Re: [asterisk-users] Problems Solved, two left
Am Wed, May 24, 2023 at 09:40:18AM -0400 schrieb Steve Matzura: > > On 5/24/2023 7:49 AM, Stefan Tichy wrote: > > Am Tue, May 23, 2023 at 07:22:22PM -0400 schrieb Steve Matzura: > > > > > 1. Still can't register my phone > > > The username and password are correct. I don't know what else to try. > > You can start a sip trace from the asterisk console. > > REGISTER sip:192.168.1.185:5060 SIP/2.0 > > Authorization: Digest username="Steve", realm="asterisk", > > [May 24 09:26:13] NOTICE[47903]: res_pjsip/pjsip_distributor.c:676 > > log_failed_request: > > Request 'REGISTER' from .' - No matching endpoint found In the endpoint section there is a parameter identify_by (default: "username,ip"). "username" means, the the from-user is used. Use "yealink" or "Steve" for both user names. Using different names for a phone make it just more complicated. You might have to change the phone configuration. > [yealink] > type = aor > contact = sip:Steve@192.168.1.185 There should be no "contact" parameter for a phone. The phone sends the required information with the register request. -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Solved, two left
Am Tue, May 23, 2023 at 07:22:22PM -0400 schrieb Steve Matzura: > 1. Still can't register my phone > The username and password are correct. I don't know what else to try. You can start a sip trace from the asterisk console. pjsip set logger on There should be a REGISTER from the phone, a Response 401 and an ACK from the phone. Then asterisk should receive another REGISTER with an additional "WWW-Authenticate" header. The response could be 401 again or 403 or something else. > 2. Asterisk can't find the extension in my inbound context. > [May 23 18:34:12] NOTICE[46582]: res_pjsip_session.c:3968 new_invite: > voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected because > extension not found in context 'voipms-inbound'. This can happen if there is no contact_user parameter. "contact_user=" sets the SIP contact header's user portion of the SIP URI this will affect the extension reached in dialplan when the far end calls you at this ; registration. The default is 's'. -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Ready to throw up my hands in defeat
Am Mon, May 22, 2023 at 04:50:49PM + schrieb TTT: > You don't say what happens when you start Asterisk, but I'll assume > your registration with your provider is failing. Some CLI commands are usefull to find out which part of the configration is working. *CLI> pjsip list endpoints the endpoint should be listed as "Not in use" *CLI> pjsip list registrations sipgate/sip:sipgate.de sipgate Registered (exp. 51s) Here the provider is called sipgate. > -Original Message- > > From: Steve Matzura [mailto:s...@noisynotes.com] > Sent: Monday, May 22, 2023 12:15 PM > [voipms-inbound] > > exten => 3115552368,1,Answer() ; fake DID number For a first test you may replace this by that line: exten => _X.,1,Goto(from-internal,100,1) (first test only. That call is not "from-internal") > i.e., where's the routing from the system answering the inbound connect > request to the actions in the extension 100 statements? "Goto" can be used to jump to another context, extension and priority. -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error
On Tue, Jan 26, 2021 at 10:12:22AM -1000, Ruisheng Peng wrote: > The self-sign asterisk.crt: I saved that file in "x.crt". openssl x509 -in x.crt -noout -text RSA Public-Key: (1024 bit) > and Letsencrypt cert.pem: I saved that file in "y.crt". openssl x509 -in y.crt -noout -enddate notAfter=Jan 29 01:24:25 2021 GMT > There were a few mentions of this problem on the web, and one said changing > the security mode of the certs to 755 fixed his problem. That makes no sense. Which version of openssl ist used on that CentOS7 box ? In "/etc/ssl/openssl.cnf" you find something like this: MinProtocol = TLSv1.2 CipherString = DEFAULT@SECLEVEL=2 You could set the level to "1" or even to "0" and restart Asterisk. -- Stefan Tichy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP TLS not working, Asterisk 16.9.0
Hi Karsten, On Thu, Apr 30, 2020 at 05:50:39PM +0200, Karsten Wemheuer wrote: > The server sends Server Hello, Certificate, Server Key > Exchange and Server Hello Done. Something in that packet seems to be unacceptable for openssl 1.1.1d as it is compiled and configured for Buster. Certificate length, Digest algorithm, ... You my change the system default settings at the bottom of "/etc/ssl/openssl.cnf", restart asterisk and try again. Keep in mind that this will affect the whole server. -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] if function when the true value has a colon in it?
