Re: [asterisk-users] Core Dump - Asterisk 1.4.24 - Elastix

On Tue, Nov 03, 2009 at 01:18:14AM -0300, Fernando Berretta wrote: > Hi, > > Yesterday I've got a core dump from Asterisk, other times I was able to > discover what this core dump was related with through gdb Ouput info,, > but this time.. I'm really lost. Could some one please help me > > GDB

[asterisk-users] Exchange 2007 UM issues with Asterisk 1.6

Hi Guys, Does anyone know how to make the custom build of Asterisk 1.6.1 work with Exchange 2007 UM? It always times out with system unavailable and won't go through. I have TCP enabled and all the trunk and outgoing settings configured. When I use TrixBox CE 2.8.0.2, it works. But custom i

[asterisk-users] Core Dump - Asterisk 1.4.24 - Elastix

Hi, Yesterday I've got a core dump from Asterisk, other times I was able to discover what this core dump was related with through gdb Ouput info,, but this time.. I'm really lost. Could some one please help me GDB output is at http://pastebin.com/m603e6a74 Any help would be appreciated. Best

Re: [asterisk-users] Asterisk as Outbound Proxy ?

"Proxy" is not the correct term to describe this scenario, but yes, it is possible in principle. Kristijan Vrban wrote: > Hello, short question: is there a possibility to use asterisk as an > outbound proxy? iam open for any suggestions, use asterisk trunk, dirty > patches, ugly workarounds, e

Re: [asterisk-users] turn the ring tone OFF during dialing

On 11/02/09 19:56, Joseph wrote: >Is the a way to turn the ring tone OFF during dialing? >When I'm in a macro mode I have to listen to ring the tone for 20sec before >macro finish and I get connected. I've found a better solution, setting musing on hold before calling party answers: m: Provide

[asterisk-users] turn the ring tone OFF during dialing

Is the a way to turn the ring tone OFF during dialing? When I'm in a macro mode I have to listen to ring the tone for 20sec before macro finish and I get connected. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- a

Re: [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6

My mental plan orginilly was: 1.- Creating a macro that acceps ARGs like. a.- agent b.- queue/s In the macro we could have the voice respose for the loging. I am using on 1.4 the following procedure. * the agents call to 21 to loging, and it is promt just for the passwd * the agents call to 22 t

Re: [asterisk-users] Asterisk 1.4 and Fax

On 11/03/2009 04:25 AM, Thomas Kenyon wrote: > Kevin P. Fleming wrote: > >> Lee Howard wrote: >> >>> I've heard of people who go to casinos and come home with a couple >>> thousand bucks winnings, too. But the truth is that invariably the vast >>> majority of people who gamble don't win.

Re: [asterisk-users] Asterisk 1.4 and Fax

2009/11/2 Doug Lytle > Christian Victor wrote: > > 2009/11/2 Doug Lytle mailto:supp...@drdos.info>> > > > > > > Faxing over SIP is never a good idea. > > > > And why would that be? I think that faxing over SIP using T.38 is a > > fantastic idea. > > As far as I know, T.38 isn't supported unde

Re: [asterisk-users] Asterisk as Outbound Proxy ?

Kristijan Vrban wrote: > Hello, short question: is there a possibility to use asterisk as an > outbound proxy? iam open for any suggestions, use asterisk trunk, dirty > patches, ugly workarounds, everything. > > What is want to build is: > > SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy

[asterisk-users] Asterisk as Outbound Proxy ?

