduce MySQL to be the backend for the configuration and call log data and put this machine behind a load balancer, so that in due course, when I need to add more machines to handle the load, I will have no need to reconfigure asterisk, or build new configurations, and if I keep the base OS install uniform, I should in theory be able to deploy more asterisk boxes very fast behind a load balancer to increase the capacity of my VoIP Farm with minimal work. *VoIP farm is my term, please do not use it in any presentations to the powers that be inside your organisation - If you wish to do so please send £10(ten) via paypal to my email address which is clearly displayed in the email headers!* Also, in theory, it allows for testing of new configuration, without having to change the configuration on multiple machines at the same time. Which is always a good thing. Any help an advice, or questions are most welcome, as I wish to turn this mountain into a mole hill, a very stable, and expandable mole hill! Thank you for your time, Mr Gabriel ___ -- 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 -- Jean-Michel Hiver - Synapse co-founder CTO GSM +262 692 828 070 ___ -- 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] no ringtone - just silence/bridging of external calls
Hello For the ringtone try progressinband=yes in sip.conf. I don't think you can bridge do a ringback at the same time, why not proxy the RTP and send the ringback yourself using the 'm' modifier? Cheers Jean-Michel. 2009/3/30, alex.mosbur...@orange-ftgroup.com alex.mosbur...@orange-ftgroup.com: Hi Community! If this issue was already topic, please excuse or delete my request... Topic 1 no ringtone: I configured a SIP registration with my SIP provider (SIPCall). Everything works fine except the ring tone for the caller. The caller hears silence until the called party takes up the phone. I used the DIAL command with the r and R option but no luck... :( Has anybody the same problem than me and a resolution for it? - Topic 2 external bridging: The prior approach was to bridge to external calls. An external SIP number terminates and will be re-routed back to a mobile phone number. The session was first packet2packet switched, which did not work. After setting reinvite=yes, the bridge works. Now I added 2 internal extensions to the mobile phone number in the DIAL command -- did not work (mobile phone rings but no communication possible; just silence). Topology: SIP Provider -- Asterisk -- SIP Provider -- Mobile phone /- ext 10 /- ext 20 The DIAL command was: Dial(SIP/06544564...@sipcall.atSIP/10SIP/20,,r) The aim is that all extensions and the mobile rings and the first pick up takes the call. During call setup music on hold would be good... It shows no errors in the debug of the CLI. I would appreciate if somebody could help me. Thanks, Alex * This message and any attachments (the message) are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ___ -- 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 -- Jean-Michel Hiver - Synapse co-founder CTO GSM +262 692 828 070 ___ -- 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] Relay Register
Not sure about this. It seems you are trying to find a solution to a problem which you do not actually describe. I.E, you have problem X, you think that doing Y might be the solution, but you don't know how to do Y (and in this case, neither do I). How about exposing underlying problem X to the list? Cheers Jean-Michel. 2009/3/24, cedric.bon...@orange-ftgroup.com cedric.bon...@orange-ftgroup.com: Good morning everybody. My question is simple. Is there a way to perform relay register with Asterisk ? More precisely, I want my clients regiter to a Proxy Registrar (OpenSIPS/Kamailio) through my Asterisk : REGISTER REGISTER Client Asterisk --- OpenSIPS So Asterisk keep a list of registered clients and only allows them to call and be called. Thank you for your answers. -- Cédric Bonnet /FT/RO/DPS/CTR/CPM/VASF Tel. +33 (0) 1 55 88 36 60 cedric.bon...@orange-ftgroup.com * This message and any attachments (the message) are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ___ -- 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 -- Jean-Michel Hiver - Synapse co-founder CTO GSM +262 692 828 070 ___ -- 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] Relay Register
Then, I don't know :-) Seems you are looking for a way to have a distributed architecture. The way I would do it is to let asterisk handle the registrations and then use something like ENUM or DUNDi (more likely ENUM since it's a more recognized standard) to know where the call should be going. Cheers Jean-Michel. 2009/3/24, cedric.bon...@orange-ftgroup.com cedric.bon...@orange-ftgroup.com: Hmm no, it is exactly what I want to do, not in order to solve an other problem. In a more global context, I am trying to study if asterisk can act as a Session Border Controller. If I ask Asterisk in the sip.conf file to manually register to the Proxy Registrar, it works for incoming and outcoming calls (I have an issue with IP-IP calls that I suggested to the list yesterday). But my problem for the relay register is a real issue. Cheers, Cédric. -- Cédric Bonnet /FT/RO/DPS/CTR/CPM/VASF Tel. +33 (0) 1 55 88 36 60 cedric.bon...@orange-ftgroup.com -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Jean-Michel Hiver Envoyé : mardi 24 mars 2009 12:19 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Relay Register Not sure about this. It seems you are trying to find a solution to a problem which you do not actually describe. I.E, you have problem X, you think that doing Y might be the solution, but you don't know how to do Y (and in this case, neither do I). How about exposing underlying problem X to the list? Cheers Jean-Michel. 2009/3/24, cedric.bon...@orange-ftgroup.com cedric.bon...@orange-ftgroup.com: Good morning everybody. My question is simple. Is there a way to perform relay register with Asterisk ? More precisely, I want my clients regiter to a Proxy Registrar (OpenSIPS/Kamailio) through my Asterisk : REGISTER REGISTER Client Asterisk --- OpenSIPS So Asterisk keep a list of registered clients and only allows them to call and be called. Thank you for your answers. -- Cédric Bonnet /FT/RO/DPS/CTR/CPM/VASF Tel. +33 (0) 1 55 88 36 60 cedric.bon...@orange-ftgroup.com * This message and any attachments (the message) are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ___ -- 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 -- Jean-Michel Hiver - Synapse co-founder CTO GSM +262 692 828 070 ___ -- 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 * This message and any attachments (the message) are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ___ -- 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 -- Jean-Michel Hiver - Synapse co-founder CTO GSM +262 692 828 070 ___ -- 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] call-limit on a per destination basis
Hello
OK I have tried this in my dialplan:
exten = _0262XX,1,Set(GROUP()=Reunion)
exten = _0262XX,2,GotoIf(${GROUP_COUNT(Reunion)} 24 ? 500)
exten = _0262XX,n,NoOp(This channel is member of group: ${GROUP()})
exten = _0262XX,n,NoOp(Number of channels is ${GROUP_COUNT(Reunion)})
exten = _0262XX,n,Set(SPYGROUP=1003)
exten = _0262XX,n,Dial(IAX2/dedibox-etang-sale/${EXTEN})
exten = _0262XX,n,Congestion()
exten = _0262XX,500,NoOp(Total channels congested, retuning NOCAV)
exten = _0262XX,501,Congestion()
However here's what i see on the CLI:
-- IAX2/dedibox-etang-sale-34 is making progress passing it to
SIP/5060-006edf50
-- IAX2/dedibox-etang-sale-6 is making progress passing it to
SIP/5060-007654f0
-- Executing [0262211...@route:1] Set(SIP/5060-0070b9d0,
GROUP()=Reunion) in new stack
-- Executing [0262211...@route:2] GotoIf(SIP/5060-0070b9d0, 21 24 ?
500) in new stack
-- Goto (route,0262211459,500)
-- Executing [0262211...@route:500] NoOp(SIP/5060-0070b9d0, Total
channels congested| retuning NOCAV) in new stack
-- Executing [0262211...@route:501] Congestion(SIP/5060-0070b9d0, )
in new stack
I am *totally puzzled* with this:
GotoIf(SIP/5060-0070b9d0, 21 24 ? 500) in new stack
-- Goto (route,0262211459,500)
What GotoIf 21 24 returns true
Any ideas?
Cheers
Jean-Michel.
2009/2/26, Klaus Darilion klaus.mailingli...@pernau.at:
I have no clue about IAX, but if IAX does not support it you can program
it yourself using the GROUP and GROUPCOUNT functions.
regards
klaus
Jean-Michel Hiver wrote:
Hello,
I use asterisk to to IAX2 trunking between London POP Reunion Island
pop. I would like to know if it's possible to do a kind of call-limit
(i.e. restrict to XX) channels but on a per dialcode and / or
destination basis.
