Re: [asterisk-users] Question on resources

Hi, Am Donnerstag, dem 04.08.2022 um 20:32 -0400 schrieb Jerry Geis: > I am running Asterisk 13.30.0 > 40 core CPU (VM) VMware. > CentOS 7 > 32 G ram > 10G vmx network > > Should be plenty of room for anything... > > Yes asterisk is running 270% CPU... > Is it not taking advantage of the 40 core

Re: [asterisk-users] Pickup with pjsip not working

sts.digium.com> > Subject: Re: [asterisk-users] Pickup with pjsip not working > > On Tue, Mar 1, 2022 at 7:16 AM Karsten Wemheuer wrote: > > Am Dienstag, dem 01.03.2022 um 06:37 -0400 schrieb Joshua C. Colp: > > > On Tue, Mar 1, 2022 at 6:14 AM Karsten Wemheuer >

Re: [asterisk-users] Pickup with pjsip not working

Am Dienstag, dem 01.03.2022 um 06:37 -0400 schrieb Joshua C. Colp: > On Tue, Mar 1, 2022 at 6:14 AM Karsten Wemheuer wrote: > > Hi *, > > > > i am currently trying to migrate from chan_sip to pjsip. I am using > > Asterisk version 18.10. > > > > In chan_si

[asterisk-users] Pickup with pjsip not working

Hi *, i am currently trying to migrate from chan_sip to pjsip. I am using Asterisk version 18.10. In chan_sip information about the pickup was sent in the XML body of the NOTIFY requests: /--- \--- If I use pjsip, the pickup information is missing: /--- \--- Many phones

Re: [asterisk-users] Voice "broken" during calls

Hi Luca, Am Samstag, den 13.06.2020, 08:28 +0200 schrieb Luca Bertoncello: > Hi! > > I have a Asterisk installation to manage my phones at home (provider > is > Deutsche Telekom). > It works, but very often the voice is "broken"... > Yesterday during a call it was very difficult to understand what

Re: [asterisk-users] SIP TLS not working, Asterisk 16.9.0

Hi Stefan, thanks a lot. It is working now. Best regards, Karsten Am Freitag, den 01.05.2020, 18:40 +0200 schrieb Stefan Tichy: > Hi Karsten, > > > On Thu, Apr 30, 2020 at 05:50:39PM +0200, Karsten Wemheuer wrote: > > > > The server sends Server Hello,

[asterisk-users] SIP TLS not working, Asterisk 16.9.0

Hi, I have problems with SIP via TLS. Asterisk works as a client. The TCP connection is established, followed by a client hello from Asterisk to the server. The server sends Server Hello, Certificate, Server Key Exchange and Server Hello Done. Than Asterisk sends back a Alert (Level: Fatal, Descr

Re: [asterisk-users] internal call record

Hi, Am Sonntag, den 10.03.2019, 12:46 +0300 schrieb Gokan Atmaca: > Hello > > Mynum: 6001 , Othernum: 6002. > > > I can record as follows. But I do not enter individual records for > each internal > required. I want to do it more smoothly with a Macro. > > Thanks. > > exten => _6001,1,NoOp() > ext

[asterisk-users] Missing audio on playback in 16.0

Hi, I am currently evaluating asterisk 16. I have noticed an issue using application playback. The beginning and the end of the audio file are missing. If I use answer and wait(1) before playback, the beginning is correct. I am using chan_sip, if this is of interest. Best regards Karsten -- ___

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

Hi Luca, Am Montag, den 21.12.2015, 18:52 +0100 schrieb Luca Bertoncello: > Hi list! > > My Problem: all calls to international numbers will be dropped after exactly > 15 minutes... > I have a VoIP-account by Deutsche Telekom. > This is what I see when I call someone (my parents) and the connect

Re: [asterisk-users] Semi OT - LDAP multi-valued attributes support in SIP phones

Hi Olivier, Am Donnerstag, den 29.01.2015, 18:07 +0100 schrieb Olivier: > Hello, > > I've just started to look at LDAP in IP telephony. > > 1. I've read parts of RFC2798 which defines inetOrgPerson class. > I could find homePhone or telephoneNumber (multi-valued) attributes > but nothing like p

Re: [asterisk-users] AMI Redirect both calls from a bridge

Hi Neil, Am Mittwoch, den 17.12.2014, 09:08 -0500 schrieb Neil Cherry: > Doe anybody know of a way to redirect both channels from a bridge to > different dial plan extensions from the using the AMI. > > Currently, as soon as I redirect one of the channels the other appears > to be dropped and

Re: [asterisk-users] Asterisk sends CANCEL to the wrong destination

Hi, Am Dienstag, den 16.12.2014, 16:32 +0100 schrieb Karsten Wemheuer: > Hi, > > I got a weird behaviour in asterisk (original found in 1.8 but it is > still the same in 11.15.0). I have three phones communicating via > OpenSIPs with asterisk. Phone A dials 100 and asterisk call

[asterisk-users] Asterisk sends CANCEL to the wrong destination

Hi, I got a weird behaviour in asterisk (original found in 1.8 but it is still the same in 11.15.0). I have three phones communicating via OpenSIPs with asterisk. Phone A dials 100 and asterisk calls SIP/phone-b. Phone B accepts the call. The User on Phone B places the call on hold, dials 200 and,

Re: [asterisk-users] IAXModem or T38Modem?

