Re: [asterisk-users] Asterisk to PBX
Hi Paul, Thanks a lot for the response. I'm a novice so pardon me for the stupid questions. I thought that maybe the PSTN lines don't allow more than 1 simultaneous calls on a line, but on GSM it might be possible. I basically want to know how Asterisk can dial out calls from the lines connected to it. Ideally I want to make out as many calls from the lines connected to my Asterisk box. I have a few related questions, again pardon me if I'm a novice. How did PBX in days when didn't have Asterisk worked? If a company wanted to give desk phones to all the employees then it would have a switchboard which would route the calls. Now in this case I'm guessing that the company had only one PSTN line, but somehow the switchboard let everyone make calls and receive calls at the same time. So is it possible to have the switchboard and have it connect to Asterisk who can there by use these lines? Paul, could you also describe a bit about hook flash? Thanks. Best Regards, Hitesh Are you talking about using hook flash to change between active calls? Or the more interesting facilities available on the 3G (and beyond) networks? regards, PaulH ___ -- 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 to PBX
Some thoughts inline: logan wrote: Hi Paul, Thanks a lot for the response. I'm a novice so pardon me for the stupid questions. I thought that maybe the PSTN lines don't allow more than 1 simultaneous calls on a line, but on GSM it might be possible. I basically want to know how Asterisk can dial out calls from the lines connected to it. Ideally I want to make out as many calls from the lines connected to my Asterisk box. I don't really understand this question - Asterisk can make calls over phone lines. And it does it well. I have a few related questions, again pardon me if I'm a novice. How did PBX in days when didn't have Asterisk worked? We used to have an NEC. If a company wanted to give desk phones to all the employees then it would have a switchboard which would route the calls. Maybe. Or maybe not. Now in this case I'm guessing that the company had only one PSTN line, Why would you guess this? We had 16 phone lines in the first business I worked in. but somehow the switchboard let everyone make calls and receive calls at the same time. Because the calls never used the phone lines. So is it possible to have the switchboard and have it connect to Asterisk who can there by use these lines? I suppose.and I think attaching a vintage jack-style switchboard would be a very fun project. Paul, could you also describe a bit about hook flash? It's a way of putting a call on hold and taking one off hold - much like your descreption of how calls work on a mobile phone. You should had a read of: http://www.asteriskdocs.org/ later, PaulH ___ -- 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] dialplan number matching
hello,
why not use execif or gotoif?
this would look like this:
exten = _X.,n,ExecIf($[${EXTEN:${LEN(${EXTEN})-1}}=3]|do would ever
you want to do
best regards
steve
Vieri schrieb:
Hi,
How can I match an extension ending with 3 (just an example but applicable
to any other digit, including * or #)?
exten = _ZX.3,n,...
exten = _ZX.#,n,...
(the above does not work)
Can regular expressions be used in the standard dialplan (end with: $)?
Thanks,
Vieri
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Re: [asterisk-users] Asterisk to PBX
Thanks Paul. Your help is much appreciated here. I don't really understand this question - Asterisk can make calls over phone lines. And it does it well. Surely, Asterisk does that well, but Asterisk needs to have multiple phone lines for that. I thought that a traditional switchboard made that happen without multiple phone lines. BTW, in Asterisk terminology a phone line means different PSTN connections to the operator, right? Why would you guess this? We had 16 phone lines in the first business I worked in. Yeah, that's fine, but even 16 phone lines don't mean you can have 16 desk phones only or 16 simultaneous calls? Thanks I will take a look at asteriskdocs. Best Regards, Hitesh ___ -- 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 to PBX
logan wrote: Thanks Paul. Your help is much appreciated here. No problem - been working on telephone systems for about 12 years now - which doesn't even make me an old hand... Surely, Asterisk does that well, but Asterisk needs to have multiple phone lines for that. I thought that a traditional switchboard made that happen without multiple phone lines. Not really - but there's something you are missing in your understanding and it will come to you soon enoughjust keep reading and asking questions. Of course, Asterisk can place many calls down a network connection/adsl/E1/DS3/etc. BTW, in Asterisk terminology a phone line means different PSTN connections to the operator, right? Once again, I don't really understand this question. Why would you guess this? We had 16 phone lines in the first business I worked in. Yeah, that's fine, but even 16 phone lines don't mean you can have 16 desk phones only or 16 simultaneous calls? We had about 40 phones. We could make 16 inbound/outbound calls, and as many internal calls as we wanted to... Thanks I will take a look at asteriskdocs. Reading is a great way to learn things. PaulH ___ -- 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 freepbx difference or solutions..
Hello, for a long time i am using asterisk 1.6 with astgui. but for production system i intend to use asterisk 1.4 which i think might be more robust. And for a more developed service options i preferd to install with freepbx. But still there are big plusses and minusses for both system. My complain about astgui+1.6 was.. For example there were no backup trunk config running on that version.Even they have it on gui screen when i check the extensions.conf it seems only supporting a single trunk.Second. there were lots of options whcih was absent on 1.6..like for voicemail it has *97 for selfvoicemail but no *98 for entering both phone+calledid, or call forwarding which most of my users uses. I need to write them from the begining which will take most of my time. And for freepbx, i had the problem like i couldnt find how to group my users for outgoing calls.For some users i want to give only local call permissions for some including international calls etc. And there are no group selection or creation for such needs. So i wonder if theres any all in one solution for that?? or is there a way to solve my needs??? ___ -- 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 freepbx difference or solutions..