On Wed, Feb 20, 2019 at 12:08:14PM -0500, Brian J. Murrell wrote:
> exten =>
> s,n,Set(EXT=${IF($[${SIP}=PJSIP]?${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,PJSIP/,)})}:${ARG2})})
>
> But that ${IF expression?tval:fval} doesn't work because tval has a :
> in it which the if function is taking as the terminator for the tval.
As workaround you could use ExecIf:
ExecIF($[${SIP}=PJSIP]?Set(EXT=.):Set(EXT=...))
--
Stefan Tichy ( asterisk3 at pi4tel dot de )
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trouble removing + sign
On Wed, Feb 13, 2019 at 06:12:58PM -0500, sean darcy wrote:
> same => n,ExecIf($["${CALLERID(num):0:1}" = "+"]?Set(CALLERID(num) =
> ${CALLERID(num):1})
ExecIf($["${CALLERID(num):0:1}"="+"]?Set(CALLERID(num)=${CALLERID(num):1}))
> ExecIf correctly finds the comparison false(the "0"), but still executes the
> appiftrue .
Executes the appiftrue or just shows what could be executed?
--
Stefan Tichy ( asterisk3 at pi4tel dot de )
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disable asterisk ssl how to
On Wed, Aug 08, 2018 at 04:30:52PM -0400, Saint Michael wrote: > I am trying to install Asterisk 11 on debian 9, ... Did you install libssl-dev (Version 1.1) or libssl1.0-dev ? -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Start audio call and enable video later
Hi, when some phone inititates an audio call and sends a re-invite with audio and video later, asterisk will send a "200 OK" response. The response contains sdp with audio and video enabled. But that seems to be all asterisk does. No other action and no re-invite to the second phone. Is this a bug or is just some feature not available? I tried with different versions of asterisk and used both chan_sip and pjsip. One of the phones used is linphone (3.6.1-3). Thanks in advance. -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk TLS 5061 not listening
On Mon, May 28, 2018 at 09:26:32AM +0200, Benjamin Marty wrote: > I'm currently trying to setup an Asterisk Box with a Let's Encrypt > certificate. > > I merged privatekey, cert and chain to one file: This is not necessary. You could use tlscertfile and tlsprivatekey. > tlsenable=yes > tlsbindaddr=0.0.0.0 > tlscertfile=/etc/asterisk/tls/a-keycert.pem > tlscipher=ALL > tlsclientmethod=tlsv1 From the ChangeLog (Asterisk 13): "Consequently please, specify 'tlsclientmethod=tlsv1' in your sip.conf only if you face a server which has problems like not falling back to TLSv1.0 automatically." > But somehow my Asterisk doesn't even start to listen on the SIPS port Are there any error messages or warnings? Which asterisk version is used? -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 100% CPU after upgrade.
On Mon, Apr 03, 2017 at 05:45:58PM -0400, Mike Diehl wrote: > The system is a 2.4Gh quad-core Xenon with 4G of RAM, so it should have > plenty Not much RAM, but as long as you don't see high iowait this should be no problem. Here I have a "Intel(R) Xeon(R) CPU X3430 @2.40GHz". > I honestly don't see any reason for this server to eat 100% of it's cpu, and > am hesitant to roll it out to production until I understand why it is. Is it 100% for each CPU core or just 100% for asterisk listet by top? Are debugging compiler flags enabled (DEBUG_THREADS) ? -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which tool to automatically restart Asterisk ?