Hi Mike, Am Montag, den 24.03.2014, 01:41 -0400 schrieb Mike Diehl: > Hi all, > I'm installing Hylafax on my Asterisk system. From what I've read, I > can either use IAXModem or T38Modem to provide the virtual fax device. > So at the risk of starting a religious war, which one should I use? > I d

Re: [asterisk-users] How to configure asterisk to only accept SIP from kamailio@localhost but exchange RTP on all interfaces?

Hi Alex, Am Dienstag, den 25.02.2014, 13:04 -0500 schrieb Alex Villací­s Lasso: > El 25/02/14 08:30, Karsten Wemheuer escribió: > > Hi Alex, > > > > Am Donnerstag, den 20.02.2014, 13:48 -0500 schrieb Alex Villací­s Lasso: > >> I have a setup with asterisk-11.7.0 and

Re: [asterisk-users] How to configure asterisk to only accept SIP from kamailio@localhost but exchange RTP on all interfaces?

Hi Alex, Am Donnerstag, den 20.02.2014, 13:48 -0500 schrieb Alex Villací­s Lasso: > I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following > the setup guide at > http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . > I want to run asterisk and kamailio on

Re: [asterisk-users] sipgate outgoing calls

Hi, Am Mittwoch, den 18.09.2013, 14:29 +0100 schrieb gpxctawjc...@irational.org: > Hello > > i am trying to setup sipgate gateway > > i can get incoming calls fine, but when i dial in and then try to dial > out i get this in asterisk command line What Sipgate product are You using? At least in

Re: [asterisk-users] Error 488 Not Acceptable Here

Hi, Am Donnerstag, den 23.05.2013, 20:48 +0200 schrieb Maximilian Grobecker: > Am 22.05.2013 16:39, schrieb Andrew Colin: > > Hi guys, > > > > Any idea why I am getting this error when someone tries to send me a T38 > > Fax? > Hi, > > Maybe you have not allowed T.38 as acceptable codec ;-) > You

Re: [asterisk-users] Asterisk and hylafax: how to debug ...

Hi, Am Dienstag, den 07.05.2013, 21:48 +0200 schrieb Sebastian Niehaus: > Am 07.05.2013 18:23, schrieb Sebastian Niehaus: > > > For some reason, t38modem tells hylafax the line is BUSY so there is no > > fax send. > > Well, I may add the log of t38modem (sorry for the ugly formatting) > Parts I

Re: [asterisk-users] CallerID external call after Attended Transfer

Hi, Am Montag, den 04.02.2013, 14:45 +0100 schrieb Jonas Kellens: > Hello, > thanks you for your answer. > The IP-phones in this case are Yealink T32G. > What setting is needed in this IP-phone ? as Kevin already written, set this in asterisk: sendrpid=pai trustrpid=yes I don't

Re: [asterisk-users] Voicemail not working with vm boxes named with a star

Hi Matthew, Am Donnerstag, den 20.09.2012, 06:27 -0500 schrieb Matthew Jordan: > - Original Message - > > From: "Karsten Wemheuer" > > To: asterisk-users@lists.digium.com > > Sent: Thursday, September 20, 2012 2:28:07 AM > > Subject: [asterisk-users

Re: [asterisk-users] Voicemail not working with vm boxes named with a star

Hi list, Am Donnerstag, den 20.09.2012, 09:28 +0200 schrieb Karsten Wemheuer: > Hi list, > > in asterisk 1.4 and maybe earlier it was possible to use voicemail > system with mailboxes starting with some special characters like *. The > line in voicemail.conf was like this: >

[asterisk-users] Voicemail not working with vm boxes named with a star

Hi list, in asterisk 1.4 and maybe earlier it was possible to use voicemail system with mailboxes starting with some special characters like *. The line in voicemail.conf was like this: *123 => , AB,,,tz=cet|attach=no| Calling exten => s,n,Voicemail(*123,su) is working in asterisk 1.4. I

