Re: [asterisk-users] Being attacked by an Amazon EC2 ...
FWIW, we're seeing similar attacks. The below is what I posted on NANOG earlier, which summarizes Amazon's stellar abuse response. I've also received an off-list e-mail from someone who was getting hit with 6Gbps of traffic from them (and was not able to reach anyone there either). Time to start blocking them at the edge. Let their customers complain to them instead. -Original Message- From: Erik L Sent: April 11, 2010 10:38 To: na...@nanog.org Subject: Seeking Amazon EC2 abuse contact Could someone from Amazon EC2 please contact me off-list regarding an abuse issue from one of their IPs? Alternatively, could someone please send me the contact details of someone there? E-mailing the abuse e-mail listed in WHOIS per their instructions, including all pertinent data, results in an auto-reply indicating to use a form on their site. Submitting the form results in There has been an error while submitting your data. Please try again later. Calling their supposed NOC (as per WHOIS) results in You have reached the legal department at Amazon...please leave a message. Thanks -- Erik Caneris Inc. Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-888-444-8843 www.caneris.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
Thomas, Yes you can do this. I actually have done this and run it as a service under the name Meetmecall. I use MSN as the user interface to record the message, create phone lists of the numbers that has to be called and to actually schedule and perform the delivery. It is possible to use it for spam but the customers I have use it to notify, remember, offer or let the callee know about something relevant, but always as part of an already existing relation. With some extra parameters used, you can start a groupcall and use all the other Asterisk magic available like doing a questionarry using a smart IVR etc. etc. I can think about a long list of useful use of this service. I have no idea about the rules and legislation in other countries but in Holland you will end up with serious trouble and extreme high penalties to pay if you actually use it for spamming. I will not send you a copy of the solution but it is based on the use of call files pointing to local channels/extensions where the Asterisk magic is combined in a working (and I think clever) way. The CDR isn't perfect but disable and enable CDR at the proper points in the dial plan and clever use of the USERFIELD variable will result in useable data for billing the users. The CDR shows that most callees, listen to the message until it ends and yes, sometime there are complaints about the use but that is very rare. About the scheduling of the calls to make. It is not Asterisk that limits you. Far before reaching the limits of Asterisk it will be the bandwidth available and the SIP trunk provider that normally doesn't allow an endless number of concurrent calls. When I started this I was working for a Norwegian company offering the dial tone on the internet and I had a server almost directly connected to the backbone of internet with more or less endless bandwidth. I did some stress testing of a call center solution and 80 concurrent calls wasn't a problem and my guess is that you can far beyond 80 calls. It is wise to move the call files one after the other or one batch after the other. Moving large numbers of call files into /var/spool/asterisk/ outgoing might sometimes result in unexpected and not intended results. There are other scenarios but this was my choice. 10.000 calls will take some time but with a 30 seconds message, 20 concurrent calls and 10 seconds average to dial after around 5,5 hours the last phone call will be dialed. If the message is just 15 seconds it will take around 3,5 hours. If you want to deliver in short time, like 10 minutes, you really have to scale up to 420 concurrent calls which doesn't sound doable unless you have real serious budgets. If you put everything in place at your side you will probably run into constraints of the SIP provider and the interconnection to the pstn. btw: Asterisk has the potential to build lots of evil features and lots of standard features can be used in an evil way. Personally I think it is kind of strange that, if a question is asked, one has to explain why the answer is not mend for evil use. We don't have to help someone out and we can refuse because of lots of reasons: no time, not an interesting question, not a single sign of any effort by the one asking the question, not willing to give something away that costs lots of time and energy, the feeling that it will be used in an evil way etc. etc. I think the tone and the content of this discussion harms the Asterisk community as a whole. with friendly regards, Erik de Wild Tripple-o: your asterisk migration partner the Netherlands On 6 feb 2010, at 03:54, Thomas Perron wrote: Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] {top|bottom|interleaved} posting, once again
Funny that a small matter like top | bottom | interleaved posting can lead to a situation that is referred to as a fight . I agree with Ira that keeping all the original lines in place is very annoying when there are more lines then needed to pick up the discussion. I seem to be a top poster by nature, it never crossed my mind that this could be part of a fight or be annoying to others . Sometimes I interleave but I never post at the bottom. Lets take our bits of freedom and consider how to post (not, top, bottom or interleave) as part of our personal style of communicating ;-) erik On 6 feb 2010, at 22:04, Ira wrote: At 11:17 AM 2/6/2010, you wrote: Actually bottom-posting without trimming anything (SCNR) is about as annoying as top-posting. Interleaved posting is fine, quoting just enough text so everyone can understand the context. Seems to me if you trim so that only the minimum amount is left it hardly matters. I don't know about everyone else, but I've already read all the prior posts and only need the smallest bit of reference to connect the answer in my mind. I never read bottom posts that are more than 20 lines from the top unless I figure there's a reason I need to. Life is just to short. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ENUM and Asterisk 1.6
Hi all,
I have a problem with 1.6.1.7-rc1 and ENUM (with an own PowerDNS server
and NAPTR record). Maybe somebody has more experience with this or can
give me some input.
The dialplan:
exten = 292,1,Set(DIAL_NUMBER=43660123456)
exten = 292,2,Set(sip=
${ENUMLOOKUP(+${DIAL_NUMBER},sip,,1,ns3.e164.xxx.com)}) ;x'ed out the
domain name starting from here
exten = 292,3,NoOp(${sip})
exten = 292,4,Hangup()
The output if I dial 292:
Connected to Asterisk 1.6.1.7-rc1 currently running on srv21 (pid =
6061)
Verbosity is at least 3
== Using SIP RTP CoS mark 5
-- Executing [...@sip:1] Set(SIP/273-98117048,
DIAL_NUMBER=43660123456) in new stack
== ast_get_enum(num='+43660123456', tech='sip',
suffix='ns3.e164.xxx.com', options='', record=1
== ENUM options(): pos=1, options='0'
== ast_get_enum() profiling: FAIL,
6.5.4.3.2.1.0.6.6.3.4.ns3.e164.xxx.com, 2 ms
-- Executing [...@sip:2] Set(SIP/273-98117048, sip=) in new
stack
-- Executing [...@sip:3] NoOp(SIP/273-98117048, ) in new stack
-- Executing [...@sip:4] Hangup(SIP/273-98117048, ) in new stack
== Spawn extension (sip, 292, 4) exited non-zero on 'SIP/273-98117048'
-- Executing [...@sip:1] Hangup(SIP/273-98117048, ) in new stack
== Spawn extension (sip, h, 1) exited non-zero on 'SIP/273-98117048'
srv21*CLI
The NAPTR record:
6.5.4.3.2.1.0.6.6.3.4.e164.xxx.com 100 10 u E2U+sip !^.*$!
sip:2...@10.0.43.105! .
The output of a dig command using the ns3.e164.xxx.com (so DNS seems to
be fine):
dig @ns3.e164.xxx.com 6.5.4.3.2.1.0.6.6.3.4.e164.xxx.com ANY
; DiG 9.5.1-P3 @ns3.e164.xxx.com
6.5.4.3.2.1.0.6.6.3.4.e164.xxx.com ANY
; (1 server found)
;; global options: printcmd
;; Got answer:
;; -HEADER- opcode: QUERY, status: NOERROR, id: 11934
;; flags: qr aa rd; QUERY: 1, ANSWER: 1, AUTHORITY: 0, ADDITIONAL: 0
;; WARNING: recursion requested but not available
;; QUESTION SECTION:
;6.5.4.3.2.1.0.6.6.3.4.e164.xxx.com. IN ANY
;; ANSWER SECTION:
6.5.4.3.2.1.0.6.6.3.4.e164.xxx.com. 600 IN NAPTR 100 10 u E2U+sip !
^.*$!sip:2...@10.0.43.105! .
;; Query time: 19 msec
;; SERVER: 10.0.50.107#53(10.0.50.107)
;; WHEN: Mon Nov 16 14:14:05 2009
;; MSG SIZE rcvd: 111
enum.conf:
[general]
;
; The search list for domains may be customized. Domains are searched
; in the order they are listed here.
;
search = ns3.e164.xxx.com
search = e164.arpa
Regards,
Erik
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to configure a coverage path for anextension???
it should look something like
exten = 4000,1,Dial(SIP/4000,30,t)
exten = 4000,2,Goto(4001,1)
exten = 4001,1,Dial(SIP/4001,30,t)
If 4000,1 is answered it will never reach 4000,2
if 4000 is busy or not available for another reason it wil goto 4001,1
hope this is useful
Erik de Wild
Tripple-o
Verstuurd vanaf mijn iPhone
Op 16 sep 2009 om 16:24 heeft Ioan Indreias indre...@gmail.com het
volgende geschreven:\
Hi Juan,
1. Please use the semicolon (;) character to comment your dialplan.
Your choice (#) is intended for something else.
2. Probably you have to add the j option of Dial application (show
application Dial), like:
exten = 4000,1,Dial(SIP/4000,20,iKkTtj)
exten = 4000,102,Dial(SIP/4001,20,iKkTtj)
3. For more hints you could check voip-info page.
HTH
Ioan Indreias
www.modulo.ro
On Wed, Sep 16, 2009 at 4:52 PM, Juan Cardoza jcard...@tpmex.com
wrote:
I comment all the lines in my extensions.conf file to work only with
the
lines you provide me Danny:
Extensions.conf
[local-sip]
#exten = _4XXX,1,Dial(SIP/${EXTEN},10,tTr)
#exten = _5XXX,1,Dial(Dahdi/1/${EXTEN})
#exten = 164,1,Dial(Dahdi/1/${EXTEN})
#exten = 0550,1,Dial(Dahdi/1/${EXTEN})
#exten = _4XXX,3,Hangup()
[incoming]
exten = 4000,1,Dial(SIP/4000,20,iKkTt) - I test this line only
and it
works
exten = 4000,s-BUSY,Dial(SIP/4001,20,iKkTt) When I add this
line the
call arrives to the 4000
#exten = _4xxx,1,Dial(SIP/${EXTEN},10,tTr)
I dont answer the call and the Asterisk server drop the call.
[Sep 16 08:50:40] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec:
Unable
to enable echo cancellation on channel 23 (No such device)
-- Executing [4...@incoming:1] Dial(DAHDI/23-1, SIP/
4000,20,iKkTt)
in new stack
-- Called 4000
-- SIP/4000-08a41440 is ringing
-- SIP/4000-08a41440 answered DAHDI/23-1
-- Accepting call from '' to '4000' on channel 0/22, span 1
-- Executing [4...@incoming:1] Dial(DAHDI/22-1, SIP/
4000,20,iKkTt)
in new stack
[Sep 16 08:50:50] WARNING[1823]: chan_dahdi.c:4035 dahdi_enable_ec:
Unable
to enable echo cancellation on channel 22 (No such device)
-- Called 4000
-- SIP/4000-08a359c8 is ringing
[Sep 16 08:50:52] NOTICE[3394]: chan_sip.c:21804
handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 4000
-- Nobody picked up in 2 ms
-- Auto fallthrough, channel 'DAHDI/22-1' status is 'NOANSWER'
-- Hungup 'DAHDI/22-1'
tp2asterisk01*CLI
What could I need to fix this???
Thanks a lot for your help.