On Fri, Feb 17, 2017 at 04:38:47PM +0100, Olivier wrote: > 3. For systemd envs where /etc/init.d files are not used anymore, what do > you recommend ? Even if /etc/init.d/ files are still used for other services, asterisk may have a unit file in /etc/systemd/system/ [Unit] Description=Asterisk . [Service] Restart=on-failure RestartSec=15s ExecStopPost=-/usr/local/bin/send-some-mail . -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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) http://i.justrealized.com/2009/allowing-self-signed-certificates-for-pfsockopen-and-fsockopen/ I did not test this, but I think it might be usefull. -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 client." With AMI connections asterisk is allways the server. > I don't know how to make the AMI ignore the self-signed certificate. The client fails to verfify the certificate. Do you use PHP 5.6? The default behavior has changed. -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TLS problem
On Fri, Aug 26, 2016 at 12:36:05PM +0200, hw wrote: > Jonathan H schrieb: > >Well, what immediately stands out is: > >"FILE * open failed!" > > Yes, and it doesn´t say which file cannot be opened. I even looked at > the source and found that at that point, you can´t simply add some > debugging output to find out. It is not a file on your computers disk. It is a connection attempt from one of your phones. If a phone is configured to use tcp without tls and connect to the port used for tls (default 5061) you will see a message like this. (just an example - different situation here) "WARNING[1170]: tcptls.c:673 handle_tcptls_connection ..." Handle ... connection ... "Problem setting up ssl connection: error::lib(0):func(0):reason(0)" ERR_get_error returns 0 if there is no error in the queue. According to the description [1] this may happen and it would be possible to find out more about the error. Try to connect using "openssl s_client". 1) https://www.openssl.org/docs/manmaster/ssl/SSL_get_error.html -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] nagios asterisk check SIP
On Fri, Jun 17, 2016 at 11:22:48AM +0200, Thomas wrote: > Iam loocking for an programm to check the SIP port of an Asterisk asterisk. > > Ome time ago I have used > #/usr/bin/sipsak > but it seemed that it is not working anymore? What is the problem with sipsak? /usr/bin/sipsak -s 'sip:10@somehostname' -r 5060 -vv -S -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI: check if the user has a Mailbox
On Thu, Apr 21, 2016 at 09:34:47PM +0200, Luca Bertoncello wrote: > On an Asterisk-Server I have some users. Just two of them have a Mailbox. > I want to write a little Web interface to manage many things and I'd like to > have a menu point for the voicemail, but just if the user has a Mailbox. Action: Getvar Variable: VM_INFO(222,exists) -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP realtime: lots of problems
Hello Michael On Thu, Oct 08, 2015 at 01:32:07PM -0400, Michael Ulitskiy wrote: > > extconfig.conf: > [settings] > ps_endpoints => pgsql,users,pjsip_endpoints_v Does it change anything if you use odbc instead of pgsql? I did some testing with chan_sip/pgsql and had much less problems when pgsql was replaced by odbc. > 1. The biggest problem: if I have small number of endpoints (roughly up to a > 100) then > asterisk loads ok and pjsip seems to be working ok (with other problems > described below). > If I have larger number of endpoints (several hundred) then intermittently > (but often) asterisk > just hangs during loading. Attempting to start asterisk with console (-c) it > never reaches the user > prompt. pjsip isn't functional (doesn't reply to any sip messages). Did you try "core show locks" in this situation? -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Voicemail MWI
>From SIP-debug output I have a little more information. In one case the first subscribe after a restart was rejected: SIP/2.0 481 Call/Transaction Does Not Exist Later subscribes from the same phone needed authentication and after this where rejected: SIP/2.0 404 Not found (no mailbox) There seems to be another problem. Even if the mailbox information is available, the phones are not notified immediatly when a new voicemail has arrived. The notification is delayed until the phone subscribes again. -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Voicemail MWI