Re: [asterisk-users] Asterisk 1.8.10

Hii Am Montag, den 11.06.2012, 16:12 -0700 schrieb motty.cruz: > Hello, > How to change ring tone on interncal call? I'm using Centos 5.8 Asterisk 1.8 > > exten => > 666,1,SIPAddHeader("Alert-Info:") > exten => 666,n,Dial(SIP/10) > > The above would not how t

Re: [asterisk-users] Asterisk 1.8 Transfer CallerID

Hi, Am Dienstag, den 08.05.2012, 14:13 +0200 schrieb Jonas Kellens: > Hello, > > when a call comes in and is answered by colleague A, this colleague A > sees the CallerID of the external calling number. > > When colleague A transfers the call to colleague B, attended or > unattended, then collea

Re: [asterisk-users] DAHDISendCallreroutingFacility

Hi, Am Samstag, den 10.03.2012, 08:42 -0800 schrieb Mehdi Shirazi: > Hi > I installed Asterisk 1.8.7 with CD ISO(Elastix 2.2) > I want to use DAHDISendCallreroutingFacility Application on a PRI link(LIBPRI > Already installed). > according to > https://wiki.asterisk.org/wiki/display/AST/New+in+1.

Re: [asterisk-users] Problem while sending SIP NOTIFY via AMI in 1.8.10-rc2

Hi, a little extension to my previous post: The phone sends "200 OK" for the NOTIFY via proxy to asterisk, but asterisk seems to ignore this. About 500 ms later, the NOTIFY is repeated by asterisk. This continues up to the final timeout (with the typical log message). Karsten -- __

[asterisk-users] Problem while sending SIP NOTIFY via AMI in 1.8.10-rc2

Hi, while testing asterisk 1.8.10-rc2 I stumbled across a weird behavior. I want to notify a snom phone to reload its configuration. For this to happen, I use the NOTIFY mechanism. I started the notify via AMI command. Asterisk is bound to udp 25060, because all phones are registered with a local

Re: [asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2

Hi, Am Dienstag, den 14.02.2012, 11:32 -0600 schrieb Kevin P. Fleming: > This does appear to be a bug in Asterisk; please open an issue in JIRA, > and post the issue number here, so we can get someone looking at this > ASAP. Thanks! > Done, issue ASTERISK-19358. If I can do anything to test so

Re: [asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2

Hi Kevin, Am Dienstag, den 14.02.2012, 09:46 -0600 schrieb Kevin P. Fleming: > On 02/14/2012 09:30 AM, Karsten Wemheuer wrote: > > Hi, > > > > I got a problem with asterisk 1.8.9.2. The same scenario is working fine > > in 1.8.8.2. > > > > Asterisk calls a

[asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2

Hi, I got a problem with asterisk 1.8.9.2. The same scenario is working fine in 1.8.8.2. Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on the same LAN, no NAT. Asterisk sends the INVITE to the proxy, the proxy sends INVITE to the phone. The phone sends 180 RINGING back to

Re: [asterisk-users] Analoge and E1 ports

Hi Bilal, Am Sonntag, den 22.01.2012, 13:06 -0800 schrieb bilal ghayyad: > Hi All; > > Is there a telephony card that contains analoge ports and E1s at the same > time? Beronet in Germany produces modular media gateways as cards to plug in a pc (PCI or PCI express) or as an external box. Each g

Re: [asterisk-users] Option 'd' of application Dial not working in 1.8.8-rc2

Hi Richard, Am Mittwoch, den 02.11.2011, 09:26 -0500 schrieb Richard Mudgett: > > Hi, > > > > in asterisk 1.8.7.0 option 'd' works as expected: Pressing a key while > > in ringing state puts the call to an one digit extension. > > > > In asterisk 1.8.8-rc2 this is not working anymore. After doin

[asterisk-users] Option 'd' of application Dial not working in 1.8.8-rc2

Hi, in asterisk 1.8.7.0 option 'd' works as expected: Pressing a key while in ringing state puts the call to an one digit extension. In asterisk 1.8.8-rc2 this is not working anymore. After doing a diff on the code it seems to me, that in version 1.8.7 there is an autoanswer in application dial

Re: [asterisk-users] Problem with video phone call, error in sdp media handling?

WAG - if you start the call in voice-mode, the video codecs aren't > loaded. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karsten > Wemheuer > Sent: Wednesday, October 19,

[asterisk-users] Problem with video phone call, error in sdp media handling?

