Re: [asterisk-users] [Asterisk-users] asterisk-1.8.0 compilation error

Hi Paul, Thanks for reply. I have some mistake send compilation logs. i have written cdr_webservice.c module and its work on asterisk-1.6.2.6 version on production server. but i want to upgrade asterisk version. # make [CC] cdr_webservice.c -> cdr_webservice.o In file included from /usr/src/aste

Re: [asterisk-users] [asterisk-ss7] Incoming calls through SS7 for data modem transmissions - possible??

Thank you Horacio and Cary. We will try receiving SS7, routing via SIP, answering on the AS5300, then looping back to itself (out PRI, in PRI ports) in order to invoke the modem termination. This way we may be able to spare the TDM cards in Asterisk and reuse the E1 ports installed in the gateway.

Re: [asterisk-users] Spam

Same here. But, can the genie ever be put back in the bottle? > Cary Fitch wrote: >> Has anyone else noticed "new spam" in the last 2-3 weeks? >> > > No, > > But I run ASSP in front of my MTA. > > Doug -- _ -- Bandwidth and Colo

Re: [asterisk-users] Spam

Cary Fitch wrote: > Has anyone else noticed "new spam" in the last 2-3 weeks? > No, But I run ASSP in front of my MTA. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." --

Re: [asterisk-users] Incoming calls through SS7 for datamodemtransmissions - possible??

Thanks Cary, The first topology we are working on should be the best way then. Asterisk will answer SS7 calls, route them to the ISDN channels to be terminated by the AS5300 as they were doing before. I think TDM-2-TDM shouldn't be that much of a problem eh? No further equipment needed? *José P

[asterisk-users] Spam

I have been pounded with new, mostly text spam in the last few weeks. Tonight I realized that the address that is being spammed is a personal one I use for this list. Has anyone else noticed "new spam" in the last 2-3 weeks? Cary Fitch -- ___

Re: [asterisk-users] DAHDI phantom pickup when ringing

Jonathan Hunter wrote: > On 24 November 2010 01:20, Lyle Giese > wrote: > > Post the revelent portions of your extension.conf. Maybe you have > a logic error somewhere. > > Thanks Lyle. > > My extensions.conf is fairly simple in this regard; I use macro-stdext

Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??

Thanks Cary, What happens is, the Telco won't allow the small company to resell the ISDN connections, meaning, they bought the trunks and DIDs, then sold dialing plans to route incoming calls through the PRIs out the Internet. This is not the issue though. We definitely have to migrate to an SS7 c

Re: [asterisk-users] Why doesn't Asterisk project document certain important features of Asterisk officially?

On Wed, Nov 24, 2010 at 8:06 PM, Paul Belanger wrote: > On 10-11-24 08:34 PM, Sherwood McGowan wrote: >> True, but then some of us registered on that site and still don't have >> the ability to edit...I thought it was a community effort? Maybe I was >> wrong >> > Once registered you will be able t

Re: [asterisk-users] Why doesn't Asterisk project document certain important features of Asterisk officially?

On 10-11-24 08:34 PM, Sherwood McGowan wrote: > True, but then some of us registered on that site and still don't have > the ability to edit...I thought it was a community effort? Maybe I was > wrong > Once registered you will be able to post comments, not edit. If you would like to become part o

Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??

I am not sure where you are and what legal conventions are involved. Are you saying the Telco (and legal restrictions) say you can’t send calls to the internet via the AS5300 but you can if Asterisk does it directly? What is the “logic” in that? Or are they saying your Telco to Asterisk tru

Re: [asterisk-users] Why doesn't Asterisk project document certain important features of Asterisk officially?

On Wed, Nov 24, 2010 at 7:20 PM, Warren Selby wrote: > On Wed, Nov 24, 2010 at 11:30 AM, Tilghman Lesher > wrote: >> >> On Wednesday 24 November 2010 11:07:40 Bruce B wrote: >> > This is not to bash the Asterisk project or Digium. Don't respond if you >> > have a difference of opinion as I am not

[asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted th

Re: [asterisk-users] Why doesn't Asterisk project document certain important features of Asterisk officially?