can you try with elastix it's same like freepbx but have some advance function like group exertions and all those On Mon, Jul 20, 2009 at 3:48 PM, Oguzhan Kayhan oguzh...@bilkent.edu.trwrote: Hello, for a long time i am using asterisk 1.6 with astgui. but for production system i intend to use asterisk 1.4 which i think might be more robust. And for a more developed service options i preferd to install with freepbx. But still there are big plusses and minusses for both system. My complain about astgui+1.6 was.. For example there were no backup trunk config running on that version.Even they have it on gui screen when i check the extensions.conf it seems only supporting a single trunk.Second. there were lots of options whcih was absent on 1.6..like for voicemail it has *97 for selfvoicemail but no *98 for entering both phone+calledid, or call forwarding which most of my users uses. I need to write them from the begining which will take most of my time. And for freepbx, i had the problem like i couldnt find how to group my users for outgoing calls.For some users i want to give only local call permissions for some including international calls etc. And there are no group selection or creation for such needs. So i wonder if theres any all in one solution for that?? or is there a way to solve my needs??? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queues load balancing
Hello I have 2 queues (queue_1 and queue_2 ) in my Asterisk, and I want to send 2/3 of the calls to queue_1 and 1/3 of the calls to queue_2 How can I do that load balancing in extensions.conf? I have something like this: exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer ; 2 in 3 calls go to queue_1 exten = 123,x,Queue(queue_1) ; 1 in 3 calls go to queue_2 exten = 123,x,Queue(queue_2) But how can I configure this call distribution? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues load balancing
Take a look at: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random You should be able to do what you want with this, it obviously won't take in to account the actual amount of people still in the queue (for example if someone hangs up while on hold). I'm sure there'd be a way of integrating this in to it using some different functions, but for a quick fix random will do just fine. Cheers 2009/7/20 Joao Gomes Pereira gomespere...@startel.pt Hello I have 2 queues (queue_1 and queue_2 ) in my Asterisk, and I want to send 2/3 of the calls to queue_1 and 1/3 of the calls to queue_2 How can I do that load balancing in extensions.conf? I have something like this: exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer ; 2 in 3 calls go to queue_1 exten = 123,x,Queue(queue_1) ; 1 in 3 calls go to queue_2 exten = 123,x,Queue(queue_2) But how can I configure this call distribution? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues load balancing
Thanks for the idea. I will try it this way: exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer exten = 123,4,Random(33:123,10) exten = 123,5,Queue(queue_1) exten = 123,6,Hangup exten = 123,10,Queue(queue_2) exten = 123,11,Hangup Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt Geraint Lee wrote: Take a look at: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random You should be able to do what you want with this, it obviously won't take in to account the actual amount of people still in the queue (for example if someone hangs up while on hold). I'm sure there'd be a way of integrating this in to it using some different functions, but for a quick fix random will do just fine. Cheers 2009/7/20 Joao Gomes Pereira gomespere...@startel.pt mailto:gomespere...@startel.pt Hello I have 2 queues (queue_1 and queue_2 ) in my Asterisk, and I want to send 2/3 of the calls to queue_1 and 1/3 of the calls to queue_2 How can I do that load balancing in extensions.conf? I have something like this: exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer ; 2 in 3 calls go to queue_1 exten = 123,x,Queue(queue_1) ; 1 in 3 calls go to queue_2 exten = 123,x,Queue(queue_2) But how can I configure this call distribution? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt http://www.startel.pt +351 304500650 sip: gomespere...@startel.pt mailto:gomespere...@startel.pt ___ -- 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 -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues load balancing
On 21/7/09 12:08 AM, Joao Gomes Pereira wrote:
Thanks for the idea.
I will try it this way:
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
exten = 123,4,Random(33:123,10)
exten = 123,5,Queue(queue_1)
exten = 123,6,Hangup
exten = 123,10,Queue(queue_2)
exten = 123,11,Hangup
Bear in mind that the Random application has been deprecated in favour
of the RANDOM function:
asterisk -rx 'show application random'
-= Info about application 'Random' =-
[Synopsis]
Conditionally branches, based upon a probability
[Description]
Random([probability]:[[context|]extension|]priority)
probability := INTEGER in the range 1 to 100
DEPRECATED: Use GotoIf($[${RAND(1,100)} number]?label)
--
Cheers,
Matt Riddell
Director
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Re: [asterisk-users] queues load balancing
Here is a brute force solution:
[global]
CALLCOUNT=0
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
exten = 123,4,Set(CALLCOUNT)=${CALLCOUNT}+1)
exten = 123,5,Gotoif($[$(CALLCOUNT} = 3]?queue2)
exten = 123,6,Queue(queue_1)
exten = 123,7,Hangup
exten = 123,8(queue2),Set(CALLCOUNT=0)
exten = 123,9,Queue(queue_2)
exten = 123,10,Hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Monday, July 20, 2009 7:37 AM
To: gomespere...@startel.pt; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] queues load balancing
On 21/7/09 12:08 AM, Joao Gomes Pereira wrote:
Thanks for the idea.
I will try it this way:
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
exten = 123,4,Random(33:123,10)
exten = 123,5,Queue(queue_1)
exten = 123,6,Hangup
exten = 123,10,Queue(queue_2)
exten = 123,11,Hangup
Bear in mind that the Random application has been deprecated in favour
of the RANDOM function:
asterisk -rx 'show application random'
-= Info about application 'Random' =-
[Synopsis]
Conditionally branches, based upon a probability
[Description]
Random([probability]:[[context|]extension|]priority)
probability := INTEGER in the range 1 to 100
DEPRECATED: Use GotoIf($[${RAND(1,100)} number]?label)
--
Cheers,
Matt Riddell
Director
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Re: [asterisk-users] queues load balancing
Danny Nicholas schrieb:
Here is a brute force solution:
[global]
CALLCOUNT=0
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
exten = 123,4,Set(CALLCOUNT)=${CALLCOUNT}+1)
...,Set(CALLCOUNT=$[${CALLCOUNT} + 1])
or
...,Set(CALLCOUNT=${MATH(${CALLCOUNT}+1,int)})
exten = 123,5,Gotoif($[$(CALLCOUNT} = 3]?queue2)
exten = 123,6,Queue(queue_1)
exten = 123,7,Hangup
exten = 123,8(queue2),Set(CALLCOUNT=0)
exten = 123,9,Queue(queue_2)
exten = 123,10,Hangup
Philipp Kempgen
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Re: [asterisk-users] Delete voicemail after couple of days
Tim, this is a partial solution. The find as written would remove
greetings, unavailable messages, etc. You would need to add a grep to get
only msg files.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Friday, July 17, 2009 5:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Delete voicemail after couple of days
- Steve Edwards asterisk@sedwards.com wrote:
On Fri, 17 Jul 2009, Miguel Molina wrote:
I think the OP caught the humor -- note the smiley. I'm sorry it
didn't translate to your language.
Oops, well I'm not a native english speaker so it's really hard to
catch
some humor of a word that I don't know or I get as misspelled.
Thanks
for the definition, now I can laugh with you guys.
Sorry for all the fuzz around this.
PD: Es como si yo te contara un chiste en español!
Si, pero el Ingles es mejor que mi espanol!
(Google translate is my friend.)
--
All the politics, list etiquette, and general bitching aside, here is how I
would do what the OP wants.
Write up a small shell script that uses 'find /var/spool/asterisk/voicemail/
-mtime +2' for a list of files older than two days assuming you want ALL
files deleted older than two days. You could always grep that output if you
only wanted to delete voicemail that is not still in the INBOX or elsewhere.