Jhon
-Mensaje original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Danny
Nicholas
Enviado el: Miércoles, 16 de Septiembre de 2009 08:09 a.m.
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: Re: [asterisk-users] How to configure a coverage path for
anextension???
In regular configuration (extensions.conf) this is one way to do it:
- exten = 4000,1,Dial(SIP/4000,20,iKkTt)
- exten = 4000,s-BUSY,Dial(SIP/4001,20,iKkTt)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan
Cardoza
Sent: Wednesday, September 16, 2009 8:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] How to configure a coverage path for
anextension???
I have been checking but nothing that clear my idea...
I have the extension 4000 and the idea is when this extension
receive a call
and the extension 4000 is busy, the call from PSTN could be send to
a second
extension, example: 4001, this need to happen only if the first
extension is
busy.
If not, the call need to be take by the first station.
Please any one how can help me on this???
Best regards
Jhon
Teleperformance values: Integrity - Respect - Professionalism -
Innovation -
Commitment
The information contained in this communication is privileged and
confidential. The content is intended only for the use of the
individual or
entity named above. If the reader of this message is not the intended
recipient, you are hereby notified that any dissemination,
distribution or
copying of this communication is strictly prohibited. If you have
received
this communication in error, please notify me immediately by
telephone or
e-mail, and delete this message from your systems.
Please consider the environmental impact of needlessly printing this
e-mail.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http
Re: [asterisk-users] features.conf : feature map == getting feature to work
just a hint. you might have # assigned the moh in feature.conf and #3 to starting the recording. check your feature.conf and makesure that # isn't assigned to anything. erik de wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone Op 7 sep 2009 om 20:40 heeft jonas kellens jonas.kell...@telenet.be het volgende geschreven:\ Hi there, I need some help with a 'custom' feature. I have following feature defined in features.conf : [applicationmap] opnemencallee = #3,self/callee,Monitor,wav,/var/samba/profiles/ jonaskl/recording,m In my dialplan : [from-HostAst] exten = s,1,Set(__DYNAMIC_FEATURES=opnemencallee) exten = s,n,Dial(SIP/grandstream,30) I want the callee to be able to press #3 to be able to record the conversation but when I press these keys on my Grandstream phone, the following is displayed on the CLI : [Sep 7 20:33:49] WARNING[10870]: res_musiconhold.c:665 get_mohbyname: Music on Hold class '/var/samba/profiles/jonaskl/ recording' not found Don't know where this comes from... I have tried the same with *3. Same output on the CLI. Yes, I have restarted Asterisk after changes in features.conf. It's not my Grandstream or the DTMF-input because *8 for picking up a ringing phone works well... When I set : opnemencallee = #*3,self/callee,Monitor,wav,/var/samba/profiles/ jonaskl/recording,m and I press #*3, nothing happens... No output on the CLI. There's not much info. I followed the instructions on voip-info.org (which are the same as in features.conf). The module res_features is loaded. Greetingz, Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] invalid extension
you should check dialstatus and gotoif. if you use both in the proper way ( see the wiki) then you have the dialplan behaviour you are looking for. erik de wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone Op 7 sep 2009 om 21:26 heeft Miguel Molina mmol...@millenium.com.co het volgende geschreven:\ Administrator TOOTAI escribió: Hello, with Asterisk 1.6.1.6 I try to hangup a call if called extension is not existing. For this purpose I would use the internal i extension but seems not to work. [MyContext] exten = s,1,NoOp(Call is treated as it should) exten = s,n,NoOp(next step) exten = s,n,NoOp(aso ...) exten = _[a-zA-Z].,1,Goto(s,1); accept exten LEN 1 alpha exten = _X.,1,Goto(s,1); accept exten LEN 1 numeric exten = i,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = i,n,Hangup ; refused, end of call What I have when calling a one digit extension -in this case h- is: == Using SIP RTP CoS mark 5 [Sep 7 18:51:03] NOTICE[6084]: chan_sip.c:18523 handle_request_invite: Call from '' to extension 'h' rejected because extension not found. == Using SIP RTP CoS mark 5 Should it not go to i extension? If I call the i or s extension it's going well. Am I missing something? Hi, The 'i' extension only works in applications like Background(), WaitExten() and everything that uses DTMF to route extensions within a context. As you can see in your call, it won't work directly because asterisk by default will reject a call that doesn't match in the context or included contexts you defined for the user. Because the call is not accepted there's no need for a hangup (in a SIP environment). If you want to explicitly hangup calls using the dialplan, for your case add a one-digit catch all and leave your good calls with a 2-digit minimum: exten = _X,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = _X,n,Hangup exten = _XX.,1,Goto(s,1); accept exten LEN 1 numeric That will be enough to hangup what you want to, adjusting it to your needs. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as the recording server
using mixmonitor might not be such a good idea. afaik the mixing of the recordings of the two channels starts after ending the call causing a high cpu load. if you have recordings going on all the time moving the 2 files that has to be mixed to a dedicated mixing server might be a good idea. after mixing it should be stored in a retrievable way. Erik de Wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone Op 8 sep 2009 om 00:25 heeft Miguel Molina mmol...@millenium.com.co het volgende geschreven:\ I imagine this setup will need those two communicating entities to be part of the pabx. But support extension 100 of PABX A (legacy) calls 101 on the same platform. I want asterisk connected to PABX A via E1/T1 to know about that call and start recording (tap) without bridging or being part of that conversation Hi, Asterisk won't work as a recording server if the call doesn't go through it. In the IP world it means that both media (RTP) and signalling must pass through asterisk, and in the E1/T1 digital or analog world it means that the call must be bridged through asterisk. A simple dialplan would explain it: exten = s,1,Answer() ;Asterisk receives the call, from the lecagy PBX or from the external link (this should be two different contexts) exten = s,n,MixMonitor(blah) ; Records the conversation, exten = s,n,Dial(Tech/peer/number...,30,rtTwhatever) ; and sends the call back to the legacy PBX or to an external link If you want to record 100% calls, you would have to route every call through asterisk, even internal PBX calls. Even if you want to tap your legacy PBX to a non-asterisk recording server like the ones suggested before in this thread, the calls must go through a link to make tapping possible and you should seek an alternate solution to the internal calls within your legacy PBX. The beauty of asterisk and open source IP-PBXs relies on the native recording capabilities which makes things really easy. When you see that asterisk works and that can do the recordings and much more, you would start thinking on making asterisk your main PBX solution and leaving that legacy PBX for minimal uses. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI commands not running !!!!!
Just a hint based on experience. Run top from de linux prompt to check if any proces causes an enormous cpu load. I once ran into the same behaviour because some asterisk related php script looped and took almost all the cpu power available. erik de wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone Op 8 sep 2009 om 09:39 heeft abdelkader abdelkader2...@gmail.com het volgende geschreven:\ Hello, I am using Asterisk 1.4.22 in a debian 4.0 with a kernel version 2.6.18-6-amd64 (SMP). The processor type is Intel(R) Xeon(R) CPU E5420 @ 2.50GHz. Sometimes, I get a strange behavior from asterisk: The CLI commands does not work and Asterisk cannot receive calls. The output of every CLI command is that command is not known (no such command). Please help me resolve this problem: what can be the cause of it? is it Asterisk or my system? and what have I to do to eliminate this problem? Thks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax For Asterisk and SendFax question
You can use the M parameter to run a macro after the channel picks up or the g parameter to jump to a given context. there is also a parameter to run an AGI script. Check the dial() cmd on the wiki for further details. Erik de Wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone Op 9 sep 2009 om 00:43 heeft Joan Antoni Terre nebh...@gmail.com het volgende geschreven:\ thanks Danny, just another stupid question, as far as I know, when a call is answered after Dial application, it doesn't execute other dialplan priorities until it's hung up, which execute h priority, so how can I make it execute a SendFAX, or whatever else, when it's answered? thanks again 2009/9/8 Danny Nicholas da...@debsinc.com That’s the general idea. The application is designed to send a TIFF over an established connection. You can detect that it is a fax or just assume so. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com ] On Behalf Of Joan Antoni Terre Sent: Tuesday, September 08, 2009 4:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Fax For Asterisk and SendFax question Hi everybody, I've installed Free Fax For Asterisk in my Asterisk box but I don't understand how it works as when using SendFax application from dialplan, I can't find how to introduce destination fax number. How this application works? Do I have to call destination fax using Dial application, detect somehow that it's a fax and then use SendFAX application specifying FAXOPTs and the path to the fax file? Many thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 TE121 cable connected to a TN2464BP AVAYA card
Not an answer to your question but maybe helpfull. If you use pri to connect two servers you might prefer the European way ( E1 ) instead of T1 because it will offer you 7 extra channels. As far as I know most pri cards support both. Be aware of the role each card has to play: one the telco role and the other the enduser role. See www.asteriskguru.com/tutorials/e1t1.HTML Erik de Wild Tripple-o Your Asterisk migration partner The Netherlands Op 4 sep 2009 om 23:23 heeft Juan Cardoza jcard...@tpmex.com het volgende geschreven:\ Hello All I am looking for the cable I need to create to connect a TE121 card with a TN2464BP card (AVAYA ISDN Card), please let me know if someone have the information about this cable, my asterisk CLI show this: pri show span 1 Status: In Alarm, Down, Active And the card is in red, so I am thinking the problem is the card. Any idea??? Thanks John Teleperformance values: Integrity - Respect - Professionalism - Innovation – Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help building dahdi for debian
perhap this is not an answer to your question but following this procedure will result in a working Asterisk system. I use this list when setting up an Asterisk system after installing a basic Debian etch network install : apt-get install mc # just my bad habbit apt-get install ssh apt-get install build-essential apt-get install module-assistant m-a prepare # based on info from asterisk-guru apt-get install bison apt-get install ncurses-dev apt-get install libssl-dev apt-get install libnewt-dev apt-get install zlib1g-dev apt-get install initrd-tools apt-get install procps # change the release number for the version needed cd /usr/src mkdir asterisk # another perhaps bad habbit cd asterisk wget http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.25.1.tar.gz wget http://downloads.digium.com/pub/telephony/dahdi-linux/releases/dahdi-linux-2.1.0.4.tar.gz wget http://downloads.digium.com/pub/telephony/dahdi-tools/releases/dahdi-tools-2.1.0.2.tar.gz wget http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-1.4.8.tar.gz tar -zvxf asterisk-1.4.24.tar.gz tar –zxvf asterisk-addons-1.4.7.tar tar -zvxf dahdi-linux-2.1.0.4.tar.gz tar -zvxf dahdi-tools.2.1.0.2.tar.gz cd dahdi-linux-2.1.0.4 make make install cd .. cd dahdi-tools-2.1.0.2 ./configure make make install make config cd .. cd asterisk-1.4.24 ./configure make make install make samples modprobe dahdi_dummy #Usually I just modprobe dahdi_dummy so MeetMe() works fine. lsmod | grep dahdi #The output should look something like #debianl:~# lsmod|grep dahdi #dahdi_dummy 5224 0 #dahdi_transcode 7848 1 wctc4xxp #dahdi 186472 19 dahdi_dummy ,xpp ,dahdi_transcode ,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp #crc_ccitt 2240 1 dahdi #rtc12372 1 dahdi_dummy cd asterisk-addons-1.4.7 ./configure make menuselect make make install make samples Usually I just modprobe dahdi_dummy so MeetMe() works fine. Why making a problem of not having the debian packages available if building a working system from source is so simple? Hope this is useful for someone \erik Date: Sat, 13 Jun 2009 09:51:24 +1000 From: Alex Samad a...@samad.com.au Subject: Re: [asterisk-users] Help building dahdi for debian To: asterisk-users@lists.digium.com Message-ID: 20090612235124.gb17...@samad.com.au Content-Type: text/plain; charset=us-ascii___ -- 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] Not receiving voicemail message in mailbox