Hello, short question: is there a possibility to use asterisk as an outbound proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly workarounds, everything. What is want to build is: SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy -> via UDP/RTP -> VoIP-Provider So

Re: [asterisk-users] Asterisk 1.4 and Fax

Dan Journo wrote: > How do these fax2email providers run their service? > I've not the faintest Idea, the provider I use afaict outsource it. > Do they all use physical lines rather than use the internet? > > Thanks > Dan > ___ -- Bandwidth and Coloc

Re: [asterisk-users] Anyone seen this before

On Mon, Nov 2, 2009 at 4:26 PM, Robert Grignon wrote: > Testing a new gateway and have a Rhino Channel Bank... Sending a test fax > and everything works fine (Receive the fax fine) But I notice this in the > log > > Google search didn't return much of anything... > > > DAHDI hook failed return

[asterisk-users] Anyone seen this before

Testing a new gateway and have a Rhino Channel Bank... Sending a test fax and everything works fine (Receive the fax fine) But I notice this in the log Google search didn't return much of anything... DAHDI hook failed returned -1 (trying 1): Device or resource busy __

Re: [asterisk-users] Tutorial for SIP user

Ciao, I installed Xlite on Windows Vista, the IP connection (ping) is working, shall I check something else ? Thanks in advance. 2009/11/1 Farooq Hussain > Dear Giancarlo, > > On which OS your are installing XLITE. If you are trying to connect XLITE > using Winodws XP please make a entry in yo

[asterisk-users] DTMF Timing and Fujitsu F9600 Switch

We're in the process of replacing an ancient Centigram-based, but Fujitsu-labelled, voicemail system with an Asterisk solution. The system will interface with a Fujitsu F9600 switch and use the SMDI module in Asterisk 1.6.1.x to communicate the calling information needed to make the interplay betwe

Re: [asterisk-users] Asterisk 1.4 and Fax

On Mon, Nov 2, 2009 at 3:38 PM, Dan Journo wrote: > How do these fax2email providers run their service? > > Do they all use physical lines rather than use the internet? If you read far enough back in the archives, you'll find somebody who claimed they used asterisk-1.4 (I think hylafax) and voip

Re: [asterisk-users] Asterisk 1.4 and Fax

David Backeberg wrote: > On Mon, Nov 2, 2009 at 12:22 PM, Dan Journo > mailto:d...@keshercommunications.com>> > wrote:asterisk app_fax (which depends on SpanDSP from Lee Howard). SpanDSP was written by Steve Underwood. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty t

Re: [asterisk-users] Asterisk 1.4 and Fax

On Mon, Nov 2, 2009 at 12:22 PM, Dan Journo wrote: > Hi, > > Does anyone have an up to date guide for setting up fax 2 email with > asterisk? > You can buy this shrink-wrapped from Cisco if you're willing to pay what they're asking. There are probably other vendors who sell that too. If you in

Re: [asterisk-users] Asterisk 1.4 and Fax

How do these fax2email providers run their service? Do they all use physical lines rather than use the internet? Thanks Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Kenyon Sent: 02 November 2009

Re: [asterisk-users] Asterisk 1.4 and Fax

Kevin P. Fleming wrote: > Lee Howard wrote: >> I've heard of people who go to casinos and come home with a couple >> thousand bucks winnings, too. But the truth is that invariably the vast >> majority of people who gamble don't win. >> >> http://hylafax.sourceforge.net/docs/fax-over-voip.pdf >>

Re: [asterisk-users] Asterisk 1.4 and Fax

Christian Victor wrote: > 2009/11/2 Doug Lytle mailto:supp...@drdos.info>> > > > Faxing over SIP is never a good idea. > > And why would that be? I think that faxing over SIP using T.38 is a > fantastic idea. As far as I know, T.38 isn't supported under 1.4 Doug -- Ben Franklin quote:

Re: [asterisk-users] Asterisk 1.4 and Fax

Lee Howard wrote: >> FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can >> also describe T.38, which is not as much of a gamble as FAX over VOIP :-) > > Does Asterisk 1.4 support T.38? Only for passthrough between SIP channels; Asterisk 1.6.0 and later also support T.38 termi