For example:
[trunk]
; reunion proper, i want to send no more than 24 channels
exten = _0262XX,1,Dial(IAX2/mytrunk/${EXTEN})
; reunion mobile, i want to send no more than 12 channels
exten = _0692XX,1,Dial(IAX2/mytrunk/${EXTEN})
exten = _0693XX,1,Dial(IAX2/mytrunk/${EXTEN})
How would you go about it? Currently my IAX2 peer definition looks like
this:
# machine in london
[mytrunk]
type=friend
host=$reunion_ip
trunk=yes
qualify=yes
context=route
# machine in reunion island
[mytrunk]
type=friend
host=$london_ip
trunk=yes
qualify=yes
context=route
I use version Asterisk 1.4.11, production environment currently doing
25,000 minutes / day (that means if i want to upgrade i need to do it on
separate servers just in case something goes wrong).
Cheers,
Jean-Michel.
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--
Jean-Michel Hiver - Synapse co-founder CTO
GSM +262 692 828 070
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[asterisk-users] [FIXED] Re: call-limit on a per destination basis
The correct syntax for GotoIf is:
exten = _0262XX,n,GotoIf($[${GROUP_COUNT(Reunion)}24]?500)
Otherwise it seems to evaluate the string number 24 which is always
true.
Duh...
Thx
JM
--
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GSM +262 692 828 070
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[asterisk-users] [HOWTO] Priorize one destination over another on a link
Hello List,
The list sorted my problem thus I shall contribute back ;-)
PROBLEM:
I am posting this example, where I have a Reunion link of 30 channels. If
i send all the traffic (proper + mobile) on the link, the less profitable
proper traffic fills the link and leaves no channel for more profitable
mobile traffic. Some kind of priority is needed to always leave space for
mobile trafic: you don't want to be terminating traffic that yields 0.001 /
min of profit when you could be terminating traffic yielding 10x as much
instead.
SOLUTION
Use asterisk grouping and conditional fonctions to dynamically limit the
proper traffic in order to always keep a few channels free.
For example, imagine you have 0 channels of mobile : allow proper to use up
to 26 channels
For example, imagine you have 5 channels of mobile : allow proper to use up
to 21 channels
For example, imagine you have 10 channels of mobile : allow proper to use up
to 16 channels
For example, imagine you have 28 channels of mobile : allow proper to use up
to 0 channels
In order to do this, you set the SAME group for both mobile and proper
channels and then you apply a conditional only on the traffic which you want
to limit.
You could also be using the same kind of technique if you had two different
classes of customers: retail and wholesale. You want wholesalers to fill
your pipes of course (with best effort), but you do not want this traffic to
affect your retail service.
IMPLEMENTATION
==
This is the implementation on my production server, seems to work well, feel
free to modify to suit up your needs.
; Reunion Proper : use a conditional statement to dynamically limit number
of channels
exten = _0262XX,1,Set(GROUP()=Reunion)
exten = _0262XX,n,NoOp(This channel is member of group: ${GROUP()})
exten = _0262XX,n,NoOp(Number of channels is ${GROUP_COUNT(Reunion)})
exten = _0262XX,n,GotoIf($[${GROUP_COUNT(Reunion)}26]?500)
exten = _0262XX,n,Dial(IAX2/dedibox-etang-sale/${EXTEN})
exten = _0262XX,500,NoOp(Total channels congested, retuning NOCAV)
exten = _0262XX,501,Congestion()
; Reunion Mobile : always gets through which increments the channel count...
and thus reduces proper capacity
exten = _0692XX,1,Set(GROUP()=Reunion)
exten = _0692XX,n,NoOp(This channel is member of group: ${GROUP()})
exten = _0692XX,n,NoOp(Number of channels is ${GROUP_COUNT(Reunion)})
exten = _0692XX,n,Dial(IAX2/dedibox-etang-sale/${EXTEN})
exten = _0693XX,1,Set(GROUP()=Reunion)
exten = _0693XX,n,NoOp(This channel is member of group: ${GROUP()})
exten = _0693XX,n,NoOp(Number of channels is ${GROUP_COUNT(Reunion)})
exten = _0693XX,n,Dial(IAX2/dedibox-etang-sale/${EXTEN})
I hope this piece of information is of use to somebody, some day!
Cheers
Jean-Michel
http://ykoz.net/
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[asterisk-users] call-limit on a per destination basis
Hello,
I use asterisk to to IAX2 trunking between London POP Reunion Island pop.
I would like to know if it's possible to do a kind of call-limit (i.e.
restrict to XX) channels but on a per dialcode and / or destination basis.
For example:
[trunk]
; reunion proper, i want to send no more than 24 channels
exten = _0262XX,1,Dial(IAX2/mytrunk/${EXTEN})
; reunion mobile, i want to send no more than 12 channels
exten = _0692XX,1,Dial(IAX2/mytrunk/${EXTEN})
exten = _0693XX,1,Dial(IAX2/mytrunk/${EXTEN})
How would you go about it? Currently my IAX2 peer definition looks like
this:
# machine in london
[mytrunk]
type=friend
host=$reunion_ip
trunk=yes
qualify=yes
context=route
# machine in reunion island
[mytrunk]
type=friend
host=$london_ip
trunk=yes
qualify=yes
context=route
I use version Asterisk 1.4.11, production environment currently doing 25,000
minutes / day (that means if i want to upgrade i need to do it on separate
servers just in case something goes wrong).
Cheers,
Jean-Michel.
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Re: [asterisk-users] IAX2 trunking scalability
Hi, I thought I'd give a follow up to this discussion for the archives... Currently I'm trunking 30 channels of g.729 traffic (no transcoding going on, the call comes in and goes out as g.729) and it takes about 350 kbps bandwith bidirectional. So on average each call takes 11.5 - 12 kbps of bandwith. The solution seems stable and the QoS is identical... so for the price (2 commodity PCs...), IAX2 trunking is well worth the effort since it reduces bandwith usage by a factor of 2. Cheers, Jean-Michel. -- Jean-Michel Hiver - YKOZ +262 (0)692 828 070 ___ --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] IAX2 trunking scalability
I used to do it, but its a while ago. (Before iax2 got some more fixes) The trick was to keep the trunks small (like 40 per trunk, just make multiple), this should no longer be needed. Cpu utilisation with trunking should be lower than without trunking. Hi Zoa, Thanks for your input. I think I'll set up two boxes and do failover + loadbalancing, just in case one box decides to crash =) Cheers, Jean-Michel. ___ --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] IAX2 trunking scalability
Le Sun, 26 Aug 2007 20:20:01 +0400, Andrew Joakimsen [EMAIL PROTECTED] a écrit: On 8/25/07, Jean-Michel Hiver [EMAIL PROTECTED] wrote: I'm already receiving the calls as g.729, so there is little gain (slightly less bandwith usage, slighly worse sound) in doing g.729 - g.723 transcoding - while doing IAX2 trunking vs NOT doing it seems to half the bandwith requirement. Main point was CPU usage. Even if you receive the calls as G729 then you need to transcode to Alaw and transcode all the alaw from the E1 to G729. Unless somehow you have an E1 using G729! I do. I use a variety of audiocodes, patton, and quescom SIP gateways. All support g.729. Cheers, Jean-Michel. -- Jean-Michel Hiver - YKOZ +262 (0)692 828 070 ___ --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 as a softswitch
Le Fri, 24 Aug 2007 20:50:05 +0400, Mark Quitoriano [EMAIL PROTECTED] a écrit: What is a good softswitch that is also open source rather than asterisk? You may want to check out freeswitch. ___ --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] IAX2 trunking scalability
So you are using an asterisk box as an E1 gateway. You want to know if switching from not using IAX trunking to using IAX trunking will have any effect? Yes it will lower your bandwidth usage a little. It will not increase the CPU load. If your system can support x calls it will be able to support the same amount of calls. On about 1/2 E1, it shows that bandwith usage has been about halved - i.e. without trunking each G.729 call takes 50 kbps (inbound + oubound) and with IAX2 trunking it takes about half of that (using trunkfreq=40). Which is good! I'm wondering wether anybody already had a IAX2 trunking ON and managed to push 3 E1s worth of traffic without issues. The best thing you can do for your system is add a TC400B card. It will also legally support G723 codec which I think sounds just fine, but will save you a bit more bandwidth. Using the hardware transcoder will greatly increase the number of calls your system would be able to handle. I'm already receiving the calls as g.729, so there is little gain (slightly less bandwith usage, slighly worse sound) in doing g.729 - g.723 transcoding - while doing IAX2 trunking vs NOT doing it seems to half the bandwith requirement. Cheers, Jean-Michel. ___ --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] IAX2 trunking scalability
Hi List, I have a 2Mbps SDSL link which gets saturated during peak time because about I have about 3 E1 worth of g729 traffic going thru. So I'm planning to use IAX2 trunking to reduce bandwith requirement and squeeze out each and every bit of this (expensive) bandwith. I've set up two boxes (debian etch), one in a remote data center (which has plenty of bandwith) and one behing the SDSL link. To make things consistent I've installed the same kernel, latest stable zapata + asterisk on both ends. I've done some tests with about 1/2 E1 (15 channels) worth of calls and so far it's been working good - and the call statistics (ASR, ACD, PDD) are roughtly the same. So far, so good! Now the big question is: how far can I expect it to scale? Has anybody successfully mounted IAX2 trunking with 3-4 E1s worth of traffic? Your experience and feedback is appreciated. Cheers, Jean-Michel. ___ --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] Re: [asterisk-biz] Server for 100 concurrent calls
[EMAIL PROTECTED] a écrit : Hi all, I'm looking at some suggestions from you techies out there. Let me explain my scenario. Im a reseller to callshops. I need to take around 100 concurrent calls. Almost all endpoints are sending G723 codec and my peers take G729. Since Digium doesn't provide g723 codecs (as far as I'm aware), and there's yet no transcoding card for Asterisk (one is supposed to be out at some point, but when... god knows), for the moment you should look into something else than Asterisk. Cheers, Jean-Michel. ___ --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] Setting up FastAGI in Asterisk?
