Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

On 11/22/2011 05:42 PM, Alex Vishnev wrote: I doubt it. Unknown headers should be ignored by UAS. is it possible to post the trace? On Nov 22, 2011, at 11:39 AM, Jonas Kellens wrote: What trace do you need ? Have you read my original post ? Asterisk SIP debug trace is posted in my original

Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

On 11/22/2011 05:31 PM, Alex Vishnev wrote: Your registration should have also have the flow PEER ASTERISK REGISTER---> <--401 REGISTER(nonce) -> <200OK Then the server controls the life of the registration and 200 Expires Header

Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

On 11/22/2011 04:37 PM, Bruce Ferrell wrote: On 11/22/2011 07:29 AM, Jonas Kellens wrote: On 11/22/2011 04:25 PM, Bruce Ferrell wrote: Jonas, May I suggest that you present us your sip.conf entry for this peer, properly redacted, of course. That might help more. What I do for "gat

Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

On 11/22/2011 04:25 PM, Bruce Ferrell wrote: Jonas, May I suggest that you present us your sip.conf entry for this peer, properly redacted, of course. That might help more. What I do for "gateways" at known addresses is to put an entry like this into the sip.conf entry: [peer] type=peer

[asterisk-users] Asterisk refuses INVITE (401) and I don't know why

Hello list, this is the communication between an Aastra 5000 PBX and Asterisk, where the Aastra makes a call to Asterisk. For some reason, Asterisk responds with 401-Unauthorized and I don't know why. Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT with this Aastra.

Re: [asterisk-users] Unable to build sip pvt data - Switching equipment congestion

ng" ? *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, November 02, 2011 10:06 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Unable to build sip p

Re: [asterisk-users] Unable to build sip pvt data - Switching equipment congestion

erisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, November 02, 2011 9:57 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Unable to build sip pvt data - Switching equi

[asterisk-users] Unable to build sip pvt data - Switching equipment congestion

Hello list, can anyone tell me what the following means (found in messages log) : /[Nov 2 11:16:21] ERROR[18407] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Nov 2 11:16:21] WARNING[18407] chan_sip.c: Unable to create RTP audio session: Address already in use [Nov

Re: [asterisk-users] Unable to build sip pvt data

Hello, is there any more feedback on this thread ? On 10/20/2011 05:10 PM, Jonas Kellens wrote: On 10/20/2011 05:07 PM, Paul Belanger wrote: On 11-10-20 10:28 AM, Jonas Kellens wrote: Hello list, what does this mean ? [Oct 20 15:23:08] ERROR[1496] chan_sip.c: Unable to build sip pvt

Re: [asterisk-users] Unable to build sip pvt data

On 10/20/2011 05:07 PM, Paul Belanger wrote: On 11-10-20 10:28 AM, Jonas Kellens wrote: Hello list, what does this mean ? [Oct 20 15:23:08] ERROR[1496] chan_sip.c: Unable to build sip pvt data for 'account9' (Out of memory or socket error) [Oct 20 15:23:08] ERROR[1496] chan_sip.c:

[asterisk-users] Unable to build sip pvt data

Hello list, what does this mean ? [Oct 20 15:23:08] ERROR[1496] chan_sip.c: Unable to build sip pvt data for 'account9' (Out of memory or socket error) [Oct 20 15:23:08] ERROR[1496] chan_sip.c: Unable to build sip pvt data for 'account10' (Out of memory or socket error) [Oct 20 15:23:08] ERRO

[asterisk-users] Asterisk sponteanous reboot : core dump file

Hello, when I try to get something out of the core dump file, I get this : [root@jonas Desktop]# gdb asterisk core.sip1.server.be GNU gdb (GDB) Fedora (7.2-51.fc14) Copyright (C) 2010 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later

[asterisk-users] Asterisk Realtime SIP : vmexten

Hello list, is the field "vmexten" available when using SIP peers in a realtime Mysql-DB ? Thanks. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for

[asterisk-users] /usr/sbin/asterisk -rx and AMI

Hello, currently I run a php script in cron which polls for information on SIP peers using the /usr/sbin/asteirsk -rx method. I notice that after some time the Asterisk interface freezes, SIP Peer registrations become unreachable and sip reload or any other command on the CLI does not respon