Hi, I try to setup a video call and I sometimes get no video. I set up a Yealink VP 2009 and a Ninja Softphone. Both a in the same LAN. Asterisk release is 1.8.7.0. Call from Yealink to the Ninja is working fine, if I start the call in video mode. In this case I can switch between voice-only and

Re: [asterisk-users] T.38 passthru on 1.8.5

Hi, Am Dienstag, den 30.08.2011, 09:44 -0400 schrieb Fabian Borot: > > Hello > We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk > 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running > Linux on 2011-08-26 21:31:22 UTC] > > The call flow is: > quintum gateway -->

Re: [asterisk-users] 1.8.5 Voicemail duration incorrect

Hi Robert, Am Donnerstag, den 25.08.2011, 13:28 -0400 schrieb Robert Huddleston: > https://issues.asterisk.org/jira/browse/ASTERISK-16981 Thank You for the link. I already found it a few hours later. I put some debug output in the code and I think I found the location of the issue, but I currentl

Re: [asterisk-users] 1.8.5 Voicemail duration incorrect

Hi, Am Mittwoch, den 24.08.2011, 13:18 -0400 schrieb Robert Huddleston: > Anyone else seen this? > > > > I saw a jira but was in feedback status.. I just checked with a voicemail of 60 seconds. It was reported in .txt-file with a duration of 19 seconds. So there is a bug. Do You have a link t

Re: [asterisk-users] Trouble with *8 Pickup

Hi, Am Montag, den 15.08.2011, 10:18 +1200 schrieb Alec Davis: > > > > If you time the *8 just right so it is being handled during > > the end of > > > the Dial then I got: > > > > > > [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: > > user_data > > > is NULL [Aug 11 16:26:18] ER

Re: [asterisk-users] Voicemail issue

Hi, it seems to be fixed in 1.8.4. At least I can't reproduce it there. Karsten Am Mittwoch, den 15.06.2011, 09:29 -0400 schrieb Mike: > The same issue was present in 1.6 a few weeks ago and is fixed in > latest 1.6. Maybe latest 1.8.4 does not have this issue. > > > > Mike > > > > From:

Re: [asterisk-users] how to know length of file in seconds

Hi, Am Dienstag, den 07.06.2011, 17:07 -0400 schrieb Paul Belanger: > On 11-06-07 02:31 AM, virendra bhati wrote: > > Hi List, > > > > Is there any way by which we can get the length of any recorded files into > > seconds ? > > > > $ sox foo.wav -e stat just a remark for people using newer(?)/ot

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

Hi randall, Am Mittwoch, den 01.06.2011, 10:00 +0200 schrieb randall: > > > i get the following errors: > > > pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel > > > of span 2 > > > > > > > > > Your telco provider has crc on or off , that is

Re: [asterisk-users] function Echo() doesn't work

Hi Felix, Am Mittwoch, den 16.02.2011, 12:47 +0100 schrieb Felix Dong: > Hi guys, > > > the function Echo() did work on CAPI, but doesn't work for SIP > connection. Can anybody help? > thanks a lot. are You trying to echo between local phones or is it a external call via some VoIP Provider? I

Re: [asterisk-users] Issues with 1.8 and BlindTransfer

Hi, Am Donnerstag, den 02.12.2010, 11:02 -0500 schrieb Bryant Zimmerman: > Replys from Bryant > > On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman > wrote: > >> I am having issues with Blind Transfer on asterisk 1.8 > > >What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS? > > Verison

Re: [asterisk-users] Asterisk 1.8.0-rc5: Blind transfer failed, SIP REFER Method

Am Donnerstag, den 21.10.2010, 16:27 +0200 schrieb Karsten Wemheuer: > Hi, > > I setup an asterisk system (version 1.8.0-rc5). While using a SIP only > environment I discovered a problem using blind transfer. The phones are > SNOM or Aastra and are using the SIP REFER Method. >

[asterisk-users] Asterisk 1.8.0-rc5: Blind transfer failed, SIP REFER Method

Hi, I setup an asterisk system (version 1.8.0-rc5). While using a SIP only environment I discovered a problem using blind transfer. The phones are SNOM or Aastra and are using the SIP REFER Method. The following is working: User A calls user B, B accepts the call, user A than transfers to user C

Re: [asterisk-users] dahdi_genconf

Hi, Am Mittwoch, den 20.10.2010, 01:54 -0200 schrieb Flavio Miranda: > > > Att, > > Flavio Roberto Miranda > MSN:flaviormira...@hotmail.com > Skype: flaviormiranda > > Just one more question, what it means the "RED" under alarms when I > type dahdi show status. It should be "OK"? the "RED"-a

Re: [asterisk-users] Kernel panic (asterisk 1.8.0-rc3, dahdi-linux-2.4)

Am Samstag, den 16.10.2010, 14:00 -0500 schrieb Shaun Ruffell: > On 10/16/10 12:47 PM, Karsten Wemheuer wrote: > > Hi, > > > > Am Freitag, den 15.10.2010, 14:34 -0500 schrieb Shaun Ruffell: > >> On 10/15/2010 04:00 AM, Karsten Wemheuer wrote: > >> > >&g