Bret Schuhmacher a écrit : Hi - is support for FastAGI built in by default or do I need to configure anything within Asterisk to make it understand how to call FAGI scripts in the dialplan that contain agi://localhost/myFAGI.agi? Works pretty much out of the box :-) ___ --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] 1 FXO termination device
Hi List, I am looking for a 1 FXO analog termination device, other than the obvious PC + FXO card, and which can achieve decent call quality. The SPA-3000 seems an option... have you got any other ideas? Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asterisk-dev] open letter
[EMAIL PROTECTED] a écrit : Dear asterisk users, the asterisk projects and ser enabled me to learn SIP, I could be insulting sometimes . I must begin my business with communigate and a French company. I consider it regrettable that asterisk and ser could not do it. I do hope that these open projects will help many people. Good luck to all. Warning, big hairy trolls are coming... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asterisk-dev] open letter
[EMAIL PROTECTED] a écrit : Dear asterisk users, the asterisk projects and ser enabled me to learn SIP, I could be insulting sometimes . I must begin my business with communigate and a French company. I consider it regrettable that asterisk and ser could not do it. I do hope that these open projects will help many people. Good luck to all. Warning, big hairy trolls are coming... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --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] callback without agi
Patricio Valarezo a écrit : Hi, it's possible to implement a callback without agi?, i'm trying this but * exits without dialing (if I hungup during s,3 wait) but if it hungs in s,4 it dials, so is there an explanation to this behavior? there is an alternative to do it? just for learning Sorry to ask, but what's wrong with AGI? Cheers, Jean-Michel. ___ --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 -- echo cancellation of an audio file
William M Conlon a écrit : I recorded an internet radio program using iTunes, and somehow got an echo. Anyone have any suggestions on how to remove echo from an existing file? Convert it to gsm, send it through asterisk (by calling yourself), activate echo canceller, and record the call? Note that you might end up with crappier audio doing that... :) ___ --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] Codec Thread
3) The G723 codec also does VAD (which Asterisk doesn't support). Shame it doesn't... if you could do IAX2 trunking with g723 5.1kbps + VAD, that'd be awesome for narrow links (which is very common in developing countries). ___ --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] Can not hear the telco System Announcement
stoffell a écrit : On 9/1/06, Xue Liangliang [EMAIL PROTECTED] wrote: Hi, all. I am from Singapore, we deployed a few PABX based on Asterisk. Here in Singapore there are two Teleco providing E1 pri service, we encountered a strange problem : when calling a number that is unavailible or line suspended, one of the E1 provider keep the call ongoing, because there are system announcement like The line currently I have something similar on a european E1. I do think this has something to do with the PBX.. (asterisk in this case) I have the same 'issue' on a BRI (ISDN) interface. The 'old' PBX (a classic PBX) did sent out the telco announcement. I have tried changing priindication, but this didn't help. I can see the hangup_cause and can play prompts according to the hangup_cause, but I would prefer using the telco announcement. Have you tried progressinband=yes? As far as understand it, it forwards early RTP (that is, stuff that is received prior to the ANSWER), which might just do the trick. I had this working when interconnecting with Chile mobile. When somebody is on the line, they have some music and a message in spanish! Needless to say, with g729 the music part sounds pretty awful (in fact it already sounds awful with g711 anyway...) NB: As far as I can tell, progressinband=yes isn't supported in chan_h323, which is a shame :( Cheers, Jean-Michel. ___ --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 codec strangeness
Hi All, I have two peers (call then peerA and peerB) on my server, both can accept g711, g729 and g723. However, when peerA initiates a request, asterisk decides to transcode g729 into ulaw when peerB could very well use g729... This behavior isn't very scalable (transcoding is CPU expensive) and also it's better to minimize the amount of transcoding wherever possible. Is there a way I can fix this? NB: if i set disallow = all and allow=g729 on peerB it all works fine, but then if peerA decides to send ulaw I'm transcoding again... Cheers, Jean-Michel. ___ --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] Connecting two asterisk servers
Crazy Boy a écrit : Hi friends, Thank you to all for your response and cooperation to me. I have a doubt. I have two asterisk servers and contains two public IPs. One * server is in Florida (USA) and second * server is in Delhi (India). 1) Is it possbile to connect these two * servers? Yes. Just have something like: [serverA] type=peer host=serverA.IP.Address In ServerB's sip.conf and [serverB] type=peer host=serverB.IP.Address In ServerA's sip.conf 2) The person who is registered with Florida * server is able to make call to another person, who is registered with Delhi * server (like Intercom)? Of course. Say user joe is registered with serverB, then within serverA's dialplan, you can use: exten = 123456,1,Dial(SIP/[EMAIL PROTECTED]) ; [EMAIL PROTECTED] has extension '123456' Within serverB's dialplan, you'd simply use: exten = 123456,1,Dial(SIP/joe) ; [EMAIL PROTECTED] has extension '123456' Cheers, Jean-Michel. ___ --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 codec strangeness
Jean-Michel Hiver a écrit :
Hi All,
I have two peers (call then peerA and peerB) on my server, both can
accept g711, g729 and g723. However, when peerA initiates a request,
asterisk decides to transcode g729 into ulaw when peerB could very
well use g729...
This behavior isn't very scalable (transcoding is CPU expensive) and
also it's better to minimize the amount of transcoding wherever
possible. Is there a way I can fix this?
NB: if i set disallow = all and allow=g729 on peerB it all works fine,
but then if peerA decides to send ulaw I'm transcoding again...
Okay, I have digged the archives a bit, and apparently I'm not the only
one having this problem. I am thinking of maybe sorting out this problem
by having:
[peerA-g711]
type=peer
host=123.123.123.123
disallow=all
allow=ulaw
allow=alaw
[peerA-g729]
type=peer
host=123.123.123.123
disallow=all
allow=g729
[peerA-g723]
type=peer
host=123.123.123.123
disallow=all
allow=g723
And then using ${SIP_CODEC} to route the call correctly maybe?
I don't think having multiple peers with the same IP address would be a
big deal for outgoing calls, but asterisk will probably we confused for
incoming calls from 123.123.123.123... what do you think?
Cheers,
Jean-Michel.
___
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Re: [asterisk-users] Asterisk codec strangeness
Mojo with Horan Company, LLC a écrit : are your codec allow= statements in the same order in each peer block? meaning does peerA have g729 at a different priority than peerB? Aah, thanks that fixed it because most of the traffic is g729. Now, if peerA does send me ulaw instead of g729 (because it choose to, say), and the order of peerB is g729, ulaw, alaw, am I still going to have the same issue? My guess is yes... Cheers, Jean-Michel. ___ --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] H264
Tomislav Parčina a écrit : As far as I can see on this web page http://www.voip-info.org/wiki-Asterisk+video Asterisk doesn't support h264 codec. I can see the same on this pages http://www.asterisk.org/features Question is, can I somehow enable H264 codec support in Asterisk? I have Grandstream GXV-3000 video IP phone which supports only h264 codec. Right now I can make only direct IP video phone calls, and I would like to make calls true Asterisk. Try to add videosupport=yes under [general] in sip.conf ? Cheers, Jean-Michel. ___ --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
Assuming you use Perl for AGI scripting, which you should be doing anyways ;-) *cough* You made a typo... you really meant to say 'python'. :P flame Python is to Perl what Pascal was to C. A nice toy ;-) /flame ___ --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: what is the real use of AEL?
