Re: [asterisk-users] problem getting dahdi-linux to work with kernel 6.1.0-10

On 7/6/2023 7:08 PM, John Covici wrote: Hi. I have run into a problem compiling dahdi-linux in kernel 6.1.0-10. Apparently there was a change, so I found a patch to fix stdbool.h but now I have an implicit declaration of pci_alloc_consistent in drivers/dahdi/wct4xxp/base.c I don't see any

[asterisk-users] problem getting dahdi-linux to work with kernel 6.1.0-10

Hi. I have run into a problem compiling dahdi-linux in kernel 6.1.0-10. Apparently there was a change, so I found a patch to fix stdbool.h but now I have an implicit declaration of pci_alloc_consistent in drivers/dahdi/wct4xxp/base.c I don't see any other references to that name anywhere. I am

Re: [asterisk-users] Problem with pjsip

On Thu, Jun 8, 2023 at 9:41 AM Yves wrote: > Hello everyone. > I allow myself to submit a problem that I can not solve with my VOIP > provider Orange in France > > [2023-06-08 13:19:03] ERROR[185091]: > res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error > configuring endpoint

[asterisk-users] Problem with pjsip

Hello everyone. I allow myself to submit a problem that I can not solve with my VOIP provider Orange in France [2023-06-08 13:19:03] ERROR[185091]: res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error configuring endpoint 'Biv_Sortie' - 'from_user' field contains invalid character

[asterisk-users] problem with asterisk multiple queues and call assigning

hi, i have 2 queues - queue1 - queue2 1 agent is in both queues queue strategy is rrmemory i have 2 calls waiting call from 12:00 in queue1 from number 777 call from 12:05 in queue2 from number 666 at 12:10 agent is free for next call i have problem in that newer call (call from 12:05) from

Re: [asterisk-users] Problem with IF

Any help? Do I need to post my issue to dev ? On Wed, Jul 14, 2021 at 10:55 AM SAMPro wrote: > Hi > I need to check the return value of a sub, the sub may return empty so I > need to check for that. If the return value isn't empty set another > variable (ARG1) . This is the code I've used in

[asterisk-users] Problem with IF

Hi I need to check the return value of a sub, the sub may return empty so I need to check for that. If the return value isn't empty set another variable (ARG1) . This is the code I've used in extension.conf, but didn't work (the CLI log is after the code). *Extension.conf:* [macro-dial] same =>

Re: [asterisk-users] Problem with OPTIONS requests.

Is `test` your default context (line context= in sip.conf)? If it is not, then try setting context=test in sip.conf and reload it. On Fri, Jul 17, 2020 at 8:34 AM John Kiniston wrote: > I've got this setup in a test context. > > [test] > exten => s,hint,SIP/7124 > exten => s,1,NoOP(Options to

Re: [asterisk-users] Problem with OPTIONS requests.

I've got this setup in a test context. [test] exten => s,hint,SIP/7124 exten => s,1,NoOP(Options to $EXTEN) same => n,Hangup() exten => _x.,hint,SIP/7124 exten => _X.,1,NoOP(Options to $EXTEN) same => n,Hangup() exten => Anonymous,hint,SIP/7124 exten => Anonymous,1,NoOP(Options to $EXTEN)

Re: [asterisk-users] Problem with OPTIONS requests.

In article , John Kiniston wrote: > > I'm implementing a SBC with my Asterisk PBX but the keeps disabling the > trunk group I've configured and I think it may be because Asterisk is > returning a 4r04 to the OPTIONS. > > I've created a test context and have put in a wildcard pattern match to

Re: [asterisk-users] Problem with OPTIONS requests.

Hey John, In one installation I have, we use several monitoring tools (nagios based and custom scripts based) and we have the following: ; Reply OK to SIP:OPTIONS [public] exten => s,1,Wait(1) same => n,Hangup : For Nagios exten => nagios,1,Wait(1) same => n,Hangup NOTES: 1- We have

[asterisk-users] Problem with OPTIONS requests.

