Re: [asterisk-users] [Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue.

Further, Before I start working on full project, I wanted to test the functionalities to be implemented. So I wrote a small test dialplan to check whether I can record a Queue call in Macro which gets executed on Member answer. My actual macro would be like this... [macro-agntanserd] exten => s,1,

[asterisk-users] Audiocodes MP-114 - modem dial not going through

I have an Audiocodes MP-114 and modem dial-out can not establish connection. Snooping with Wireshark I get a lot of entries like this: No. TimeSourceDestination Protocol Info 2934 27.221559 10.0.0.11010.0.0.109RTP PT=ITU-T G

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On Thu, May 5, 2011 at 11:02 PM, Matt Riddell wrote: > On 6/05/11 3:14 PM, Ira wrote: > >> At 03:00 PM 5/5/2011, you wrote: >> >> Yes, but in my world there is one Atom powerd Linux box running Asterisk, 4 or 5 Windows machines and 2 Macs. If I want to test, it has to be on my producti

Re: [asterisk-users] [Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue.

Thank you very much for your response and suggestion. I raised the question because in my project I don't want to record all the Queue calls. I just want to record calls connected with some specific members. --AM On Thu, May 5, 2011 at 11:10 PM, Carlos Chavez wrote: > On Thu, 2011-05-05 at 18:1

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On 6/05/11 3:14 PM, Ira wrote: At 03:00 PM 5/5/2011, you wrote: Yes, but in my world there is one Atom powerd Linux box running Asterisk, 4 or 5 Windows machines and 2 Macs. If I want to test, it has to be on my production box and I'm more than happy to run beta software on that box. My comment

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

At 03:00 PM 5/5/2011, you wrote: Yes, but in my world there is one Atom powerd Linux box running Asterisk, 4 or 5 Windows machines and 2 Macs. If I want to test, it has to be on my production box and I'm more than happy to run beta software on that box. My comment is just that the protocol for m

Re: [asterisk-users] Why is PQMSTATUS empty?

On Thu, May 5, 2011 at 8:20 PM, Louis Carreiro wrote: > Hmmm I'll have to check that out tomorrow AM. So you're calling > PauseQueueMember() to actually pause the user first? Something like > PauseQueueMember(,${EXTEN})? > > If the PQMSTATUS is "Success" instead of "Already Paused", wouldn't

[asterisk-users] how to let the call play audio when the dial fail

Hi all, I want to play an audio hint to caller when his dial fail rather than the current sound "dodo". The caller use asterisk to do the call so I want to setup this asterisk to achieve play specific audio when the dial fail or time out. How to setup to achieve that? Thanks! -- __

Re: [asterisk-users] Why is PQMSTATUS empty?

Hmmm I'll have to check that out tomorrow AM. So you're calling PauseQueueMember() to actually pause the user first? Something like PauseQueueMember(,${EXTEN})? If the PQMSTATUS is "Success" instead of "Already Paused", wouldn't it always respond with "Success" therefore not allowing the logic

Re: [asterisk-users] Why is PQMSTATUS empty?

There may be a more elegant way to do it, but what I do is simply call PauseQueueMember() first, then check the result of PQMSTATUS. I forget the exact results but I think it's either 'Success' or 'Already Paused', and then I base my next step in logic based on that result. Thanks, --Warren Se

Re: [asterisk-users] Why is PQMSTATUS empty?

Well that makes sense Is there anyway to determine the user's state ([un]paused) in the dial plan prior to using the [Un]PauseQueueMember() app? I'd like to be able to keep it very simple for my users by instructing them to dial one key code to pause or unpause. v/r, Me On Thu, May 5, 201

[asterisk-users] feedback mechanism

Hi All, I would like to write a script to run on peers to monitor my resources such as whether a card was removed and send a signal to LB so it can resize the capacity configuration for that peer, but I have no idea which event in Asterisk should be monitored when a card was remove or added? Som

Re: [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!