Re: [asterisk-users] Asterisk 1.4 and Fax

Kevin P. Fleming wrote: > Lee Howard wrote: > >> I've heard of people who go to casinos and come home with a couple >> thousand bucks winnings, too. But the truth is that invariably the vast >> majority of people who gamble don't win. >> >> http://hylafax.sourceforge.net/docs/fax-over-voip.pd

Re: [asterisk-users] Asterisk 1.4 and Fax

Lee Howard wrote: > I've heard of people who go to casinos and come home with a couple > thousand bucks winnings, too. But the truth is that invariably the vast > majority of people who gamble don't win. > > http://hylafax.sourceforge.net/docs/fax-over-voip.pdf > > Everyone wants to see if the

[asterisk-users] Unexpected control subclass '-1'

I have been getting the following message every time I make a call for the past few months: [Nov 2 13:08:18] WARNING[9859]: file.c:1273 waitstream_core: Unexpected control subclass '-1' Everything seems to be working so I do not know if this is important. I am using Asterisk 1.4.

Re: [asterisk-users] Asterisk 1.4 and Fax

2009/11/2 Doug Lytle > Dan Journo wrote: > > > > I need to get it up and running before we can put in the order to > > transfer the fixed line number over to SIP. > > > > Faxing over SIP is never a good idea. > And why would that be? I think that faxing over SIP using T.38 is a fantastic idea.

Re: [asterisk-users] Asterisk 1.4 and Fax

I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, how

Re: [asterisk-users] Asterisk 1.4 and Fax

I've heard mixed reports. Some say they've had no problems, some say that faxes fail most of the time. I want to try it out and see how it goes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Se

Re: [asterisk-users] Cannot make calls

Thank you , I did that sip set debug ip 198.163.0.103 SIP Debugging Enabled for IP: 198.163.0.103 I checked the /var/log/asterisk files and there is no information there. Could you please inform where am I suppose to see the debug information ? tks Jair Ott Rose wrote: you can

Re: [asterisk-users] DAHDI/ZAP overlap dialing

--- On Mon, 11/2/09, Martin wrote: > I'd double check that you really have overlapdial=yes for > those > channels ... it should be declared > before channel => keyword in > zapata.conf/chan_dahdi.conf I declared overlapdial in zapata.conf: switchtype = euroisdn signalling = pri_cpe overlapdial

Re: [asterisk-users] Asterisk 1.4 and Fax

Dan Journo wrote: > > I need to get it up and running before we can put in the order to > transfer the fixed line number over to SIP. > Faxing over SIP is never a good idea. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, dese

[asterisk-users] Execute Macro AFTER connecting to a channel

Is the a way to execute the macro AFTER connecting to the channel: M(x[^arg]) - Execute the Macro for the *called* channel before connecting to the calling channel. doesn't work for me as I need to listen to the macro progress as it is sending DTMF tone and respond from the connected

Re: [asterisk-users] Remote IP Phone's

On Monday 02 November 2009 11:46:49 am Steve Edwards wrote: > Un-top-posting... > > > On Mon, 2 Nov 2009, Danny Nicholas wrote: > >> You can restrict on IP address, MAC address and port type and that's > >> just what I know. If someone want's through bad enough you're going to > >> have a problem,

Re: [asterisk-users] Asterisk 1.4 and Fax

Sorry Steve, Forgot to remove it before sending the email. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 November 2009 17:35 To: Asterisk Users Mailing List - Non-Commercial Discussion S

Re: [asterisk-users] Asterisk 1.4 and Fax

We want to disconnect our PSTN fax line and transfer the number over to our asterisk server. I need to get it up and running before we can put in the order to transfer the fixed line number over to SIP. Thanks Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:a

Re: [asterisk-users] Remote IP Phone's

Un-top-posting... > On Mon, 2 Nov 2009, Danny Nicholas wrote: > >> You can restrict on IP address, MAC address and port type and that's >> just what I know. If someone want's through bad enough you're going to >> have a problem, but you can at least slow down or stop casual hackers. > > Aren't