In the above, Jean-Michel puts it right on the table: of what
possible use is AEL? Why am I bothering to waste my time with it? It's a
valid question! It deserves some discussion!
First of all I'd like to thank for all the good answers and valid points
people have made to this question.
snip/
Sorry for the diversion. My answer to Jean-Michel's straightforward
question goes along some different lines than rushowr. I never really
cared how fast/efficient the extension engine was-- it's obvious I'm
not writing stuff for thousands of concurrent users like rushowr. But in
the majority of cases, it's the apps that are run from AEL that take up
all the execution time. As long as AEL execution time is pretty minimal
between priorities, it's probably going to be OK. (users of dialplans
for intensively loaded sites may HIGHLY disagree!)
I admit it, I haven't done the test. Using FastAGI is just enough speed
for me :-)
My first reason for getting excited about AEL, enough so, to rewrite the
parser to make it more user-friendly, and add a few bells and whistles,
was that it provided an opportunity to code dialplans with higher
level constructs than gotos. Truly, AEL is to extension.conf format,
as programming languages are to assembler.
While I can see that it is a nice thing to have, in my opinion it makes
configuration files look less and less like configuration files and
more like a programming language. And I don't know, I doubt that mixing
these two things look very good.
Now to be honest, I'll probably be using AEL (assuming it's there to
stay) for complicated dialplan constructs, but #includes are going to be
good to avoid having a conf file that looks like a mix of apples and
potatoes :)
But, in this, Jean-Michel
is right to ask: we already have several possible programming languages,
perl, C, java, PHP, ruby, and so on. The one advantage AEL holds over
all of them, is that its structure parallels the data model used in
Asterisk. That data model is composed mainly of contexts, extensions,
and priorities. Because AEL follows that structure, it is easier to
write dialplans than it would be in other languages. Most AGI scripts
don't have to deal with anything above the priority level... and if
you do want to generate an entire dialplan from the innards of a perl
script, I doubt it would easier to read and understand than an AEL
script, nor do I think it would be anywhere near as concise.
That's where I beg to differ. How is the code snippet more concise /
readable than, say:
#!/usr/bin/perl
use FictionalAPI; # imports NoOp, Verbose...
sub loop {
my $iterations = 100;
my $time1 = time;
NoOp('hello') for (1..$iterations);
my $time2 = time;
my $diff = $time2 - $time1;
my $prisecs = 4 * $iterations / $diff;;
Verbose(The time diff is $diff seconds);
Verbose(Which means the priorities/sec = $prisecs);
SayNumber($prisecs);
}
That assumes you have FictionalAPI, which is why I highlighted the need
to clean, well defined API and good IPC communication between Asterisk
and external systems / program.
The next reason I spent time on it was code quality. There is no lint
yet for extensions.conf. We've seen little things like misspellings
of exten = into extem = silently drop that priority, which may
take months to spot and fix. Not that a thorough linter couldn't be
written, but I did add tons of checks to the AEL parser, to spot common
errors at compile time, rather than find them at run-time in a
production dial-plan.
Now that's awesome. The OCamel language has a pretty crazy type infering
compiler (i.e. the compiler infers the type of variables from their use
in the code rather than a declaration) you might want to take a look at.
This is as good as type checking goes IMHO :)
Haven't we all taken a course on software
quality, or read some articles at least? How much do errors cost
if found at compile time, as opposed to the cost of finding them at
run time? The earlier the phase at which errors are found and
eliminated, the cheaper the error. Right? and so, AEL is a tool for
you to reduce your costs of generating dial plans. I might add here,
that other languages also do similar checks at parse time; but some of
the checks that AEL will do, are specific to the underlying data model.
You won't necessarily get those kind of checks out of perl or php.
Valid point. However, Perl has a nice fix: the Test::More suite and
things like Mock::Object, which let you write pretty comprehensive test
suites to do some kind of code quality (any decent CPAN module has a
test suite...). That being said, you are right, type checking is
important and it saves time. I wish Perl had Ocamel parser / compiler
features :)
[BTW, haven't you ever stopped, after you have finished writing a
dialplan, just as you are about to put it in production on a live
asterisk server with tons of
Re: [asterisk-users] Call file do 2 outbound call
Daniel Hikel a écrit : Hello, I am not so really familar with asterisk at the moment, but i am working hard on it. Please could anybody advise me how to write a call file for the queue to do 2 outbounds call and connect both via my SIP interface. Why not write two .call files if you want two calls? ___ --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
Douglas Garstang a écrit : 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... Apache is an HTTP server, not a generic TCP server. Plus, using Perl's Net::Server is really easy. Besides, if you don't need to use AGI on a separate box, you can use perperl and get very fast execution times without having to worry about using a server at all. Assuming you use Perl for AGI scripting, which you should be doing anyways ;-) Cheers, Jean-Michel. ___ --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] NuFone chan_h323
Hi List, I would like to know if there is an option similar to progressinband=yes (which as I understand, forwards early RTP) with chan_h323. Any ideas? Cheers, Jean-Michel. ___ --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] Recent additions to the Digium Asterisk development team
Steve Murphy joined our development team at the beginning of June. Steve (murf on IRC/Mantis) had rewritten Asterisk's expression parser and the AEL language parser as a volunteer community member, along with various other bug fixes and improvements. Which makes me think, what is the real use of AEL. I have taken a look at it, and it makes asterisk's config file almost as unreadable as SER. What exactly does AEL do that a well written AGI / FastAGI app doesn't? I would think (but I'm surely wrong) that it would be better to do work on having well defined APIs that allow us to script Asterisk (such as AGI and the Manager interface) rather than invent Yet Another Pseudo Programming Language - which is going to be an endless task... Don't you think? That being said, just like the rest of the community, I'm very happy with Kevin's exciting announcement! Cheers, Jean-Michel. ___ --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] High Availability with PRI failover
Anybody has first-hand experience with any (or both) of these options? Are there any other possibilites that I'm missing? Some other foneBRIDGE-like product I still haven't heard of? Thanks in advance Another option would be to get a carrier grade VoIP - PRI gateway. I have an Audiocodes 4 E1s (extensible to 8 E1s) working here, and it'll use whatever channels are up so if I have a red alarm on one of the T2s it's not a big deal. Features power redundancy and network redundancy as well, which is nice. Built in codec translation and adequate echo cancellation is a plus. It also means you can add more servers without adding PRI cards (since you communicate with the gateway using SIP). Of course this doesn't protect you from the gateway going down, but usually these things are fairly robust, solid state devices... they're a little bit expensive, but in my experience worth every penny. Cheers, Jean-Michel. ___ --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 and NAT
Lincoln Zuljewic Silva a écrit : Hello all. I'm having a little problem here with NAT, and I already read a lot of documentation on web, but I still cant understand how to get asterisk and external (on internet) sip clients connected. So you have an Asterisk that is behind NAT, and you want to connect it to other NATted devices? Cheers, Jean-Michel. ___ --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: Hints to help me debug cdr_odbc not inserting
Tomislav Parčina a écrit : In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... How can I see why asterisk isn't attempting an insert ? Or if it is, why don't I see any errors? Does the user asterisks connects the db with have the proper permissions on the cdr table? Have you checked that the table specified in cdr_odbc.conf exists? What do you see when you go on the Asterisk CLI, type set verbose 10 and attempt a call? Cheers, Jean-Michel. ___ --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 do you harden an Asterisk install?
shadowym a écrit : I remember reading a small write up somewhere. I think it was on the Asterisk Wiki. I can't find it anymore. It's probably a bit dated by now but some of it would still be relevant. Can anyone recommend a good guide or even some of their own suggestions. Maybe use a solid-state fanless computer, with no moving parts? It means a low power consumption CPU (probably Via), a good thermal design, and a solid state disk (flash disk or CF + Adapter). Cheers, Jean-Michel. ___ --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 do load balancing (1:1) with IAX and two different ISPs
Ken Dresdell a écrit : Hello folks, Does anyone have an idea how I could setup a load balancing (1:1) solution with IAX and two different Internet service providers. The idea is to increase the bandwidth between offices with cheap Internet access (DSL/Cable). Do you want to load balance all your LAN / WAN traffic or just VoIP traffic? Cheers, Jean-Michel. ___ --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] 1000s of extensions in one context?