On Wed, Nov 24, 2010 at 11:30 AM, Tilghman Lesher wrote: > On Wednesday 24 November 2010 11:07:40 Bruce B wrote: > > This is not to bash the Asterisk project or Digium. Don't respond if you > > have a difference of opinion as I am not looking for personal opinions > > but rather JUST WONDERING THE

Re: [asterisk-users] DAHDI phantom pickup when ringing

Tzafrir, On 24 November 2010 18:12, Tzafrir Cohen wrote: > > Can you replicate those phantom answers without calling all channels? > > Try: > > originate DAHDI/7 application Echo > > Does that line answer without you picking up the phone? Or does it > require a combination of several channels r

Re: [asterisk-users] action at registering or de-registering

On Wed, Nov 24, 2010 at 4:24 PM, Hans Witvliet wrote: > On Wed, 2010-11-24 at 15:47 -0600, Sherwood McGowan wrote: >> On Wed, Nov 24, 2010 at 3:08 PM, Hans Witvliet wrote: >> > On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote: >> >> On Asterisk 1.8 when a SIP peer resgisters or unregisters i

Re: [asterisk-users] action at registering or de-registering

On Wed, 2010-11-24 at 15:47 -0600, Sherwood McGowan wrote: > On Wed, Nov 24, 2010 at 3:08 PM, Hans Witvliet wrote: > > On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote: > >> On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates > >> a PeerStatus event. I don't know if this is

Re: [asterisk-users] Avoided deadlock Error

Othe problem is small time my hdd is full of recording -- Sent from my BlackBerry® VoIP, Windows/Linux Administration and Network Management US Numbers: 561-886-0664 Nicaragua Mobile: +505.8488.6876 -Original Message- From: Stefan Schmidt Sender: asterisk-users-boun...@lists.digium.com D

Re: [asterisk-users] Avoided deadlock Error

Am 24.11.2010 13:48, schrieb Bayardo Sanchez: > My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem > is : > > Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided > deadlock for '0x861f6d8', 9 retries! > Nov 24 06:45:01 WARNING[3335]: channel.c:780 ch

Re: [asterisk-users] action at registering or de-registering

On Wed, Nov 24, 2010 at 3:08 PM, Hans Witvliet wrote: > On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote: >> On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates >> a PeerStatus event. I don't know if this is in 1.4/1.6 as well, but >> should be easy enough to test. >> >> He

Re: [asterisk-users] Audiocodes firmware

On 10/14/10 15:38, Bryant Zimmerman wrote: >For which device models? > > > From: "Mark Murawski" >Sent: Thursday, October 14, 2010 3:26 PM >To: asterisk-users@lists.digium.com >Subject: [asterisk-users] Audiocodes firmware > >Does anyone have links to the mo

Re: [asterisk-users] Someone has hacked into our system

On 11/24/10 10:39, Gary Kuznitz wrote: >> Look for "allowguest" default is "yes" >> I change it to allowguest=no >> In addition you might want to restrict some countries in your dial-plan, >> here is my list: > >This would be great. Can I put this anyplace in extensions.conf? >Or does it need t

Re: [asterisk-users] action at registering or de-registering

On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote: > On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates > a PeerStatus event. I don't know if this is in 1.4/1.6 as well, but > should be easy enough to test. > > Here is an example of what I see on the manager interface durin

Re: [asterisk-users] Someone has hacked into our system

Thank you for the reply. On 23 Nov 2010 at 18:51, John (John Novack ) commented about Re: [asterisk-users] Someone has hacked into our : > > > Gary Kuznitz wrote: > > Thank you for the reply... > > > > Comments below... > > On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher > us...@lists.digiu

Re: [asterisk-users] Recommended *WRT router to run Asterisk?

On Fri, 19 Nov 2010 10:15:40 -0500, jon pounder wrote: >What is nice is when the $50 hardware and the $1000 hardware run exactly >the same software so other than the drivers for the hardware itself, >everything else behaves the same way and its easy to move around >configurations to grow. (I am

Re: [asterisk-users] Someone has hacked into our system

On 23 Nov 2010 at 16:54, Joseph (Joseph ) commented about Re: [asterisk-users] Someone has hacked into our : > On 11/23/10 14:18, Gary Kuznitz wrote: > >Thank you for the reply... > > > >Comments below... > >On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher >us...@lists.digium.com>) commente

Re: [asterisk-users] DAHDI phantom pickup when ringing

On Sun, Nov 21, 2010 at 11:13:00PM +, Jonathan Hunter wrote: > Hi, > > I've been experiencing trouble with my DAHDI channels for some time and have > finally decided to try and resolve the issue. > > Essentially, the problem I am having is that when a call comes in, and my > DAHDI phones ther

[asterisk-users] Disable connected line updates for dahdi PRI channel

Hi, Starting in Asterisk 1.8.0, Asterisk supports connected line updates. This is fantastic for SIP. How can I prevent them from being sent to a PRI channel? I'm having problems when a call is answered by an internal SIP extension, then transferred (blind or attended) to another internal SIP

Re: [asterisk-users] kernel: dahdi: Detected time shift.

I know there was a patch for dahdi to fix server lockups on time shift. (not sure what version, but if you changed the time, the server would just go crash.) Do you have the latest version ? Check your ntpd settings to make sure your time isn't bouncing all over the place. William

Re: [asterisk-users] kernel: dahdi: Detected time shift.