Re: [asterisk-users] Mysql dialplan statement not executed

On 09/14/2011 02:51 PM, Jonas Kellens wrote: Hello, I do the following in a macro in the dialplan : exten => s,n,MYSQL(Connect connid localhost user password AsteriskDB) exten => s,n,MYSQL(Query resultid ${connid} UPDATE custDB SET active=1 WHERE routeID=${ARG1} AND nr=1) exten =>

[asterisk-users] Mysql dialplan statement not executed

Hello, I do the following in a macro in the dialplan : exten => s,n,MYSQL(Connect connid localhost user password AsteriskDB) exten => s,n,MYSQL(Query resultid ${connid} UPDATE custDB SET active=1 WHERE routeID=${ARG1} AND nr=1) exten => s,n,MYSQL(Disconnect ${connid}) But nothing changes in m

[asterisk-users] How does AMI work with events ?

Hello list, I don't really understand how AMI works. I read some information and examples on the net, but they all show how you login to the AMI, give an action and receive a response. The end. I guess you just re-run the script every time you want the action to be executed. How then does th

[asterisk-users] Cisco SPA 941 and auto-answer with SIPheader Call-Info

Hello, I'm trying to page the Cisco SPA 941 by adding the SIP-header Call-Info: answer-after=0 dialplan : exten => _*XX*,n,SIPAddHeader("Call-Info: answer-after=0") SIP debug : INVITE sip:testcorp6@192.168.1.106:5064 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK6090dca4;rport

[asterisk-users] Asterisk spontaneous reboot

Hello, Today I restart the MySQL-DB (/sbin/service mysqld restart) and I could no longer connect to asterisk (/usr/sbin/asterisk -r) for a few seconds. There is now a core dump present in /tmp : -rw--- 1 root root 88M Aug 26 08:07 core.sip.pbx.tld-2011-08-26T08:07:35+0200 How can I g

[asterisk-users] DISA password

Hello list, how can I give a simple password to the DISA-application ? The following : exten => 1000,n,DISA('123456',from-TEST) results in : [Jul 23 13:47:51] WARNING[2357]: app_disa.c:255 disa_exec: DISA password file '123456' not found on chan SIP/test6-0006 Kind regards, Jonas. --

[asterisk-users] Call to *2*999... : IP-phone configration

Hello list, I am unable to call *2*999... because my phone automatically sends the number after I press *. So my IP-phone calls *2. Now this is a Cisco, but that's not my question. Does anyone know what setting I need to adjust so my phone (but actually any IP-phone) accepts an * in the midd

[asterisk-users] Get second cipher in an extension

Hello list, how can I get the second character/cipher of an extension ? If I have : exten => 12345,n,NoOP() How can I get "2" ? If I have : exten => 787,n,NoOP() How can I get "8" ? Thanks ! Kind regards, Jonas. -- _ -

[asterisk-users] Asterisk and Audiocodes PRI card

Hello list, can anyone tell me if this card : http://www.audiocodes.com/product/ipm-260-sip is compatible with Asterisk (DAHDI) for use as PCI PRI card ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Queue log in MySQL DB

On 06/08/2011 09:10 AM, Satish Barot wrote: Set queue_log = no in logger.conf. By default it is set to 'yes'. [SATISH] Will there then still be queue logging at all ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation P

[asterisk-users] Queue log in MySQL DB

Hello list, I have configured extconfig.conf to save queue log into my MySQL-DB. I notice however that there is still logging too in /var/log/asterisk/queue_log. How can I disable queue logging into text files ? Kind regards, Jonas. -- __

Re: [asterisk-users] Hints custom:abcdef

In other words : is it correct to say that hints need to be unique, even if they are defined in different contexts ? On 05/20/2011 12:07 PM, Jonas Kellens wrote: Hello list, I want certain devices to monitor certain extensions/SIPaccounts and other devices to monitor other extensions

[asterisk-users] Hints custom:abcdef

Hello list, I want certain devices to monitor certain extensions/SIPaccounts and other devices to monitor other extensions/SIPaccounts. Therefore I do the following : [from-TEST1] include => test1-blf [from-TEST2] include => test2-blf [test1-blf] exten => 10,hint,SIP/testcorp1 exten => 20