Re: [asterisk-users] Asterisk 1.4 and Fax

On 2 Nov 2009, at 17:22, Dan Journo wrote: > Does anyone have an up to date guide for setting up fax 2 email with > asterisk? So you can fax them obnoxiously long signatures? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- a

Re: [asterisk-users] Remote Party ID

On Mon, Nov 02, 2009 at 03:18:03PM -, Dan Journo wrote: > > Does anyone know how to set the remote party id? I guess someone does. If you provide more details you'll have better chances of getting a good answer. Version of Asterisk? Why do you want to set the remote party ID? How have you

Re: [asterisk-users] Remote IP Phone's

Actually, both. You can (AFAIK) specify 5060, 1 etc and UDP/TCP, etc. Of course, I have been wrong at least once before :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Monday, Novembe

Re: [asterisk-users] Remote IP Phone's

Un-top-posting... > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Connor > Spiess > > I am wondering what people are doing for security when registering IP > phone's remotely if you do not have the equipment to do a VPN tunnel at > the remote site. The phone I would be working w

Re: [asterisk-users] DAHDI/ZAP overlap dialing

I can only tell you that it worked before... > Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Extension '1004' in context > 'from-pstn-deviate-custom' from '7034' does not when you have overlapdial turned on it should have checked if there's a potential matching extension which you have it righ

[asterisk-users] Asterisk 1.4 and Fax

Hi, Does anyone have an up to date guide for setting up fax 2 email with asterisk? Thanks Dan IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for sup

Re: [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6

I have sent this before but here is how I do agent login and queue: ; Agent login logout exten => *20,1,Verbose(2,Doing agent login/logout) exten => *20,n,Answer() exten => *20,n,wait(.0.5) exten => *20,n,Read(AgentNumber,agent-user) exten => *20,n,Set(UserID=${DB(ExtenToUser/${AgentNum

Re: [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6

To avoid boring everybody else to death with the discussion, I created a mailing list for that on Google Groups - see http://tinyurl.com/yjtf62s Thanks l. 2009/11/2 Lenz Emilitri > We were thinking about doing something similar as well. A lot of people are > asking for this. If there is anybod

Re: [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6

We were thinking about doing something similar as well. A lot of people are asking for this. If there is anybody else interested, we could share the load I was thinking about creating a context like @agents, so that when you do the log-on you basically add Local/1...@agents as a member of the

Re: [asterisk-users] Dialstatus

On 11/02/09 07:28, Steve Edwards wrote: >On Mon, 2 Nov 2009, Patrick Plattes wrote: > >> you can do print the dialstatus to the console e.g.: >> exten => s,n,NoOp(${DIALSTATUS}) >> >> More info: >> http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp > >A "better practice" would be to use verbose()

[asterisk-users] Nagios check_asterisk_peers needs rights to question the Asterisk-server

When executing the following command : [r...@nagios ~]# /usr/local/nagios/libexec/check_nrpe -H ip_address -c check_asterisk_peers I get the following output : NRPE: Unable to read output Somewhere Nagios does not have enough rights to question Asterisk about the sip peers. These are the right

Re: [asterisk-users] DAHDI/ZAP overlap dialing

--- On Sat, 10/31/09, Martin wrote: > On Sat, Oct 31, 2009 at 5:27 AM, > Tzafrir Cohen > wrote: > > I'm not sure if handling of overlap hasn't changed > since. > > > > But can you provide a trace of how Asterisk sees > things? e.g. 'pri > > intense debug span 1' > > > > the intense debug is ov

Re: [asterisk-users] SNOM 870

>> The mortality rate on power supplies, diplays and the number or >> broken >> receiver hook swicthes on the lot of Snom 360's i bought 3 years >> ago is >> outright embarrassing. >> > That's odd. We've had Snom 190s, 320s, and 360s running day in day out > for years with not a single issue.