On Wed, Sep 16, 2015 at 04:44:45PM -0400, Nick Olsen wrote: > I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, > These are loaded into asterisk without the mailbox info. Leading to > "Received SIP subscribe for peer without mailbox" notices. And non-working > MWI. > > Occasionally, It just works. But only on a peer or two at a time. And > it'll stop working after a few minutes. Here it seems to be the other way round. Occasionally I see that peers have lost there mailbox setting and don't get notify messages with voicemail information. It is Asterisk 13.5.0 "sip prune realtime peer ..." "sip show peer ... load" After this the setting is restored, but until now I have no idea why this happens. The database field mailbox remains unchanged. Could you post the Realtime SIP Settings? -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 11.14 - voicemail incorrect duration
Hi Dominique On Mon, Jan 26, 2015 at 04:37:23PM +0100, Dominique Haeber wrote: So, from 15:24:04 to 15:24:10 there are 6 seconds. But asterisk only count 2. What can be the reason? It is not silence. Are you sure? The value for silencethreshold (140) is unusually large. -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Hi, some more information could be usefull. On Wed, Apr 30, 2014 at 12:15:03PM +0200, Administrator TOOTAI wrote: after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like sip reload or iax2 reload does nothing. Is Asterisk fully booted? There should be such a message for each AMI connection. Is it possible to unload chan_sip.so and to load it again? Are there error messages related to sip and iax modules? -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to tell Asterisk to to send Ringing signals as into RTP
On Wed, Jan 15, 2014 at 03:11:46PM +0100, Olivier wrote: 2. Are the setings bellow sufficient to implement the above method (from experience, I've gathered mixed results and I would appreciate any input that would confirm I'm on the right or the wrong track) or shall add more magic somewhere (Answer(), Progress(), ...) ? Dial(SIP/foo,45,r(ring)) Asterisk will start sending audio packets to the caller before the callee answers. sip.conf: progressinband=yes progressinband=never -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up realtime config.
On Sun, Oct 10, 2010 at 06:29:27PM -0600, Mike Diehl wrote: Asterisk replied: Peer test not found. So it looks like I'm missing something pretty basic. I would suggest to check extconfig.conf. -- Stefan Tichy ( asterisk2 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI 2.2.1, Asterisk 1.6.2.6 - Channel unacceptable (6)
Hi, it was some configuration error. I droped the old config and started with the sample file to build a new one. Therefore I do not know which parameter in chan_dahdi.conf caused the problem. Anyway, now it is working. The only remaining problem is that no caller ID is available for incoming calls. -- Stefan Tichy ( asterisk2 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI 2.2.1, Asterisk 1.6.2.6 - Channel unacceptable (6)
Hi, we have a problem with a Digium TE122 card, DAHDI 2.2.1 and Asterisk 1.6.2.6 in the UK. Outgoing calls do work without problems, but each incoming call fails: Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Originator) Message type: RELEASE (77) [08 02 82 86] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Channel unacceptable (6), class = Normal Event (0) ] Processing IE 8 (cs0, Cause) q931.c:3801 q931_receive: call 1 on channel 1 enters state 0 (Null) Channel 0/1, span 1 got hangup, cause 6 The dial plan just consists of a Ringing() and a Wait(2). Most configuration errors probably should affect outgoing calls as well. A working configruation example would be very helpfull. Any help appreciated -- Stefan Tichy ( asterisk2 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.0-beta4 - Monitor / MixMonitor Recording