Anyways, then use -exec to rm the files. If the goal was to remove all
files, it might look something like this:
#!/bin/bash
find /var/spool/asterisk/voicemail/ -mtime +2 -exec rm {}\;
Run that from cron once a day/hour/whatever and you're set.
rant
It still amazes me how often posters are unable to get a simple answer to a
question and instead are inundated with 'you top posted', 'you didn't ask
the question right', 'your spelling was wrong', etc... I mean, is this list
just a really big bridge with a bunch of trolls(no pun intended) waiting to
pounce on people just wanting to get to the other side where Asterisk
Enlightenment awaits?
And of course because I've diverted from the norm and possibly hurt
someone's ego, I expect a full backlash or smarmy remarks etc. Thank you in
advance.
/rant
--Tim
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Re: [asterisk-users] Count Available Queue members
Tilghman Lesher wrote:
My understanding of QUEUE_MEMBER_COUNT just give a total of agents in the
queue.
Synopsis Count number of members answering a queue
It may or may not be the answer to the OP's question, depending upon what he
meant by available. Without clarification, it's impossible to know.
And along with this, even if it just returns the total number of queue members
available, that could be useful if you're calling via something like a Local
channel, where you could use the GROUP() and GROUP_COUNT() functions.
Place a GROUP() in the Local channel prior to calling the member, and then you
can do:
exten = start,n,Set(MEMBERS_TO_TAKE_CALLS=$[${GROUP_COUNT(agents)} -
${QUEUE_MEMBER_COUNT(queue)}])
Something like that anyways.
Leif Madsen.
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
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[asterisk-users] callforward with asterisk-gui.problem with stdexten
Hello, i am trying to enable call forwarding on asterisk 1.6 with
asterisk-gui
If i set my stdexten as follows (with the lines i marked) everything seems
like working.
But if i make any change on asterisk-gui and apply it.. it recreates the
macro-stdexten and deletes my configuration regarding to it.
So where should i add my call-forward configuration???
Where am i making a mistake??
[macro-stdexten]
exten = s,1,Set(__DYNAMIC_FEATURES=${FEATURES})
exten = s,2,GotoIf($[${FOLLOWME_${ARG1}} = 1]?5:3)
adding starts from here
exten = s,3,Set(temp=${DB(CFIM/${ARG1})})
exten = s,4,Dial(Local/${te...@default/n) ; Unconditional forward
exten = s,5,Set(DB(lastcaller/${ARG1})=${CALLERID(num)}) ; Note the last
caller
--ends here
exten = s,6,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,7,Goto(s-${DIALSTATUS},1)
exten = s,8,Macro(stdexten-followme,${ARG1},${ARG2})
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
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[asterisk-users] callforward with asterisk-gui.problem with stdexten
Hello, i am trying to enable call forwarding on asterisk 1.6 with
asterisk-gui
If i set my stdexten as follows (with the lines i marked) everything seems
like working.
But if i make any change on asterisk-gui and apply it.. it recreates the
macro-stdexten and deletes my configuration regarding to it.
So where should i add my call-forward configuration???
Where am i making a mistake??
[macro-stdexten]
exten = s,1,Set(__DYNAMIC_FEATURES=${FEATURES})
exten = s,2,GotoIf($[${FOLLOWME_${ARG1}} = 1]?5:3)
adding starts from here
exten = s,3,Set(temp=${DB(CFIM/${ARG1})})
exten = s,4,Dial(Local/${te...@default/n) ; Unconditional forward
exten = s,5,Set(DB(lastcaller/${ARG1})=${CALLERID(num)}) ; Note the last
caller
--ends here
exten = s,6,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,7,Goto(s-${DIALSTATUS},1)
exten = s,8,Macro(stdexten-followme,${ARG1},${ARG2})
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
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[asterisk-users] What am I doing wrong?
Hi Gang, I've got the latest SVN branch of 1.4 downloaded onto SUSE 11.0. Everything is happy EXCEPT, I can't get fax to be recognized by make menuselect. I tried copying app_rxfax.c and app_txfax.c to the apps directory and starting again from ./configure, but no joy. Any suggestions? Danny Nicholas ___ -- 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] Digium TDM400P in Soekris net5501-70?
Hello - I've been running Asterisk (quite happily!) for several years now using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM). I'm also running another old PC running m0n0wall as a firewall. Between these two boxes, that run 24x7, I'm drawing a lot more power than needed and hoping to make a dent in my monthly electric bill by consolidating the two into a single box with efficient power supply, low power processor, and no spinning HD platters. Main question is whether anyone knows if the Digium TDM400P should be compatible with the 3.3V PCI slot in the Soekris Net5501-70 box? Soekris' description for the net5501-70 says, in part, it has support for one or two low-power standard PCI board I see on my Digium card that it requires a molex connector supplying voltage. The Net5501 has a small 4-pin molex header on the board, I wonder if a small to regular sized molex power cable would do the job to supply this card. If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? -- Brian McEntire Photographer Owner B Scott Photography (240) 358-6655 studio www.bscottphoto.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] perhaps libpri issue (thought it was a dahdi issue )
On Sat, Jul 18, 2009 at 12:05:51PM -0400, Jerry Geis wrote: / I am current running on a production system // zaptel 1.4.12.1 // libxpri 1.4.1 // asterisk 1.4.25 // // The above configuration works. // // I tried to update to dahdi 2.2.0, libpri 1.4.7 and asterisk 1.4.25 // This did not work. calls came in but not out. I dropped back to the // initial configuration. / libpri 1.4.7? Why not 1.4.10? / // Today, I dried only updating libpri on the original configuration. // I removed /usr/lib/libpri* installed libpri 1.4.7, recompiled asterisk // 1.4.25 // and incoming calls worked and outgoing calls gives me the same error I got // at my last upgrade attempt. // // -- Called g1/317XXX // -- Channel 0/18, span 1 got hangup, cause 99 // -- Hungup 'Zap/18-1' // == Everyone is busy/congested at this time (1:0/0/1) // // // I again removed /usr/lib/libpri* , reinstalled libpri 1.4.1, recompiled // asterisk // and everything worked again. // // What might me the issue here I originally thought it was a dahdi // issue but // it appears to be a libpri issue. / Over the weekend I tried 1.4.10.1 and same thing. Channel 0/18, span 1 got hangup, cause 99 Drop back to 1.4.7 and it worked again. What next? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delete voicemail after couple of days
Un-top-posting...