Re: [asterisk-users] SNOM 870

Remco Barendse wrote: > On Fri, 30 Oct 2009, hbk wrote: > > >> Hi, >> >> I have played with the 820 for some weeks, mostly love it excellent speech >> quality. Even got the "mini" browser running >> showing my favorite webcam, this could be put to real use too:) >> >> Issues so far: >> Some emb

Re: [asterisk-users] Skyp SIP? - what is free for a home *

There is something called Opensky that claims to allow making and receiving skype calls on your SIP device (including Asterisk). However I haven´t tested it yet. http://latestgeeknews.blogspot.com/2009/02/opensky-skype-interface-gateway-who.html 2009/11/1 hbk > Hi, > > I get confused about all

Re: [asterisk-users] Asterisk Fax Module

Just my .02; You shouldn't use outlying features like fax on rc releases - these aren't usually but can be (b)leading edge stuff. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Monday, November 02, 2009

Re: [asterisk-users] MySQL CDR

What version of asterisk are you installing? Thanks, --Warren Selby On Mon, Nov 2, 2009 at 5:59 AM, Dan Journo wrote: > Hello, > > > > Does anyone know where I can get an up to date guide on installing > CDR_MSQL? > > > > VOIP-Info has old information. > > > > Many thanks > > Dan > > > > > > __

Re: [asterisk-users] Asterisk Fax Module

Probably after 1.6.2 has been officially released beyond the release candidate stage. Thanks, --Warren Selby On Mon, Nov 2, 2009 at 4:14 AM, Khaled W Chehab wrote: > When we can expect to have a res_fax and res_fax_degium module for > asterisk V 1.6.2 > > > > Regards > > > > > > *Khaled Chehab

Re: [asterisk-users] SNOM 870

On Fri, 30 Oct 2009, hbk wrote: > Hi, > > I have played with the 820 for some weeks, mostly love it excellent speech > quality. Even got the "mini" browser running > showing my favorite webcam, this could be put to real use too:) > > Issues so far: > Some embarrassing crashes while speaking, wa

Re: [asterisk-users] Remote IP Phone's

Hello Connor, You might be able to start with this link: http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret You can also go even further if you know the IP address of where the phones are coming from by using the permit/deny options: http://www.voip-info.org/wiki/index.php?page=Asterisk

Re: [asterisk-users] supermicro hardware + sangoma

Hi,   Try Berofix / beronet - tested with Tylersburg Supermicro mb's - works perfectly well.   Jacek   -Original Message-From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of marek cervenkaSent: Monday, November 02, 2009 1:38 PMTo: ast

Re: [asterisk-users] Remote IP Phone's

You can restrict on IP address, MAC address and port type and that's just what I know. If someone want's through bad enough you're going to have a problem, but you can at least slow down or stop casual hackers. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-bou

Re: [asterisk-users] Dynamic DNS trunk --- SOLVED

dnsmgr.conf: enable=yes refreshinterval=300 regards. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 30, 2009 3:28 AM To: 'Asterisk Users Mailing List - Non-Com

[asterisk-users] Remote IP Phone's

Hi all, I am wondering what people are doing for security when registering IP phone's remotely if you do not have the equipment to do a VPN tunnel at the remote site. The phone I would be working with mainly is the Polycom lineup. Thanks, Connor Spiess _

Re: [asterisk-users] Async Agi problem

because read=agi lets you read agi events, not send agi actions, agi in write= must be set too if you want to send agi commands. On Mon, Nov 2, 2009 at 3:25 AM, Robert Bielik wrote: > Robert Bielik skrev: > > Ok, now pretty much everything is up 'n running, however when I try to > send an ANSWER

Re: [asterisk-users] Dialstatus

On Mon, 2 Nov 2009, Patrick Plattes wrote: > you can do print the dialstatus to the console e.g.: > exten => s,n,NoOp(${DIALSTATUS}) > > More info: > http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp A "better practice" would be to use verbose() -- an application with greater functionality wr