I have it up and running o my system with this line in voicemail.conf and a symlink named sendmail to the actual msmtp program. mailcmd=/usr/sbin/sendmail -v -t -f lmy e-ail name@gmail.com This is the install log based on info provided by others and the install process itself. Hope it is useful /Erik Step 1 Installing needed packages/libs on your system install this packages (I'm not sure if all the packages are needed but with this packages it works) apt-get install libwww-perl apt-get install openssl apt-get install libcrypt-ssleay apt-get install libnet-ssleay-perl apt-get install libcrypt-ssleay-perl Step 2 download msmtp download msmtp van sourceforge (http://sourceforge.net/projects/ msmtp/) to /usr/src/ Step 3 bunzip2 msmtp.tar.bz2 tar -xvf msmtp.tar cd /usr/src/msmtp Step 4 built msmtp ./configure make make install Step 5 check if msmtp is on the system and if the output looks like below. # msmtp --version msmtp version 1.4.9 TLS/SSL library: GnuTLS Authentication library: GNU SASL Supported authentication methods: plain cram-md5 digest-md5 gssapi external login IDN support: enabled NLS: enabled, LOCALEDIR is /usr/share/locale System configuration file name: /etc/msmtprc User configuration file name: /root/.msmtprc Copyright (C) 2006 Martin Lambers and others. This is free software. You may redistribute copies of it under the terms of the GNU General Public License http://www.gnu.org/licenses/gpl.html. There is NO WARRANTY, to the extent permitted by law. Step 6 Make a symlink from /usr/local/bin/msmtp to /usr/sbin/sendmail (the name of the symlink is sendmail) # ln -s /usr/local/bin/msmtp /usr/sbin/sendmail Step 7 Add /root/.msmtprc (be aware of the dot) to the system with only owner read and write permissions and with this lines (adjust to your x...@gmail.com account). This way it works for a gmail account defaults logfile /var/log/msmtp.log account default from xx@gmail.com protocol smtp host smtp.gmail.com port 587 user xxx@gmail.com password password auth on tls on tls_certcheck on tls_trust_file /root/cert.pem Step 8 certificate file copy the certificate file to the root directory /root/cert.pem copied on system (see attachement) Step 9 configuration of /etc/asterisk/voicemail.conf Add this to /etc/asterisk/voicemail.conf as a replacement of the mailcmd = line mailcmd=/usr/sbin/sendmail -v -t -f your_gemail_name@gmail.com and uncomment attach = yes Add a vociemailbox to the system in [default] of voicemail.conf [default] ; Define maximum number of messages per folder for a particular context. ;maxmsg=50 500 = 1234,name,e-mail adress step 10 adding a test extension to the system Add an extension to /etc/asterisk/extension.conf to test de setup something like exten = 888,1,Answer() exten = 888,n,Voicemail(500) If you call 888 with in internal phone you enter the voicemail routine and a recording will be made. After finishing you will receive an e- mail with the recording as an attachement. And you are done Message: 2 Date: Sat, 16 May 2009 21:47:58 +0200 From: jonas kellens jonas.kell...@telenet.be Subject: Re: [asterisk-users] Not receiving voicemail message in mailbox To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 1242503278.3667.4.ca...@localhost.localdomain Content-Type: text/plain; charset=us-ascii I have put the following in my voicemail.conf-file : mailcmd=/usr/local/bin/msmtp -d --syslog=on -d and syslog=on are to debug some information, because I am still not receiving my voicemail-messages via mail as an attachment ! I don't know which mailcommand I need to put here to make Asterisk use msmtp as 'mailing server'. It is currently not working... The logfile /root/.msmtp.log is not mentioning anything. I think this is because Asterisk is really not using msmtp to send the message. Can someone help me figure this out... ? Jonas. ___ -- 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] cepstral vs festival
Somewhat off-topic, but I'll mention briefly that it's a multi-city service and you can get more info at http://www.trafficondemand.ca/ I believe that it's still considered beta for non-Toronto. You have Kitchener/Waterloo! Yay dials Oh. No traffic. Boo-urns. Hehe...working on it ;) I'd definitely like to know when you start populating the traffic part of K/W (and separate out london, it's a poor choice to group. Kitchener/Wwaterloo/Cambridge sure... but London? That's a common Torontonian thing to do. :-) Agreed. I advised the client against that, during design, but here we are. Hopefully he requests us to change this soon. On an unrelated note, I always find the Toronto is the centre of the universe attitude quite amusing. Some clients who call us for DSL qualifications, when asked Where are you located? respond with Bathurst Shephard. No sir, what city and province? -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival (MRCP)
John: However, that doesn't mean that it shouldn't be implemented. This is an area in which I think there is a disproportionate amount of non- discussion, since many people who would use or be interested in MRCP simply don't participate in the Asterisk project because it doesn't meet their needs out of the gate. Therefore, we see few people asking for it, in a self-fulfilling loop. Is MRCP something that is significantly lacking in Asterisk? Is it a difficult protocol to implement? Is there anyone here on -dev with the experience to do it? I don't know whether it's significantly lacking nor how difficult it is to implement, but it's certainly nice to have. It would increase the appeal of Asterisk to those used to working with MRCP-compatible resources in other platforms. That said, it can be argued that it's best to keep Asterisk simple and free of extra features. If its core purpose does not consist of interfacing with ASR and TTS engines, then some would argue that it's best to keep such features to a separate platform. Regards, -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
Thanks. Unfortunately no SIP/IAX access at this time, only by dialing one of the TNs. However, I'll bring it up with the client and see if they'd want us to configure that. Somewhat off-topic, but I'll mention briefly that it's a multi-city service and you can get more info at http://www.trafficondemand.ca/ I believe that it's still considered beta for non-Toronto. Regards, -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith (lists) [EMAIL PROTECTED] Sent: Thursday, December 04, 2008 10:43 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] cepstral vs festival On December 2, 2008 07:55:00 pm Erik (Caneris) wrote: Nuance would say no :) I'd say maybe. Call up +14164854854, it's a recent project we did for a That's pretty cool! Is there any SIP or IAX access to this (aside from dialing a POTS number) ? -A. ___ -- 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] cepstral vs festival
Festival sucks. Cepstral sucks less. The End. In my experience, it depends on the specific app, who's paying, and who's going to be the victim, err...user listening to it. This is the difference between domain/context specific phrases/words to pronounce vs. general stuff, a client on a tight budget or not, the users being internal vs. customers/public, and so on. Cepstral is a $30 TTS engine. It's not too bad, but you'll find mostly things like Realspeak deployed in large scale professional deployments, such as those used by the big boys, telcos/banks/airlines. We deployed Cepstral recently for a client, for a phone-in service used by the general public, and I can tell you that there was quite a bit of work in teaching it with SSML how to pronounce stuff. Again, it really depends on your specific situation. You should definitely try out those two at least and also ensure that the client/stakeholders are aware of limitations. There's a certain expectation of it will speak perfectly these days, followed by disappointment and blame when reality hits them. Regards, -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Eric Fort [EMAIL PROTECTED] Sent: Tuesday, December 02, 2008 3:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cepstral vs festival I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? Thanks, Eric ___ -- 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] cepstral vs festival
Erik - Have you found RealSpeak to be worth the cost? Actually my last note was probably a bit misleading because in the particular cases I mentioned RealSpeak, the platform wasn't Asterisk and Cepstral wasn't even on the radar. Can Cepstral, with the hourly $ spent on tuning, be made to be a reasonable substitute? Nuance would say no :) I'd say maybe. Call up +14164854854, it's a recent project we did for a client using Asterisk, Cepstral, and a lot of custom code. It's a free phone-in service that allows folks to get local traffic, weather, news, commuter transit, border crossing wait times, and more. There's obviously quite a bit of domain-specific, dynamic, constantly changing text, so this is certainly an example of pushing it to the max. Just think of all the street names it has the potential to mispronounce. It's a work in progress, but it's very promising. Definitely an example of a lot of hourly $ spent on tuning as you put it. My results: The RealSpeak sample was more clear than the Cepstral. Depends on what you mean by more clear. As Brent Davidson mentions, make sure you're comparing 8khz to 8khz, or similar. If you mean it pronounces things better, then I agree. That being said, I'd really be interested in hearing if anyone has done a RealSpeak-to-Asterisk conduit. I wasn't able to quickly uncover how they interact with third-party systems - is it VoIP? A C library? Some sort of HTTP socket? The more methods we can get working with Asterisk, the better, because not every implementation of a voice system has the same requirements... MRCP is the standard for interfacing with ASR and TTS engines (including RealSpeak) in other platforms. Brief Googling reveals a previous flame war on asterisk-dev regarding MRCP. I have no idea if it's implemented in Asterisk now. Regards, -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax with asterisk
Hi! I'm new to this list. I tried to search the list archive for a solution on my current setup, but couldn't find any. We have an asterisk connected directly to the PSTN with 2 E1 lines through a Sangoma A102d interface. We also have a regular FAX machine. My question is how to get the fax service handled by asterisk? I want to cancel the analog line I have for the FAX machine today. What would be the best solution? Fax machine and asterisk is on the same LAN, not much load, with high end switches etc. Can I expect good results with using our existing FAX machine, connected to asterisk through an ATA box? Best Regards, Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
Hi Giorgio, Thanks for your answer. Your setup is exactly what we're thinking of. We have 1100 DID's, so that shouldn't be a problem at all. Which ATA box are you using? Erik On Sep 23, 2008, at 2:06 PM, Giorgio Incantalupo wrote: Hi Olivier, We DO NOT use faxdetect because it does not work properly. That's why we link a PRI DID to it, so when people call that DID the fax machine gets direct fax data without passing thru faxdetection. Giorgio Incantalupo. Olivier wrote: 2008/9/23 Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi Erik, we use an ATA device connected to the fax machine. If you want to receive faxes, since Asterisk fax detection is not reliable Hi, Which fax detection did you used, then ? , use one DID to link it directly to the ATA: you lose a number but you gain a fully-working fax! Giorgio Incantalupo. Erik Haider Forsen wrote: Hi! I'm new to this list. I tried to search the list archive for a solution on my current setup, but couldn't find any. We have an asterisk connected directly to the PSTN with 2 E1 lines through a Sangoma A102d interface. We also have a regular FAX machine. My question is how to get the fax service handled by asterisk? I want to cancel the analog line I have for the FAX machine today. What would be the best solution? Fax machine and asterisk is on the same LAN, not much load, with high end switches etc. Can I expect good results with using our existing FAX machine, connected to asterisk through an ATA box? Best Regards, Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk
Hi Matthew, Thanks for your suggestion. The problem is that most of our users would not feel comfortable with using software fax solutions. So we will have to stick with the old fax machine. Our reception takes care of the fax machine, receiving and sending faxes. This one fax is shared by ~ 600 employees. Erik On Sep 23, 2008, at 3:25 PM, Matthew Marion wrote: Hey Erik, You can also check out pika technologies which supply chan_pika. This comes with a fax application that will let you do your faxes in asterisk (even using non-pika boards). Works pretty good... pikatechnologies.com mattm On Tue, Sep 23, 2008 at 4:51 AM, Erik Haider Forsen [EMAIL PROTECTED] wrote: Hi! I'm new to this list. I tried to search the list archive for a solution on my current setup, but couldn't find any. We have an asterisk connected directly to the PSTN with 2 E1 lines through a Sangoma A102d interface. We also have a regular FAX machine. My question is how to get the fax service handled by asterisk? I want to cancel the analog line I have for the FAX machine today. What would be the best solution? Fax machine and asterisk is on the same LAN, not much load, with high end switches etc. Can I expect good results with using our existing FAX machine, connected to asterisk through an ATA box? Best Regards, Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Trunk and normal
Bilal - I think you're perhaps confusing two meanings of the word trunk. In this case, trunk is referring to the trunk of the SVN development repository, not SIP or IAX trunks. This can be seen as the main development area for asterisk. On Tue, Sep 2, 2008 at 10:22 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Sorry, but I did not find in the below link anything answering the difference between the trunk and not trunk version? When to use asterisk trunk and asterisk normal? Regards Bilal --- On Tue, 9/2/08, Dan Julius [EMAIL PROTECTED] wrote: From: Dan Julius [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk Trunk and normal To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, September 2, 2008, 9:33 AM Hi, checkout http://svnbook.red-bean.com/en/1.4/svn.tour.importing.html#svn.tour.importing.layout this explains about versioning Dan On Tue, Sep 2, 2008 at 3:56 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; I see and hear about the Trunk version, and sometimes when I ask about something (like media timeout for SIP trunk), then they say ur asterisk vesion should be trunk version. What is the difference between Trunk version and not Trunk version? And how can I obtain the Trunk version? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erik Anderson http://andersonfam.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Trunk and normal
Usually you'd only need to go to the trunk to get features that haven't made it into the stable tarballs yet. On Tue, Sep 2, 2008 at 10:37 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Yes I mean the trunk for the development, when I have to select such version and when I can use the normal? Regards Bilal --- On Tue, 9/2/08, Erik Anderson [EMAIL PROTECTED] wrote: From: Erik Anderson [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk Trunk and normal To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, September 2, 2008, 11:33 AM Bilal - I think you're perhaps confusing two meanings of the word trunk. In this case, trunk is referring to the trunk of the SVN development repository, not SIP or IAX trunks. This can be seen as the main development area for asterisk. On Tue, Sep 2, 2008 at 10:22 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Sorry, but I did not find in the below link anything answering the difference between the trunk and not trunk version? When to use asterisk trunk and asterisk normal? Regards Bilal --- On Tue, 9/2/08, Dan Julius [EMAIL PROTECTED] wrote: From: Dan Julius [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk Trunk and normal To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, September 2, 2008, 9:33 AM Hi, checkout http://svnbook.red-bean.com/en/1.4/svn.tour.importing.html#svn.tour.importing.layout this explains about versioning Dan On Tue, Sep 2, 2008 at 3:56 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; I see and hear about the Trunk version, and sometimes when I ask about something (like media timeout for SIP trunk), then they say ur asterisk vesion should be trunk version. What is the difference between Trunk version and not Trunk version? And how can I obtain the Trunk version? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erik Anderson http://andersonfam.org -- Erik Anderson http://andersonfam.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to monitor Asterisk logs ?
On Tue, Jul 15, 2008 at 3:22 PM, Olivier [EMAIL PROTECTED] wrote: Hi, How can I be notified anytime a given warning message appears in Asterisk logs ? Oliver - This is a project I've had my eye on for a while: http://www.splunk.com I've never used it, nor have I set it up, but from reading the feature list, it looks like it's able to keep an eye on any number of log files and notify you if it sees an error. Unless they have built-in asterisk support (which I doubt), I'd bet you'd need to specify some regex rules for what constitutes an error. Anyway - report back if you end up giving it a try. I've wanted to get it set up for several months now, but haven't been able to due to lack of play time in my work schedule. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] First-time queue app: verifying human member?
Good evening all - for the first time, I'm implementing my first-ever queue in asterisk. Overall, it's a pretty simple setup, 4 static members, very low call volume, etc. The one thing that has stumped me so far, though, is the following... This is a queue I'm setting up for contacting our IT support staff off-hours. As such, I've just added the cell phone numbers of our staff as members. I'd like to somehow verify that it's an actual human answering the phone when a member is dialed and not their mobile phone's voicemail. Is that possible? I'd envision just requesting that the member press 1 or something to accept the call. I currently have the timeout in queues.conf set low enough so that the call will never automatically roll over to that member's mobile voicemail, but I can't guaranty that the staff member won't just hit Ignore on their phone and send it directly to voicemail. Ideas? Thanks! -Erik -- Erik Anderson http://andersonfam.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Openfire Asterisk-IM Plugin Performance Observation
On Fri, Jun 20, 2008 at 12:47 PM, JR Richardson [EMAIL PROTECTED] wrote: So now the PBX is over 1.2 Gig for the installation. Typical PBX installs are under 600 Meg. This makes me wonder about server stability, reliability and performance as uptime creeps on and user count increases over 50 to 100+. Increased data on the hard drive won't really have an affect on reliability or performance. Can anyone give me feedback on real world experience with this type of setup and any performance issues that my arise? I can't speak directly to the asterisk + openfire situation. I can, however, say that I've been running openfire for nearly a year now on a very highly-loaded server (other than openfire, it's running nagios and cacti, monitoring about 300 devices around our network) - the load average on this 5-year single processor old dell server is pegged near 1.00 24x7. I haven't had a single problem with openfire, and I have between 50 and 100 open sessions at any one time. In the year that I've been running openfire, I've only had to restart it once, and that was to upgrade the software. It takes very little CPU, and a modest amount of RAM. Is it better for production to run Openfire on a separate server than the PBX? What's your definition of better. Is it better to not have all your eggs in one basket? Is it better to only need to purchase one server? Is it better to only have one server to manage/update/etc versus two? My biggest concern is deploying a 100+ user environment with high call volume and high chat volume. Java seems to be a bit resource hungry with the user notifications and call pop ups. I would hate to have the IM server walking over Asterisk and affecting call quality or PBX stability. Speaking personally, I'd have no problems putting openfire and asterisk on the same box. If needed, you could even just nice the openfire process down to a lower priority than asterisk - it's not as latency-sensitive as asterisk is. I'd doubt you'll need to do that, though. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Browser based VoIP client? None of them are very full featured
On Wed, Jun 4, 2008 at 5:52 PM, Bob G [EMAIL PROTECTED] wrote: None of them have features like hold, transfer, voice mail, dtmf, conference as far as I know none of them has caller ID Only 1ezphone.com has all that and the buttons are programmable for CRM features. Hrm: - no apparent compatibility with any service other than that which is offered via 1ezphone - Frequent spammy emails. - Dubious claims on website: ...we are going to make the only phone portal you will every want. - Some poor person's info revealed on the User Account page: http://1ezphone.com/profile.html - Revelation of someone's call history: http://1ezphone.com/callhistory.html# I, for one, won't be giving this a try any time soon. -Erik ___ -- 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] using gtalk received instant messages in the dialplan
I have been doing some reading about gtalk and asterisk. Most of it is pointed to enable using gtalk for making phonecalls. Would it be possible to use gtalk instant messaging input (just some text send to the gtalk account configured on an asterisk box) into the dialplan. This way you could use gtalk im to trigger all kind of events like sending an sms, adding sip entries to the system, start conferences etc. etc. The basic question is: is it possible to store the received Gtalk message into a variable that can be used to trigger events in a dialplan (which isn't actually a dial plan anymore) or doesn't anything like that exists at this moment. Is this just a crazy thought or does the idea of triggering events in the dialplan via Gtalk im input make sense. I was thinking about a call with the im message as a variable in it starting a local channel that goes into the relevant part of the dialplan. examples: sms: Could you please call Mark 0612345678 for sending an sms sip_entry: 500 cdwtg_34$ ALAW snom320 for adding a sip entry to sip.conf with number 500 and password cdwtg_34$ for a snom320 conference: 0591234567 0201234567 0612345678: for setting up a conference between this numbers All the logic has to be in the dialplan or scripts but it all should start with receiving a message send by a gtalk client. My personal opinion is that it would make a great and easy to explain user interface that can be used from every pc and every pda or smartphone. ___ -- 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] Dutch Asterisk mailing list
As far as I know there is no Dutch Asterisk mailing list but there is a Dutch Asterisk forum. See http://forum.asteriskportal.nl/ It is not an answer to your question but you are more then welcome to join the forum. Erik de Wild Tripple-o Your Asterisk migration partner ___ -- 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] nokia 770 has a build in mic, asterisk and iphone
was almost tempted to try it, but time was short at the time, and holding an N770 to my head seemed a bit silly.. (built in mic and speakers, but no socket for an external mic) Gordon I run Asterisk on a Nokia 770 and as a mini pbx it run pretty smooth. I used it, just for fun and demonstration, with a sip account and 4 concurrent sip calls were running smooth. For those just making and receiving phonecalls now and then it is a very useble pbx. a little bit off topic: The Nokia 770 has an internal mic. It is the little hole next to the hole for the power supply. It looks like a reset whole but it isn't and I'm afraid you will ruin the mic if you try to reset your Nokia 770 this way. With the Gizmo client for the Nokia 770 you can make (partly free) phonecalls and yes, you have to keep the Nokia 770 against your ear. about Asterisk and iPhone I run Asterisk on an iPhone too. It fascinates me to have Asterisk running on small devices like the Nokia 770 and the Iphone. It was running but it took almost 100 % of the cpu power, turning the Iphone into a device to keep your hands warm during cold days. Something most have gone wrong with building Asterisk . As soon as Asterisk is up and running it is actually the same as running it on a normal server. For using Asterisk as the device pbx (routing incoming calls, moh, queueing, transfer, voicemail etc.) I think there should also be a sip phone available that can be registered on the local Asterisk. The first sip phone (Fring) is a smart proof of concept but certainly not ready for normal use. With a sip phone available you can make outbound calls. sip phone - asterisk - sip provider - pstn-net And receive inbound calls the other way around. I'm very interested in using my iPhone as a sip phone with PBX functionalities but it seems not to be an easy task to make a stable sip phone for the iPhone with Apple trying everything to prevent this from happening. ___ -- 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] Dutch Asterisk mailing list?