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Friday, July 17, 2009 5:33 PM
Write up a small shell script that uses 'find
/var/spool/asterisk/voicemail/ -mtime +2' for a list of files older than
two days assuming you want ALL files deleted older than two days. You
could always grep that output if you only wanted to delete voicemail
that is not still in the INBOX or elsewhere. Anyways, then use -exec to
rm the files. If the goal was to remove all files, it might look
something like this:
#!/bin/bash
find /var/spool/asterisk/voicemail/ -mtime +2 -exec rm {}\;
Run that from cron once a day/hour/whatever and you're set.
On Mon, 20 Jul 2009, Danny Nicholas wrote:
Tim, this is a partial solution. The find as written would remove
greetings, unavailable messages, etc. You would need to add a grep to get
only msg files.
Skip the grep...
sudo find /var/spool/asterisk/ -mtime +2 -name msg* -exec rm {} \;
(Note the space after the closing brace.)
But Carlos had the best answer.
On Fri, 17 Jul 2009, Carlos Chavez wrote:
I did not catch all the messages on this thread but why not use the
messages-expire.pl script included in Asterisk for this simple task?
It will delete and renumber all messages and you can program how many
days before a message is deleted.
Why re-invent the wheel. This script is way more flexible than a 1-liner.
(Note that Asterisk will renumber the messages for you automagically in
case you decide to cobble up your own script.)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Digium TDM400P in Soekris net5501-70?
Look into AstLinux as one possible solution for both Asterisk and a firewall on the 5501, with no hard drive. John Novack Brian McEntire wrote: Hello - I've been running Asterisk (quite happily!) for several years now using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM). I'm also running another old PC running m0n0wall as a firewall. Between these two boxes, that run 24x7, I'm drawing a lot more power than needed and hoping to make a dent in my monthly electric bill by consolidating the two into a single box with efficient power supply, low power processor, and no spinning HD platters. Main question is whether anyone knows if the Digium TDM400P should be compatible with the 3.3V PCI slot in the Soekris Net5501-70 box? Soekris' description for the net5501-70 says, in part, it has support for one or two low-power standard PCI board I see on my Digium card that it requires a molex connector supplying voltage. The Net5501 has a small 4-pin molex header on the board, I wonder if a small to regular sized molex power cable would do the job to supply this card. If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? -- Dog is my co-pilot ___ -- 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-dev] MeetMe feature request: bypass pincode
Emrah wrote:
This is an asterisk-users question, and would have been more appropriate to
have
asked there.
Instead of setting up your conferences in meetme.conf, you could set them up
dynamically in the dialplan, and then you can control whether the user is
prompted for a pin or not when joining the conference, based on whatever
logic
you want.
Something like the following could work (untested):
exten = start,1,NoOp()
exten = start,n,Set(PIN=1234)
exten = start,n,Set(USER_MUST_ENTER_PIN=${IF($[${CUT(CHANNEL,-,1)} !=
SIP/myself]1:0)})
exten = start,n,MeetMe(7070,d${IF($[${USER_MUST_ENTER_PIN} =
1]?,${PIN})})
exten = start,n,Hangup()
Thanks for your answer.
However what I would like to achieve is a little bit more complicated.
It involves the manager to originate a call and put a participant in the
conference. I made a workaround with an agi script but I would
definitely prefer a sexier method to do it (like I said with an option
to the application).
Thanks for your hint anyway.
I'm not sure how that restricts you from using a dialplan trick, since the
originate you're doing from your AMI interface is still going to execute
dialplan (or could).
Additionally, I don't remember any of that information in your original email.
You should give all relevant information so people can give you a good answer.
I'm reasonably confident there are enough methods to get what you want though
without a new configuration option.
However, without more information about how what you're really trying to do, I
can only speculate.
Leif Madsen.
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
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[asterisk-users] Event Log
I am using an IBM Server, after while in the MBR it said that Event logs are full, so after clearing it, the asterisk can't run. i think it deleted a file, so which file i have to create again. and what's its chmod. Thanks _ With Windows Live, you can organize, edit, and share your photos. http://www.microsoft.com/middleeast/windows/windowslive/products/photo-gallery-edit.aspx___ -- 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] Event Log
Probably /var/log/asterisk/messages 0644. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T Sent: Monday, July 20, 2009 1:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Event Log I am using an IBM Server, after while in the MBR it said that Event logs are full, so after clearing it, the asterisk can't run. i think it deleted a file, so which file i have to create again. and what's its chmod. Thanks _ With Windows Live, you can organize, edit, and share your photos. http://www.microsoft.com/middleeast/windows/windowslive/products/photo-gall ery-edit.aspx ___ -- 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] Digium TDM400P in Soekris net5501-70?
At 10:09 AM 7/20/2009, you wrote: If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? I just built a box for my Asterisk system using an Intel Motherboard with an Atom 330, 5400 RPM HD, TDM 400 with 4 red cards and the cheap PS hat came with the case. Draws 43 watts according to my Kill-A-Watt and except for the TDM 400 which I already had it cost under 250 as parts from NewEgg. The only annoyance might be it has only one ethernet port with the only easy place to put another being a USB port. 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
Re: [asterisk-users] What am I doing wrong?
Hi Nicholas! Perhaps, there are other ways as I describe here, but I use this way successfully about 4 years - install latest spandsp version - went to root directory of your svn asterisk - type make distclean (because there are preconfigured things in downloaded version) - change to following file of your asterisk directory /build_tools/menuselect-deps in the last line of this file, insert this *SPANDSP=1* - after them, change back to root directory of asterisk and open the file makeopts last line, insert *SPANDSP_LIB=-lspandsp - *after them, you can type make menuselct and expected entries could be shown in menue - make - make install One hint: since 1 year, I use SIP (t38) for incoming fax and using ISDN for outgoing fax, because there are many old faxes, which have no G38 Support and can not recieve fax over t38 regards, Kare Danny Nicholas schrieb: Hi Gang, I've got the latest SVN branch of 1.4 downloaded onto SUSE 11.0. Everything is happy EXCEPT, I can't get fax to be recognized by make menuselect. I tried copying app_rxfax.c and app_txfax.c to the apps directory and starting again from ./configure, but no joy. Any suggestions? Danny Nicholas ___ -- 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 am I doing wrong?