Re: [asterisk-users] Kernel panic (asterisk 1.8.0-rc3, dahdi-linux-2.4)

Hi, Am Freitag, den 15.10.2010, 14:34 -0500 schrieb Shaun Ruffell: > On 10/15/2010 04:00 AM, Karsten Wemheuer wrote: > > > I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with > > dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an > > olde

[asterisk-users] Kernel panic (asterisk 1.8.0-rc3, dahdi-linux-2.4)

Hi, I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN card. The system starts without any errors. I discovered a severe issue. The kernel pan

Re: [asterisk-users] Asterisk 1.8: Warning messages in CLI while putting a SIP-Call on hold

Am Mittwoch, den 06.10.2010, 15:11 +0200 schrieb Karsten Wemheuer: > Hi, > > while testing current release candidate 1.8.0-rc2 I stumbled on a weird > behavior. I did not find any hints in the archives or at the bug > tracker. > > Two SIP-Clients are connected (both on

[asterisk-users] Asterisk 1.8: Warning messages in CLI while putting a SIP-Call on hold

Hi, while testing current release candidate 1.8.0-rc2 I stumbled on a weird behavior. I did not find any hints in the archives or at the bug tracker. Two SIP-Clients are connected (both on the local net, no NAT). The RTP stream flows directly between the phones. If I set phone A on hold, the musi

Re: [asterisk-users] HDLC Bad FCS (8) on Primary D-channel

Hi Olivier, Am Freitag, den 11.06.2010, 14:27 +0200 schrieb Olivier: > > > 2010/6/11 Karsten Wemheuer > Hi, > > Am Freitag, den 11.06.2010, 11:54 +0100 schrieb Gareth Blades: > > Olivier wrote: > > > Hello, >

Re: [asterisk-users] HDLC Bad FCS (8) on Primary D-channel

Hi, Am Freitag, den 11.06.2010, 11:54 +0100 schrieb Gareth Blades: > Olivier wrote: > > Hello, > > > > I've got a running system in which logs are full of messages such as: > > [Jun 10 07:24:14] NOTICE[2414] chan_dahdi.c: PRI got event: HDLC Bad FCS > > (8) on Primary D-channel of span 2 > > >

Re: [asterisk-users] SIP RTP ports not released when channel is hung up

Hi, Am Donnerstag, den 18.02.2010, 10:49 +0100 schrieb Armin Schindler: > On Tue, 16 Feb 2010, Armin Schindler wrote: > > On Tue, 16 Feb 2010, Marcus Hunger wrote: > >> Hi, > >> > >> did you see this one: https://issues.asterisk.org/view.php?id=16774 ? It > >> looks related to your issue. > > >

Re: [asterisk-users] Trouble getting feature codes to work

Hi, Am Donnerstag, den 21.01.2010, 21:08 -0500 schrieb hugolivude: > Hi, > > I'm having trouble getting feature codes to work in Asterisk 1.4.21.2. > Features.conf contians this: > > blindxfer=## > atxfer=*2 > automon=*1 > disconnect=** > > I'm really most interested in getting disconnect to wo

Re: [asterisk-users] DTMF detection on dahdi with b4xxp (again, some more details)

Hi, Am Dienstag, den 05.01.2010, 15:38 +0100 schrieb Christian Theune: > Hi, > > I tried again getting DTMF detection on my ISDN devices with dahdi going > again. I used the channel debug to see whether asterisk sees the frames > and detects them as DTMF. > > Interestingly here's what works: >

Re: [asterisk-users] Unable to set TOS to 184?

Hi Bart, Am Donnerstag, den 29.10.2009, 16:36 -0700 schrieb Bart Fisher: > I don't understand this message: > > [2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos: > Unable to set TOS to 184 You did not tell us, which version of asterisk You are running. The kernel restricts setti

Re: [asterisk-users] Calling non-extension numbers issue

Hi, Am Montag, den 29.06.2009, 10:35 -0400 schrieb Kayton Sapale: > That's the strange thing. Nothing shows when monitoring the service > in debug. On the phone, however, I do see a "connection time-out" > error. I guess this might indicate that the device is attempting to > connect to the serv

Re: [asterisk-users] Asterisk / Hylafax

Hi Michael, Am Samstag, den 13.12.2008, 23:09 +1300 schrieb Michael: > For some odd reason the call registration issue doesn't seem to stop it > working, except a few seconds after Hylafax answers the call it hangs up, I > suspect because Asterisk only supports T38 pass through. > > Here is my

Re: [asterisk-users] One Way Audio Problem

Hi, Am Donnerstag, den 16.10.2008, 09:37 +0800 schrieb GNUbie: > Hello Karsten, > > On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer <[EMAIL PROTECTED]> wrote: > > > > Please post Your sip.conf. > > Which IP-Address do You configure in the snom for Your aster