Richard wrote: >> No, conference scheduling is not a feature that we have built >> directly into ConfBridge, and I'm debating on what it would look >> like. > Scheduling isn't built into MeetMe either, but the fact that it > dynamically reads from a database means that you can write external > pr

Re: [asterisk-users] Why is PQMSTATUS empty?

PQMSTATUS is set only after you run the application PauseQueueMember(). Thanks, --Warren Selby, dCAP On May 5, 2011, at 2:11 PM, Louis Carreiro wrote: > Hey all! > > I'm trying to do a bit of logic here so that a user only has to dial one code > to pause/unpause in a queue (e.g. *0 will (un)

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On 6/05/11 8:35 AM, Ira wrote: At 01:07 PM 5/5/2011, you wrote: I am not saying using production servers to test, rather reproducing your production setups in a test environment. You would then create test plans or test cases of the features you use in Asterisk. Once documented, for each and eve

Re: [asterisk-users] missed call notification

Make it to Astricon this year, I'll buy you a drink and you can take the test there! That's what I did last year. :) Thanks, --Warren Selby, dCAP On May 5, 2011, at 1:27 PM, Sherwood McGowan wrote: > Heheh, well Warren, I'm just a quick draw I guess ;-) Hey, at least you have > dCAP by your

Re: [asterisk-users] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer

On 11-05-05 05:14 PM, Mark G Thomas wrote: Hi, I think this must be a bug introduced with 1.6.2.17.something. When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or 1.6.2.18, my AEL Dial() commands with the "U" options fail with the following error: [May 3 12:05:54] ERROR[6300] app

[asterisk-users] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer

Hi, I think this must be a bug introduced with 1.6.2.17.something. When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or 1.6.2.18, my AEL Dial() commands with the "U" options fail with the following error: [May 3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non-existent

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

At 01:07 PM 5/5/2011, you wrote: I am not saying using production servers to test, rather reproducing your production setups in a test environment. You would then create test plans or test cases of the features you use in Asterisk. Once documented, for each and every RC of Asterisk you go thr

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On 11-05-05 04:11 PM, Mark Deneen wrote: On Thu, May 5, 2011 at 4:07 PM, Paul Belanger wrote: On 11-05-05 12:30 PM, Ira wrote: At 07:56 AM 5/5/2011, you wrote: So how can we fix this? How can we get more people involded? What makes projects like FedoraTesting[3] and DebianTesting[4] popula

Re: [asterisk-users] estimated queue hold time

Hi again, Perhaps I could execute a macro while joining the queue and evaluate the QUEUEHOLDTIME variable there? If so can I in the macro remove a member from the queue it just joined ??? thanks in advance On 5 May 2011 19:41, Tiago Geada wrote: > Hello list, > > I'm looking for a way to ha

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On Thu, May 5, 2011 at 4:07 PM, Paul Belanger wrote: > On 11-05-05 12:30 PM, Ira wrote: > >> At 07:56 AM 5/5/2011, you wrote: >> >>> So how can we fix this? How can we get more people involded? What >>> makes projects like FedoraTesting[3] and DebianTesting[4] popular? How >>> can the Asterisk pr

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On 11-05-05 12:30 PM, Ira wrote: At 07:56 AM 5/5/2011, you wrote: So how can we fix this? How can we get more people involded? What makes projects like FedoraTesting[3] and DebianTesting[4] popular? How can the Asterisk project reproduce their success? Well, it's not a lot of people willing to

Re: [asterisk-users] receive faxes

On 05/06/2011 02:09 AM, David Backeberg wrote: T.38 has a boatload of problems, and most of those problems are because people who aren't employed by Digium did not read the specs, or they did read the specs, but felt like they had to violate the specs to get their code to work with a different b

[asterisk-users] Why is PQMSTATUS empty?