What is the most reliable method for Asterisk to detect the Called ID for incoming calls on an analog line in the Netherlands? In Holland you have to pay to receive cid info on the incoming line. If you don't pay at the moment you can start with that. There are 2 ways for a provider to deliver the cid,ETSI en FSK. In Holland (with a couple of other countries) ETSI is used so if you have a phone that only supports FSK the CID will never work. I still have a couple of ETSI - FSK converters catching dust. So if you pay for CID but your phone doesn't support and you have a pot line connected to your Asterisk server I can provide you with a solution for a couple of EUR. If you use the proper card maybe you can adjust the settings so it supports ETSI instead of FSK. I used X100P cards and needed the convertor to get proper CID If the Dutch mailing list starts I will join ;-) Erik de Wild Tripple-o ___ -- 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] Not hearing first prompt
This does the trick for me. Sorry for not posting it earlier
As long as there is no answer the channel takes a second pause by
jumping to (wait). As soon as the status is not NOANSWER anymore the
routine jumps to (go_on) and plays the special_message or any other
message of our choice.
exten = s,1(answer),Answer()
exten = s,n(gotoif),Gotoif($[ ${DIALSTATUS} : NOANSWER]?wait:go_on)
exten = s,n(wait),Wait(1)
exten = s,n,Goto(gotoif)
exten = s,n(go_on),Playback(special_message)
Erik de Wild
Tripple-o
Your Asterisk migration partner
Another solution that works for me is to add Playback(silence/1)
just
before whatever you are about to do. Something about the playback
command opens the channel up.
-Brent
Sherwood McGowan wrote:
Alan Lord wrote:
Sherwood McGowan wrote:
snip /
Hrm...I have encountered this before and sometimes doing an
explicit
Answer() then a Wait(2), then calling the service can help.
Hope this is helpful
Sherwood McGowan
Bingo!
Thanks a bunch. That sorted it.
Al
Fantastic! Very glad I could help.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel 1.4.10 doesn't build on debian etch epia itx system
Today I have been messing around with updating my residential phonesystem (it was running a 1.0 version from years ago). I have downloaded the last source packages for zaptel-1.4.10.1and asterisk-1.4.19.2. Zaptel doesn't want to build. After a long time of making this is the output that stops it suddenly. Does it makes sense to try another lower version of Zaptel, do I miss a package or should I change a line in the Makefile like I had to do to build Asterisk (Proc=i586 instead of Proc=uname -m which result in i686. The updated box is now running without zaptel and it seems to work ok but I would like to add ztdummy for conferences. Any suggestion to solve this problem is very welcome. Friendly regards, Erik de Wild output uname -a Linux debian 2.6.18-6-486 #1 Sun Feb 10 22:06:33 UTC 2008 i686 GNU/Linux # gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE- DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o zttool zttool.o -lnewt # Can't locate Config_heavy.pl in @INC (@INC contains: /etc/perl /usr/ local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/ share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/ site_perl .) at /usr/lib/perl/5.8/Config.pm line 65. # make[2]: Entering directory `/usr/src/asterisk/zaptel-1.4.10.1/kernel/ xpp/utils' # cc -I../.. -o print_modes -g -Wall print_modes.c # ./print_modes init_fxo_modes # for i in zt_registration xpp_sync lszaptel xpp_blink zapconf zaptel_hardware; do perl -I./zconf -c $i || exit 1; done # Can't locate File/Basename.pm in @INC (@INC contains: ./zconf /etc/ perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/ perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/ local/lib/site_perl .) at zt_registration line 11. # BEGIN failed--compilation aborted at zt_registration line 11. # make[2]: *** [perlcheck] Error 1 # make[2]: Leaving directory `/usr/src/asterisk/zaptel-1.4.10.1/kernel/ xpp/utils' # make[1]: *** [utils-subdirs] Error 2 # make[1]: Leaving directory `/usr/src/asterisk/zaptel-1.4.10.1' # make: *** [all] Error 2 ___ -- 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] Problems passing variables from a macro
I pass a value from a macro by storing the value needed to the $
{MACRO_RESULT} variable. This is returned and because of this
available after finishing the macro. I'm not sure that it works in the
way you are looking for but it works for me.
Erik de Wild
Tripple-o
___
-- 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] Require a Touch-Tone to Connect? proof of concept with meetme()
I have read the post about the touch tone before to connect so
transfered calls don't end up in voicemail boxes of mobile phones. I
have done some work last year on transfering an inbound call to
different extensions by using meetme() and local channels so a whole
group can start talking. I end up with a remarkably low number of
lines and it is actually working . It is just a proof of concept that
can be complemented with voiceprompts and a mechanism to make sure
that just one extra line enters the conference room. If you have
improvements please share them on the mailing list. I hope someone
will find this usefull.
Below are the actual Asterisk lines. It is pure old fashioned Asterisk
without any additional AGI scripts or whatever.
With friendly regards,
Erik de Wild
Tripple-o
Your Asterisk migration partner
; this is where te inbound call is routed to with exten =
whatever,n,Goto(inbound_forking,s,1)
;;
[inbound_forking]
; this are the three local channels used for dialing the external or
local numbers. In this
; example all the numbers are external
exten = s,1,Dial(local/[EMAIL PROTECTED]local/[EMAIL PROTECTED]local/
[EMAIL PROTECTED],20)
; this is where the inbound call is routed to the conference room,
notice the /n
;;
exten = s,n,Dial(local/[EMAIL PROTECTED]/n,10)
[meetme]
;; extension for the inbound call
;;;
exten = inbound,1,MeetMe(9000,qM1)
exten = inbound,n,Hangup()
;
; extensions for the three (or more) different outbound lines
; that are routed into a macro
exten = intern1,1,Dial(SIP/3120/0031621xx,
20,M(meetme_test))
exten = intern2,1,Dial(SIP/3120/0031642xx,
20,M(meetme_test))
exten = intern3,1,Dial(SIP/3120/0031556xx,
20,M(meetme_test))
; this is the macro for joining the conference room
; first it read the number of
[macro-meetme_test]
exten = s,1,Set(ROOMNUMBER=9000)
exten = s,n,MeetMeCount(${ROOMNUMBER}|COUNT)
exten = s,n,Wait(2)
exten = s,n,SayNumber(${COUNT}) ; as long as the number is 0 or 1
it makes sense to join
exten = s,n,Authenticate(1) ; here is the one
touch needed before you can join. A voicemail box of a mobile can't do
that ;-)
exten = s,n,Meetme(9000)
exten = s,n,Hangup()
___
-- 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] Minimum upload speed for Asterisk?
On Thu, May 1, 2008 at 3:19 PM, Frank Tarczynski [EMAIL PROTECTED] wrote: Is 384kB up too slow? Probably not. Is there any guidance for the minimum upload speed for an Asterisk box? I'm guessing this is for just a few calls at a time, correct? I'd guess that rather than these quality issues being caused by cramped bandwidth, they're actually being caused by latency issues. Have you ever checked the latency of the connection between your asterisk server and your SIP/IAX endpoint? If it's really high (say 300ms+) or if the latency is really erratic, you'll have quality issues. You didn't mention whether you are doing traffic shaping on your upstream connection, so I'll assume you're not. That would be something good to look into - with traffic shaping, you can prioritize your VoIP traffic over all other types of network traffic. -erik ___ -- 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] Newbie Polycom: Where is SoundPointIPWelcome.wav used?
On Tue, Apr 8, 2008 at 12:06 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: When I downloaded the sip and bootrom from Polycom website, I noticed a file called SoundPointIPWelcome.wav. However, I have no idea where and when it was used. I played the wav file but I have never heard the phone using this wav file before. Does anyone know what it is used for? It's played at the completion of the boot process. It's always been very quiet on the models I've worked with. -erik ___ -- 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] Problems with DELL 1600
On Wed, Apr 2, 2008 at 9:37 AM, Ruben Zamora [EMAIL PROTECTED] wrote: I just want to know if anyone have problems with server DELL 1600, Like: Hangup Call. Give us some more details of your setup and you'll probably have better chances of getting an answer. -Erik ___ -- 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] IVR Asterisk Voice Recognition - Asterisk with lumenvox
On Wed, Apr 2, 2008 at 10:24 PM, Al Baker [EMAIL PROTECTED] wrote: Clearly all of this not feasible in a IVR environment, so, in the absence of all this, just how good , and how sophisticated of a voice recognition can one achieve ? Have you ever called Google 411? 1-800-GOOG-411 It'll blow your mind ;-) ___ -- 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] FYI about my Mona Vie business venture
On Mon, Mar 24, 2008 at 1:56 PM, BerkHolz, Steven [EMAIL PROTECTED] wrote: I am not going to go into a sales pitch. This is just an FYI to this opportunity. Sorry, but one man's opportunity is another man's sales pitch. To sign up to be a distributor , which is required to make money, is $54 A case of Mona Vie is $120. A case will last 2 people a month. (you only take 2 ounces a day) This may seem like a lot, but: 1. You will not need to buy any vitamins. 2. My brother-in-law is already making $200 a month, after being in the system for a month, So his cost for the Mona Vie is covered and he is making $80 a month. 3. As more people sign up, the amount he gets back will increase. I am very excited with this, both in the health benefits I am already seeing, and the income potential. Sure looks like a sales pitch to me... This is spam, pure and simple. Please stop abusing the list for your own business opportunities. -Erik ___ -- 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] phpagi
On Wed, Mar 19, 2008 at 12:48 PM, Carlos Carvalhar [EMAIL PROTECTED] wrote: How do I install phpagi? http://phpagi.sourceforge.net/ Since phpagi is really just a set of php libraries, all you need to do to install is dump it somewhere and add that location to your php include_path. -Erik ___ -- 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] RES: phpagi
On Wed, Mar 19, 2008 at 1:31 PM, Carlos Carvalhar [EMAIL PROTECTED] wrote: But when I download the gz file it doesn't uncompress as php files, the phpagi-2.14.gz file returns a phpagi-2.14 file...and I tried with winrar and 7-zip that usually uncompress gzip files without problem. How can I get the php files of the class phpagi? How did you download it? $ wget http://superb-east.dl.sourceforge.net/sourceforge/phpagi/phpagi-2.14.tgz $ tar zxvf phpagi-2.14.tgz $ cd phpagi-2.14 $ ls -Erik ___ -- 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] Is Asterisk ready for Prime-Time?