I'm implementing a SBC with my Asterisk PBX but the keeps disabling the trunk group I've configured and I think it may be because Asterisk is returning a 4r04 to the OPTIONS. I've created a test context and have put in a wildcard pattern match to try and catch those options but it doesn't seem to

Re: [asterisk-users] problem with logger: syslog vs. file

On Wed, 3 Jun 2020, Fourhundred Thecat wrote: On 2020-06-03 17:21, Steve Edwards wrote: How about:     syslog.local0   = error,verbose,warning no debugging detail.     syslog.local0   = debug,error,verbose,warning include debugging detail.

Re: [asterisk-users] problem with logger: syslog vs. file

> On 2020-06-03 17:21, Steve Edwards wrote: How about:     syslog.local0   = error,verbose,warning no debugging detail.     syslog.local0   = debug,error,verbose,warning include debugging detail. currently, the above has no effect on logging. As

Re: [asterisk-users] problem with logger: syslog vs. file

On Wed, 3 Jun 2020, Fourhundred Thecat wrote: On 2020-06-03 12:18, Tony Mountifield wrote: In article <88f96e46-e6bb-a7ef-bebb-5588ef6cd...@gmx.ch>, However, the conversation would then be: should both logging types include line number and function? should both logging types omit them? should

Re: [asterisk-users] problem with logger: syslog vs. file

> On 2020-06-03 12:18, Tony Mountifield wrote: In article <88f96e46-e6bb-a7ef-bebb-5588ef6cd...@gmx.ch>, However, the conversation would then be: should both logging types include line number and function? should both logging types omit them? should it be a configuration option in logger.conf

Re: [asterisk-users] problem with logger: syslog vs. file

In article <88f96e46-e6bb-a7ef-bebb-5588ef6cd...@gmx.ch>, Fourhundred Thecat <400the...@gmx.ch> wrote: > > On 2020-06-02 17:48, Tony Mountifield wrote: > > In article <94191802-6c9c-bdab-615b-001786a2a...@gmx.ch>, > > Fourhundred Thecat <400the...@gmx.ch> wrote: > >> > On 2019-11-16 03:29,

Re: [asterisk-users] problem with logger: syslog vs. file

> On 2020-06-02 17:48, Tony Mountifield wrote: In article <94191802-6c9c-bdab-615b-001786a2a...@gmx.ch>, Fourhundred Thecat <400the...@gmx.ch> wrote: > On 2019-11-16 03:29, Fourhundred Thecat wrote: case LOGTYPE_SYSLOG: snprintf(buf, size, "%s[%d]%s: %s:%d in %s: %s",

Re: [asterisk-users] problem with logger: syslog vs. file

In article <94191802-6c9c-bdab-615b-001786a2a...@gmx.ch>, Fourhundred Thecat <400the...@gmx.ch> wrote: > > On 2019-11-16 03:29, Fourhundred Thecat wrote: > > Hello, > > > > I am logging directly into file and also to syslog. > > Here is snippet from my /etc/asterisk/logger.conf: > > > >

Re: [asterisk-users] problem with logger: syslog vs. file

> On 2019-11-16 03:29, Fourhundred Thecat wrote: Hello, I am logging directly into file and also to syslog. Here is snippet from my /etc/asterisk/logger.conf: messages => notice,warning,error,verbose syslog.local0 => notice,warning,error,verbose But the logs look different:

[asterisk-users] problem with logger

Hello, I am logging directly into file and also to syslog. Here is snippet from my /etc/asterisk/logger.conf: messages => notice,warning,error,verbose syslog.local0 => notice,warning,error,verbose But the logs look different: VERBOSE[7609][C-0013] pbx.c: NOTICE[3042] chan_sip.c:

Re: [asterisk-users] problem with new install with asterisk 15.7.4

On Mon, Oct 7, 2019, at 10:23 AM, John Covici wrote: > hmmm, is asterisk 16 long term support? I thought only the od > numbered releases were long term support. Asterisk 13 and 16 are both LTS releases. The upcoming 17 will be a standard release. You can always consult the wiki[1] to know what

Re: [asterisk-users] problem with new install with asterisk 15.7.4

hmmm, is asterisk 16 long term support? I thought only the od numbered releases were long term support. On Mon, 07 Oct 2019 08:02:51 -0400, George Joseph wrote: > > [1 ] > [2 ] > Oh, I forgot to mention that Asterisk 15 went End-Of-Life last Thursday. :) > You should use Asterisk 16. > >