Re: [asterisk-users] Light indicator managed by Asterisk

Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, May 12, 2011 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Light indicator managed by Ast

Re: [asterisk-users] Light indicator managed by Asterisk

On 05/12/2011 07:12 PM, Carlos Chavez wrote: On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote: Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even

Re: [asterisk-users] Light indicator managed by Asterisk

On 05/12/2011 07:24 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, May 12, 2011 12:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [asterisk-users] Light indicator managed by Asterisk

On 05/12/2011 07:12 PM, Carlos Chavez wrote: On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote: Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even

Re: [asterisk-users] Light indicator managed by Asterisk

On 05/12/2011 06:58 PM, Andrew Latham wrote: On Thu, May 12, 2011 at 12:50 PM, Jonas Kellens wrote: Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even

[asterisk-users] Light indicator managed by Asterisk

Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a cer

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!!! [

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

e cause code!!! Jonas. On 05/05/2011 10:57 AM, Jonas Kellens wrote: 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 0

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

On 05/05/2011 10:22 AM, Jonas Kellens wrote: 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_si

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

y 5 10:08:38] DEBUG[16215] chan_sip.c: Hangup call SIP/expsom7-043f, SIP callid 21cd242d0d4075961fbd21bd45012727@The_IP [May 5 10:08:38] DEBUG[16215] chan_sip.c: Updating call counter for outgoing call [May 5 10:08:38] DEBUG[16215] chan_sip.c: Hanging up channel in state Ringing (not UP)/ On 05/05/2011

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

0x11be6140 and write factory 0x11be6b78 both fail to provide 160 samples [May 5 10:13:51] DEBUG[16223] audiohook.c: Failed to get 160 samples from read factory 0x11be6140 On 05/03/2011 04:03 PM, Jonas Kellens wrote: Hello, I see a lot of these messages in the debug log : /[May 3 15:47:09] DEBUG

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

Hello, I see a lot of these messages in the debug log : /[May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples from write factory 0xae17e18 [May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples from write factory 0xae17e18 [May 3 15:47:09] DEBUG[19081] audiohoo

Re: [asterisk-users] Asterisk unresponsive

On 04/18/2011 06:36 PM, Paul Belanger wrote: On 11-04-18 09:46 AM, Jonas Kellens wrote: Asterisk freezed and only a reboot of the whole server fixed this. Any command on the Asterisk CLI was not executed because Asterisk was too busy processing all of these messages that you see in the debug

Re: [asterisk-users] Registrations stops after 403 FORBIDDEN

On 04/18/2011 05:33 PM, Warren Selby wrote: On Mon, Apr 18, 2011 at 4:54 AM, Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello list, I have in sip.conf : So are my settings wrong ? What does sip show settings look like from the CLI? vps*CLI> sip show

Re: [asterisk-users] Asterisk unresponsive

On 04/18/2011 03:58 PM, Terry Brummell wrote: http://lmgtfy.com/?q=audiohook.c%3A+Failed+to+get+160+samples+from+read+factory This should tell me that there are others who experience this same problem in some kind of form and that there is no real answer to it ? Hence why I seek for an answ

[asterisk-users] Asterisk unresponsive

Hello list, I've got a whole lot of these in my debug log : [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr

[asterisk-users] Registrations stops after 403 FORBIDDEN

Hello list, I have in sip.conf : /maxexpiry=60 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) defaultexpiry=120

Re: [asterisk-users] Realtime SIP & peer status

On 04/15/2011 11:53 AM, Adolphe Cher-aime wrote: Registry type Event will give you information about your peer. Adolphe Cher-aime From my Iphone I don't find information on how this event tells me whether the SIP peer is occupied with a call or not. How can I capture the notify messages (as

Re: [asterisk-users] Realtime SIP & peer status

On 04/13/2011 09:18 PM, Rob Coward wrote: Rather than add extra overhead to your dialplan and the asterisk server, why not make use of the AMI and have a background process listening for the various events and updating your database accordingly ? See http://www.voip-info.org/wiki/view/aster