Hey all! I'm trying to do a bit of logic here so that a user only has to dial one code to pause/unpause in a queue (e.g. *0 will (un)pause depending on the users's state). My logic looks fine to me but every time ${PQMSTATUS} shows up empty. Here's the extensions.conf part exten => *0,1,NoOp

Re: [asterisk-users] receive faxes

spandsp works fine on our PRI line context FAX { s => { NoOp(INFO: Getting fax in ); Answer(); Set(TIMEOUT(absolute)=600); // 10 min Wait(3); if("${CALLERID(num)}"="") { //

[asterisk-users] estimated queue hold time

Hello list, I'm looking for a way to have the estimated hold time on a queue prior to joining it. someone suggested to me to Queue() first for 1 sec, read variable QUEUEHOLDTIME, validade it and Queue() again. But as we're using real time configuration that would mean a event ENTERQUEUE and a LE

Re: [asterisk-users] missed call notification

Heheh, well Warren, I'm just a quick draw I guess ;-) Hey, at least you have dCAP by your name! I've been at this 6-7 years and still haven't gotten off my butt and taken the tests :D On Thu, May 5, 2011 at 1:20 PM, Warren Selby wrote: > And Sherwood beats me to the punch again :). > > Thanks, >

Re: [asterisk-users] asterisk-1.8 crash if no extension specified in Dial

This issue has been resolved in latest branch 1.8 and will be resolved 1.8.5 version. Thanks for report. -S > Date: Thu, 5 May 2011 19:47:50 +0200 > From: supp...@accm.at > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] asterisk-1.8 crash if no extension specified

Re: [asterisk-users] missed call notification

You guys awesome! its working now only i need to modify script and do some trimming Thanks a lots again.. -S From: wcse...@selbytech.com Date: Thu, 5 May 2011 13:20:59 -0500 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] missed call notification And Sherwood beats me to th

Re: [asterisk-users] missed call notification

And Sherwood beats me to the punch again :). Thanks, --Warren Selby, dCAP On May 5, 2011, at 1:15 PM, Sherwood McGowan wrote: > No, the variables are channel specific except for when they're inherited, > which doesn't affect you here > > On Thu, May 5, 2011 at 1:02 PM, satish patel wrote: >

Re: [asterisk-users] missed call notification

I forget the term, but basically the variables you set on a current active channel are only accessible on that channel. In this case the variables are specific to the specific call in progress. Thanks, --Warren Selby, dCAP On May 5, 2011, at 1:02 PM, satish patel wrote: > After google i fou

Re: [asterisk-users] missed call notification

No, the variables are channel specific except for when they're inherited, which doesn't affect you here On Thu, May 5, 2011 at 1:02 PM, satish patel wrote: > After google i found something and i tried following. I set variable > before Dial and its give me proper value in "h" extension but now

Re: [asterisk-users] receive faxes

On Thu, May 5, 2011 at 1:43 PM, vip killa wrote: > The majority of open source projects out are NOT run by commercial > institutions... Postfix kicks butt. But only because IBM paid for development, for a long number of years, and because they hired somebody who had a really good idea how to impr

Re: [asterisk-users] missed call notification

After google i found something and i tried following. I set variable before Dial and its give me proper value in "h" extension but now question is if multiple user dial multiple extension then will it overwrite current variable value ? exten => s,1,Set(_CALLED_EXT=${ARG2}) exten => s,n,Dial(

Re: [asterisk-users] missed call notification

Could you please tell me how ( Syntax ) and where in macro ? I am not expert in dialplan variables. I appreciate your help Date: Thu, 5 May 2011 12:44:19 -0500 From: sherwood.mcgo...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] missed call notification if you sa

Re: [asterisk-users] asterisk-1.8 crash if no extension specified in Dial

I had that problem too, I wastesting with asterisk 1.8.3.2 and come across this: Call from one extension to another with: [macro-internal-call];ARG1=extension to call exten => s,1,Set(TOCALL=${DB(SIP/${ARG1})}) exten => s,2,Dial(SIP/${TOCALL},60,tT) ... As I had no entry in the asteriskdb, so