[asterisk-users] Remote Party ID

Hello, Does anyone know how to set the remote party id? Thanks Dan Journo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.co

[asterisk-users] change L(x[:y][:z]) parameter of DIAL command after call is bridged

Hi, is there any way from outside change x,y an z after a call is bridged? maybe with AMI interface? best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update o

Re: [asterisk-users] IVR

As I understand this thread, you want two different contexts based on the number you dial. If you dial 1703... the big10 context should be executed. If 1567... then Cleveland is executed. Is this correct? If so Then this is what the two lines in [default] should read: > exten => _1703.,1,Goto(bi

Re: [asterisk-users] GSM and Wav format

FWIW, I convert all of my files to WAV for Web reading using SOX. Sox will let you put all your files into the compressed gsm format for storage (sox file.wav file.gsm), then you can just reverse the process for presentation (sox file.gsm /tmp/file.wav) _ From: asterisk-users-boun...@

Re: [asterisk-users] pattern matching DID

On Sun, 2009-11-01 at 18:50 -0500, Thomas Perron wrote: > Where is everyone located? I am in Virginia, USA There are literally thousands of people on this mailing list, so I doubt it's worth having everyone tell you where they're from. That being said, I'm also in Virginia (near Fredericksburg),

Re: [asterisk-users] asterisk 1.6.0 seems to have improper dial status when dialing dahdi extension

svn 6466 from trunk. Tzafrir Cohen wrote: > On Sun, Nov 01, 2009 at 04:13:22PM -0500, cov...@ccs.covici.com wrote: > > Hi. When I dial a Dahdi extension using asterisk 1.6.0, and there is no > > answer, the extension hangs up, but the dial status is busy instead of > > no answer. How do I get

Re: [asterisk-users] Xorcom device not showing up in /proc

On Mon, Nov 02, 2009 at 02:28:36PM +0100, Loic Didelot wrote: > Hello, > I now have this which looks better. > r...@pbx1:~# lsusb > Bus 002 Device 010: ID e4e4:1162 > Bus 002 Device 001: ID : > Bus 001 Device 004: ID 0403:e6c8 Future Technology Devices > Bus 001 Device 003: ID 0403:60

Re: [asterisk-users] Xorcom device not showing up in /proc

Hello, I now have this which looks better. r...@pbx1:~# lsusb Bus 002 Device 010: ID e4e4:1162 Bus 002 Device 001: ID : Bus 001 Device 004: ID 0403:e6c8 Future Technology Devices Bus 001 Device 003: ID 0403:6001 Future Technology Devices Bus 001 Device 001: ID : r...@pbx1:

Re: [asterisk-users] GSM and Wav format

On Mon, Nov 02, 2009 at 05:11:53PM +0500, ABBAS SHAKEEL wrote: > Hello, > > Let me explain a scenario > > There are different Asterisk Servers at different Remote locations. > Recording in different formats for FIVE seconds reveals that ".WAV" (wav49, wav/gsm) should be playable by most systems.

Re: [asterisk-users] Xorcom device not showing up in /proc

On Mon, Nov 02, 2009 at 10:38:45AM +0100, Loic Didelot wrote: > Hi, > can I reset and reload the firmware while the kernel modules are loaded? Yes, there shouldn't be a problem with that. > > Should the files appear in /proc once the firmware has been loaded > correctly or do I need to unload/re

Re: [asterisk-users] GSM and Wav format

After conversion from .wav to .mp3 the size remains almost the same. On Mon, Nov 2, 2009 at 5:46 PM, ABBAS SHAKEEL wrote: > Thanks Patrick. > > First: I dont do that intentionally. > > Thanks for suggestion. Let me investigate it. > > > On Mon, Nov 2, 2009 at 5:34 PM, Patrick Plattes wrote: > >>