Re: [asterisk-users] problem with new install with asterisk 15.7.4

Oh, I forgot to mention that Asterisk 15 went End-Of-Life last Thursday. :) You should use Asterisk 16. On Mon, Oct 7, 2019 at 5:58 AM George Joseph wrote: > > > On Fri, Oct 4, 2019 at 1:19 PM John Covici wrote: > >> Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10 >>

[asterisk-users] problem with new install with asterisk 15.7.4

Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10 system and I am running into the following problem. I need to install meetme (I know its old), and I have dahdi installed and the configure script answers yes to all the edahdi questions, but the app_meetme says depends on

Re: [asterisk-users] Problem with the DB() function (Ira)

>So the new install is coming along. I hooked up the new box for a couple of >hours and got a bunch more problems worked out. And yet some still remain. I >have this subroutine I call occasionally: >. >. >. >Also, when I installed asterisk it did not set itself up to start when the >machine

Re: [asterisk-users] Problem with the DB() function

Hello Jean-Denis, Sunday, March 3, 2019, 11:28:02 AM, you wrote: >> But the third line which should print those 80 characters back to the screen >> prints an empty string. What might I be doing wrong. It's worked from >> version 2 or 3 through 13 but it seems to be broken in 16. > This looks

Re: [asterisk-users] Problem with the DB() function

Le 02/03/2019 à 18:10, Ira a écrit : > exten => 1,1,set(DB(forwards/calls)=${home_in}) >same => n,set(DB(forwards/number)=1) >same => n,verbose(${DB(forwards/calls)}) >same => n,return > > I can see the code running on the console and it prints out the first line > with ${home_in}

[asterisk-users] Problem with the DB() function

Hello Ira, So the new install is coming along. I hooked up the new box for a couple of hours and got a bunch more problems worked out. And yet some still remain. I have this subroutine I call occasionally: exten => 1,1,set(DB(forwards/calls)=${home_in}) same => n,set(DB(forwards/number)=1)

[asterisk-users] Problem receiving calls with Telmex in Mexico...

    Hi.  I am having a problem when trying to receive calls via en E1 from  Telmex using MFC/R2 (MX Variant).  Outgoing calls are fine.  We are using a PBXact system with a Digium TE420 (5th Gen) card.  Here is a log from the call: [10:46:37:707] [Thread: 140631230322432] [Chan 1] - Call

Re: [asterisk-users] Problem with AudioCodes MP-114 ATA

hi, quite unlikely (besides of an defect) that the behaviour of your AudioCodes or Asterisk changed "from alone"... something must have changed. What does the logs say (from asterisk... do you see register-events? and from you AudioCodes?) The AudioCodes Devices can export and restore their

[asterisk-users] Problem with AudioCodes MP-114 ATA

All; I have an AudioCodes MP-114 four FXS ATA that recently stopped registering to my PBX. I'm pulling my hair out here trying to figure out the root cause without much success. Does anyone have a sample config file that I could use as a sample? Any insight at all would be greatly

Re: [asterisk-users] Problem with DAHDI

Luca Bertoncello schrieb: > But if I try to call another VoIP-phone it rings but no voice will be > transferred... Got it! A "little" firewall problem... :( Regards Luca Bertoncello (lucab...@lucabert.de) --

Re: [asterisk-users] Problem with DAHDI

Tzafrir Cohen schrieb: > This means that you have configured a dahdi channel in > /etc/asterisk/chan_dahdi.conf . The default configuration does not > include one. Do you have any DAHDI device on the system? I think not... > If /dev/dahdi/channel itself does not

Re: [asterisk-users] Problem with DAHDI

On Thu, Feb 15, 2018 at 06:55:16PM +0100, Luca Bertoncello wrote: > Hi again! > > I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI > with Armbian/Debian 9. > > First test was to call a test service that say the time. Works! > Second test was to record my voice and play

[asterisk-users] Problem with DAHDI

Hi again! I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI with Armbian/Debian 9. First test was to call a test service that say the time. Works! Second test was to record my voice and play it again. Works! Third test was to call the other VoIP-phone. It does NOT

Re: [asterisk-users] Problem using Android SIP-Client

Harel Cohen schrieb: > Is the Sophos a home router or professional one? In many cases what home Of course the professional firewalls (we have two Sophos in Cluster, to manage our two SDSLs) > router does by default needs to be configured manually on professional one. > E.G.