Re: [asterisk-users] missed call notification

if you saved ${_CALLED_EXT} to the value of ${EXTEN} from within the macro, you'd get 's'do it while you still have the called number as the EXTEN On Thu, May 5, 2011 at 12:42 PM, satish patel wrote: > > Also check for CANCEL, since this should be the status if the caller > hangs up before t

Re: [asterisk-users] receive faxes

The majority of open source projects out are NOT run by commercial institutions... they are run by people committed to a better product (not to making money)... they are maintained by people who have the resources to host a repository (which does not take a lot of resources) and a community of dedi

Re: [asterisk-users] missed call notification

Also check for CANCEL, since this should be the status if the caller hangs up before the call is picked up. But CANCEL is return nothing [macro-stdexten] exten => s,1,Dial(${ARG2}&iax2/${ARG1},20,t) ; Ring the interface, 20 seconds maximum, call screening option (or use P for data

Re: [asterisk-users] [Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue.

On Thu, 2011-05-05 at 18:16 +0530, Asterisk Man wrote: > Hi, > > I have a simple Queue(named 1) and one Member(SIP/1119) logged into > it. Now when a caller is placed into Queue and gets connected with > Member, I want to record the call. It does record the call when I use > MixMonitor() before pl

Re: [asterisk-users] missed call notification

You want me to do this in macro-stdexten ? I have following dialplan. I have used "h" extension in original context because you can't you "h" inside macro right ? [macro-stdexten] exten => s,1,Dial(${ARG2}&iax2/${ARG1},20,t) ; Ring the interface, 20 seconds maximum, call screenin

Re: [asterisk-users] missed call notification

Set a variable ${_CALLED_EXT} to ${EXTEN} before you hang up the call, then reference that variable in your h exten. Thanks, --Warren Selby, dCAP On May 5, 2011, at 11:59 AM, satish patel wrote: > Hi All, > > I am using > http://www.theschmandts.org/blog/2007/05/05/email-notifications-for-m

[asterisk-users] missed call notification

Hi All, I am using http://www.theschmandts.org/blog/2007/05/05/email-notifications-for-missed-calls-in-asterisk/ to implement missed call feature. and i modify script to grab email address from voicemail.conf But i am not able to see DEST extension in this script ? what would be the vari

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

From: "Ira" Sent: Thursday, May 05, 2011 12:38 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind? At 07:56 AM 5/5/2011, you wrote: >So how can we fix this? How

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

At 07:56 AM 5/5/2011, you wrote: So how can we fix this? How can we get more people involded? What makes projects like FedoraTesting[3] and DebianTesting[4] popular? How can the Asterisk project reproduce their success? Well, it's not a lot of people willing to run beta software on their p

Re: [asterisk-users] asterisk for g729 to g711

On 05/05/2011 12:36 PM, Woody Dickson wrote: Does anyone know if Asterisk is a good tool to be used for a large quantity of g711 and g729 transcoding? Well, what is "large?" Transcoding is a fairly CPU-bound process. In principle, the answer to your question is probably more yes than no, be

[asterisk-users] asterisk for g729 to g711

Hi, Does anyone know if Asterisk is a good tool to be used for a large quantity of g711 and g729 transcoding? What is the best alternative for that? -- Woody Dickson woodydick...@gmail.com US and Worldwide Termination Contact me for the following offering USA On

Re: [asterisk-users] receive faxes

It isn't any better than the so called t.38 support in Asterisk that only drops calls. Gee I wonder why, maybe so they can sell their fax product? On Wednesday, May 4, 2011, Steve Edwards wrote: > On Wed, 4 May 2011, vip killa wrote: > > > screw that i just got hylafax to work with IAXMODEM..