Re: [asterisk-users] GSM and Wav format

Thanks Patrick. First: I dont do that intentionally. Thanks for suggestion. Let me investigate it. On Mon, Nov 2, 2009 at 5:34 PM, Patrick Plattes wrote: > Hi, > > at first: why do you use capitals for your name? Don't do that if you > don't have a very good reason. > > You can convert wav to m

[asterisk-users] supermicro hardware + sangoma

hi, i have new supermicro server (centos5, 2.6.30.9/2.6.27.34/2.6.18-distro kernels, wanpipe 3.5.6) card is: 1 . AFT-A101-SH : SLOT=4 : BUS=8 : IRQ=11 : CPU=A : PORT=1 : HWEC=32 : V=36 and i have this in log irq 17: nobody cared (try booting with the "irqpoll" option) Pid: 0, comm: swapper Not

Re: [asterisk-users] GSM and Wav format

Hi, at first: why do you use capitals for your name? Don't do that if you don't have a very good reason. You can convert wav to mp3 on the recording server and then send it to the central system. Bye, Patrick On Mon, Nov 2, 2009 at 1:11 PM, ABBAS SHAKEEL wrote: > > Hello, > > Let me explain a

[asterisk-users] GSM and Wav format

Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to t

[asterisk-users] MySQL CDR

Hello, Does anyone know where I can get an up to date guide on installing CDR_MSQL? VOIP-Info has old information. Many thanks Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list T

Re: [asterisk-users] Dialstatus

Hi, you can do print the dialstatus to the console e.g.: exten => s,n,NoOp(${DIALSTATUS}) More info: http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp Bye, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

Re: [asterisk-users] SNOM 870

- "Garth van Sittert" wrote: | We have the 870 working great in our test environment so far. | | | Garth van Sittert | BSC (Physics & Comp Sci) | Technical Director | BitCo | 08600 24826 | www.bitco.co.za | | | | --[ UxBoD ]-- wrote: | > Anybody tried one with Asterisk yet ? Views ? | >

Re: [asterisk-users] SNOM 870

We have the 870 working great in our test environment so far. Garth van Sittert BSC (Physics & Comp Sci) Technical Director BitCo 08600 24826 www.bitco.co.za --[ UxBoD ]-- wrote: > Anybody tried one with Asterisk yet ? Views ? > > Best Regards, > > > ___

Re: [asterisk-users] hardware requirements for asterisk

On Mon, 2009-11-02 at 09:37 +, aster...@opensourcesolution.in wrote: > hello friends > friend i had just finished my chapters of asterisk. ill be > configuring asterisk in for home for r/d purpose. i am having p4 > machine with 1 GB RAM, ill be configuring asterisk on centos 5.3, the > only d

[asterisk-users] Asterisk Fax Module

When we can expect to have a res_fax and res_fax_degium module for asterisk V 1.6.2 Regards Khaled Chehab NGN Eng. Untitled Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com

Re: [asterisk-users] hardware requirements for asterisk

aster...@opensourcesolution.in wrote: > hello friends > friend i had just finished my chapters of asterisk. ill be configuring > asterisk in for home for r/d purpose. i am having p4 machine with 1 GB > RAM, ill be configuring asterisk on centos 5.3, the only doubt which i > am having is whic

Re: [asterisk-users] Xorcom device not showing up in /proc

Hi, can I reset and reload the firmware while the kernel modules are loaded? Should the files appear in /proc once the firmware has been loaded correctly or do I need to unload/reload the kernel modules? Best regards, Loic. On Mon, 2009-11-02 at 10:38 +0200, Tzafrir Cohen wrote: > On Mon, Nov 0

[asterisk-users] hardware requirements for asterisk

hello friends friend i had just finished my chapters of asterisk. ill be configuring asterisk in for home for r/d purpose. i am having p4 machine with 1 GB RAM, ill be configuring asterisk on centos 5.3, the only doubt which i am having is which hardware ill have to buy to configure asterisk. i