On Mon, 20 Jul 2009, Danny Nicholas wrote: Any suggestions? Sorry. I can't resist :) Asking a question with a useless Subject. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Event Log
No, this file is still existed, i think it's another file. Thanks From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 20 Jul 2009 13:20:23 -0500 Subject: Re: [asterisk-users] Event Log Probably /var/log/asterisk/messages 0644. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T Sent: Monday, July 20, 2009 1:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Event Log I am using an IBM Server, after while in the MBR it said that Event logs are full, so after clearing it, the asterisk can't run. i think it deleted a file, so which file i have to create again. and what's its chmod. Thanks With Windows Live, you can organize, edit, and share your photos. _ More than messages–check out the rest of the Windows Live™. http://www.microsoft.com/windows/windowslive/___ -- 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] Event Log
Asterisk -vc should tell you what it wants to be able to start. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T Sent: Monday, July 20, 2009 1:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Event Log No, this file is still existed, i think it's another file. Thanks _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 20 Jul 2009 13:20:23 -0500 Subject: Re: [asterisk-users] Event Log Probably /var/log/asterisk/messages 0644. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T Sent: Monday, July 20, 2009 1:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Event Log I am using an IBM Server, after while in the MBR it said that Event logs are full, so after clearing it, the asterisk can't run. i think it deleted a file, so which file i have to create again. and what's its chmod. Thanks _ With Windows Live, you can organize, edit, and share http://www.microsoft.com/middleeast/windows/windowslive/products/photo-gall ery-edit.aspx your photos. _ check out the rest of the Windows LiveT. More than mail-Windows LiveT goes way beyond your inbox. More than http://www.microsoft.com/windows/windowslive/ messages ___ -- 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] Digium TDM400P in Soekris net5501-70?
Thanks Ira - I may yet still go with a standard Intel solution, but I think there could be major power savings to be had going with a smaller box like a Soekris if it can work. A good rule of thumb for 24x7 devices is $1 per watt per year, so 45 watts, while good, will still be $45 per year. I don't know what a Soekris would draw, but without a power supply fan, and using a CF card rather than a conventional HD, I'm hoping the power use would be much reduced. I will look into AstLinux. I'm actually hoping to run a VM (like VMWare) on this solution and run the firewall (m0n0wall) inside the VM. M0n0wall is a tiny distro that runs from a CD (or can run from a CF card), so I think it would still run fine inside a VM. On Mon, Jul 20, 2009 at 2:20 PM, Irai...@extrasensory.com wrote: At 10:09 AM 7/20/2009, you wrote: If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? I just built a box for my Asterisk system using an Intel Motherboard with an Atom 330, 5400 RPM HD, TDM 400 with 4 red cards and the cheap PS hat came with the case. Draws 43 watts according to my Kill-A-Watt and except for the TDM 400 which I already had it cost under 250 as parts from NewEgg. The only annoyance might be it has only one ethernet port with the only easy place to put another being a USB port. 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 -- Brian McEntire Photographer Owner B Scott Photography (240) 358-6655 studio www.bscottphoto.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] Digium TDM400P in Soekris net5501-70?
At 12:47 PM 7/20/2009, you wrote: I may yet still go with a standard Intel solution, but I think there could be major power savings to be had going with a smaller box like a Soekris if it can work. A good rule of thumb for 24x7 devices is $1 per watt per year, so 45 watts, while good, will still be $45 per year. I don't know what a Soekris would draw, but without a power supply fan, and using a CF card rather than a conventional HD, I'm hoping the power use would be much reduced. There is a version running on a Blackfin based board that claims to only use 4 watts, http://blog.astfin.org/ I thought about it, but didn't want to be stuck to that hardware and OS version and it's not cheap. 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
Re: [asterisk-users] Digium TDM400P in Soekris net5501-70?
On Mon, Jul 20, 2009 at 01:09:30PM -0400, Brian McEntire wrote: Hello - I've been running Asterisk (quite happily!) for several years now using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM). I'm also running another old PC running m0n0wall as a firewall. Between these two boxes, that run 24x7, I'm drawing a lot more power than needed and hoping to make a dent in my monthly electric bill by consolidating the two into a single box with efficient power supply, low power processor, and no spinning HD platters. Main question is whether anyone knows if the Digium TDM400P should be compatible with the 3.3V PCI slot in the Soekris Net5501-70 box? Hi I have a the same setup you mention here, except I have a tdm410 card. I have a cf boot and a SSD card as well. Running Debian for firewall and asterisk server. Works well I have 3 vpn tunnels and a 6to4 tunnel ending on this machine, 2 fxs + 1 fxo. from my collectd graphs it looks like it really only spike when I am taking readins :) One catch the case that comes from soekris is too tight to put the molex on, I had to solder it to the connectors underneath. all fine though I am not sure about running a vm on this box though - I have some thing similiar at another site, but a bigger box. Alex Soekris' description for the net5501-70 says, in part, it has support for one or two low-power standard PCI board I see on my Digium card that it requires a molex connector supplying voltage. The Net5501 has a small 4-pin molex header on the board, I wonder if a small to regular sized molex power cable would do the job to supply this card. If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? -- Expense Accounts, n.: Corporate food stamps. signature.asc Description: Digital signature ___ -- 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] Vote on whether SipPhone should support ISN routing.
Should SipPhone support ISN routing for their 747 ITAD? Cast a vote: http://forums.gizmo5.com/viewtopic.php?t=10197 Meanwhile if you're interested, you can use the Nerd Vittles 'bandit' ITAD #1089 to call a SipPhone/Gizmo5 subscriber via ISN, which I think is clever (Karl tips his hat to Ward Mundy) and it's also really, really funny. ___ -- 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] Digium TDM400P in Soekris net5501-70?