On Wed, Mar 19, 2008 at 4:38 PM, Bill Andersen [EMAIL PROTECTED] wrote: Although this is a users list, I think it is more of a list for Asterisk resellers. I'd be interested in how many of you are simply using Asterisk as your phone system and NOT selling your services or an Asterisk based solution? Anyone? Just a user? /me raises hand. That being said. As just a user of Asterisk, it is clear that if I want to continue with Asterisk, it looks like I really need to learn the ins-and-outs of Asterisk and ditch my pre-packaged solution. Off to Amazon for to find TFOT (I want the hard copy :) Agreed - I'm sure you'll be much more happy with the stability of your vanilla asterisk implementation (assuming you're running on a stable OS and server-class hardware) as well as being much more comfortable with what's going on behind the scenes. -Erik ___ -- 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] Druid Open Source Edition
On Mon, Mar 17, 2008 at 12:09 PM, Brett Crapser [EMAIL PROTECTED] wrote: Then I noticed how all the asterisk files/directorys had been 777'ed. Ouch - I think I'll pass as well. -- Erik Anderson http://andersonfam.org ___ -- 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] Mail Server
On Thu, Mar 13, 2008 at 4:04 PM, Mike Hammett [EMAIL PROTECTED] wrote: I need to setup a small mail server on a local network. It only needs SMTP ability as it's just so Asterisk can send out emails. The machine has sendmail installed. My primary mail server seems to be rejecting the messages. Some research says something isn't configured properly. What do I have to do so the outside world accepts emails from my Asterisk box? It is behind a NAT. Does your ISP provide an SMTP server you can use? If so, it's usually easiest to set that up as a smarthost and tell sendmail to send through that server. If this isn't an option, you need to make sure that your asterisk server has a valid publicly-available DNS record (and reverse DNS). That's most likely the reason the remote server is rejecting these emails. -erik ___ -- 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] DID number
On Sun, Mar 2, 2008 at 3:21 AM, Mike [EMAIL PROTECTED] wrote: Just curious if anyone has suggestions on how one can get a near FREE(I hope) DID number. Hey Mike - give IPKall a try: http://www.ipkall.com/ They'll give you a free Washington state DID along with free SIP to your asterisk server. -Erik ___ -- 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] Newbie on VoIP
On Mon, Mar 3, 2008 at 12:40 AM, NOC Ph [EMAIL PROTECTED] wrote: [NOCPH] I have to open the SIP port and web. Another question, the SIP port is 5060 UDP, how about the conference? Does it use the same port also? That's a good start, but you'll also need to open the RTP ports as well - these usually fall in the 10k-20k udp range. 5060/udp is used for call signalling only, the actual voice data can use a variety of ports, depending on how you're set up. You can specify what RTP ports you want to use in your rtp.conf. -erik ___ -- 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] test
On Tue, Feb 26, 2008 at 10:59 AM, Joel Solanki [EMAIL PROTECTED] wrote: checking wheather my mail goes to asterisk users mailling list or not ACK. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I call cheap to UK cell phones
On Tue, Feb 26, 2008 at 1:19 PM, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Greetings, How can I call cheap to UK cell phones. I am located in Toronto, Canada, but need to call UK cell phones both from Toronto and London. I'd guess you could get an account with one of these providers: http://voip-info.org/wiki/view/VOIP+Service+Providers+Business+Europe#UnitedKingdom -erik ___ -- 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] Had it with Dell Garbage
On Tue, Feb 26, 2008 at 2:10 PM, Matt [EMAIL PROTECTED] wrote: I've had it with Dell server garbage.They seem to change RAID controllers as much as I change socks, and then the controllers don't work with Linux, unless you load a new driver.They sell servers with a PCI-e slot in them, but then you get it and find out the RAID controller is using the PCI-e slot! Their sales folks are dumber than rocks, and they change them more often than I change underwear. [end rant]. Ouch! :-) I can't speak to the PCIe issue, but I've never in my life had compatibility issues with the Dell RAID controllers. What kernel are you on? Can anyone recommend an IBM or Gateway server that you have used with Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has room for one or two PCI-express interface cards? Gateway server? Ew. Have you looked into the new Sun servers? I've been researching them lately, and they have some compelling offerrings. They also offer full support for linux as well... -erik ___ -- 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] Multiple Asterisk Servers. One Conference
On Wed, Feb 20, 2008 at 8:49 PM, Klaverstyn, David C [EMAIL PROTECTED] wrote: I currently have about 10 Asterisk servers scattered around the place each hosting their own dynamic conference centre. Is there any way that when people join these conference centres on each server that somehow Asterisk bridges the conference centres on each server to form one large conference? In theory, this wouldn't be difficult at all. I'd imagine it could go something like this: set up one central conference server. Each branch server would call an extension (zap/sip/iax/whatever) on the main server, which in turn would dump it into a certain meetme room. Alternatively, you could have the central server call out to the branch servers and join them to the meetme room. In practice, though, I have no idea how the audio quality would be. -erik ___ -- 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] restart asterisk daily
On Thu, Feb 14, 2008 at 8:38 PM, Al lists [EMAIL PROTECTED] wrote: Always rely on free -m to see how much free memory you have not top. You could install and use htop - it's a much more functional (and informative) version of top. It shows the difference between shared/buffer/cache memory. -erik ___ -- 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] restart asterisk daily
On Thu, Feb 14, 2008 at 9:37 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: It also consumes more CPU. True, a fraction more. If you have that little overhead on your server, though, that this would cause a problem, you probably should upgrade your hardware, IMHO. -eriik ___ -- 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] Semi-OT: bluetooth conference phone?
All - I've been trying to pick out a bluetooth conference phone that I could use with a softphone along with my asterisk server. I've been looking at the TrendNet TVP-SP4BK. Have any of you used this device or any other bluetooth conference phone? How have your experiences been? Thanks! -Erik -- Erik Anderson http://andersonfam.org ___ -- 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] Cannot hear voice through SIP Phone from one side
On Feb 5, 2008 2:32 PM, Sanjoy Rath [EMAIL PROTECTED] wrote: The Asterisk server is a linux server. There is no firewall between the servers. It is in a DMZ. My bet is that it's not a *true* DMZ. You're still dealing with NAT, and that's what's causing the one-way audio. This topic has been discussed ad nauseam on the list and is documented quite well on the wiki - search there and you'll most likely find the answers you're looking for. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to hookup to cell phone for outbound calls?
On Feb 5, 2008 2:37 PM, Drew Gibson [EMAIL PROTECTED] wrote: How about http://www.mgamble.ca/oss/iphone_asterisk/ ? Hah! Cool, but quite ridiculous. :-) ___ -- 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] Server Compatibility List for Asterisk
It is my understanding that the cast majority of the compatibility issues went away with the recent chipset change on the digium cards. Soa compatibility list really isn't needed. I've run the digium cards on all manner of Dell hardware (from old-school desktops all the way to the high end servers) and have never had issues. On 1/31/08, broadband Voice [EMAIL PROTECTED] wrote: Digium has a compatibility list of servers, however, it has not been updated since 2006. One of the servers on the list has since been taken out of production by Dell. Here are the remaining servers on the list: HP Proliant DL360IBM x206IBM x346 Does anyone has a most recent list and I will be adding the digium cards for T1 the 220 series with echo cancellation? -- Erik Anderson http://andersonfam.org ___ -- 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] Join me on Last.fm!
Classy. On Jan 25, 2008 2:37 PM, Sina Owolabi [EMAIL PROTECTED] wrote: Hi asterisk-users@lists.digium.com, Add me as a friend on Last.fm so we can share our music taste :) Check out what I'm listening to. A personal note from me: boo! Signing up is free and takes less than a minute. Just click here to automatically accept my add. Visit my music profile and leave me a shout! I'll see you around, - Sina Owolabi PS: I'm shina01 on Last.fm. You received this message because someone (Sina Owolabi) who knows you sent you an invitation to join them on Last.fm. Your address was not saved and we will never contact you unsolicited. For more information, see our privacy policy at: http://www.last.fm/help/privacy.php. ___ -- 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 -- Erik Anderson http://andersonfam.org ___ -- 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] Replacement for Allison
On Jan 24, 2008 10:14 AM, Matt [EMAIL PROTECTED] wrote: That worked... hrmm not that great... anyone know of any decent sounding recording of Allison for Asterisk? What's your definition of decent sounding? IMHO and that of many of my co-workers, the default Allison recordings sound great...not sure exactly what you're looking for. -erik ___ -- 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] I am having a problem connecting my X-Lite to my Asterix box
On Jan 20, 2008 7:14 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote: I have added two extentsions. I am try to test connecting X-lite to the server. I have two extension one 1000 with password 1234 and one 2000 with password 2000. Andrew - could you send us the relevent sections of your sip.conf? That would be quite helpful in helping you troubleshoot this problem. Also, please post any messages that appear on the asterisk console when you try and register your x-lite phone. -Erik ___ -- 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] I am having a problem connecting my X-Lite tomy Asterix box
On Jan 20, 2008 7:47 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote: Here are my log information. [Jan 20 12:34:00] NOTICE[2637] chan_sip.c: Registration from 'Andrewsip:[EMAIL PROTECTED]' failed for '192.168.3.116' - Device does not match ACL [Jan 20 12:35:33] NOTICE[2637] chan_sip.c: Registration from 'Andrewsip:[EMAIL PROTECTED]' failed for '192.168.3.116' - Device does not match ACL I am not a Linux guy I am a Windows Programmer I can not get to the sip.conf? Are you using asterisk or trixbox? If asterisk, just open up /etc/sip.conf in an editor... ___ -- 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] I am having a problem connecting my X-Lite tomy Asterix box
On Jan 20, 2008 8:06 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote: Windows XP. Andrew - you're going to need to get us your sip.conf before we can really assist you any further. ___ -- 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] Anyone Using a Dell PowerEdge T105 in Production
On Jan 16, 2008 6:39 PM, Steve Totaro [EMAIL PROTECTED] wrote: Unbeatable price for a low end Asterisk server (or any server for that matter) http://configure.us.dell.com/dellstore/config.aspx?c=uscs=04kc=6W300l=enoc=bednv4ks=bsd I wonder if anyone has any experience with this box and Digium or Sangoma hardware? Any compatibility issues? If not, I might stock up on them. Wow - that *is* a great price. I don't have any of this particular box in production, but I do have 2 PowerEdge SC440s (one step up from the T105) running asterisk along with Sangoma PRI cards. They're working great. I really only have two issues with these low-end servers: 1. You can't order 'em with RAID support. I'm getting around this by using software RAID1 in linux, but I'd much prefer having a hardware RAID controller. 2. The Dell DRAC remote management cards aren't compatible with these low-end server motherboards. I've become *completely* addicted to the DRAC cards on the high-end PowerEdges, to the point that I now refuse to order a server without a DRAC card. That said, I'm sure this server would run a small/medium asterisk install just fine. -Erik ___ -- 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] Anyone Using a Dell PowerEdge T105 in Production
On Jan 16, 2008 7:28 PM, Steve Totaro [EMAIL PROTECTED] wrote: You can add the raid option for $199. I think I might pickup about ten of them at this price. I can always resell them as general purpose servers or even workstations if Asterisk/Zaptel/Linux does not like the boxen. Ahh - nice. That wasn't an option when I ordered the SC440. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Traffic Shaping
On Jan 10, 2008 8:24 AM, Drew Gibson [EMAIL PROTECTED] wrote: It has 5 ports! Although the ports are labeled as 1 Internet port and 4 LAN ports, each can be assigned to a VLAN of your choosing and you can use them as you please (at least you can under openWRT). Yup - you can do the same with DD-WRT. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Traffic Shaping
On Jan 9, 2008 8:33 PM, Matt Riddell [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Does anyone know of a cheap (very cheap) dual port traffic shaping box (i.e. sub $100) that can be configured for IAX/SIP? Pick up a Linksys WRT54GL and install dd-wrt on it. That will traffic shape any type of traffic you want. I have installed several of these around the country and they work great for prioritizing VoIP traffic. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Traffic Shaping
On Jan 9, 2008 9:40 PM, Matt Riddell [EMAIL PROTECTED] wrote: Heh yeah that's what I was thinking of doing. What's the traffic shaping like? Can I specify max bandwidth etc or use hfsc shaping? DD-WRT will do both HTB and HFSC shaping, though I've only ever used HTB. Here's the dd-wrt wiki page on its QoS implementation: http://www.dd-wrt.com/wiki/index.php/Quality_of_Service Looks like they don't recommend HFSC currently due to some lag issues. That might have been fixed, though, in the more recent firmware builds. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dialplan date and time operations
Thanks!
I got it now!