Re: [asterisk-users] receive faxes

> I don't believe you really understand what Open Source means...it > does not mean "FREE". Actually, it DOES mean "free", especially since Asterisk is under the GPL. But, as RMS often says, "that's 'free' as in 'free speech', not 'free beer'". That problem doesn't exist in French, where there a

Re: [asterisk-users] Discussion: Test platform

On Thu, May 5, 2011 at 10:07 AM, Olle E. Johansson wrote: > > So how can we fix this? How can we get more people involded? What makes > projects like FedoraTesting[3] and DebianTesting[4] popular? How can the > Asterisk project reproduce their success? > Give them something that tests their ow

Re: [asterisk-users] receive faxes

On Wed, May 4, 2011 at 9:01 AM, vip killa wrote: > screw that i just got hylafax to work with IAXMODEM...i refuse to pay > digium a dime... supposed to be open-source right? > I don't believe you really understand what Open Source means...it does not mean "FREE". It means that the source is

Re: [asterisk-users] Auto dialing Polycoms and other SIP phones

Pan, Thank you, that makes sense. I’ll investigate further. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pan B. Christensen Sent: Thursday, May 05, 2011 11:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discus

[asterisk-users] Asterisk 10 / Trunk and RecieveFax "F" Option

I have been using sendfax and recievefax with 1.8.x.x version I have a patch that Kevin Fleming wrote to allow the forced shutoff of T.38 "F" option. This was considered a new feature so it is not in new releases of 1.8.x and I have not been able to get a patch working for the current releases

Re: [asterisk-users] Queues, pickup and transfers

2011/5/5 Olivier > Hi, > > If my memory serves me right, up to Asterisk 1.6, Queue app internals kept > the application from working some other apps such as PickUp. > > I wonder if such things are possible (and if possible, still keep useful > Queue Logs ie logs in which picked up or transfered c

Re: [asterisk-users] [Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue.

On Thu, May 5, 2011 at 7:46 AM, Asterisk Man wrote: > Hi, > > I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now > when a caller is placed into Queue and gets connected with Member, I want to > record the call. It does record the call when I use MixMonitor() before > placi

Re: [asterisk-users] Auto dialing Polycoms and other SIP phones

Hello Mike, It is possible for Polycom phones to auto-answer an incoming call with speakerphone. I don’t have the details available right now, but it requires changing the phone’s configuration and sending a custom sip header with the INVITE. Great care should be taken when implementing this, a

[asterisk-users] Queues, pickup and transfers

Hi, If my memory serves me right, up to Asterisk 1.6, Queue app internals kept the application from working some other apps such as PickUp. I wonder if such things are possible (and if possible, still keep useful Queue Logs ie logs in which picked up or transfered calls are shown as such): 1- a c

[asterisk-users] Does IAX2 support call completion or callback ?

Hi ALL, Does IAX2 support call completion feature ? I meant between SIP <---> IAX2 ? or IAX2<--->IAX2 -S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Discussion: Test platform

> > Here is the thing, there is nothing stopping 'the community' today from doing > this. In fact, we already have a testsuite [1] in place, running each > subversion commit and producing results for the last year. But this is only > one type of testing; automated, we also have unit tests bui

Re: [asterisk-users] receive faxes

Where's the dislike button? On May 4, 2011 7:03 AM, "vip killa" wrote: > screw that i just got hylafax to work with IAXMODEM...i refuse to pay > digium a dime... supposed to be open-source right? > > On Wed, May 4, 2011 at 9:52 AM, Danny Nicholas wrote: > >> -- >>

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On 11-05-04 06:01 PM, Matt Riddell wrote: On 3/05/11 4:01 AM, Hans Witvliet wrote: Just a thought If "Digium" / "the community" realy want an objective way of deciding whether can/should migrate to any other version, you realy need a feature-matrix (pethaps starting from version 1.2.*) And for

Re: [asterisk-users] Orginate not working well with PSTN lines

5 maj 2011 kl. 16.35 skrev Gilles: > On Tue, 3 May 2011 13:48:34 -0500, Shaun Ruffell > wrote: >> I know this thread is dead but: I do not believe this should go into the >> DAHDI >> kernel modules. > > I agree. It's just too bad Dahdi is unable to report how an outgoing > call is doing: Still