Thanks for the reply Alex. I'm not too scared of the soldering iron (I own one, but my work with it isn't pretty ;-) But can you confirm, are you just using the small power header on the board to supply power to the pci card? I was wondering if I was going to have to snake an another wall wort into the box to power the card, would be good if I don't have to do that! Not 100% sure I could run a VM on it, but the new net5501 board comes with 512MB ram and I think a 500-ish MHz processor, way more than what I'm currently using to run m0n0wall, so even if the VM takes a bite out of it, it should be fine, hardest part might be configuring the VM to boot monowall from CF. Can you partition a CF card? (ie, one partition for the monowall firmware and the other for the stripped down linux install to run Asterisk?) On Mon, Jul 20, 2009 at 4:44 PM, Alex Samada...@samad.com.au wrote: On Mon, Jul 20, 2009 at 01:09:30PM -0400, Brian McEntire wrote: Hello - I've been running Asterisk (quite happily!) for several years now using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM). I'm also running another old PC running m0n0wall as a firewall. Between these two boxes, that run 24x7, I'm drawing a lot more power than needed and hoping to make a dent in my monthly electric bill by consolidating the two into a single box with efficient power supply, low power processor, and no spinning HD platters. Main question is whether anyone knows if the Digium TDM400P should be compatible with the 3.3V PCI slot in the Soekris Net5501-70 box? Hi I have a the same setup you mention here, except I have a tdm410 card. I have a cf boot and a SSD card as well. Running Debian for firewall and asterisk server. Works well I have 3 vpn tunnels and a 6to4 tunnel ending on this machine, 2 fxs + 1 fxo. from my collectd graphs it looks like it really only spike when I am taking readins :) One catch the case that comes from soekris is too tight to put the molex on, I had to solder it to the connectors underneath. all fine though I am not sure about running a vm on this box though - I have some thing similiar at another site, but a bigger box. Alex Soekris' description for the net5501-70 says, in part, it has support for one or two low-power standard PCI board I see on my Digium card that it requires a molex connector supplying voltage. The Net5501 has a small 4-pin molex header on the board, I wonder if a small to regular sized molex power cable would do the job to supply this card. If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? -- Expense Accounts, n.: Corporate food stamps. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkpk1yMACgkQkZz88chpJ2NvmgCg3+4zJhQBcnQzxMPeQ1N+KXn1 XBMAnjtAOUjpC/++2acwVuHcYOpPQG21 =eRW6 -END PGP SIGNATURE- ___ -- 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 -- Brian McEntire Photographer Owner B Scott Photography (240) 358-6655 studio www.bscottphoto.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] How to restrict registrations by useragent?
Hi, I have an extension which I want to use only for x-lite, and don't want anybody to register IP phones on it. I can see that 'sip show peer 3547' shows softphone's id. Is there a way to restrict registrations on this extension by useragent id? I googled but so far couldn't find any way to do it. -- Zeeshan A Zakaria ___ -- 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] 2 Problems with 1.6.2
Have you solved this issue? When I restart the machines I can't make an outgoing DAHDI call until I get an incoming call on that same line. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira Sent: viernes, 17 de julio de 2009 07:47 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 2 Problems with 1.6.2 At 01:09 PM 7/17/2009, you wrote: Sorry, that's the most frequent problem that people have with MWI in 1.6, so it was worth mentioning. I would suggest that you file a bug report on https://issues.asterisk.org. It would be helpful if you would include SIP debug output for both a machine that is working, as well as a machine that is not working. So I'd be more than happy to file a bug report and include all the SIP debug anyone might need but it's been so many years since I did it that I've no idea how anymore. So I grabbed a cordless handset, sat down at the console, typed sip set debug ip 192.xxx.xxx.xx and called that voicemail box to leave a message. The instant I hung up a notify message was sent to my phone, but the red light did not come on. If you remind me the how, I'll grab that message and post it here. Thanks so much for the help. 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 Checked by AVG - www.avg.com Version: 8.5.375 / Virus Database: 270.13.18/2243 - Release Date: 07/17/09 18:00:00 ___ -- 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] Error: Invalid SIP message - rejected , no call id
On about 25% of inbound calls to a ring group, picking up any one extension as it rings results in dead air. Some details regarding my VoIP network to make the following logs more readable: 192.168.7.130 resolves to the trixbox host. 192.168.7.135 resolves to endpoint 812. 192.168.7.137 resolves to endpoint 811. 192.168.7.138 resolves to endpoint 810. 192.168.7.139 resolves to endpoint 813. 192.168.7.140 resolves to endpoint 817. 24.136.116.102 is the address of the pbx. 66.23.129.253 is the address of my VoIP provider's peering host. Very verbose asterisk logging of such a failed inbound call returns snippets such as the following two examples: - [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 0: ACK sip:18502296...@phonehome.admiralenvelope.com SIP/2.0 (57) [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.7.140:5060;branch=z9hG4bK8ef20feeb668f72f (66) [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 2: From: Shipping sip:8...@phonehome.admiralenvelope.com ;tag=4aeafc6270620b72 (77) [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 3: To: sip:18502296...@phonehome.admiralenvelope.com ;tag=as7823cf0c (66) [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 4: Contact: sip:8...@192.168.7.140:5060;transport=udp (51) [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 5: Supported: path (15) [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 6: Proxy- Authorization: Digest username=817, realm=asterisk, algorithm=MD5, uri=sip:18502296...@phonehome.admiralenvelope.com, nonce=12f646df, response=e77e7b202fc6a0bc5930460db8243292 (191) [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 7: Call-ID: 9dd235bb45bb9...@192.168.7.140 (39) [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 8: CSeq: 61074 ACK (15) [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 9: User-Agent: Grandstream GXP2000 1.1.6.16 (40) [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 10: Max-Forwards: 70 (16) [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 11: Allow: INVITE ,ACK ,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 12: Content-Length: 0 (17) [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 13: (0) [Jul 16 16:17:38] VERBOSE[3214] logger.c: --- (13 headers 0 lines) --- [Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Invalid SIP message - rejected , no callid, len 763 [Jul 16 16:17:42] VERBOSE[3214] logger.c: --- SIP read from 192.168.7.135:5060 --- - [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.7.130:5060;branch=z9hG4bK4ddb9288;rport (64) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 2: From: Sales: (14) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 3: To: sip:8...@192.168.7.137:5060;transport=udp ;tag=5d9dbfef4e870100 (67) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 4: Call-ID: 238f32201de94e3336a339d650b71...@192.168.7.130 (55) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 6: User-Agent: Grandstream GXP2000 1.1.6.16 (40) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 7: Contact: sip:8...@192.168.7.137:5060;transport=udp (51) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 8: Allow: INVITE ,ACK ,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 9: Content-Type: application/sdp (29) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 10: Supported: replaces, timer (26) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 11: Content-Length: 212 (19) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 12: (0) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: v=0 (3) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: o=811 8002 8000 IN IP4 192.168.7.137 (36) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: s=SIP Call (10) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: c=IN IP4 192.168.7.137 (22) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: t=0 0 (5) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: m=audio 5008 RTP/AVP 0 101 (26) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: a=sendrecv (10) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: a=ptime:20 (10) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: a=rtpmap:101 telephone- event/8000 (33) [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: a=fmtp:101 0-11 (15) [Jul 16 13:43:42] VERBOSE[3214] logger.c: --- (12 headers 11 lines) --- [Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Invalid SIP message - rejected , no callid, len 766 [Jul 16 13:43:42] VERBOSE[3214] logger.c: --- SIP read from 192.168.7.135:5060 --- I performed a tcpdump of UDP packets during one of these failed inbound calls. Of approx. 3000 packets logged, almost all packets are a repeat of the following
Re: [asterisk-users] Digium TDM400P in Soekris net5501-70?