On 11/24/2010 11:02 AM, Shaun Ruffell wrote: > On 11/24/2010 10:43 AM, Jonas Kellens wrote: >> The only thing I have is a high level of mentionning of "kernel: dahdi: >> Detected time shift." in /var/log/messages. >> >> >> What is causing this kernel message ? Could this be the cause of the >> serv

Re: [asterisk-users] kernel: dahdi: Detected time shift.

On 11/24/2010 10:43 AM, Jonas Kellens wrote: > Hello list, > > I'm experiencing a lot of server freezes lately. The server just... freezes. > > I notice in the log files (/var/log/asterisk/messages & > /var/log/messages) that logging stops at the time the server hangs. > Logging continues when th

Re: [asterisk-users] kernel: dahdi: Detected time shift.

On Wed, Nov 24, 2010 at 11:43 AM, Jonas Kellens wrote: > Hello list, > > I'm experiencing a lot of server freezes lately. The server just... freezes. > > I notice in the log files (/var/log/asterisk/messages & /var/log/messages) > that logging stops at the time the server hangs. Logging continues

[asterisk-users] Why doesn't Asterisk project document certain important features of Asterisk officially?

Hi Everyone, I am wondering why documentation of some of the vital parts of Asterisk is hosted on voipinfo.org (unreliable is some parts) and not on asterisk.org? For example the list of AMI events are not well documented and one has to guess which version supports which event. The documentation f

[asterisk-users] IPv6: What You Need to Know Now

Yes, the thanksgiving holiday is here (in the USA)! But also, the fear of running out of IP addresses next year has raised its ugly head and since we don't do Thanksgiving in Europe, we have some serious talking to do about this problem. This Friday at 12 Noon EST, Olle Johansson will be joining u

[asterisk-users] kernel: dahdi: Detected time shift.

Hello list, I'm experiencing a lot of server freezes lately. The server just... freezes. I notice in the log files (/var/log/asterisk/messages & /var/log/messages) that logging stops at the time the server hangs. Logging continues when the server has been restarted (which is the only solution

Re: [asterisk-users] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk

2010/11/20 Olivier > > Depending on what telco Charlie is connected to would change the CallerId > presented to Charlie from being Alice's or Bob's Cid. > > When a call is forwarded, Charlie's telco receives different ANI and CID : > some (seems to) favor ANI and some CID. > > An interesting thin

Re: [asterisk-users] [Asterisk-users] asterisk-1.8.0 compilation error

On 10-11-24 06:09 AM, RAJNIKANT VANZA wrote: > make[1]: *** [cdr_webservice.o] Error 1 > make: *** [cdr] Error 2 > What is cdr_webservice.o ? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.or

[asterisk-users] Originate Response.

Hi to all. I am conducting several tests with the Asterisk manager and I noticed something that I believe to be a problem. When I generate a call with the Action Originate with the Async option true, the event OriginateResponse returns normally. But when I generate a call in the same way, without

Re: [asterisk-users] SPA942 on speaker phone does not hang up?

It is the phone itself: go to Regional tab and scroll down to Reorder Delay and make it 255. That tells it not to play re-order tone and just hangup. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Wednesday, Nove

Re: [asterisk-users] action at registering or de-registering

On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates a PeerStatus event. I don't know if this is in 1.4/1.6 as well, but should be easy enough to test. Here is an example of what I see on the manager interface during a register/unregister: Event: PeerStatus Privilege: system,all

[asterisk-users] Avoided deadlock Error

My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem is : Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x861f6d8', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85a6420', 9 ret

[asterisk-users] DTMF CallerID

Hi, Does anyone know if CID is already working with Digium TDM800P card using DTMF signalling? (I'm brazillian) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live intr

Re: [asterisk-users] Asterisk 1.6 and Music on Hold

On Wed, Nov 24, 2010 at 5:35 AM, Olivier CALVANO wrote: > 2010/11/24 Sherwood McGowan : >> On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO wrote: >>> Hi >>> >>> i have a small problems on Asterisk 1.6 with the MusiconOld : >>> >>> musiconhold.conf: >>> >>> [Sound_1] >>> mode=quietmp3 >>> directo

Re: [asterisk-users] SPA942 on speaker phone does not hang up?

On Wed, Nov 24, 2010 at 5:16 AM, Cassius Smith wrote: > Hello all, > I am using Linksys SPA942 in my current installation activity. I see a > peculiar behavior: A call is made and the SPA942 uses its speaker. When the > far end of a call hangs up , the SPA942 stays off hook, and after a time > pla

Re: [asterisk-users] Asterisk 1.6 and Music on Hold

2010/11/24 Sherwood McGowan : > On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO wrote: >> Hi >> >> i have a small problems on Asterisk 1.6 with the MusiconOld : >> >> musiconhold.conf: >> >> [Sound_1] >> mode=quietmp3 >> directory=/var/lib/asterisk/moh/Sound_1 >> >> in extensions.conf: >> >> exte

[asterisk-users] TDM calls fall after some minutes

Hello, it's my first post on this list, I hope not to bore youwith my novice questions.. We're using a TDM400 with 3 fxo modules connected to pstn. Call goes inbound/outbound correctly, after playing a bit on some dahdi-channels.conf/chan_dahdi.conf options. The "big" problem is that after

[asterisk-users] SPA942 on speaker phone does not hang up?