Re: [asterisk-users] Problem using Android SIP-Client

Hi, Is the Sophos a home router or professional one? In many cases what home router does by default needs to be configured manually on professional one. E.G. a home router will allow outgoing sessions and create a return path by default where professional one won't. Two things I would look for: 1.

Re: [asterisk-users] Problem using Android SIP-Client

the issue is quiet sure codec based: [Oct 24 15:32:41] NOTICE[3731]: channel.c:4280 __ast_read: Dropping incompatible voice frame on SIP/messagenet-028e of format gsm since our native format has changed to 0x8 (alaw) shorter: Dropping incompatible voice frame on SIP/messagenet-028e of

Re: [asterisk-users] Problem using Android SIP-Client

Luca Bertoncello schrieb: Hallo again > I configured an user for my mobile phone and I can call, but as soon > as the other party answer, I get this error in Log: > > [Oct 24 15:32:41] NOTICE[3731]: channel.c:4280 __ast_read: Dropping > incompatible voice frame on

Re: [asterisk-users] Problem using Android SIP-Client

You should try another SIP client, just to check it. (Zoiper or cSipSimple, for example). Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 24

[asterisk-users] Problem using Android SIP-Client

Hi list! I use Asterisk 1.8.30.0 on an OpenWRT-Router (I know, not the last version, but I can't upgrade). It always runned very well, and it runs very well with our home phones, too, but now I have problems using the native Android SIP-Client... I configured an user for my mobile phone

[asterisk-users] Problem with rport (CGNAT) going from Linux kernel 3.16 to 4.9

Hello, I operate an Asterisk server (v11.13.1) on Debian stable, and it's rock-solid. The other day, however, I accidentally upgraded the kernel from the stable 3.16.0 to 4.9.0. Subsequently, audio stopped working. Below you can find my analysis while running the 4.9.0 kernel. 888 is a simply

Re: [asterisk-users] Problem "re-parking" calls

> All; > I have a problem with regards to “re-parking” calls and I was > hoping someone could shed some light on the topic. Consider this scenario: > > (1) An inbound call comes in and the attendant answers it > (2) The attendant places the call on hold and the caller is sent to >

[asterisk-users] Problem "re-parking" calls

All; I have a problem with regards to "re-parking" calls and I was hoping someone could shed some light on the topic. Consider this scenario: (1) An inbound call comes in and the attendant answers it (2) The attendant places the call on hold and the caller is sent to extension 701 (3)

[asterisk-users] Problem "re-parking" calls

All; I have a problem with regards to "re-parking" calls and I was hoping someone could shed some light on the topic. Consider this scenario: (1) An inbound call comes in and the attendant answers it (2) The attendant places the call on hold and the caller is sent to extension 701 (3)

Re: [asterisk-users] Problem setting up ssl connection

On Fri, Oct 28, 2016 at 02:07:24PM +0200, Jonas Kellens wrote: > I use PHP 5.6.27. > > So I should be looking inside php.ini ? Web search: php self signed certificate fsockopen User contributed notes in [1] or the example in [2] 1) http://php.net/manual/en/function.fsockopen.php 2)

Re: [asterisk-users] Problem setting up ssl connection

On 26-10-16 23:24, Stefan Tichy wrote: On Wed, Oct 26, 2016 at 04:57:15PM +0200, Jonas Kellens wrote: if it is indeed manager.conf that I need to edit then the problem is that I see no param : tlsdontverifyserver=yes A comment copied from sip.conf.sample: "If set to yes, don't verify the

Re: [asterisk-users] Problem setting up ssl connection

On Wed, Oct 26, 2016 at 04:57:15PM +0200, Jonas Kellens wrote: > if it is indeed manager.conf that I need to edit then the problem is > that I see no param : tlsdontverifyserver=yes A comment copied from sip.conf.sample: "If set to yes, don't verify the servers certificate when acting as a