Roger Schreiter a écrit : Hi, is several 1000s of extensions in a context a problem? Not at all in my experience. Cheers, Jean-Michel. ___ --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] CDR Call Status
Hi List, When using cdr_csv, the call status are plain strings, i.e. NO ANSWER, ANSWERED, BUSY, etc. However, when using cdr_odbc, the call status are integer. Is there some docs somewhere that would let me know what the integers map to? Cheers, Jean-Michel. ___ --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] g729.1 + g723.1 codec conversion
Hi List, I'm a little bit annoyed with digium's g.729 licenses as they don't work out of the box on my FreeBSD platform, plus I need g.723 codec conversion anyway. Is there any software (even commercial would be OK) or dedicated hardware which does this, and potentially echo cancellation also? I'm looking at capacities ranging between 30 and 120 channels. Cheers, Jean-Michel. ___ --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] Limiting a group of phones available channels
Hi List I have 10 separate SIP phones, and I wish to limit the simultaneous available channels to 5 maximum for these. How would you go about it without setting up a separate * box? 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
Re: [Asterisk-Users] Most stable Asterisk version
shadowym a écrit : Hi there, I am getting ready to set up a production Asterisk system. It needs to be stable. Upgrading, patching, rebooting, troubleshooting etc. are pretty much NOT an option once this thing is deployed. Like any phone system, it is expected to just work. Try FreeBSD's Asterisk port. It has been working rock-solid for me so far. It's been a few weeks now with no issues (fingers crossed)... But I admit that it does _JUST_ softswitching (i.e. call routing, load balancing and database CDR collection) and hence has the smallest possible feature set (search google: voip-info asterisk slimming). Another option is to buy Digium's commercial edition of Asterisk, which is supposed to be just what you describe. Best Regards, 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
Re: [Asterisk-Users] Echo Cancellation
Douglas Garstang a écrit : General question. If you install a Digium card in an Asterisk system, and install zaptel drivers, do this give any benefit of echo cancellation? Our PSTN gateway is a separate Audiocodes box, so the zaptel card wouldn't actually be connected to anything. I'm wondering though doing this would help, in general, with echo cancellation. a) No it won't unless you connect it to a TDM circuit b) I have an audiocodes too (mediant 2000 4E1), and I've found the echo cancellation to be superb. I'm surprised to see you're having issues! 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
Re: [Asterisk-Users] Echo Cancellation
a) No it won't unless you connect it to a TDM circuit b) I have an audiocodes too (mediant 2000 4E1), and I've found the echo cancellation to be superb. I'm surprised to see you're having issues! Me too, given we're on fiber gigabit ethernet, with only a few test calls in progress! It's strange. I'm on 100 Mbps network with tens of thoushands of minutes per day. Pehaps it needs to warm up - or maybe your network is just too good :) More seriously, since this hardware is carrier grade (translate: quite expensive), if I was you I would contact / enquire audiocodes about it. There is probably an option that isn't set somewhere. 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
Re: [Asterisk-Users] Signaling and media
Johansson Olle E a écrit : 26 jun 2006 kl. 07.10 skrev Martin Joseph: On Jun 25, 2006, at 4:11 PM, Jean-Michel Hiver wrote: Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? SIP streams are signalling... Sorry, I was talking about the media. Have you tested the ACL features in sip.conf - accept/deny ? Any pointers on these ACLs? -- 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
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? Quel bordel, sacrebleu! -- 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
[Asterisk-Users] Signaling and media
Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? Thanks, 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
Re: [Asterisk-Users] Don't use CDRTool From AG-projescts
[EMAIL PROTECTED] a écrit : hello to all, I advice you to not use CDRtool from ag-projects : Fisrt ag-projects talk about is product like a gpl software however they don't provide at least some documentation for non commercial users . And they don't have to. I have not dealt with them before, nor do I have any connections with them, yet I find your libellous accusations completely unacceptable. Regards, 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
Re: [Asterisk-Users] AGI: Dial and Recording my own CDR
Peter Beckman a écrit : Hi folks -- I have a FastAGI Perl script running, handling calls. It works great. At one point I have a Dial() command. If the called party hangs up, Dial() returns 0, and when I call my own recordCdr() function using the channel variables ANSWEREDTIME, DIALEDTIME and DIALSTATUS, everything is fine. However, if the called party picks up, and then the dialing party hangs up Dial() returns -1, ANSWEREDTIME and DIALEDTIME == 0 (or something like that) and DIALSTATUS returns AGI::No Response. How do I make sure to get the right billing information if the dialing party hangs up? I think you're supposed to use DeadAGI rather than AGI. 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
Re: [Asterisk-Users] IAX FXS.. Any experience with...
Paul Hales a écrit : Steve Jones wrote: http://www.x100p.com/products_2.htm Anyone ever use this box? How’s it compare with the Iaxy? I’d like to buy one or the other.. The Iaxy is appealing because to me, it seems less “no name”, but this one says that it supports using hostnames, whereas apparently the iaxy only supports IP addresses?? That’s appealing to the dynamic DNS guy in me! J It works fine. I've tried both, and I'd say this FXS beats the IAXy hands down. Good stuff. 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
Re: [Asterisk-Users] AGI: Dial and Recording my own CDR
So if I change my AGI call from AGI to DeadAGI, when the caller hangs up, the variables I mentioned above will pretend like the call is still going, and record the right info? Hey, worth a shot. Yep, I think so. Let us know if it works with FastAGI... I've never tried FastAGI before but it's definitely something I'm going to look at! 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
Re: [Asterisk-Users] ODBC cdr tearing my hair out
Julian Lyndon-Smith a écrit : svn trunk. I'm trying to get cdr to work with my odbc database. I have followed a checklist that I had previously but still can't get it to work. There are no errors (verbose 40 and debug 40), I get [cdr_odbc.so] = (ODBC CDR Backend) == Parsing '/etc/asterisk/cdr_odbc.conf': Found Hi, Do you see a message on the CLI saying: cdr_odbc: Query Successful! ? If not, what do you see? *CLI cdr status CDR logging: enabled CDR mode: simple CDR registered backend: cdr-custom CDR registered backend: csv CDR registered backend: cdr_manager As I said there are no errors, but the cdr odbc does not show up :( Have you got: ; Database Call Detail Records load = cdr_odbc.so ; ODBC CDR Backend - Requires N/A In your modules.conf? what is really strange is that I have also set up the same odbc database for func_odbc, and registered my custom SQL functions and can access these (the db manager shows that this session is connected) cdr.conf == [general] enable=yes cdr_odbc.conf == [general] dsn=mydsn username=myuser password=mypassword loguniqueid=yes dispositionstring=yes table=PUB.cdr ;cdr is default table name usegmtime=no ; set to yes to log in GMT I don't know if it will help, but I will share my config with you. I can get realtime sip_friends, iax_friends, and CDRs to work. For some reason, no realtime extensions though :-( I use unixODBC. /usr/local/etc/asterisk/cdr_odbc.conf [global] dsn=PostgreSQL username=asterisk_db password=* loguniqueid=yes usegmtime=yes /usr/local/etc/odbcinst.ini [PostgreSQL] Description = PostgreSQL driver for Linux Win32 Driver = /usr/local/lib/libodbcpsql.so Setup = /usr/local/lib/libodbcpsqlS.so FileUsage = 1 /usr/local/etc/odbc.ini [PostgreSQL] Description = Connection to asterisk_db Driver = PostgreSQL Trace = Yes TraceFile = /tmp/sql.log Database= asterisk_db Servername = localhost UserName= Password= Port= 5432 Protocol= 6.4 ReadOnly= No RowVersioning = No ShowSystemTables= No ShowOidColumn = No FakeOidIndex= No ConnSettings= I must be missing something really really obvious here and would appreciate any help I can't see exactly what your problem is. I used to have the same problem but then realized that cdr_odbc was doing cdr logging GMT and I was selecting yesterday but in my time zone and thought there was a problem was actually everything was working fine :/ Good luck! 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
Re: [Asterisk-Users] T1 Copper or T1 Fiber Line
[EMAIL PROTECTED] a écrit : Thanks for the inso... So T1 lines in the United States also use copper lines from the company to the telephone exchange in some installations? What's the benefit to the subscriber to this? I don't think there is any difference. The E1s I've got at home are brought with copper HD-SDSL and they work just fine. 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
[Asterisk-Users] Echo Cancelling VoIP traffic
Hi List, I know that the zaptel modules have echo cancellation, but is this possible to do this on VoIP - VoIP traffic as well? I'm toying with a SIP gateway which has apparently a terrible call quality and would like to know if there is any way asterisk can help with this. 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
Re: [Asterisk-Users] OPENSER / SER and Asterisk
Santosh Rao a écrit : asterisk has a extremely cool documentation. The wiki has everything a newbie like me could hope for.. with samples and everyhting./. where as we are having a very dificult time finding proper documentation or samples and stuff like thtt for SER.. may be if someone good with SER could update ther voip-info/wiki and write some basics abt the ser.cfg or somethjing .. then it would be great. You can find some very good SER tutorials on onsip.org. You need to subscribe though, but it's free. 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
Re: [Asterisk-Users] OPENSER / SER and Asterisk