Re: [asterisk-users] cli commands missing

Hi Eric, Am Sonntag, den 12.10.2008, 18:06 -0700 schrieb Eric Fort: > resolve.conf and dns is working. The problem persists. > /var/log/asterisk/messages shows a few notices and warnings on > res_smdi.c, res_musiconhold.c, and usbradio.c. when I disable loading > of these in modules.conf aster

Re: [asterisk-users] One Way Audio Problem

Hi, Am Montag, den 13.10.2008, 10:00 +0800 schrieb GNUbie: > Hello Gordon, > > On Mon, Oct 13, 2008 at 2:22 AM, Gordon Henderson > <[EMAIL PROTECTED]> wrote: > > > > You mention the SIP phone being inside the LAN. Where is the Asterisk box? > > It is the main gateway of the IP phones and my lapt

Re: [asterisk-users] Read one or X DTMF

Hi Ruddy, Am Freitag, den 12.09.2008, 21:38 -0400 schrieb Ruddy Gbaguidi: > But user just needs to enter * instead of *# > We are doing this because 80% of the callers already have an account, > so, instead of playing : > "If you have an account, press 1, if not press 2" > > we prefer to play >

Re: [asterisk-users] Read one or X DTMF

Hi Ruddy, Am Freitag, den 12.09.2008, 13:22 -0400 schrieb Ruddy Gbaguidi: > Thanks for the hint. Sorry about that. > If I use your soution, I cannot make any difference between a user > pressing * and a user that reach the timeout because he didn't enter any > digit. > In both cases, I will have

Re: [asterisk-users] Extension not found

Hi Michel, Am Freitag, den 12.09.2008, 17:41 +0300 schrieb michel freiha: > Dear All, > > I have the following scenario...When a customer dial 111 number a beep > message will iplay in order to record and playback his voice...Else > he'll be routed to another call flow as you can see in the conte

Re: [asterisk-users] Read one or X DTMF

Hi, Am Freitag, den 12.09.2008, 11:03 -0400 schrieb Ruddy Gbaguidi: > Hi all > I'm just having a problem now and I don't have any idea how to do this. > > It is pretty simple. When a customer calls, to speed up the navigation > in the dialplan, I want something like > > "Welcome. Please enter y

Re: [asterisk-users] problem with iaxmodem!

Hi, Am Mittwoch, den 06.08.2008, 17:24 +0200 schrieb Nadjia Boumédiène: > My iax.conf looks like this: > > [iaxmodem] > type=friend > host=127.0.0.1 > secret=x > context=fax-out > permit=127.0.0.1 > disallow=all > allow=ulaw > > after editing inittab I reload it by running: "/sbin/init q" >

Re: [asterisk-users] Dialplan Action on Authentication

Hi David, Am Sonntag, den 20.07.2008, 11:57 +0200 schrieb David Ashwood: > Morning guys and gals, > > > > I’d like to be able to run some code when a device (soft/hardphone) > authenticates to Asterisk. > > I remember reading somewhere that there’s the possibility of part of a > dialplan can

Re: [asterisk-users] asterisk seg fault

Hi, Am Mittwoch, den 25.06.2008, 08:42 -0400 schrieb Jerry Geis: > I am running asterisk from svn check out from yesterday Jun 24. > I started with 1.4.20, then 1.4.21 then svn. > > I am getting: > pcm_local.h:389 snd_pcm_channel_area_addr assertion bitsofs %8 = 0 failed > segment fault. > > I a

[asterisk-users] Problem with SIP, attended transfer and GROUP_COUNT

Hi, maybe someone can give me a hint to solve the following issue. I want to limit the calls to a specific SIP-destination. Disabling callwaiting at the phones is not an option, because it should be configured via the * database. My solution uses GROUP_COUNT, which works fine most of the time. In

Re: [asterisk-users] Ring back when free?

Hi, Am Freitag, den 04.04.2008, 13:03 + schrieb Tony Mountifield: > In article <[EMAIL PROTECTED]>, > Faraz R. Khan <[EMAIL PROTECTED]> wrote: > > Thinking out loud: write a asterisk call file (when the calling user > > presses 5) which keeps on trying to connect the two. > > I thought about

Re: [asterisk-users] Two phones fail to agree on codec, asterisk at fault?

Hi Martin, Am Freitag, den 28.03.2008, 14:27 +0100 schrieb martin f krafft: > [...] > So calls are going via an asterisk bridge and the symptoms of my > problem are: > > 1 if C450IP calls softphone, they can talk fine > 2 if softphone calls C450IP, voice only goes from C450IP to > softphon

Re: [asterisk-users] Mistake in the wiki's description of cmd Pickup() ?