Re: [asterisk-users] Problem setting up ssl connection

On 26-10-16 15:03, Dan Jenkins wrote: On Wed, Oct 26, 2016 at 1:46 PM, Jonas Kellens > wrote: Hello I keep getting the following error when trying to connect to the Asterisk server using AMI : $socket =

Re: [asterisk-users] Problem setting up ssl connection

On Wed, Oct 26, 2016 at 1:46 PM, Jonas Kellens wrote: > Hello > > > I keep getting the following error when trying to connect to the Asterisk > server using AMI : > > $socket = fsockopen("tls://11.22.33.44","5039", $errno, $errstr, 5); > > Erorr on CLI : > > [Oct 26

[asterisk-users] Problem setting up ssl connection

Hello I keep getting the following error when trying to connect to the Asterisk server using AMI : $socket = fsockopen("tls://11.22.33.44","5039", $errno, $errstr, 5); Erorr on CLI : [Oct 26 14:38:19] ERROR[2992]: tcptls.c:609 handle_tcptls_connection: Problem setting up ssl connection:

Re: [asterisk-users] Problem with REMAINDER? 957%60 be 15 remainder 57 not 15 remainder -3 ?

I saw this on the bug list first and sent a reply, but for the archives I'll copy it here, too. REMAINDER() calls libm's remainder(3) or remainderl(3), infix % calls fmod(3) or fmodl(3). remainder(3) is defined to round the quotient to the nearest int (always using round-to-even, notsithstanding

Re: [asterisk-users] Problem with REMAINDER? 957%60 be 15 remainder 57 not 15 remainder -3 ?

Yes! That's the one. Thank you. That's a good workaround. The following test dialplan shows the bug (feature?) exten => 7,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN}) same => n,Set(seconds=57) same => n,While($[${seconds} <= 400]); same => n,Set(minutes=$[FLOOR(${seconds} / 60)])

Re: [asterisk-users] Problem with REMAINDER? 957%60 be 15 remainder 57 not 15 remainder -3 ?

All I can tell you is where -3 comes from. >From http://www.voip-info.org/wiki/view/Asterisk+Expressions : REMAINDER(x,y) computes the remainder of dividing x by y. The return value is x - n*y, where n is the value x/y, rounded to the nearest integer. If this quotient is 1/2, it is rounded to the

[asterisk-users] Problem with REMAINDER? 957%60 be 15 remainder 57 not 15 remainder -3 ?

I'm not mathematically gifted, but shouldn't 957%60 be 15 remainder 57? Google and my desktop calculator certainly think so. So where am I going wrong here? The following code exten => 7,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN}) same => n,Set(myNum=957) same =>

Re: [asterisk-users] problem with DTMF detection on calls created with Originate AMI command

i'm using Asterisk 1.6.2.9-2+squeeze12 2016-06-30 22:14 GMT+02:00 Richard Mudgett : > > > On Thu, Jun 30, 2016 at 3:00 PM, nik600 wrote: > >> Dear all >> >> i'm creating an outgoing call to number xxx with this command: >> >>

Re: [asterisk-users] problem with DTMF detection on calls created with Originate AMI command

On Thu, Jun 30, 2016 at 3:00 PM, nik600 wrote: > Dear all > > i'm creating an outgoing call to number xxx with this command: > > http://host:port/mxml?action=Originate=Local/xxx@to-external > =testDTMF=cRETEUNICA=1 > > wich points correctly to this portion of dialplan: > >

[asterisk-users] problem with DTMF detection on calls created with Originate AMI command

Dear all i'm creating an outgoing call to number xxx with this command: http://host:port/mxml?action=Originate=Local/xxx@to-external =testDTMF=cRETEUNICA=1 wich points correctly to this portion of dialplan: [cRETEUNICA] exten => testDTMF,1,Answer exten => testDTMF,n,Read(digito,,1) exten =>

[asterisk-users] Problem sending and receiving faxes

Hi list! I installed Hylafax on a Ubuntu-Server 14.04. On this server runs Asterisk 11.7.0, too and it was configured like my own Asterisk server at home, but it does not work... :( So, I configured Asterisk to connect to Deutsche Telekom and it does! Then I configured iaxmodem to speak with the

Re: [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.