On Mon, 20 Jul 2009, Brian McEntire wrote: A good rule of thumb for 24x7 devices is $1 per watt per year, so 45 watts, while good, will still be $45 per year. In San Diego, CA we pay $0.33 per kWh (Over 200% of Baseline rate). With 8,760 hours in a year. That works out to $2.98 per watt or $130.86 for a 45 watt device. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Digium TDM400P in Soekris net5501-70?
I still don't see what you gain by using m0n0wall and a separate Asterisk install. I can't think of one thing that you would need a separate m0n0wall instance to do that AstLinux can't do on it's own. The web interface has become quite completely in the last few releases. Traffic shaping, firewall, vpn support etc. I don't understand how a VM does anything more than complicate an otherwise simple set up. Darrick Brian McEntire wrote: Thanks for the reply Alex. I'm not too scared of the soldering iron (I own one, but my work with it isn't pretty ;-) But can you confirm, are you just using the small power header on the board to supply power to the pci card? I was wondering if I was going to have to snake an another wall wort into the box to power the card, would be good if I don't have to do that! Not 100% sure I could run a VM on it, but the new net5501 board comes with 512MB ram and I think a 500-ish MHz processor, way more than what I'm currently using to run m0n0wall, so even if the VM takes a bite out of it, it should be fine, hardest part might be configuring the VM to boot monowall from CF. Can you partition a CF card? (ie, one partition for the monowall firmware and the other for the stripped down linux install to run Asterisk?) On Mon, Jul 20, 2009 at 4:44 PM, Alex Samada...@samad.com.au wrote: On Mon, Jul 20, 2009 at 01:09:30PM -0400, Brian McEntire wrote: Hello - I've been running Asterisk (quite happily!) for several years now using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM). I'm also running another old PC running m0n0wall as a firewall. Between these two boxes, that run 24x7, I'm drawing a lot more power than needed and hoping to make a dent in my monthly electric bill by consolidating the two into a single box with efficient power supply, low power processor, and no spinning HD platters. Main question is whether anyone knows if the Digium TDM400P should be compatible with the 3.3V PCI slot in the Soekris Net5501-70 box? Hi I have a the same setup you mention here, except I have a tdm410 card. I have a cf boot and a SSD card as well. Running Debian for firewall and asterisk server. Works well I have 3 vpn tunnels and a 6to4 tunnel ending on this machine, 2 fxs + 1 fxo. from my collectd graphs it looks like it really only spike when I am taking readins :) One catch the case that comes from soekris is too tight to put the molex on, I had to solder it to the connectors underneath. all fine though I am not sure about running a vm on this box though - I have some thing similiar at another site, but a bigger box. Alex Soekris' description for the net5501-70 says, in part, it has support for one or two low-power standard PCI board I see on my Digium card that it requires a molex connector supplying voltage. The Net5501 has a small 4-pin molex header on the board, I wonder if a small to regular sized molex power cable would do the job to supply this card. If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? -- Expense Accounts, n.: Corporate food stamps. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkpk1yMACgkQkZz88chpJ2NvmgCg3+4zJhQBcnQzxMPeQ1N+KXn1 XBMAnjtAOUjpC/++2acwVuHcYOpPQG21 =eRW6 -END PGP SIGNATURE- ___ -- 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] DAHDI Error and poor audio quality
-- I know it doesn't really sound very helpful to blame the entire server manufacturer, but some others might agree, brand spanking new and shiny might not be the best thing for Asterisk, especially these cards. There's nothing wrong with brand spanking new and shiny, as long as it is not certain name brand manufacturers who find a need to 'distinguish' themselves in the marketplace by making motherboards that aren't fully standards compliant. I've had far fewer problems with Dell, for example. Yes it is DL580 from HP. I wanted something big for the type of load to be used but now im very convinced that it wasn't a best shot! The major problem I have been facing with both dell and HP, is the kernel panic!! This one however doesn't give the panic but HDLC and D-channels disconnection does not want to dis appear. Digium has advised me to downgrade to dahdi 2.1.0.4 libpri 1.4.10.1 and monitor the situation. Busy doing that Shall update ___ -- 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 to PBX
Sadly, at the end of the day the answers will probably be no, no, no and no. PaulH logan wrote: Hi, I'm an absolute newbie and wanted to know the following. I want to have a setup where I have a PSTN line connected to my Asterisk box and want to know if it is possible to make more than one simultaneous outbound call through that VoIP gateway? Can Asterisk do this magic of concurrent calls on one PSTN line?? If I put it in other words then can I receive more than one simultaneous call on a PSTN number through Asterisk (the dialplan would forward those calls to different extensions) and the phone line still be able to receive more calls? Do I need some special hardware for the above or a simple SIPURA3000 would be good enough? Please pardon me if this is not the correct list for this question. Thanks. Best Regards, Hitesh ___ -- 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] Digium TDM400P in Soekris net5501-70?