Hello all, I am using Linksys SPA942 in my current installation activity. I see a peculiar behavior: A call is made and the SPA942 uses its speaker. When the far end of a call hangs up , the SPA942 stays off hook, and after a time plays a fast busy. The user then has to press the line presence butt

[asterisk-users] [Asterisk-users] asterisk-1.8.0 compilation error

Hi all, I want to upgared from asterisk-1.6.2.6 version to asterisk-1.8.0 version. When i execute "make" command for compilation i have seen below errors. In file included from /usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/cdr.h:31 /usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/

Re: [asterisk-users] Asterisk 1.6 and Music on Hold

On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO wrote: > Hi > > i have a small problems on Asterisk 1.6 with the MusiconOld : > > musiconhold.conf: > > [Sound_1] > mode=quietmp3 > directory=/var/lib/asterisk/moh/Sound_1 > > in extensions.conf: > > exten => 0532xx,1,Answer > exten => 0532xx,2,Musi

Re: [asterisk-users] Contradiction in GROUP() function

On 24 November 2010 10:12, Steve Davies wrote: > I am confused. In Asterisk 1.2 and 1.4, in the code there is an error: > "Setting a group requires an argument (group name)" > > But the syntax is shown as: "Syntax: GROUP([category])" > > The [category] square brackets indicate to me an "optional"

[asterisk-users] Asterisk 1.6 and Music on Hold

Hi i have a small problems on Asterisk 1.6 with the MusiconOld : musiconhold.conf: [Sound_1] mode=quietmp3 directory=/var/lib/asterisk/moh/Sound_1 in extensions.conf: exten => 0532xx,1,Answer exten => 0532xx,2,MusicOnHold(Sound_1) exten => 0532xx,3,Dial(SIP/ACCOUNT001,180,t) exten => 0532xx,4,

[asterisk-users] Contradiction in GROUP() function

I am confused. In Asterisk 1.2 and 1.4, in the code there is an error: "Setting a group requires an argument (group name)" But the syntax is shown as: "Syntax: GROUP([category])" The [category] square brackets indicate to me an "optional" parameter, which contradicts the error. Verison 1.6 is no

Re: [asterisk-users] astcanary ?

On 11/24/2010 10:28 AM, --[ UxBoD ]-- wrote: Hello, I notice that the following proces is running : astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 1527 What is this ?? Kind rega

Re: [asterisk-users] action at registering or de-registering

On Wed, Nov 24, 2010 at 1:20 AM, Hans Witvliet wrote: > Hi all, > > Perhaps someone has dealt with it before. > > I want to activate a bunch of my own scripts after someone has registred > om my asterisk, or when his cient has de-registerded. > > have been skimming through AGI and AMI, and seen a

Re: [asterisk-users] Asterisk 1.6 and Username in Dial

No you can't On Wed, Nov 24, 2010 at 2:34 AM, Olivier CALVANO wrote: > Hi > > i don't see a answer at my question > > Bye > Jerome > > > > > > 2010/11/9 Olivier CALVANO : >> Hi >> >> In Asterisk 1.6/realtime Mysql, we can't put a username/password in a >> Dial Command ?: >> >> 'Dial', 'SIP/Userna

Re: [asterisk-users] astcanary ?

- Original Message - Hello, I notice that the following proces is running : astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 1527 What is this ?? Kind regards, Jonas. You are running Asterisk with priority set. Check /etc/asterisk/asterisk.conf for the line

Re: [asterisk-users] usage of account code in CDR

We use it to determine who is the caller. Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com Find us on Facebook -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digi

Re: [asterisk-users] Asterisk 1.6 and Username in Dial

Hi i don't see a answer at my question Bye Jerome 2010/11/9 Olivier CALVANO : > Hi > > In Asterisk 1.6/realtime Mysql, we can't put a username/password in a > Dial Command ?: > > 'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r' > > Thanks > Olivier > -- ___

Re: [asterisk-users] DAHDI phantom pickup when ringing

On 24 November 2010 01:20, Lyle Giese wrote: > Post the revelent portions of your extension.conf. Maybe you have a logic > error somewhere. > > Thanks Lyle. My extensions.conf is fairly simple in this regard; I use macro-stdexten: [macro-stdexten]; exten => s,1,NoOp('${CALLERID(NAME)}' [${CAL

[asterisk-users] astcanary ?

Hello, I notice that the following proces is running : astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 1527 What is this ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-