On 2016-02-17 16:28, Richard Mudgett wrote: On Wed, Feb 17, 2016 at 5:56 PM, Ernie Dunbar wrote: On 2016-02-17 15:32, Richard Mudgett wrote: On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar wrote: Hi everyone. We have an Asterisk server running

Re: [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.

On Wed, Feb 17, 2016 at 5:56 PM, Ernie Dunbar wrote: > On 2016-02-17 15:32, Richard Mudgett wrote: > >> On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar >> wrote: >> >> Hi everyone. >>> >>> We have an Asterisk server running Debian Squeeze, with

Re: [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.

On 2016-02-17 15:32, Richard Mudgett wrote: On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar wrote: Hi everyone. We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with some minor source code

Re: [asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.

On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar wrote: > Hi everyone. > > We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 > (basically, the Debian Stable version for Squeeze, but with some minor > source code changes specific to our site). We're

[asterisk-users] Problem compiling res_fax_spandsp.c on Debian server.

Hi everyone. We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with some minor source code changes specific to our site). We're trying to upgrade to 11.13.1 (The Debian Stable version for Jessie), but I've run

Re: [asterisk-users] Problem with Cisco CUBE when dialling into Asterisk 13 server

On Tue, Sep 1, 2015 at 2:02 AM, Brendan Ord wrote: > Hello, > > > > This is a problem with my Cisco CUBE (2811), so apologies for this being > kind of off-topic. It is acting as a border for my Asterisk 13 server > though J > > > > Rather than re-type the details of

[asterisk-users] Problem with Cisco CUBE when dialling into Asterisk 13 server

Hello, This is a problem with my Cisco CUBE (2811), so apologies for this being kind of off-topic. It is acting as a border for my Asterisk 13 server though :) Rather than re-type the details of my problems, I have a post in the Cisco community with running-configs and various debugs

Re: [asterisk-users] Problem no voice

On Wednesday 15 Jul 2015, Luca Bertoncello wrote: But it seems, that I found the problem, adding: disallow=all allow=g729 to the configuration of the peer for this number... You need the following; disallow=all allow=alaw in the configuration for *every* device. There is literally no

[asterisk-users] Problem no voice

Hi list! I have 4 numbers on my Asterisk 1.8. 3 work perfectly, the 4.th not. I'm not sure, when it finish to work, since a month ago it runs without any problem... Well, if I will be called on this number I can't hear anything and in Asterisk I see these: [Jul 15 18:59:55] WARNING[8752]:

Re: [asterisk-users] Problem no voice

jg webaccounts...@jgoettgens.de schrieb: How is the 4th phone configured? It's not a phone, just a number routed on a phone that receives calls for other number, too (without any problem). You could also enable SIP debugging to get more information about the problem. I already set core set

Re: [asterisk-users] Problem no voice

I have 4 numbers on my Asterisk 1.8. 3 work perfectly, the 4.th not. I'm not sure, when it finish to work, since a month ago it runs without any problem... Well, if I will be called on this number I can't hear anything and in Asterisk I see these: [Jul 15 18:59:55] WARNING[8752]:

Re: [asterisk-users] Problem asterisk voicemail message records

On Mon, Jun 8, 2015 at 9:56 AM, Igor Potjevlesch igor.potjevle...@gmail.com wrote: Hello! I've got a little problem with Asterisk (11.14.1), the voicemessages are kinda limited to 40 seconds (average) aproximately; because when a message reach this long I got a cut in the file (*.wav) after

[asterisk-users] Problem asterisk voicemail message records

Hello! I've got a little problem with Asterisk (11.14.1), the voicemessages are kinda limited to 40 seconds (average) aproximately; because when a message reach this long I got a cut in the file (*.wav) after I got this message: WARNING[15035][C-21ef]: format_wav_gsm.c:418 wav_read: Short