Darrick - You seem adamant, and I will look deeper into the firewall in Astlinux! :-) The one thing running monowall in a VM would do for me is (in theory) make it very simple to move my existing, working m0n0wall configuration. I've been running it for a while, it serves a bunch of DHCP clients, does a little NAT, and has 20 or so specific rules for what can talk to what across the LAN, WAN, and DMZ segments of the firewall. If Astlinux can do all that, and I can grok it easily, it might be easier than running m0n0wall inside a VM. I suppose the other thing running m0n0wall inside a VM might do is a little extra security. If the firewall is in a VM and the asterisk part is running on the hardware without access to the LAN ports (which are all owned by the VM) then it *might* make the asterisk install a little more secure or less exposed to automated attacks. Not saying this is a high payoff for me, but another potential pro for a VM setup. On Mon, Jul 20, 2009 at 7:55 PM, Darrick Hartmandhart...@djhsolutions.com wrote: I still don't see what you gain by using m0n0wall and a separate Asterisk install. I can't think of one thing that you would need a separate m0n0wall instance to do that AstLinux can't do on it's own. The web interface has become quite completely in the last few releases. Traffic shaping, firewall, vpn support etc. I don't understand how a VM does anything more than complicate an otherwise simple set up. Darrick Brian McEntire wrote: Thanks for the reply Alex. I'm not too scared of the soldering iron (I own one, but my work with it isn't pretty ;-) But can you confirm, are you just using the small power header on the board to supply power to the pci card? I was wondering if I was going to have to snake an another wall wort into the box to power the card, would be good if I don't have to do that! Not 100% sure I could run a VM on it, but the new net5501 board comes with 512MB ram and I think a 500-ish MHz processor, way more than what I'm currently using to run m0n0wall, so even if the VM takes a bite out of it, it should be fine, hardest part might be configuring the VM to boot monowall from CF. Can you partition a CF card? (ie, one partition for the monowall firmware and the other for the stripped down linux install to run Asterisk?) On Mon, Jul 20, 2009 at 4:44 PM, Alex Samada...@samad.com.au wrote: On Mon, Jul 20, 2009 at 01:09:30PM -0400, Brian McEntire wrote: Hello - I've been running Asterisk (quite happily!) for several years now using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM). I'm also running another old PC running m0n0wall as a firewall. Between these two boxes, that run 24x7, I'm drawing a lot more power than needed and hoping to make a dent in my monthly electric bill by consolidating the two into a single box with efficient power supply, low power processor, and no spinning HD platters. Main question is whether anyone knows if the Digium TDM400P should be compatible with the 3.3V PCI slot in the Soekris Net5501-70 box? Hi I have a the same setup you mention here, except I have a tdm410 card. I have a cf boot and a SSD card as well. Running Debian for firewall and asterisk server. Works well I have 3 vpn tunnels and a 6to4 tunnel ending on this machine, 2 fxs + 1 fxo. from my collectd graphs it looks like it really only spike when I am taking readins :) One catch the case that comes from soekris is too tight to put the molex on, I had to solder it to the connectors underneath. all fine though I am not sure about running a vm on this box though - I have some thing similiar at another site, but a bigger box. Alex Soekris' description for the net5501-70 says, in part, it has support for one or two low-power standard PCI board I see on my Digium card that it requires a molex connector supplying voltage. The Net5501 has a small 4-pin molex header on the board, I wonder if a small to regular sized molex power cable would do the job to supply this card. If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? -- Expense Accounts, n.: Corporate food stamps. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkpk1yMACgkQkZz88chpJ2NvmgCg3+4zJhQBcnQzxMPeQ1N+KXn1 XBMAnjtAOUjpC/++2acwVuHcYOpPQG21 =eRW6 -END PGP SIGNATURE- ___ -- 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 -- Brian McEntire Photographer Owner B
Re: [asterisk-users] 2 Problems with 1.6.2
Sebastian wrote: Have you solved this issue? When I restart the machines I can't make an outgoing DAHDI call until I get an incoming call on that same line. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira Sent: viernes, 17 de julio de 2009 07:47 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 2 Problems with 1.6.2 At 01:09 PM 7/17/2009, you wrote: Sorry, that's the most frequent problem that people have with MWI in 1.6, so it was worth mentioning. I would suggest that you file a bug report on https://issues.asterisk.org. It would be helpful if you would include SIP debug output for both a machine that is working, as well as a machine that is not working. So I'd be more than happy to file a bug report and include all the SIP debug anyone might need but it's been so many years since I did it that I've no idea how anymore. So I grabbed a cordless handset, sat down at the console, typed sip set debug ip 192.xxx.xxx.xx and called that voicemail box to leave a message. The instant I hung up a notify message was sent to my phone, but the red light did not come on. If you remind me the how, I'll grab that message and post it here. Thanks so much for the help. 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 Checked by AVG - www.avg.com Version: 8.5.375 / Virus Database: 270.13.18/2243 - Release Date: 07/17/09 18:00:00 ___ -- 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 I have the same issue == https://issues.asterisk.org/print_bug_page.php?bug_id=14577 ___ -- 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] Digium TDM400P in Soekris net5501-70?
Brian McEntire wrote: Darrick - You seem adamant, and I will look deeper into the firewall in Astlinux! :-) Brian, I am one of the developers, so I happen to like what we've done. There have been some huge changes to the web interface and the overall project in the past year or so. http://www.astlinux.org The one thing running monowall in a VM would do for me is (in theory) make it very simple to move my existing, working m0n0wall configuration. I've been running it for a while, it serves a bunch of DHCP clients, does a little NAT, and has 20 or so specific rules for what can talk to what across the LAN, WAN, and DMZ segments of the firewall. If Astlinux can do all that, and I can grok it easily, it might be easier than running m0n0wall inside a VM. The firewall part of Astlinux is Arno's IPtables firewall. The web interface can handle most (if not all) of what you're trying to do. We've exposed a few more options in our svn trunk, but that's undergoing some big changes right now to support dahdi. I'm running an image based on that right now, but it will probably be another week or so before trunk is stable enough for general use. If there's something you need that's not exposed in the web interface, ask and someone on our mailing list can get you going in the right direction. If you have any problems/questions, ask over on our mailing list or in the #astlinux channel on freenode. I suppose the other thing running m0n0wall inside a VM might do is a little extra security. If the firewall is in a VM and the asterisk part is running on the hardware without access to the LAN ports (which are all owned by the VM) then it *might* make the asterisk install a little more secure or less exposed to automated attacks. Not saying this is a high payoff for me, but another potential pro for a VM setup. That could very well be the case, but I highly doubt you're going to like the results of using a net5501 as a virtual machine host. The hardware was never really intended for that purpose. Darrick ___ -- 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] 2 Problems with 1.6.2
At 03:30 PM 7/20/2009, you wrote: Have you solved this issue? When I restart the machines I can't make an outgoing DAHDI call until I get an incoming call on that same line. I've not and from the responses it's sounds like a known problem 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
[asterisk-users] how to use patgen and pattest for PRI card?
hello: I wan to use the test tools-patgen and pattest for pri cards. according to Tzafrir Cohen from http://docs.tzafrir.org.il/man/pattest.8.html, i still does not know how to use that. do i need to connect two pri cards with two servers, or use a cable to connect two cards in one server? please give me a more details in term of cables and configurations. thanks! Chris ___ -- 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 Call Transsfer
*I have the following scenario : ** ** kamailio User(222) --- Asterisk GW Call to kamailio user **--1234786 ** ** | ** ** | ** ** | -- 1234567 ** ** Firstly extension 222 will call the asterisk (this call made for internal DB query and rule checking for extension) now asterisk again transfer that call to another kamailio user 1234786 number and this works fine** without no problem , after that 222 wants to transfer that call to another extension 1234567** , then 222 will transfer 1234786to 1234567 ( the problem ** now started --) how can i achive this i dont know i have tried feature.conf dial with Tt option and also set variable GOTO_ON_BLINDXFR=woohoo^s^1 before daling can anybody know how to achive this call transsfer within this scenario regars Dhaval ** * ___ -- 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