voip-info/wiki and write some basics abt the ser.cfg or somethjing .. then it would be great. I haven't read the tutorials, so I could be wrong, but I doubt they'd be very much use. They probably don't do more than give a basic overview, and I'm sure they don't touch things like avpops. Yeah, but he mentioned he wanted some basics about SER. So... Other than that, I agree, SER's documentation is terrible. What SER really needs is a wrapper around it that lets you write Asterisk dialplans (with a very minimalistic set of commands) and converts it into a ser.conf file. Now that would be terrific :) 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
Re: [Asterisk-Users] VOCAL + Asterisk
Akpome Akpoguma a écrit : I want to start a community based voip network projcet and am thinkimg of using VOCAL and asterisk gateways. my question is, has anyone bench marked asterisk vs VOCAL? is it a wise idea to use VOCAL + Asterisk or Asterisk all the way.am expecting 1000 - 5000 users.. Apparently the way to do it is to use SER to handle all the SIP fluff (REGISTERs mostly) and then use Asterisk as a gateway for PSTN access. I used to do it but then realized that I didn't need it since I work mostly with wholesalers and don't have that much SIP signaling to deal with anyway. Then after that, you can use Asterisk simply as a B2BUA. As long as you don't do any transcoding, you will be fine. Currently I have 15-20k minutes daily going through an Asterisk box which just does two things: - Keep CDRs records in a database (using cdr_odbc) - Does dialplan functions (prefix manipulation, access control using contexts), load balancing (to balance traffic between multiple gateways, using Macros and Random()), and least cost routing. The machine is rather low-end (Sempron 2400+, 1 Gb RAM) but the load average is only about 0.2 and the CPU usage is around 10%. So for a low price tag of around €500 per unit, I can easily afford to have a second machine which can quickly take over if the main is down. All the transcoding and the echo cancellation is being handled by proprietary SIP VoIP gateways such as Audiocodes (excellent hardware, I recommend it). I use FreeBSD + Postgresql + unixODBC + Asterisk. I have set up a minimal Asterisk (use autoload = no in your modules.conf) and Asterisk's memory footprint is around 28M. The machine doesn't swap or hang... after a lot of research it seems I've found the combination which works for me! It took me two days + one sleepless night to set up though :) 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
Re: [Asterisk-Users] extensions.conf
Douglas Garstang a écrit : -Original Message- From: Moises Silva [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 1:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] extensions.conf No limit in code imposed. Not sure about performance penalty for a file that big, have you considered using ARA (Asterisk Realtime Architecture)? On 13 Jun 2006 21:06:52 +0200, andrutto [EMAIL PROTECTED] wrote: Hi Does anyone know how big extensions.conf can be? I am trying to set up Asterisk which will have about 45 000 lines in extensions.conf. Is there any limitation about the amount of lines in that file? Write a perl script that generates a mock 45,000 extensions.conf file, with 45,000 incrementing extensions, throw in a couple of contexts. Start Asterisk and see what happens. Actually i've done 50,000+ line dialplans using my Asterisk::LCR dialplan generator, and asterisk has been just fine with it. 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
[Asterisk-Users] Detecting gateways which time out
Hi List, I would like to know if there is a way to detect gateways which time out (because of network problems or hardware failure for instance) when you send traffic to them. So when you do: Dial(SIP/[EMAIL PROTECTED]) If a call couldn't get through because the gateway has timed out, i want to do something about it. The idea would be to suspend gateway which time out for 60 minutes, and then if calls still don't go through, suspend them permanently and send an email alert. Any ideas? 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
Re: [Asterisk-Users] Callback Application: Suggestions Please.
Tigran Kocharyan a écrit : Dear Asterisk Comunity, I'm thinking about developing a callback application based on the following scenario: 1. Customer Calls the outgoing number which is a PSTN line connected to my Zap channel 2. Asterisk captures the Caller ID and calls back the customer. 3. As soon as the customer picks up the phone, asterisk plays a promt to enter the Destination number. 4. Asterisk Connects the Outgoing number through another channel (SIP/IAX/ZAP) and bridges the call. 5. After the completion, I should see the Disconnect Reason and the Duration for each leg of the call. The first two steps are quite evident. Now the trick comes on step 3. How to Dial out a number and listen for DTMF tones? http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out It looks like you want to do a calling card application. Why not try astcc.agi? It works great... After this, maybe park the call, or send it to conference room, then create the second leg of the call, then bridge the call... Do you think Asterisk reached to a level whereas it is possible to achieve the solution? Sure it has. -- 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
[Asterisk-Users] Looking for postpaid quality A-Z termination
Hi List, After quite a bit of struggle, it looks like I'm all ready to roll out prepaid cards on my small island. I now have a 4 E1s with a bit of spare capacity in order to accept incoming calls, and I can route Reunion Island mobile and fix through my own installations. For all other destinations, I need a carrier. I need good wholesale prices to Comoros, Mauritius, Madagascar, India, China / HK, France, UK. I am looking for some quality A-Z SIP / g.729 termination providers who are willing to work on a postpaid basis. This is because I have enough work as it is, and I don't want to be checking for account balance all the time. Since I work on a postpaid basis with my own clients (for VoIP termination), it would fit my business model better. I am going to start selling the cards very progressively, so don't expect volume to be very high straight away. Since I want to work with postpaid providers, this is a good thing since it will give us time to build some trust as volumes progressively go up. Eventually I plan to be selling under 12 month around 1,000 cards per week, which would amount to roughly 1,500 EUR worth of call termination (weekly). I understand that doing postpaid is considered risky by many. However I have been operating in the VoIP industry for a year and a half and have a profitable and cash flow positive business, with good cash reserves. I invite you to check the archives to see that I've been around doing VoIP for a little while now. If you are interested in building this business relationship with me, please let me know. Best Regards, 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
Re: [Asterisk-Users] Looking for postpaid quality A-Z termination
Martin Joseph a écrit : What part of NON-COMMERCIAL do you not understand? Ooops. I really thought I _did_ send it to the biz list. I guess I was a bit too fast with Thunderbird's auto-complete feature. Sincere apologies to the list. 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
Re: [Asterisk-Users] Change g729 payload
Attilla De Groot a écrit : Hi All, I have a SIP provider that tells me that my RTP stream uses a 20bytes payload in the g729 coded data. And they would like that we change this to 30bytes (3 frames). But maybe I'm wrong but isn't a certain payload just a standard for a codec ? You're wrong :) And if I'm wrong, how can I change the payload for my g729 calls in Asterisk. I had the same problem. Unfortunately this value is hard coded in Asterisk's code. I don't know if recent versions of Asterisk support this. 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
Re: [Asterisk-Users] Asterisk::AGI and DIALEDTIME
JP Carballo a écrit : Jean-Michel Hiver wrote: Hi List, In one of my AGIs (using DeadAGI) I grab the answered time using: my $res = $agi-exec (DIAL $dialstring); my $answeredtime = $agi-get_variable (ANSWEREDTIME); However this information differs from what's written in the Master.csv file (which happens to be the correct value!) Any ideas why? On my system, answeredtime returns the time elapsed since the call was answered by the destination. The time elapsed stored in Master.csv is from the time the current incoming call (channel) was answered. There are two values in Master.csv: the first one is the total time of the call (including the ringing bits and everything), the second one being the time which has been effectively answered (billable time). This second value is the correct one and differs from what I expect. Ah well, no worries. I've setup cdr_pgsql.conf and will process the CDRs every minute or so with a cron job. It's a bit patchy but what can you do :) 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
[Asterisk-Users] Asterisk crashes at startup
Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas? 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
Re: [Asterisk-Users] Asterisk crashes at startup
Robert Rawlinson a écrit : You may have file damage. Run the file repair. Rob, thanks for your response. Which tool would you use to do that? -- 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
Re: [Asterisk-Users] Asterisk crashes at startup
Brian C. Fertig a écrit : run asterisk with asterisk -c and see if it gives anymore information. You can also get it to produce a core dump and see if it gives you anymore information. That's pretty much what I did, and it says: [func_uri.so] = (URI encode/decode functions) == Registered custom function URIDECODE == Registered custom function URIENCODE == Manager registered action DBGet == Manager registered action DBPut == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk cleanly ending (0). 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
[Asterisk-Users] Asterisk::AGI and DIALEDTIME
Hi List, In one of my AGIs (using DeadAGI) I grab the answered time using: my $res = $agi-exec (DIAL $dialstring); my $answeredtime = $agi-get_variable (ANSWEREDTIME); However this information differs from what's written in the Master.csv file (which happens to be the correct value!) Any ideas why? I'm using asterisk 1.2.7.1 and the lastest asterisk-perl distrib. 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
[Asterisk-Users] Recent debian packages?