[asterisk-users] Problem with NAT - Part 2

Hi again! I decided, just for fun, to install Asterisk on a server of mine (available in Internet) and to log on my mobile phone on this server. This Server communicate with my Asterisk at home and if I call a phone at home from my mobile phone (logged on the Asterisk on the second server), it

[asterisk-users] Problem with SIP-TLS

Hi list! I'm trying to configure my Asterisk to accept SIP-TLS connections, too. I followed this HowTo: http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/ But as soon I try to connect to my Asterisk using SIP-TLS I get on Asterisk-CLI: == Problem setting up ssl

Re: [asterisk-users] Problem with SIP-TLS

ricky gutierrez xserverli...@gmail.com schrieb: compilation problems with the module srtp , check the module module show like srtp Now available on OpenWRT... :( Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ --

Re: [asterisk-users] Problem with SIP-TLS

2015-06-05 14:29 GMT-06:00 Luca Bertoncello lucab...@lucabert.de: I think it is a problem on Asterisk for OpenWRT... :( Regards Luca Bertoncello (lucab...@lucabert.de) compilation problems with the module srtp , check the module module show like srtp -- rickygm

Re: [asterisk-users] Problem with SIP-TLS

ricky gutierrez xserverli...@gmail.com schrieb: Hi lucas , dou you try this: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial Tested right now. Same problem... I think it is a problem on Asterisk for OpenWRT... :( Regards Luca Bertoncello (lucab...@lucabert.de) --

Re: [asterisk-users] Problem with SIP-TLS

2015-06-05 12:21 GMT-06:00 Luca Bertoncello lucab...@lucabert.de: Hi list! I'm trying to configure my Asterisk to accept SIP-TLS connections, too. I followed this HowTo: http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/ But as soon I try to connect to my Asterisk

[asterisk-users] Problem with realtime mysql I can't seem to resolve

Hello I have already several Asterisk servers running with similar configuration, but now I stumble into a problem. I have mysql configuration res_config_mysql.conf : [MyAsteriskDB] dbhost = 127.0.0.1 dbname = MyAsteriskDB dbuser = astadmin dbpass = mysecret dbport = 3306 dbsock =

Re: [asterisk-users] Problem with realtime mysql I can't seem to resolve

On 22May, 2015, at 03:51, Jonas Kellens jonas.kell...@telenet.be wrote: Realtime seems to be loaded : *CLI realtime mysql status general configured for asterisk on socket file /var/lib/mysql/mysql.sock with username asterisk. MyAsteriskDB connected to MyAsteriskDB@127.0.0.1, port 3306

[asterisk-users] problem in h323 trunk to cisco router

hello every body, i want to have h323 trunk between cisco 2800 and asterisk 11.13.1 with ooh323 module. i configured both side and have successful call from cisco to asterisk. but when call comes from asterisk to cisco, my phone rings but no audio is heard and call is disconnected after 5

[asterisk-users] Problem with odbc connector with cdr

Hello, I'm stuck with getting cdr records stored in MySql database. I have a working realtime environment and have verified that the db connection works fine when used via res_config_mysql.conf. I'd appreciate Your help on how to get the odbc connector working as I think there's something wrong

Re: [asterisk-users] Problem with odbc connector with cdr

I notice you have MySQL-asterisk as your definition in your odbc.ini but you are trying to connect to simply 'MySQL' with your 'isql' command. Does isql work with 'MySQL-asterisk' as the DSN instead of simply 'MySQL' ? I have machines that use /etc/odbc.ini and machines that use

Re: [asterisk-users] Problem with odbc connector with cdr

Thanks John, At first got an error using MySQL-asterisk, but then I removed /etc/ ini files and used the DSN in /usr/local/etc/odbc.ini, that did the trick for isql. I must have created the files /etc/ while following a guide online. Nice! After some meddling with the Asterisk conf files to have

Re: [asterisk-users] Problem with Cisco Phones

Apparently this is a known problem past v8 firmware: http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update- version-9/ I've done some more playing about and what I've noticed is that even when using TCP SIP on the 8.x Firmware conferencing doesn’t work - making it use UDP