Re: [asterisk-users] Orginate not working well with PSTN lines

On Tue, 3 May 2011 13:48:34 -0500, Shaun Ruffell wrote: >I know this thread is dead but: I do not believe this should go into the DAHDI >kernel modules. I agree. It's just too bad Dahdi is unable to report how an outgoing call is doing: Still ringing, busy, answered. --

Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

Thanks Danny - We are replacing 1.2.x and with just basic features There are some bug in 1.8.3.3 thats why i thought go with latest branch. cant wait for 1.8.4.x release -S From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 5 May 2011 09:17:14 -0500 Subject: Re: [asterisk

Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Thursday, May 05, 2011 9:14 AM To: asterisk-users Subject: [asterisk-users] Asterisk 1.8 latest branch safe for production ? Hi All, Just wondering is it sa

[asterisk-users] Asterisk 1.8 latest branch safe for production ?

Hi All, Just wondering is it safe to use asterisk 1.8 latest branch on production ? http://svn.asterisk.org/svn/asterisk/branches/1.8/ Revision 317100 -S -- _ -- Bandwidth and Col

Re: [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2

> In that case it suggests it is some setting you have applied to the > phones that is causing it. I just called Aastra tech support. I'm always VERY impressed that the first person who picks up the phone is very technical. He said that they've had reports of this issue. The problem goes away

Re: [asterisk-users] audiohook.c: Failed to get 160 samples from write factory

On 05/05/11 14:16, Olle E. Johansson wrote: We've had that for quite some time. There's an option to Dial() and one for Queue() to enable it. Check the documentation. /O yes my only problem with the 'c' option for the Dial command is that it still seems to add the Reason header if the ca

Re: [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2

> In that case it suggests it is some setting you have applied to the > phones that is causing it. Can you post the local.cfg & server.cfg > files from the phone (removing the passwords from there first)? Sure: local.cfg is checksums, server information, and: contrast level: 3 ringer volume:

Re: [asterisk-users] audiohook.c: Failed to get 160 samples from write factory

On 05/05/2011 03:11 PM, Paul Hayes wrote: On 05/05/11 14:04, Jonas Kellens wrote: Hello list, what does this mean : [May 5 *14:58:12*] DEBUG[8770] chan_sip.c: This call was answered elsewhere[May 5 14:58:12] DEBUG[8770] chan_sip.c: ### It's the cause code, buddy. The cause code!!! [May 5

[asterisk-users] Auto dialing Polycoms and other SIP phones

Hi, Is there a reliable way to auto-dial SIP phones (specifically Polycom) with some sort of TAPI driver in Windows? I am aware of SIPTAPI, which makes the user's phone ring, and when picked up dials the desired number, but I (and more to the point, many of my customers) find this annoying. I

Re: [asterisk-users] audiohook.c: Failed to get 160 samples from write factory

5 maj 2011 kl. 15.11 skrev Paul Hayes: > On 05/05/11 14:04, Jonas Kellens wrote: >> Hello list, >> >> >> what does this mean : >> >> [May 5 *14:58:12*] DEBUG[8770] chan_sip.c: This call was answered >> elsewhere[May 5 14:58:12] DEBUG[8770] chan_sip.c: ### It's the cause >> code, buddy. The

Re: [asterisk-users] audiohook.c: Failed to get 160 samples from write factory

On 05/05/11 14:04, Jonas Kellens wrote: Hello list, what does this mean : [May 5 *14:58:12*] DEBUG[8770] chan_sip.c: This call was answered elsewhere[May 5 14:58:12] DEBUG[8770] chan_sip.c: ### It's the cause code, buddy. The cause code!!! [May 5 14:58:12] DEBUG[8770] chan_sip.c: This call

Re: [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2

On 05/05/11 13:41, Richard Kenner wrote: Asterisk does indeed send an Options before the OK but my 57i doesn't seem to mind. That's odd. It does for me. Perhaps you need to upgrade firmware on the Aastra phone? The problem occured when I DID upgrade it! Precisely to the one you mentioned.