Re: [asterisk-users] Realtime SIP & peer status

On 04/13/2011 11:46 AM, Andrew Thomas wrote: Fair enough. Then if this is really what you want I guess an AGI is the best way to go. As for load - well, that depends on how many concurrent connections you figure on having [and of course the platform it's all on]. Platform : CentOS 5.5 Asterisk

Re: [asterisk-users] Realtime SIP & peer status

On 04/13/2011 11:28 AM, Andrew Thomas wrote: Maybe I should have asked 'why do you want to put the status in to a mySQL database'? BTW - extensions.conf has mySQL functions built in - so no external script is actually needed. Well, I read out this information in a website which serves as a co

Re: [asterisk-users] Realtime SIP & peer status

On 04/13/2011 11:20 AM, Ishfaq Malik wrote: On Wed, 2011-04-13 at 11:09 +0200, Jonas Kellens wrote: On 04/13/2011 10:57 AM, Ishfaq Malik wrote: On Wed, 2011-04-13 at 10:15 +0200, Jonas Kellens wrote: Hello, I'm using SIP realtime with MySQL DB. Is it possible to ge

Re: [asterisk-users] Realtime SIP & peer status

On 04/13/2011 10:57 AM, Ishfaq Malik wrote: On Wed, 2011-04-13 at 10:15 +0200, Jonas Kellens wrote: Hello, I'm using SIP realtime with MySQL DB. Is it possible to get the status of the SIP peer (free / calling) from this realtime DB ? If not, is there another way to obtain the call

[asterisk-users] Realtime SIP & peer status

Hello, I'm using SIP realtime with MySQL DB. Is it possible to get the status of the SIP peer (free / calling) from this realtime DB ? If not, is there another way to obtain the call state of a SIP peer ? Kind regards, Jonas. -- __

[asterisk-users] SIP register and contact header

Hello, I define SIP registrations as follow in sip.conf : register => number:passwd@sip-server example : register => 33:mypass@ip_sip_server But apparently the SIP 'contact' header in the SIP REGISTER looks like this : /Contact: / How come ? And how to change this so it reads : /

Re: [asterisk-users] Registration from '"000000" x 1000

On 04/02/2011 02:08 PM, Steve Davies wrote: On 2 April 2011 09:46, Jonas Kellens wrote: Hello list, I often see the following in my message log : [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" ' failed for '184.106.109.168' - No mat

[asterisk-users] Registration from '"000000" x 1000

Hello list, I often see the following in my message log : [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" ' failed for '184.106.109.168' - No matching peer found [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" ' failed for '184.106.109.168' - No

[asterisk-users] Get phone number from SIP header PAI

Hello list, I want to get the phone number out of the following P-Asserted-Identity header : /"BlaBlaBla" "/ I do the following in the dialplan : /exten => _XXX.,n,Set(PY=${SIP_HEADER(P-Asserted-Identity)}) exten => _XXX.,n,Set(PY2=${CUT(PY,@,1)})/ This gives me : /"BlaBlaBla" _XXX.,n,Set

Re: [asterisk-users] Asterisk 1.6.2.10 & CDR custom added field

On 03/25/2011 08:19 AM, Tilghman Lesher wrote: On Thursday 24 March 2011 04:50:48 Jonas Kellens wrote: On 03/24/2011 10:45 AM, Rizwan Hisham wrote: You have to use adaptive cdr for this functionality. In 1.8 the conf file for adaptive cdr is cdr_adaptive_odbc.conf. The sample conf

Re: [asterisk-users] Asterisk 1.6.2.10 & CDR custom added field

On 03/24/2011 10:45 AM, Rizwan Hisham wrote: You have to use adaptive cdr for this functionality. In 1.8 the conf file for adaptive cdr is cdr_adaptive_odbc.conf. The sample conf file should tell you everything. If you are using some other cdr engine then you will have to jump into the code o

[asterisk-users] Fwd: Asterisk 1.6.2.10 & CDR custom added field

rds, Jonas. Original Message Subject:[asterisk-users] Asterisk 1.6.2.10 & CDR custom added field Date: Tue, 22 Mar 2011 14:05:23 +0100 From: Jonas Kellens Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing Lis

[asterisk-users] Asterisk 1.6.2.10 & CDR custom added field

Hello list, I have added an extra field "mycolumn" to the cdr table in my MySQL-DB. I simply try to add a value to this column by doing the following in the dialplan : exten => 600,n,Set(CDR(mycolumn)="myvalue") But this value is not written to the column 'mycolumn' together with the other