Hi Stefan, Am Dienstag, den 05.02.2008, 10:30 +0100 schrieb Stefan Guenther: > Hi, > > according to the description of Pickup() on page > http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup > > I can use this command to pickup a call at a certain extensions. > > When I try this with e.g. >

Re: [asterisk-users] GROUP_COUNT and Attended transfer

Hi Paul, Am Dienstag, den 05.02.2008, 10:10 +1100 schrieb Paul Hales: > With some of the phones (snom, for example) you can turn off mwi, so the > phone will only accept one call at a time. Much easier. > > PaulH Thanks for Your answer. Unfortunaly turning call waiting off is not an option for m

[asterisk-users] GROUP_COUNT and Attended transfer

Hi, I want to use GROUP_COUNT to limit calls to a specific destination. From somewhere on the wiki I am using the following context: exten => 200,1,Set(GROUP()=${CALLERID(num)}) exten => 200,n,GotoIf($[${GROUP_COUNT(${EXTEN})} >= 1]?BLOCK) exten => 200,n,Set(OUTBOUND_GROUP=${EXTEN}) exten => 200,

Re: [asterisk-users] Cisco power injector with GXP2000 phones

Hi, Am Donnerstag, den 06.12.2007, 11:30 -0500 schrieb Jon Pounder: > Quoting Ricardo Carvalho <[EMAIL PROTECTED]>: > > > I only see one explanation to my problem... > > > > GXP2000 phones only implement PoE mode A of the IEEE 802.3af protocol, and > > the power injector does only PoE mode B of t

Re: [asterisk-users] get egress SIP call Id

Hi Ray, Am Dienstag, den 09.10.2007, 10:10 -0500 schrieb Ray Chen: > Hi Philipp, > > Thank you for your response to my question. I am working on a > project which uses Asterisk as the voice engine. I need to get > the ingress and egress sip call id for a ca

Re: [asterisk-users] Configuration files inside SQLite3

atabase > which has ODBC compatibility. Thus, sqlite qualifies. If you are using > trunk, you won't even need to use ODBC, because Asterisk has native > support for sqlite. Are You shure the native support of asterisk is for SQLite3 as the original poster asks for? AFAIK * su

Re: [asterisk-users] Musiconhold instead ringing

Hi, Am Samstag, den 08.09.2007, 09:44 + schrieb wassim darwish: > > From: [EMAIL PROTECTED]> > To: asterisk-users@lists.digium.com> Date: Fri, 7 Sep 2007 19:10:04 > -0500> Subject: Re: [asterisk-users] Musiconhold instead ringing>> On > Friday 07 Sept

Re: [asterisk-users] Slow list

Hi, Am Donnerstag, den 05.07.2007, 14:58 -0400 schrieb Andrew Kohlsmith: > On Thursday 05 July 2007 2:38 pm, Doug Lytle wrote: > > Already did that. I use ASSP for filtering. Digium and associated > > mailing lists are white listed. There was only 1 attempt for deliver > > and there were no del

RE: [asterisk-users] Log CODECS in CDR's

Hi Morgan, Am Freitag, den 11.05.2007, 10:32 +0100 schrieb Morgan Gilroy: > Thanks for the pointers, I know about the Set(CDR..) function but I need > the codec that was negotiated in the Dial (once I have that its easy to > stick it into the cdrs as you pointed out). > Ie a call comes in as G729

Re: [asterisk-users] BeroNet HFC-4S card is now detected as only 2 ports

On Fri, Apr 06, 2007 Tzafrir Cohen wrote: > On Fri, Apr 06, 2007 at 01:18:24AM +0200, Henrik Woffinden wrote: > > Hello list, > > > > After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently > > detected as 2 ports instead of 4. > > > > I still load the driver as "modprobe qozap ports

Re: [asterisk-users] chan_capi and only one B channel usable?

On 03/22/2007, Torge Szczepanek wrote: > Hello list! > > I have a Asterisk 1.2.10 running using the package from Backports.org > for Debian Sarge. > > I have setup chan_capi (0.6.5 from Backports) and it seems that I am > only able to use on B-Channel. > > When trying to place the second call I

Re: [asterisk-users] Mini-ITX board + FXO PCI card?

Hello, Am Donnerstag, den 15.02.2007, 10:55 +0800 schrieb Leo Ann Boon: > > > > 1. The smallest mini-ITX case I found that accepts a PCI card is the > > Travla C138: If you used a mini-ITX with a Digium TDM400P, do you know > > if it fits? I didn't find its width, and apparently, the C138 will n

Re: [asterisk-users] volume control in VoIP

Hi, On Fryday, 2007-02-02 François Delawarde wrote : > Don't you think it could be an interesting feature in Asterisk? It > already does transcoding, why not gain when voice flow passes through it? > > François. On a "SIP-to-SIP"-Call Asterisk is not neccessarily in the voice flow, so this does

Re: [asterisk-users] No D-channels available! Using Primary channel 16 as D-channel anyway!