Re: [asterisk-users] Problem with Cisco Phones

If I remember correctly, 9.x firmware dropped UDP support altogether. On Thu, Jan 22, 2015 at 4:31 AM, Jordan Cook - Gyron Networks jordan.c...@gyron.net wrote: Apparently this is a known problem past v8 firmware: http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update-

Re: [asterisk-users] Problem with Cisco Phones

Next step is packet capture to see if there is a clue as to the cause of the failure in the SIP signalling. On Tue, Jan 20, 2015 at 10:41 AM, Jordan Cook - Gyron Networks jordan.c...@gyron.net wrote: We were using G722 - I thought similarly and tried a call with alaw. Same problem occurred,

Re: [asterisk-users] Problem with Cisco Phones

Next step is packet capture to see if there is a clue as to the cause of the failure in the SIP signalling. Right, I see the following when running SIP Debug. Looks to me like the phones are expecting the server to do the conference mixing, which I guess it would do in CallManager? --- SIP

Re: [asterisk-users] Problem with Cisco Phones

Apparently this is a known problem past v8 firmware: http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update-version-9/ On Tue, Jan 20, 2015 at 11:16 AM, Jordan Cook - Gyron Networks jordan.c...@gyron.net wrote: Next step is packet capture to see if there is a clue as to

[asterisk-users] Problem with Cisco Phones

Possibly slightly off topic, has anyone ever had Cisco 79xx Series phones come up with cannot complete conference errors when trying to conference two calls together? This message may be private and confidential. If you have received this message in error, please notify us and remove it from

Re: [asterisk-users] Problem with Cisco Phones

We were using G722 - I thought similarly and tried a call with alaw. Same problem occurred, any other ideas? I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can only do a single G729 channel, and if you require G729 for the second leg of a conference, it will fail.

Re: [asterisk-users] Problem with Cisco Phones

I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can only do a single G729 channel, and if you require G729 for the second leg of a conference, it will fail. On Tue, Jan 20, 2015 at 10:03 AM, Jordan Cook - Gyron Networks jordan.c...@gyron.net wrote: Possibly slightly

[asterisk-users] Problem with Read() ?

All, I have a weird situation here and haven't been able to turn up any useful information in searches, so I thought I'd post to the list. Essentially, I have a customer who wants us to forward some of their calls to various cell phones. Normally, I'd use FollowMe() for this (that's how most of

[asterisk-users] Problem reload queue dynamical members

Guys, I have a problem. I have a queue on asterisk 1.8 that members are added dynamically via the AMI QueueAdd. When you run the CLI a reload app_queue.so all members who were in the queue disappear. This is a bug or some parameter that I do not know? Would have another way to do the reload queue

Re: [asterisk-users] Problem reload queue dynamical members

On Fri, Jun 6, 2014 at 9:03 AM, Eduardo Leones edua...@ypytecnologia.com.br wrote: Guys, I have a problem. I have a queue on asterisk 1.8 that members are added dynamically via the AMI QueueAdd. When you run the CLI a reload app_queue.so all members who were in the queue disappear. This is

Re: [asterisk-users] Problem reload queue dynamical members

Josh, thanks for the feedback. That problem can also occur with dynamic members, would not be just for those who work with realtime? tks 2014-06-06 10:14 GMT-03:00 Josh Metzger joshdmetz...@gmail.com: On Fri, Jun 6, 2014 at 9:03 AM, Eduardo Leones edua...@ypytecnologia.com.br wrote:

Re: [asterisk-users] Problem building Asterisk-12.2.0

What distro are you building on? CentOS 5.10. Both have the libraries listed in install_prereq. Indeed it has all but 2 or 3 of those libraries (none related to uuid), but after running that script, it was still missing what it needed for uuid. Unfortunately, there's no upgrade path from

Re: [asterisk-users] Problem building Asterisk-12.2.0

Richard Kenner skrev 2014-04-27 12:27: What distro are you building on? CentOS 5.10. e2fsprogs-devel is the package that provides uuid.h on centos 5 /niklas -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Problem building Asterisk-12.2.0

e2fsprogs-devel is the package that provides uuid.h on centos 5 I tried that first and it didn't seem to. I'm pretty sure I needed uuid-dce-devel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

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