Re: [asterisk-users] Trunk form asterisk1 to asterisk2 fails

On 03/16/2011 08:39 PM, Jonas Kellens wrote: On Asterisk server 2 I see the following when I make a call with a Grandstream IP-phone, registered at Asterisk server 1 : [Mar 16 20:32:44] WARNING[1680]: chan_sip.c:12872 check_auth: username mismatch, have , digest has [Mar 16 20:32:44

[asterisk-users] Trunk form asterisk1 to asterisk2 fails

Hello, When I want to send a call from asterisk-server 1 to asterisk-server 2, it fails. On Asterisk server 1 : register => user:passwd@asterisk1 ; Test TRUNK [trunk2] type=peer host=asterisk1 username=user ;defaultuser=user secret=passwd disallow=all allow=alaw allow=gsm qualify=yes canreinv

Re: [asterisk-users] Extract Remote-Party-ID from incoming INVITE indialplan

On 03/16/2011 02:56 PM, Andrew Latham wrote: On Wed, Mar 16, 2011 at 10:50 AM, Jonas Kellens wrote: is it possible to extract the Remote-Party-ID from an incoming call in the dialplan ? Is there some kind of function for this ? Kind regards, Jonas. 1.8 Documentation on Connected

Re: [asterisk-users] Extract Remote-Party-ID from incoming INVITE indialplan

On 03/16/2011 02:07 PM, Danny Nicholas wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, March 16, 2011 2:39 AM *To

[asterisk-users] Extract Remote-Party-ID from incoming INVITE in dialplan

Hello list, is it possible to extract the Remote-Party-ID from an incoming call in the dialplan ? Is there some kind of function for this ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-dig

Re: [asterisk-users] SIPAddHeader not working

On 03/15/2011 12:39 PM, Steven Howes wrote: On 15 Mar 2011, at 11:30, Jonas Kellens wrote: On 03/15/2011 12:24 PM, Steven Howes wrote: On 15 Mar 2011, at 09:08, Jonas Kellens wrote: I also notice the presence of a "Remote-Party-ID" SIPheader... Where does this come from ?! N

Re: [asterisk-users] SIPAddHeader not working

On 03/15/2011 12:24 PM, Steven Howes wrote: On 15 Mar 2011, at 09:08, Jonas Kellens wrote: I also notice the presence of a "Remote-Party-ID" SIPheader... Where does this come from ?! Not from my dialplan... sendrpid in your sip.conf Steve Not really : [3259] type=peer ho

Re: [asterisk-users] SIPAddHeader not working

On 03/14/2011 05:06 PM, Steven Howes wrote: On 14 Mar 2011, at 15:58, Jonas Kellens wrote: dialplan : exten => 67121212,1,NoOp() exten => 67121212,n,Set(CALLERID(all)="3259" <3259>) exten => 67121212,n,SIPAddHeader(P-Preferred-Identity: ) exten => 6712121

Re: [asterisk-users] SIPAddHeader not working

Header(P-Preferred-Identity:) SIPAddHeader(Privacy: id) That works for me in the UK. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: 11 March 2011 15:06 To: Asterisk Users Mailing

Re: [asterisk-users] SIPAddHeader not working

Ext. 2003 *From*: "Jonas Kellens" *Sent*: Wednesday, March 09, 2011 9:18 AM *To*: brya...@zktech.com, "Asterisk Users Mailing List - Non-Commercial Discussion" *Subject*: Re: [asterisk-users] SIPAddHeader no

Re: [asterisk-users] SIPAddHeader not working

On 03/09/2011 02:09 PM, Bryant Zimmerman wrote: *From*: "Jonas Kellens" *Sent*: Wednesday, March 09, 2011 4:18 AM *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" *Subject*: [asterisk

[asterisk-users] SIPAddHeader not working

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0...@sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" ;tag=729476

[asterisk-users] Difference mohsuggest & mohinterpret

Hello list, what is the difference between mohsuggest & mohinterpret when defining a SIP peer ?! If a certain SIP peer puts another channel on hold, what field then determines the moh class that Asterisk will choose to play to that channel ? If I take the test and call from peer A to peer