Hi, maybe I am a little bit late with this answer. I take a look at Your config and the debug output. > > Provider <--te11xp---> asterisk <---te11xp--> nortel merridian > option 11c > > zapata.conf > - > context=from-pstn > switchtype=dms100 > s

[Asterisk-Users] Q: Status of feature Call Deflection / Partial Rerouing in chan-capi and zaphfc

Hello, AFAIK the feature CD (call deflection) is only possible on point-to-multipoint links, is this correct? I've heard about the feature "partial rerouting" which should do the same on point-to-point-links. Is this implemented in either bristuff or chan-capi(-cm)? Thanks in advance, Karsten

Re: [Asterisk-Users] Problem with chan-capi: outgoing calls on two lines

Hello Armin, Am Mo, den 27.02.2006 schrieb Armin Schindler um 20:23: > On Mon, 27 Feb 2006, Karsten Wemheuer wrote: > > Hello, > > > > On Mo, 27 Feb 2006, Armin Schindler wrote: > > > This is not a bug, just normal behaviour. > > > chan_capi does not kno

Re: [Asterisk-Users] Problem with chan-capi: outgoing calls on two lines

Hello, On Mo, 27 Feb 2006, Armin Schindler wrote: > On Mon, 27 Feb 2006, Karsten Wemheuer wrote: > > > > In detail: > > When all lines are connected, the first two calls are placed on line 1 > > (which is on controller 1). The next two calls are placed on line 2 (on &g

[Asterisk-Users] Problem with chan-capi: outgoing calls on two lines

Hello, while testing the following scenario, I ran into trouble: One * box with two AVM active controllers in Point-to-Point-Mode is connected to another * box with ZapHFC/Quad-BRI cards using bristuff in NT-mode. All is working fine, I can call from one box to the other and vice versa. But if I'

Re: [Asterisk-Users] distorted native music on hold

Hi, On Mon, Jan 16 2006 Louis-David Mitterrand wrote: > Hello, > > Using asterisk-1.2.1 I am trying to convert my music-on-hold files from > .wav to alaw: > > % sox moh.wav -r 8000 -c 1 moh.al resample -ql > > The file sounds fine when listened with: > > % sox mox.al -t ossdsp /dev

Re: [Asterisk-Users] MoH trouble with latest bristuff (0.3.0-PRE-1f) -> SOLVED!

Hi, I answer to my own posting... On Sun, Jan 15 2006 Karsten Wemheuer wrote: > Hi, > > I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When > I activate music-on-hold on a SIP-to-SIP connection, the music sounds > like in a fast-forward play mode. On

[Asterisk-Users] MoH trouble with latest bristuff (0.3.0-PRE-1f)

Hi, I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When I activate music-on-hold on a SIP-to-SIP connection, the music sounds like in a fast-forward play mode. On the *-console I can see much lines like this: -- Silence suppression is disabled (option_silence_suppression=0 c

Re: [Asterisk-Users] CAPI unable to handle busy()

Hello Armin, On Mo, 2 Jan 2006 Armin Schindler wrote: > I don't think it is necessary to exclude it. Just build chan_capi-cm and > overwrite chan_capi.so as well as remove the app_capi* modules from your > installation. > > Armin Many thanks, it is working. Karsten __

Re: [Asterisk-Users] Limit concurent calls per MSN on BRI (bristuff/zaphfc)?

Hello, On Tue, 03.01.2006 Pisac wrote: > How can I limit incoming concurent calls per MSN on BRI ISDN? > > I'm using bristuff/zaphfc with imediate=no in zapata.conf, and I have 4 > MSNumbers. I want that incoming caller (to my MSNumber 1234) get busy if > that number is already in use through ano

Re: [Asterisk-Users] CAPI unable to handle busy()

Hello Armin, On Mo, 02.01.2006 Armin Schindler wrote: > On Mon, 2 Jan 2006, Karsten Wemheuer wrote: > > Hello, > > > > first of all, I say "Happy New Year" to this list! > > > > While using asterisk 1.2.1 with bristuff-0.3.0-PRE-1d (which include

[Asterisk-Users] CAPI unable to handle busy()

Hello, first of all, I say "Happy New Year" to this list! While using asterisk 1.2.1 with bristuff-0.3.0-PRE-1d (which includes chan_capi 0.4.0-PRE1), I ran into the following problem. I want to signal "busy" to an incoming call, but that doesn't work. The dialplan looks like this: exte

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