Re: [asterisk-users] audiohook.c: Failed to get 160 samples from write factory

Hello list, what does this mean : [May 5 *14:58:12*] DEBUG[8770] chan_sip.c: This call was answered elsewhere[May 5 14:58:12] DEBUG[8770] chan_sip.c: ### It's the cause code, buddy. The cause code!!! [May 5 14:58:12] DEBUG[8770] chan_sip.c: This call was answered elsewhere[May 5 14:5

[asterisk-users] [Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue.

Hi, I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now when a caller is placed into Queue and gets connected with Member, I want to record the call. It does record the call when I use MixMonitor() before placing the caller into Queue, but not when MixMonitor() is used in m

Re: [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2

> Asterisk does indeed send an Options before the OK but my 57i doesn't > seem to mind. That's odd. It does for me. > Perhaps you need to upgrade firmware on the Aastra phone? The problem occured when I DID upgrade it! Precisely to the one you mentioned. > Or turning off qualify for this pee

Re: [asterisk-users] SIP secruity: username and password

On Thursday 05 May 2011, bilal ghayyad wrote: > Hi All; > > When the endpoint register on Asterisk or initiate a call, so they exchange > the sip username and password. What is the possibility that this will be > capture by the hacker and how to avoid this problem? If the two devices are connected

Re: [asterisk-users] SIP secruity: username and password

On 05/05/2011 08:21 AM, Olle E. Johansson wrote: Because they HAVE TO. In the 401/407 reply, there's a challenge (nonce) which is an important part of the MD5 Digest Auth scheme. I meant more to contrast with how some UACs will attempt to re-cycle old Authorization credentials in re-registrat

Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

Not sure if you are issuing DHCP at the access point or from a central control >From a central control should allow seamless roaming within different APs, >assuming easy auth to the AP, the only issue you get is when the handset >dithers between choosing signals from one or the other, and thats

Re: [asterisk-users] SIP secruity: username and password

5 maj 2011 kl. 14.17 skrev Alex Balashov: > Bilal, > > On 05/05/2011 08:08 AM, bilal ghayyad wrote: > >> When the endpoint register on Asterisk or initiate a call, so they >> exchange the sip username and password. What is the possibility that >> this will be capture by the hacker and how to av

Re: [asterisk-users] SIP secruity: username and password

Thanks Alex for clearing up the bit about the NONCE, that's what I was trying to remember when I said CallID :) Good explanation by the way! :) On Thu, May 5, 2011 at 7:17 AM, Alex Balashov wrote: > Bilal, > > > On 05/05/2011 08:08 AM, bilal ghayyad wrote: > > When the endpoint register on Aste

Re: [asterisk-users] SIP secruity: username and password

Bilal, On 05/05/2011 08:08 AM, bilal ghayyad wrote: When the endpoint register on Asterisk or initiate a call, so they exchange the sip username and password. What is the possibility that this will be capture by the hacker and how to avoid this problem? Strictly speaking, there is no inheren

Re: [asterisk-users] SIP secruity: username and password

Little to none...SIP is set up so that the packet contains identifiable data (the username) but the authentication is performed with a digest of the username password [domain] and [CalliD] (I think I got that right) On Thu, May 5, 2011 at 7:08 AM, bilal ghayyad wrote: > Hi All; > > When the endp

Re: [asterisk-users] SIP secruity: username and password

5 maj 2011 kl. 14.08 skrev bilal ghayyad: > Hi All; > > When the endpoint register on Asterisk or initiate a call, so they exchange > the sip username and password. What is the possibility that this will be > capture by the hacker and how to avoid this problem? We never exchange passwords in

[asterisk-users] SIP secruity: username and password

Hi All; When the endpoint register on Asterisk or initiate a call, so they exchange the sip username and password. What is the possibility that this will be capture by the hacker and how to avoid this problem? Regards Bilal -- ___

Re: [asterisk-users] receive faxes

I would never choose to use it. Our system is built on top of it (before I ever got here) and it would be too great a task to change it not to mention management would not go for a change. On Wed, May 4, 2011 at 6:09 PM, Matt Riddell wrote: > On 5/05/11 3:02 AM, vip killa wrote: > >> Honestly Di

Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

What happens when base stations are moved such as at any time, handsets are either within range of one (or two or more) access point ? Is the call ended when the handset moves from one access point to another ? Are those access points 80211r compliant ? -- _

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

5 maj 2011 kl. 12.04 skrev Paul Hayes: > On 05/05/11 05:41, Cary Fitch wrote: >> >> >>Flavio E. Goncalves >>www.asteriskguide.com >> >>Compare to which version of Windows… Patches are a never ending process >> >>Cary Fitch >> >> > > I think th

Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

On 5/05/11 10:46 PM, Shawn L wrote: Yes, I'm talking about mid-call. I do have rtptimeout and qualify set, both to 30 seconds, which should be plenty of time. I set them both because if a phone moves out of range, and never comes back, asterisk was keeping the channel open way to long. Yeah,

Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

Yes, I'm talking about mid-call. I do have rtptimeout and qualify set, both to 30 seconds, which should be plenty of time. I set them both because if a phone moves out of range, and never comes back, asterisk was keeping the channel open way to long. On Wed, May 4, 2011 at 7:50 PM, Matt Riddell

Re: [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2

On 05/05/11 04:37, Richard Kenner wrote: I recently tried to update my Aastra 57i to version 3.2 and ran into a problem. It won't properly register and says "contact mismatch". I added "sip contact matching: 2" to aastra.cfg, but that didn't help. When I look at the SIP trace, but I see is the

Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

On 05/05/11 00:02, Ira wrote: Not that it applies but I recently installed a Snom M3 and it seems to behave like you want. When I walk out of range and then back in the call is usually still there. I've not tested past that so it might hang up after an unknown timeout. Ira The difference her

[asterisk-users] Could not place calls through IAX

Hello, I have some problems in placing calls through IAX... It does not work :) in the asterisk console I can't see nothing about dialplan enter or so, IAX debbugging seems to be unuseful... this is my configuration: [612] type=friend secret=123456 notransfer=yes disallow=all allow=gsm allow=ula

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On 05/05/11 05:41, Cary Fitch wrote: Flavio E. Goncalves www.asteriskguide.com Compare to which version of Windows… Patches are a never ending process Cary Fitch I think this attitude is half the problem. Asterisk is not a desktop computer

Re: [asterisk-users] best current version and motherboard/CPU compatibilities

On 04/05/11 18:17, || dave cantera Mobile wrote: paul, doug, I had several AMD athlons 64bit... no problems running centos, suse. they seem solid on 1.4.xx... had a few intel celerons and P4s. they were good as well. guess I was Lucky back then! thanks for supporting the list! daveC don't get

Re: [asterisk-users] audiohook.c: Failed to get 160 samples from write factory

Hello list, can it be that this has something to do with MixMonitor : [May 5 10:49:41] DEBUG[19790] func_audiohookinherit.c: Set audiohook MixMonitor to be inheritable [May 5 10:49:41] DEBUG[19869] audiohook.c: Read factory 0x95dc590 and write factory 0x95dcfc8 both fail to provide 160 sampl

Re: [asterisk-users] audiohook.c: Failed to get 160 samples from write factory

Hello, and this is what happened before : /[May 5 10:08:38] DEBUG[16215] channel.c: Hanging up channel 'SIP/expsom10-0442' [May 5 10:08:38] DEBUG[16215] chan_sip.c: This call was answered elsewhere[May 5 10:08:38] DEBUG[16215] chan_sip.c: ### It's the cause code, buddy. The cause

Re: [asterisk-users] audiohook.c: Failed to get 160 samples from write factory

Hello, is there yet any feedback on this ?? What is causing this and how to overcome this ? Using Asterisk 1.6.17.2, but had it on the past also with version 1.6.2.10, 1.6.2.11 and 1.6.2.16.1 I had it again today : [May 5 10:13:51] DEBUG[16223] audiohook.c: Failed to get 160 samples from

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