Here is a sample for a delayed callback after a caller gets to a users
voicemailbox. Purpose: Reminder for people that they got a message on their
v. box.
exten = 1002,1,Answer
exten = 1002,2,Set(CHANNEL(musicclass)=default)
exten = 1002,3,Queue(test|t|||5)
exten = 1002,4,Voicemail(b1205)
exten = 1002,5,System(echo -e Channel: SIP/we-static\\nCallerID:
VOICEMAIL
8500\\nContext: test\\nExtension: 444 /tmp/${UNIQUEID}.call)
; add 15 minutes (in seconds 900) to the epoch time
exten = 1002,6,Set(newepoch=${MATH(${EPOCH} + 900 |int)})
; write it out for debugging purpose
exten = 1002,7,NoOp(${newepoch})
exten = 1002,8,System(touch -t ${STRFTIME(${newepoch},,
%Y%m%d%H%M)} /tmp/${
UNIQUEID}.call)
exten =
1002,9,System(mv /tmp/${UNIQUEID}.call /var/spool/asterisk/outgoing
/)
exten = 1002,10,Hangup
Kind Regards,
Erik
Am Mittwoch, 2. Januar 2008 18:59 schrieb Tilghman Lesher:
On Wednesday 02 January 2008 09:34:24 Erik Wartusch wrote:
No it's even simpler. ( I dont need an IF case)
I just want to add e.g. 15 minutes to the current date / time:
So simply said:
${STRFTIME(${EPOCH},,%Y%m%d%H%M)} + 15 minutes!
My question was how can I do that.? Of yourse e.g. if it's 23.57 pm and I
add 15 minutes the day should increase +1 and the hours start with 0:x
the minutes with 12 ( and not 72 as the normal addition would result).
${STRFTIME($[${EPOCH} + (15 * 60)],,%Y%m%d%H%M)}
___
--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] Asterisk dialplan date and time operations
Hi all, Im using Asterisk 1.4.11 and I want to proceed some time and date operations in my dial plan. (for a time shifted callback). Should look like: CURRENT TIME + x minutes. Of course it should increase the hours for example in this case: 10.59 + 5 minutes = 11.04 I guess I've to use the math function in 1.4 but how can I manage easily the time operations? Kind Regards, Erik ___ --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 dialplan date and time operations
Thanks Doug. Yes. Thats what I plan to do and allready knew about. But I want to know wether there is an easy option to operate with times (add x minutes to a time) or not? My research until now wasn't successful and I'm wondering that nobody before had this problem... (date and time manipulation) Cheers, Erik Am Mittwoch, 2. Januar 2008 13:23 schrieb Doug Lytle: Erik Wartusch wrote: Hi all, Im using Asterisk 1.4.11 and I want to proceed some time and date operations in my dial plan. (for a time shifted callback). If you'll be using call files to do this, you can 'touch' them to a future date and Asterisk will not act on them until that date is reached. Doug ___ --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 dialplan date and time operations
daveC,
No it's even simpler. ( I dont need an IF case)
I just want to add e.g. 15 minutes to the current date / time:
So simply said:
${STRFTIME(${EPOCH},,%Y%m%d%H%M)} + 15 minutes!
My question was how can I do that.? Of yourse e.g. if it's 23.57 pm and I add
15 minutes the day should increase +1 and the hours start with 0:x the
minutes with 12 ( and not 72 as the normal addition would result).
Kind Regards,
Erik
Am Mittwoch, 2. Januar 2008 16:02 schrieb dave cantera:
erik,
you can start here:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime
http://www.asteriskguru.com/tutorials/gotoiftime.html
daveC
Erik Wartusch wrote:
Hi all,
Im using Asterisk 1.4.11 and I want to proceed some time and date
operations in my dial plan. (for a time shifted callback).
Should look like:
CURRENT TIME + x minutes.
Of course it should increase the hours for example in this case:
10.59 + 5 minutes = 11.04
I guess I've to use the math function in 1.4 but how can I manage easily
the time operations?
Kind Regards,
Erik
___
--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
--
===
Erik Wartusch
Deuromedia Technologies GmbH
Barichgasse 40-42
1030 Wien
Austria
Phone: +43 16986442 1205
Fax: +43 16986442 200
email: [EMAIL PROTECTED]
www.deuromedia.com
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Best firmware for Linksys Router thatis SIP AWARE
On Nov 28, 2007 9:44 AM, Dovid B [EMAIL PROTECTED] wrote: So do I. I set SIP to high how ever the calls are still bad. I guess I need to read up a bit more on the firmware and how to set it up correctly. Are the calls poor quality in both directions or on just one of the legs of the call? Implementing QoS on your router will really only help network traffic going *out* of your network. In otherwords, you can really only affect your upload traffic. One thing to consider is that you may just have a poor-quality internet connection. Have you done a VoIP speed test? Here's the one that I use: http://www.voipreview.org/voipspeedtester.aspx This sort of test is ideal for VoIP because unlike most other speed tests, it measures latency, jitter, etc. -erik ___ --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] Sangoma Question
On Nov 28, 2007 10:52 AM, Jeremy Mann [EMAIL PROTECTED] wrote: Do sangoma cards use the standard Zaptel drivers? Or do they have to be compiled externally like Rhino cards? Sangoma maintains a patchset that gets applied to the stock zaptel drivers before compilation. They provide automated tools that will take care of the patching/compiling/installing/configuring for you. -erik ___ --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] Urgent question.
You should be able to issue a stop gracefully command to asterisk. That'll cause it to stop accepting new calls, but will let existing calls continue until complete. -erik On Nov 27, 2007 12:06 PM, Alex Balashov [EMAIL PROTECTED] wrote: In other words, what I need is a way for the upstream switch to somehow think that the B channels are out of service, but without actually taking the B channels out of service and dropping the existing calls. From within asterisk, zaptel, wanpipe, whatever. Is that possible? On Tue, 27 Nov 2007, Alex Balashov wrote: Our provider gives us four PRIs as a trunk group hunt group. Meaning, the provider's switch will cycle through B channels in span 1, 2, 3, ... until it finds one that is available. I have moved spans 2-4 onto another machine. But we have one remaining box with a PRI full of calls and I don't know what to do with them; the box is failing, but dropping them by simply yanking the PRI is not acceptable from a business POV. Sending Congestion() or Busy() in the dial plan wouldn't work because the far-end switch would simply pass that onto the subscriber, rather interpreting it to mean that the B channel is unavailable and it should go on to other T1s in the trunk group. Any ideas? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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 -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Best firmware for Linksys Router that is SIP AWARE
On Nov 26, 2007 7:51 AM, Dovid B [EMAIL PROTECTED] wrote: Hi, I am currently playing with DD-WRT and I like it. I am looking for something that is more SIP Aware. Anyone know one those that are out there ? Dovid - what exactly are you hoping this sip aware firmware will do that dd-wrt doesn't? I've been using dd-wrt in combination with various SIP ITSPs for several years and have had no problems - just add the necessary port forwards and a few traffic shaping rules and it works just fine. I do know that they (the dd-wrt people) have a voip edition of dd-wrt available. I'm not sure what additional functionality it has over the standard version, though. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Best firmware for Linksys Router that is SIP AWARE
On Nov 26, 2007 8:29 AM, David Boyd [EMAIL PROTECTED] wrote: I struggle with the traffic shaping rules, would you be willing to provide additional details as to what you have done in past? Any additional information would be greatly appreciated. Sure - I use the default HTB traffic scheduler. The number one tricky thing about traffic shaping that most people miss is that they don't set their uplink speed correctly. For 99% of the use cases out there, you have no control of your downlink speeds, so there's not a whole lot you can do for that - you really only have control of your uplink packets. So - do a bunch of speed tests and then set your uplink speed to about 80% of your max upload speed. That will ensure that there's always a bit of overhead and that your link itself will never be the uploade bottleneck. After doing this, just start classifying traffic. Here's a synopsis of the rules I use: - DNS - high priority - SIP - express priority - RTP - express priority - HTTP/https - bulk priority - (other p2p applications) - bulk priority Putting those rules in place should make a big difference. You can also specify a specific ethernet jack on the router that will get high priority if that would help in your setup. HTH- Erik ___ --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] Filesharing + video + voice supported Soft phone
On Nov 26, 2007 3:07 PM, Bob Gibson [EMAIL PROTECTED] wrote: VMukti.com I have a few comments for you: 1. Your webserver has been throwing 500 errors all afternoon. 2. It appears that all you've been doing with your time all day is spamming the list with VMukti.com. 3. Do you really think you're convincing any people to check out this product by doing this? Please go away until you can figure out a way to contribute in a meaningful way. -Erik ___ --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] p2p t1 with sangoma hw
On Nov 17, 2007 11:49 PM, Michael J. Liberatore [EMAIL PROTECTED] wrote: I figured that one side would be pri net and the other would be pri cpe, well I chose pri cpe and the next question was asking for a switch type, national isdn 2, att, nortel, etc - that sounds really wrong. Pick national and make sure it's set at both ends. (this is also known as national isdn 2) So basically I am at a stand still, any help would be great, would it be pri net on both sides? If its suppsoed to be pri cpe on one side and pri net on the otherside then what would the switch type be? All verizon told me is that its b8zs/esf, that's it. One end of your T1 link will need to be pri_net and one will need to be pri_cpe. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - best policy for logs
On Nov 15, 2007 12:55 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: In my experience, it's easier to combine them all into one syslog server, and then utilize tools to filter them apart when necessary, since there are more tools to do that than to *combine* them when that is necessary, which it often is. Agreed - I have all of my servers send their syslogs to /var/log/messages on one central logging server. If you want to examine a device-specific log, just use tail + grep. That said, any system logger worth it's salt will make it extremely easy to have device-specific log files if you prefer. -erik ___ --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-stat problem
On Nov 14, 2007 4:15 PM, Richard Cahilig [EMAIL PROTECTED] wrote: Hi, I installed asterisk-addons and asterisk-stats, Its working now except of one problem. The problem is there is no call logs when you open the cdr report. The message is when you open the cdr report is: - Call Logs - Back to Top No data found !!! 1 / 1 Did I missed something in the configuration of mysql-addons or asterisk-stat? Here is my asterisk-stats page: http://203.115.187.91/cdr, the username is admin and the password is password. Thank you very much. Richard - just click search when you go to one of the report pages. It doesn't do the query manually. -erik ___ --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] Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11
Thx John !! Hmm I found now on voip-info.org a lot of Beta releases which should fix my problems... Kind of strange whats going on with Grandstream devices and their firmware ... If you install the latest official release you can expect a few troubles with Asterisk 1.4.11 (one way audio -- randomly, dropped calls). So you have to install the BETAS whether you want or not... That you have to use unique ports is a rumour and not SIP standard. As John said -- IP:Port must be unique . I definitely not understand why I should use random ports. Kind Regards, Erik I`m using several GXP2020 phones with newest Firmware 1.1.4.18. I had issues with phone locking up using 1.1.4.18. I've now gone to 1.1.4.22 and have eliminated that. Asterisk Version: 1.4.11. Me too. Still testing 1.4.13 on a non-production system. I use on every phone the 1 as local port and in the rtp.conf From my knowledge of IP I don't think this is a problem since the address/port would be unique. However the example config I originally had from Grandstream indicated that each phone should use a different port and recommended to use the random port option on the phones. I have since assigned the port number on each phone to 1 plus the extension number. This was done to create a unique port number and to help with troubleshooting when using Wireshark or tcpdump. I set this in the config file for each phone. John ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is wrong with this mailing list
On Nov 13, 2007 11:21 PM, Mohammad Shokuie [EMAIL PROTECTED] wrote: Anyone knows what is wrong with this mailing list its a while all my new posts appear as a reply (branch) for others post, is there any hints i could prevent this issue?? I believe your posts are all showing up correctly for me. That said, this sort of thing can happen frequently if, instead of composing a new email to the list, you hit Reply to an existing message and just change the subject line. -erik ___ --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] What is wrong with this mailing list
On Nov 13, 2007 11:44 PM, Mohammad Shokuie [EMAIL PROTECTED] wrote: HI Erik, thanks for your post, Actually im sending new posts not replying but if you see them correct, how come its wrongly viewed for me. Are you using a speciall software to view mailing lists? Im just using firefox not a special one! You're using firefox? How so? I'd recommend either a good email client (Thunderbird) or a good web email interface (gmail). (I'm using gmail's web interface) ___ --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] Grandstream GXP2020 + Asterisk 1.4.11
Hi, I`m using several GXP2020 phones with newest Firmware 1.1.4.18. Asterisk Version: 1.4.11. It happens several times that users complain that the caller cannot hear the transmitted voice from the phones Also now it happens quite often that callers on hold beeing dropped. Environment: ISDN with chan_misdn and SIP internal calls. No NAT no DNS name (only IPS configured). I configured in sip.conf and on the phone now that alaw is preferred. As I saw in the FMW Bug list that GSM is not a good option Also I set the canreinvite=no as it is also configured in a Grandstream manual. I use on every phone the 1 as local port and in the rtp.conf I allowed a range from 1 - 5. As far my SIP knowledge is up to date the local port has not to differ from phone to phone or I´m wrong? Any idea or useres which had the same problems and fixed it? My sip.conf: [test1] type=friend context=outgoing username=test1 secret=987454 qualify=yes host=dynamic nat=yes canreinvite=no disallow=all allow=alaw allow=ulaw callerid=Test 0 insecure=very Kind Regards, Erik ___ --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] Two PRI setup questions
Yep - as Doug mentioned, give esf framing and national switchtype a try. I have a PRI from ATT in one of my offices, and use this setup. -erik On 11/1/07, Lutgring, Sam [EMAIL PROTECTED] wrote: I am in the process of implementing a new ISDN pri and have a couple of questions. This is a full 24 channels (23 B and 1 D) delivered over a T1 interface. The interface looks good and is not showing any errors. Any help that you can provide would be greatly appreciated. 1) What switchtype should be configured in the zapata.conf file when ATT is using CUSTOM? My understanding is that this equates to the dms100 in Asterisk, is this right? The D channel is coming up just fine, but ATT tells me that they cannot see the B channels. When I try to make a call I get a slow busy and the debug shows an ISDN cause code of 34, no circuit available. 2) Is there a way to see the idle status of a B channel? When ATT tells me they don't see the B channels coming up, is there a way that I can see this in Asterisk??? Thanks in advance. ___ --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 -- Erik Anderson http://andersonfam.org ___ --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] Need T1 crossover cable?