Hi, I'd like to use the convenience of apt packaging, but debian sarge's default asterisk is something like 1.0.7. Are there any apt repositories which provide newer versions of the software? Thanks! -- 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
Re: [Asterisk-Users] Recent debian packages?
Stefan Reuter a écrit : Jean-Michel Hiver wrote: I'd like to use the convenience of apt packaging, but debian sarge's default asterisk is something like 1.0.7. Are there any apt repositories which provide newer versions of the software? sure: http://pkg-voip.buildserver.net/debian Thanks! I'm giving this one a try :) -- 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
Re: [Asterisk-Users] Junghanns GSM card
Andrew Nowrot a écrit : Hi, Does anyone have some experience with junghanns GSM cards? I want to know if I can use this cards to send SMS directly from Asterisk box. They look terrible to me. From the picture on their website it looks like you need one antenna per GSM channel (most gateways use 1 antenna per 4 ports). Plus it would suck to have to open your computer to swap the SIM cards... Apparently the current market pricing for VoIP - GSM gateways is around 500 euros per port. Now if you can beat this using their hardware, they might be onto something. Cheers, Jean-Michel. Mail scanne par Orange Caraibe. ___ --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] Compare to Skype
This is only an issue if your SIP phone has a poor/nonexistent jitter buffer. I agree with that. Asterisk should just forward any RTP immediately and let endpoints handle the jitter buffer - unless asterisk is the endpoint itself (e.g. with phones plugged in its fxs ports). ___ --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] Looking for input on which way to go with smallbusiness setup
Kerry Garrison a écrit : You will kick yourself up and down the block for not using IP phones in the end. What are you going to spend? $65 for an ata and $25 for a phone? Spend the extra money and get SNOM or Linksys phones. You then need to figure out how to get 12 analog lines into Asterisk. Using 3 TDM400's is not really an option as you will spend countless hours trying to figure out interrupt issues. The next best option is 3 Mediatrix 1204 Gateways, this will set you back about $1,800 and will be maxed out. The best option, sticking with analog lines that is, would be a Rhino CB-24 FXO Channel Bank connected to a Rhino R1T1 card in the asterisk server. This bundle will hit you for about $2,000 but will only be half full. True, but using IP Phones means you need to do proper QoS at the level of your LAN or install a second LAN (which kinda defeats the purpose) as otherwise a network card going bananas could kill your phone system. So you need to get a proper QoS enabled switch, which is quite expensive too... ___ --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] Don't see my post
John Rich a écrit : Hi Folks, I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that can handle high volumes. This mailing list is not moderated (as far as I can tell) _but_ your message would be more appropriate on the -biz list anyways. 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
Re: [Asterisk-Users] Bandwidth Management
Andy Tan a écrit : Hi Alex, thanks for the suggestion. Did some checks, and thought that I could set a global variable to track the utilized bandwidth. Wish that there are plans for support to include variables like SIP_CODEC in other protocols. Actually this sounds like a really nice idea. It would be cool to have a way to start using less intensive bandwith codecs for new calls when bandwith reaches a certain threshold. For example: - 0-40% bandwith: g711 - 40-60% bandwith: g729 - 60%-80% bandwith: g723 - 80%-100% bandwith: drop new calls, or maybe use lpc10 It wouldn't help in SOHO usage but when using Asterisk as a call termination gateway, it would help making the most out of available bandwith. g711 is certainly better than g729 when you have the bandwith, and i'm pretty sure that even lpc10 sounds better when on non-saturated bandwith compared with g729 with some packet loss... How would you go about implementing this? 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
Re: [Asterisk-Users] asterisk credit card processing
Joseph a écrit : Is there a way somehow to implement Asterisk with Credit Card Processing (IVR system)? Yes, using AGI. ~google asterisk voip-info agi 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
Re: [Asterisk-Users] QuesCom 400 IP/GSM
Giordano Grandis a écrit : Hi ll, anyone never used this QuesCom 400 IP/GSM with Asterisk ? Should it works? Yes, it works fine. 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
Re: R: [Asterisk-Users] QuesCom 400 IP/GSM
Giordano Grandis a écrit : Ok, perfect. And what protocol is it use ? SIP or SCCP ? SIP. 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
Re: [Asterisk-Users] AGI Scripts Terminate too Soon
snip HOWEVER, if the CALLER hangs up the call, it seems as if Asterisk immediately kills the AGI script. My script seems to terminate immediately and therefore execution does not continue after the Dial() command. /snip http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI Cheers, Jean-Michel. ___ --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] AGI Scripts Terminate too Soon
I am red-faced! The TFOT book explicitly says this on page 158, on the box titled, AGI(), EAGI(), DeadAGI(), and FastAGI(): The DeadAGI() application is also just like AGI(), but it works correctly on a channel that is dead (i.e., a channel that has been hung up). As this implies, the regular AGI() application doesn't work on dead channels. Jean-Michel, thanks for pointing out the (almost) obvious! No Problem. Astcc uses DeadAGI as well and it works well if either parties hang up, which is what you want :) Cheers, Jean-Michel. ___ --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] Is Asterisk a PBX?
Zach A a écrit : Hi everybody, This question is confusing me for some time. From selling point of view to a customer, calling asterisk a PBX doesn't look right. According to the definitions of PBX or PABX, Asterisk is not just PBX but much more than that. My question is, how should I introduce Asterisk to a customer? I don't want to call it a PBX. Call it a telephony e-server or something. Some people like that BS. 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
Re: [Asterisk-Users] OT: VoIP over bonded link
Colin Anderson a écrit : It's stupid. Don't ever connect 2 different building with copper. Just wait until you get some kind of lightening hit or electrical fault, but make sure you are no where near it. Use fibre. Thanks for the reply. Unfortunately, the conduit for the provisioning of the new building is unsuitable for fibre (too many sharp bends) and we can't core out the concrete and put in a new conduit because of obstacles in the way that make laying new conduit impractical, so we are stuck with (existing) copper. We already have copper-to-copper connections of different types (electrical, security etc) between the buildings so a lightning strike is going to hose us no matter what. That aside, does anyone have opinions on my original question as to the suitability of bonded links for VoIP? You might have a little bit of jitter, but that's what jitter buffer is all about. IMHO it would be fine for VoIP but as it has been pointed on the list it would be wise to prioritize correctly both ends of your aggregated link. PS: What about Wifi for your link? With a couple of well placed high-gain outdoor antennas you could cover the distance and have similar throughput... It could be significantly cheaper too! 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
Re: [Asterisk-Users] TDMoIP and Asterisk
I think you missed the point of my question, which was to know if there was any attempt to make Asterisk talk TDMoIP directly, so that I wouldn't have to have any E1 hardware in the Asterisk server at all to satisfy my failover requirements. I guess your answer is an indirect to my knowledge, it hasn't been done. To my knowledge, it hasn't been done. The fonebridge would do what I want, but I can't get it in Australia, and if I imported one, I wouldn't legally be able to attach it to the phone system. No, the phone bridge does TDMoE, not TDMoIP. 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
Re: [Asterisk-Users] TDMoIP and Asterisk
James Harper a écrit : Does anyone know anything more about using Asterisk and TDMoIP together? Well, these boxes have an E1 interface. So you should be able to use a Digium or Sangoma card to connect to them and be happily doing TDMoIP... 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
Re: [Asterisk-Users] TDMoIP and Asterisk
James Harper a écrit : I want to do it the other way around. Asterisk---TDMoIPRADE1Telco You'll need 2 RAD boxes, i.e. Asterisk - RAD - TDMoIP - RAD - E1 - Telco 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
Re: [Asterisk-Users] PSTN connection via IP/ethernet
Nick Hoffman a écrit : Hey guys. If my Asterisk box connects to the PSTN using SIP and IP over ethernet and doesn't require any authentication, what sort of a trunk would need to be created? Something like Dial (SIP/number@ip_address) ? 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
Re: [Asterisk-Users] Better i18n for Asterisk?