Re: [asterisk-users] Xorcom device not showing up in /proc

Hi, I am using this version zaptel-1.4.12.9.svn.r4653. Best regards, Loïc. On Mon, 2009-11-02 at 10:38 +0200, Tzafrir Cohen wrote: > On Mon, Nov 02, 2009 at 09:33:21AM +0100, Loic Didelot wrote: > > Hello, > > I remember having a similar problem sometime back but I do not remember > > the soluti

Re: [asterisk-users] asterisk 1.6.0 seems to have improper dial status when dialing dahdi extension

On Sun, Nov 01, 2009 at 04:13:22PM -0500, cov...@ccs.covici.com wrote: > Hi. When I dial a Dahdi extension using asterisk 1.6.0, and there is no > answer, the extension hangs up, but the dial status is busy instead of > no answer. How do I get this to work -- do I need to update dahdi? The > car

Re: [asterisk-users] SIP Peers still ping with SIP OPTIONS on a reload

I have the same result with Asterisk 1.4.21 on a Debian Lenny server -- -- -- Marc LEURENT lf...@leurent.eu Le mercredi, 28 octobre 2009 12.27:59, Marc Leurent a écrit : > Hello, when I remove a peer from my sip.conf and just do a reload, the peer > is still ping with SIP OPTIONS until I restart

Re: [asterisk-users] Forward DID to another server

Alex, You forgot to clip the extra > from the quote, shame on you! On Mon, Nov 2, 2009 at 9:47 AM, Alex Balashov wrote: > Tzafrir Cohen wrote: > >> Top-posting, on top of your other sins. >> >> Please spare us this capital punishment. > > An entirely fair point. > > Nevertheless, I eagerly await

Re: [asterisk-users] Forward DID to another server

Tzafrir Cohen wrote: > Top-posting, on top of your other sins. > > Please spare us this capital punishment. An entirely fair point. Nevertheless, I eagerly await your similarly convicted petitions aimed at curbing illiterate, obnoxious and indolent attempts to get others to do extensive work

Re: [asterisk-users] Xorcom device not showing up in /proc

On Mon, Nov 02, 2009 at 09:33:21AM +0100, Loic Didelot wrote: > Hello, > I remember having a similar problem sometime back but I do not remember > the solutions. My Xorcom FXS bank is not showing up in /proc. > > Here some helpful output: > r...@pbx1:~# zaptel_hardware > usb:002/006 xpp_

[asterisk-users] PSTN Line Parameters Checking

I have Asterisk and Digium AEX808B What are please the commands that i can run on Asterisk to get the information about the connected lines from PSTN to see the parameters of them and as well the corresponding files in Asterisk that i can change into, to tune these parameters to be matched t

Re: [asterisk-users] Forward DID to another server

On Mon, Nov 02, 2009 at 02:49:05AM -0500, ALEX BALASHOV wrote: > In a manner of speaking. Top-posting, on top of your other sins. Please spare us this capital punishment. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:t

Re: [asterisk-users] Forward DID to another server

On Mon, Nov 02, 2009 at 12:19:41PM +0530, DHAVAL INDRODIYA wrote: > hello all, > > i have 2 asterisk boxes on that 1 have public IP Address and another is only > have local IP address > > now on public IP there are some 7 DID forwarded , now i want to forward 3 > DID out of 7 DID to local machin

[asterisk-users] Xorcom device not showing up in /proc

Hello, I remember having a similar problem sometime back but I do not remember the solutions. My Xorcom FXS bank is not showing up in /proc. Here some helpful output: r...@pbx1:~# zaptel_hardware usb:002/006 xpp_usb- e4e4:1161 Astribank-modular USB-firmware pci::01:05.0 wct4x

Re: [asterisk-users] Async Agi problem

Robert Bielik skrev: > Ok, now pretty much everything is up 'n running, however when I try to send > an ANSWER (or any) command to *, it replies with > org.asteriskjava.manager.response.ManagerError "Permission Denied". In > manager.conf for the *-java client, I have > > read = > system,call,lo