[asterisk-users] DTMF and Snom

Hello list, I'm having some troubles with DTMF tones. When pressing numbers on a Snom phone, the DTMF-signal takes too long. I have the following in sip.conf : dtmfmode = rfc2833 which works well for Grandstream, Yealink and Cisco phones. But not for Snom. Snom support tells me I should

Re: [asterisk-users] Realtime queues not playing prompts

Why is "queue_thankyou" played, even if the field is set to NULL ? Why is "queue_thereare" and "queue_callswaiting" not playing ?? Kind regards, Jonas. On 02/11/2011 12:48 PM, Jonas Kellens wrote: Hello list, I'm using realtime queues and noticing

[asterisk-users] Realtime queues not playing prompts

Hello list, I'm using realtime queues and noticing that prompts are not played as expected. Database : announce = queue_youarenext = queue_youarenext queue_thereare = queue_thereare queue_callswaiting = queue_callswaiting queue_holdtime = queue_thankyou = queue_reporthold = announce_frequency

[asterisk-users] How to load new musiconhold classes ?

Hello, I've defined some new musiconhold classes in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [908001] mode=files directory=/var/lib/asterisk/moh/908001 random=yes ; [101001-1] mode=files directory=/var/lib/asterisk/moh/101001/1 random=yes ; [101001-2]

[asterisk-users] Musiconhold priority

Hello list, what musiconhold class has priority : - field "musiconhold" of the SIPaccount and field "musiconhold" of a queue or - Set(CHANNEL(musicclass)=) ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Pickup local/.... not working

On 01/26/2011 05:38 PM, Sherwood McGowan wrote: I think you're missing something in your explanation... the code represented in your email shows no reason for a Local channel to be recreated. Goto commands do not result in Local channel creation, nor does the Dial command Well, what happens is

Re: [asterisk-users] Pickup local/.... not working

On 01/26/2011 04:26 PM, Danny Nicholas wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, January 26, 2011 9:22 AM *To

[asterisk-users] Pickup local/.... not working

Hello list, is it possible that it is not possible to pickup a local channel ?? [Jan 26 16:13:43] -- Executing [10@sub-pickup:24] Pickup("SIP/voip5-0750", "Local/329596@default-505a;2@PICKUPMARK") in new stack [Jan 26 16:13:43] NOTICE[29658]: app_directed_pickup.c:265 pickup_exec: No

[asterisk-users] Unable to insert cdr-data into mysql-DB

Hello list, I keep on getting the error : ERROR[1707] cdr_addon_mysql.c: Cannot connect to database server 127.0.0.1: (1045) Access denied for user 'asteriskcdr'@'localhost' (using password: YES) I have a 'cdr' table in my MySQL-DB. On this table the user 'asteriskcdr' has select, insert,

Re: [asterisk-users] context problem

On 01/20/2011 05:23 PM, Jeroen Eeuwes wrote: Hi Jonas, What else can I try ? Yeah, Asterisk always assumes that from 1 ip address there can only be inbound number. Not very user-friendly. I think I've used something like this: exten => s,1,Set(CALL-TO=${SIP_HEADER(TO)}) exten =>

Re: [asterisk-users] context problem

On 01/20/2011 04:29 PM, Danny Nicholas wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Thursday, January 20, 2011 9:20 AM *To

Re: [asterisk-users] context problem

On 01/20/2011 04:43 PM, Jose P. Espinal wrote: Jonas Kellens wrote: [snip] register => 119909:pas...@sip.prov.org/52525252 register => 119909:pas...@sip.prov.org/59595959 [TRUNKin] exten => _52525252,1,NoOp(context TRUNKin - 52525252) exten => _52525252,n,GoTo(blabla,525252

Re: [asterisk-users] context problem

On 01/20/2011 04:29 PM, Danny Nicholas wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Thursday, January 20, 2011 9:20 AM *To

[asterisk-users] context problem

Hello list, Asterisk 1.6.16.1 I have the following registrations : register => 119909:pas...@sip.prov.org/52525252 register => 119909:pas...@sip.prov.org/59595959 [119909] type=friend host=sip.prov.org username=119909 defaultuser=119909 secret=passwd context=TRUNKin extensions.conf : [TRUNKi