Could we have run into another Americanism here? OK, back to being English and bashing the French ;-} Nice french 'tache :) Anyway, the problem at hand is not the french language here. The problem is that any worthwhile i18n mechanism must pass the yota master test. Speak like this, it should be able to. If your i18n system can do that then you're pretty sure it's flexible enough to deal with all kinds of different languages. As for time, this is not so much i18n but more of a localization problem. It would be nice to have a framework for this too, because well, hearing that the message was received at sept oh huit doesn't make any sense either :) 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
Re: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - long delay between answering and ringing
Anthony Rodgers a écrit : Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000 answers the SIP call from the Asterisk server (immediately upon dialing, according to the Asterisk CLI) and the ringing tone begins (the remote phone begins ringing at that same time). The delay is enough for users to think that the phone isn't working - not what you want for 911! Any ideas? You could use the 'r' flag in your Dial() command to simulate a ringing tone instantly. This is less than ideal though. Have you done some SIP traces (using ngrep for examples) to look when the SIP 'ringing' signal is actually being sent? 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
Re: [Asterisk-Users] Asterisk native sounds now available!
Kristian Kielhofner a écrit : Hello everyone, As I promised at eTel last week, I have finished up work on my Asterisk Native Sounds project. Here's a little diddy from astlinux.org: --- Asterisk Native Sounds are a collection of audio prompts for Asterisk. They will improve quality, reduce CPU usage, reduce latency, and (in some cases) eliminate the need for G729 licenses! The Asterisk Native Sounds are a collection of alternative sounds prompts for Asterisk. Here's how it works. I had Allison Smith (the voice of Asterisk) re-record all of the sound prompts present in Asterisk 1.2. She provided them to me in the best audio format possible. I then converted them into several native Asterisk sound formats. Why would I do all of this? What tools did you use to convert the sounds in all possible formats? Asterisk's sound files become quickly limited, and it would be nice to have a way to build your own IVRs native formats. 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
[Asterisk-Users] Better i18n for Asterisk?
Hi List, Do you know if there are any plans to improve i18n for Asterisk? The current i18n way of doing it with asterisk is very limited and most of the time does not work. For example, take voicemail: message received at seven 30 am might sound good in English. But: message recu a sept trente apres-midi sounds terrible in French, because you *need* to say sept heure trente and not sept trente. Is there a way to fix this / improve the situation (other than write own voicemail AGI)? 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
Re: [Asterisk-Users] change languages from an IVR
Mark Phillips a écrit : A customer of mine wants an IVR where the first 3 choices are 1 English 2 Spanish 3 French I can build the IVR but how do I get the system prompts to then speak the selected langauge. For example, a caller has selected Spanish and so is routed to the Spanish part of the IVR. At some point he breaks out of the IVR to leave a VM. How does the system know to continue offering him Spanish? asterix:~# asterisk -rx 'show application setlanguage' -= Info about application 'SetLanguage' =- [Synopsis]: Sets user language [Description]: SetLanguage(language): Set the channel language to 'language'. This information is used for the syntax in generation of numbers, and to choose a natural language file when available. For example, if language is set to 'fr' and the file 'demo-congrats' is requested to be played, if the file 'fr/demo-congrats' exists, then it will play that file, and if not will play the normal 'demo-congrats'. Always returns 0. -- 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
Re: [Asterisk-Users] One way audio - it doesn't make sense
What ports am I missing? Could the problem be entirely something else? Somehow I had the feelings that calls going out (since they originate from the device behind the NAT) would not be a problem, but calls coming in could be. I really would appreciate a hint from somebody who knows better than I do (i.e. anybody) Pehaps you have set your device to use an outgoing codec which is not supported out of the box by asterisk, such as g.729? ulaw or gsm should work. Check your codec config in your sip.conf as well. For debugging purposes, you should use ulaw everywhere (assuming your ISP supports it). Also, are you having any messages on the asterisk command line? Log onto your server, type in: asterisk -r set verbose 100 set debug 100 And let us know what you're seeing on the CLI. 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
Re: [Asterisk-Users] IP PAX gateway to PSTN
Sam a écrit : Hi, If I setup an IP PAX gateway to handle VoIP calls to a traditional phone line, I am wondering how each VoIP call to the PSTN connection get charged by a local Telecom. I am not really sure to understand the question. But assuming you are having: (remote phone) - internet - PSTN gateway - end phone The connection charge is going to be PSTN gateway - end phone. However note that in certain VoIP-backwards countries this scheme is illegal, and the telco might ask you to pay the international call termination charge if they find out you're doing this. 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
Re: [Asterisk-Users] IP PAX gateway to PSTN
Thanks for the answer. Is this PSTN gateway is something for a VoIP company to setup in order to connect their VoIP calls to the Telco's PSTN then to the end phone? I don't think Australia treat this as illegal. But I m not sure how much the Telco will charge from IP PAX (or PSTN) gateway to end phone. Assuming that there are 1000 VoIP calls thru Telco's PSTN to end phones, how will these calls get calculated? is the charge will be per-call basis? Sam, I am still unsure to understand your question :-/ How much your telco is going to charge you for the PSTN calls depends on your arrangement with the telco... Usually, with proper volume interconnects (say you order a PRI line), these calls are charged per second. As your volume increases you will usually be in a position to negociate better rates. 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
Re: [Asterisk-Users] IP PAX gateway to PSTN
Sam, I am still unsure to understand your question :-/ How much your telco is going to charge you for the PSTN calls depends on your arrangement with the telco... Usually, with proper volume interconnects (say you order a PRI line), these calls are charged per second. Do I really need PRI T1 line when I initially setup VoIP network? It largely depends what you are trying to achieve. But if you want to become a VoIP carrier for your area, yes. 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
Re: [Asterisk-Users] re: questions about sip requests to asterisk 1.2
In asterisk 1.2 asterisk completely ignores the request (even at most verbose level) and an ngrep shows a not found returned to SER. anyone have any idea why this is happening, bug/feature, or how to get it to work the way it did in 1.09? I want to upgrade but I don't want to lose this functionality. Since I use 1.0.9 and use exactly the same scheme, I am interested on how to upgrade as well. 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
Re: [Asterisk-Users] Slightly OT: OpenPBX.org and Freeswitch
Michael Collins a écrit : This is slightly OT in that it isn’t specifically *-related, but I was wondering what the members of the * user community felt about these two subjects. I’ve been perusing the OpenPBX.org mail list and the current hot topic is the fact that their project has come to a grinding halt. They are concerned that they don’t have enough people working on their project. They feel that * has improved since the fork but they still have the same complaints: the Asterisk core is a “pig,” the core is huge and messy with ugly code, and finally that Asterisk development is throttled by the “Digistapo.” Please take your big hairy troll back home. Keep this for the IRC or your local LUG beer ranting. -- 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
Re: Need to terminate 7 lines (was: [Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 10)
Kevin Steil a écrit : Need help...I need to install a card to terminate 7 lines...I have not order the phone lines yet...I can either do analog lines 1FBs or order a fractional T1... If I was you, I would go for the T1 since you can expand it when it's needed. any suggestions on what hardware would be easier to install and configure... If you plan on having only one T1 I think you should get a single T1/E1 Digium card. That being said I'm in a bad position to recommend it since I went with a proprietary gateway myself (boo!). also if I went with a T1...do I need an external CSU/DSU or anything or does it just plug into the T1 card...thanks.. As far as I'm aware you don't. 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