On 10/26/07, Michelle Dupuis [EMAIL PROTECTED] wrote: I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My Sangoma A102D shipped with 2 T1 cables - which I assume are straight through. Do I need to make crossover cables for this scenario? Yes - a crossover *is* needed in this configuration. -erik ___ --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] Compatibility Issues with dell poweredge 1950 and TE110P card
On 10/23/07, Joseph Begumisa [EMAIL PROTECTED] wrote: Has anyone had any compatibility issues with a TE110P card installed on a Dell Poweredge 1950? I noted the following error on the LCD display of the Dell Poweredge 1950: E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. The Dell hardware owners manual states that it means the system BIOS has reported a PCI parity error on a component that resides in PCI configuration space at bus 0, device 4, function 0 and advises that the PCI expansion card be removed and reseated. I had this error on a 1950 while testing a Sangoma quad-port card. Re-seating the PCI expansion board seemed to solve the problem. -erik ___ --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] Extensions.conf for basic IVR?
On 10/22/07, Vincent [EMAIL PROTECTED] wrote: 2008 might be a good year to update * - The future of telephony :-) Version 2 of TFOT was just released a few weeks ago... http://downloads.oreilly.com/books/9780596510480.pdf -- Erik Anderson http://andersonfam.org ___ --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] A linksys SPA921 behind NAT and firewall
On 10/20/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: If you are trying to use non-complied (XML) profiles... don't even bother wasting your time. Why is that? I'm using the xml-style config and they're working just fine. -- Erik Anderson http://andersonfam.org ___ --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] Best USB Handset and Softphone Combination
On 10/19/07, Mike Clark [EMAIL PROTECTED] wrote: Do they play well with Vista? Hah - I have no idea. We installed Vista on one laptop here when Dell started shipping it. That lasted about 3 days and 10 support tickets from the user. Then we reverted back to XP. Haven't touched Vista since. -erik ___ --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] Best USB Handset and Softphone Combination
On 10/19/07, Steve Totaro [EMAIL PROTECTED] wrote: Any advice on softphones, handsets, or practical experience with this sort of deployment? It would be very nice if there was a central way of provisioning the phones. I've deployed several setups internally using X-Lite and these headsets: http://www.newegg.com/Product/Product.aspx?Item=N82E16826275009 Haven't heard of a single problem thus far. -erik ___ --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] What web GUI are people happy with?
On 10/17/07, shadowym [EMAIL PROTECTED] wrote: Ok so you use templates. I understand that. The problem is some people on here seem to be claiming they type it all in from scratch in like 3 minutes. Just call me out if you feel the need to. Please don't try and hide behind the some people on here type of comments. Call me out directly if you feel the need. I can take it :-) So...I don't feel the need to prove myself to you. I have a fairly good grasp on the conf file syntax, and with a well-thought out and well documented goal, it's not unreasonable for me to say that I can type out a config from scratch in 30 minutes. After working in vim for as long as I have, you learn to use the many shortcuts that it provides for text manipulation, copy buffers, moving blocks of code around, etc. I also use a syntax highlighting rule file for asterisk configs, so any typos I make are immediately evident. It's really remarkable how this discussion has turned into a pissing match. I could really care less if you have a hard time believing my statements. I'm not trying to push CLI on you or anyone. Yes - I recommend that people give it a try before going to a GUI, but I fully recognize that vanilla asterisk text configuration isn't for everyone. -Erik P.S. By the way - don't misquote me. I said nothing about laying down a config in 3 minutes. ___ --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] What web GUI are people happy with?
On 10/16/07, shadowym [EMAIL PROTECTED] wrote:
I don't do text editing so please indulge me. Why would someone want to do
that when a GUI makes life so much easier?
On a practical note, If someone was deploying 2 or 3 of these a week, most
of which have 5-10+ extensions doing all kinds of fancy things like call
queues, parking, forwarding, followme, voicemail to email etc. etc. how
practical is it to type all this in by hand making sure to get ever single
space, ., ,, {}, [] etc. exactly right which NEVER happens. So then
you have to spend more time debugging the conf files.
Even with a bunch of pre-made templates it seems like an awful lot of
unnecessary heavy lifting when a GUI can make it so much easier and
efficient.
This is *very* much a to each their own issue. You say that a web
GUI is more efficient - I say that vi is more effecient. You say that
using a text editor is more error-prone - I say that a web GUI is more
likely to mess things up in a difficult way to troubleshoot.
Use what works for you and don't worry about it.
:-)
___
--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] What web GUI are people happy with?
On 10/16/07, shadowym [EMAIL PROTECTED] wrote: So how long would it take you to vi a 20 extension office with custom dialplan involving a medium level of complexity? Including time to debug etc. Well - there's a large amount of subjectivity in your question, but perhaps I'll answer with not long. I don't know - 20 sip extensions, maybe 5 minutes. Probably another 30 for the dialplan and debugging. My point still stands - use what you're comfortable with. I spend the vast amount of my day working through an SSH console into various linux servers, so it would only make sense that for me (and many other CLI geeks), it doesn't make sense to use a GUI. I actually get a little put out when I have to switch over to my browser or another GUI tool to get things done. So - the CLI is what works for me. I'm not going to push that on you or anyone as the definitive best management tool for asterisk. -erik ___ --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] Affordable SIP Trunk for Home PBX ?
On 10/12/07, D4rk F1ber [EMAIL PROTECTED] wrote: Curious what others are using, and if anyone can make some recommendations? Not sure if this has been covered already on the list, and not sure if recommending companies are allowed, so maybe I need get replies off list? There are quite literally hundreds of VoIP service providers out there: Here's a list of some of them: http://voip-info.org/wiki/view/VOIP+Service+Providers+Residential Billing schemes usually fall into one of two categories. They'll either bill you a flat monthly fee for an unlimited plan or one with a large number of minutes. Or...they'll bill you on a per-minute, usage-only basis. The only provider I've had direct experience with is Teliax. I'm on an outgoing-only plan with them and it's been perfect so far. They bill something like $0.025/minute. If you want incoming calls as well, there's a per-month DID charge. If you are just wanting to receive incoming calls, check out IPKall - they'll give you a DID and a SIP trunk to your PBX for incoming calls. -erik ___ --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] Is there real benefits on a SMP machine for Asterisk?
On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote: I don't think there is a formula like cpu usage = loadavg / #cpus A loadavg of 3 says that there are 3 processes waiting to be executed. Anyway, I'll admit that a loadavg of 3 /might/ be ok. Here's a quote from this page: http://en.wikipedia.org/wiki/Load_%28computing%29#Unix-style_load_calculation For systems with multiple CPUs, the number needs to be divided by the number of processors in order to get a percentage. - Erik ___ --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] Is there real benefits on a SMP machine for Asterisk?
On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote: I wouldn't be too happy about a system with a loadavg of 3. The system he mentioned had 8 cores, though. So a load average of 3 is less than 50% usage. -erik ___ --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] Is there real benefits on a SMP machine for Asterisk?
On 10/11/07, Raúl Gómez C. [EMAIL PROTECTED] wrote: At this point I was wondering if Asterisk gets real benefits on systems with several cores (up to 8 in Dell PE2950) for a system that will handle up to 35 simultaneous SIP call with 10 FXO ports and 2 FXS for analog phones/fax (Sangoma A400D PCI card). For this load level (even with high-load transcoding), a multi-core machine certainly would not be needed. That said, it certainly wouldn't hurt anything to add on extra cores, especially if they're free ;-) -erik ___ --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] anyone using SIP trunks from Time Warner Telecom?
On 10/8/07, Forrest Beck [EMAIL PROTECTED] wrote: I was told that Asterisk was supported when we looked at the service. Hey Forrest - thanks for the information. Might you be able to send along the contact information for the TW rep who told you that asterisk was supported? I've been in conversation with our Sales rep today, and he's quite adamant that they currently only support Cisco Call Manager and CCM Express. I believe they're using CCM to provice the SIP trunks - if this is indeed the case, I don't see interoperability with asterisk as a problem. Thanks -Erik ___ --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] When does the future arrive?
On 10/9/07, Hans Witvliet [EMAIL PROTECTED] wrote: Hi all, Probably this is the wrong place to ask, but is there an estimated time of arrival of the future? i.e. TFOT--next generation dealing with * -1.4 I attended a workshop some time ago, and the book was part of the package The Future, my friend, is here. http://downloads.oreilly.com/books/9780596510480.pdf Enjoy! ___ --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