[asterisk-users] audiohook.c: Write factory 0x153cf678 was pretty quick last time, waiting for them

Hello list, what does this mean in the debug-log : [Jan 19 15:11:04] DEBUG[1475] audiohook.c: Write factory 0x153cf678 was pretty quick last time, waiting for them. [Jan 19 15:11:04] DEBUG[1701] audiohook.c: Read factory 0x14fe5ef0 was pretty quick last time, waiting for them. [Jan 19 15:11:04

Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'

On 01/14/2011 02:40 PM, Andrew Latham wrote: On Fri, Jan 14, 2011 at 7:55 AM, Jonas Kellens wrote: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] D

Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'

On 01/14/2011 02:22 PM, Thorsten Göllner wrote: Am 14.01.2011 12:50, schrieb Jonas Kellens: On 01/14/2011 12:44 PM, Thorsten Göllner wrote: Am 14.01.2011 11:55, schrieb Jonas Kellens: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654

Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'

On 01/14/2011 12:44 PM, Thorsten Göllner wrote: Am 14.01.2011 11:55, schrieb Jonas Kellens: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654]

[asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'

Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlo

Re: [asterisk-users] queue_log in MySQL database

On 01/13/2011 05:25 PM, James Lamanna wrote: Hi Jonas, On Thu, Jan 13, 2011 at 8:19 AM, Jonas Kellens wrote: Hello, can /var/log/messages/queue_log be saved in a MySQL database ?? So it would be easier to work with... I don't think Asterisk has this support built-in...mayb

[asterisk-users] queue_log in MySQL database

Hello, can /var/log/messages/queue_log be saved in a MySQL database ?? So it would be easier to work with... Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

[asterisk-users] Fail2Ban & CSF

Hello list, anyone knows if fail2ban works together with CSF (http://www.configserver.com/cp/csf.html) ?? I use CSF for blocking port scanning and blocking of IP-adresses. I wonder if fail2ban will overwrite rules in iptables of CSF and vica versa. Kind regards, Jonas. -- __

[asterisk-users] Show voicemail in GUI

Hello list, I have a management user interface written in php for controlling some functions of Asterisk PBX. I use realtime a lot. Is there a way to easily get the details of a voicemail account and the messages that have been left ? In use realtime voicemail, but how to get the messages

Re: [asterisk-users] DTMF-troubles with Snom

wrote: Jonas What is the dtmf setting on all peers involved in the call? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 *From*: "Jonas Kellens" *Sent*: Wednesday, January 05, 2011 4:55 PM *To*:

[asterisk-users] dtmf-troubles with Snom

Hello list, I'm having DTMF-troubles with a Snom phone. I want to know if it's the Snom or Asterisk that makes the trouble. I'm playing a prompt, then make a choice for "2" : [Jan 5 17:06:38] VERBOSE[29172] file.c: [Jan 5 17:06:38] -- Playing '/var/lib/asterisk/sounds/vprompts/10900

[asterisk-users] DTMF-troubles with Snom

Hello list, I'm having DTMF-troubles with a Snom phone. I want to know if it's the Snom or Asterisk that makes the trouble. I'm playing a prompt, then make a choice for "2" : [Jan 5 17:06:38] VERBOSE[29172] file.c: [Jan 5 17:06:38] -- Playing '/var/lib/asterisk/sounds/vprompts/10900

[asterisk-users] Add Privacy: id to SIP-invite

Hello list, is it possible to add the field Privacy: id to a SIP INVITE message ? INVITE sip:32444666...@1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP1 .2.3.4:5060 From: "R321113" ;tag=2096790244 To: Call-ID: 3b040826e909d311880a009033060...@192.168.12.40

[asterisk-users] Go from CALLINGout to just CALLING

Hello list, how can I go from CALLINGout to just CALLING ? I've tried : exten => s,n,Set(newVAR=${CUT(CALLINGout,,3)}) or exten => s,n,Set(newVAR=$[CUT(CALLINGout,,3)]) But no result : [Jan 4 11:10:12] -- Executing [...@from-s:34] NoOp("SIP/s2-003b", "newVAR=") in new stack Aster

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