On Thu, Aug 20, 2009 at 08:45:58PM +0200, Stefan Tichy wrote: Recording a call without mixing fails. User hit '*1' to record call. filename: wav,auto-1250793354-24-22,m Not even the Monitor application or the Monitor manager command do work. ast_monitor_change_fname seems to return -1 all the time without printing the warning Cannot change monitor filename of channel ... -- Stefan Tichy ( asterisk2 at pi4tel dot de ) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.2.0-beta4 - Monitor / MixMonitor Recording
MixMonitor seems to work: -- User hit '*3' to record call. filename: auto-1250792853-24-22 == Begin MixMonitor Recording SIP/snom2-084c4ec8 /var/spool/asterisk/monitor/auto-1250792853-24-22.wav exists now. Recording a call without mixing fails. User hit '*1' to record call. filename: wav,auto-1250793354-24-22,m TOUCH_MONITOR_OUTPUT is set to auto-1250793354-24-22 but /var/spool/asterisk/monitor/ does not contain any new file. Dial options are twhx and in features.conf there is: [featuremap] automon = *1 automixmon = *3 Is there anything else to follow to get monitoring working for asterisk 1.6 or is it a bug in the 1.6.2.0-beta4 ? Thanks in advance -- Stefan Tichy ( asterisk2 at pi4tel dot de ) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ztdummy and Asterisk
On Sat, Nov 01, 2008 at 11:03:09PM -0200, Aldo D. Sudak wrote: Loading zaptel hardware modules: ztdummy. Running ztcfg: done. There is no need to run ztcfg if you just want to use ztdummy. The call to ztcfg is probably just part of some standard init script. -- Stefan Tichy ( asterisk2 at pi4tel dot de ) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OpenStage20 Problem
Hi, is anyone Siemens OpenStage 20 SIP phone connected to asterisk 1.4 ? Since V1 R4.11.0 the phone shows Number unavailable each time an outgoing call gets connected. To users this looks like an error message. It is a bit confusing. This problem did not occur when V1 R3 was used, but this had a lot of bugs. -- Stefan Tichy ( asterisk2 at pi4tel dot de ) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mpg123 problem
On Sun, Jun 22, 2008 at 12:24:22AM -0700, fateme fatah wrote: I want to install mpg123-0.59r on my asterisk server.I downloaded it in /usr/src then untared it and I typed these command : Just have a look at www.mpg123.org and fetch the up to date version. 0.59r is probably available with your distribution but it is known to cause some problems. -- Stefan Tichy ( asterisk2 at pi4tel dot de ) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ResetCDR Options v, a - Asterisk 1.4
show application ResetCDR shows 3 option values which may be used. w works as expected, but what about v and a? v -- Save CDR variables - but theese are saved anyway a -- Store any stacked records - but what are stacked records? I found some ResetCDR examples using option w or no option. In which situation one of the other options does make sense? Any help appreciated -- Stefan Tichy ( asterisk at pi4tel dot de ) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4, h.323, OpenCom 1010
Hi, there is an OpenCom 1010 (Software Version 4) acting as Gatekeeper and configured for h.323 connections. Asterisk 1.4.11 with ooh323 from addons 1.4.2 detects the gatekeeper, but until now phonecalls are not possible. To be accurate one test call was successfull, but I did not manage to get it working again. Is it a good idea to use ooh323 in this situation? Should I try some other h.323 module? Is there anyone who has this combination working? Tanks in advance -- Stefan Tichy ( asterisk at pi4tel dot de ) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Non-USASCII chars in sip.conf?
On Fri, Sep 28, 2007 at 03:40:09PM +0200, Per Jessen wrote: This must have been asked before, but googling didn't help much. How do I define a callerid that contains non-USASCII characters? E.g. ä, ö, ü, å, ø, æ etc. ? Use UTF-8 Encoding. -- Stefan Tichy ( asterisk at pi4tel dot de ) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_alsa - no sound / strange sound - 1.4.9
Hi some problem with chan_alsa. Depending on the configuration I don't get any sound output (output_device not set in alsa.conf - same as output_device=default) or very strange output (output_device=hw:0,0) when dialing into something like exten = 10,1,Answer exten = 10,n,Playback(soundfile) exten = 10,n,Hangup Other alsa applictions do work without problems and for example this simple test does work: aplay -D default /usr/share/sounds/alsa/Front_Center.wav Installed is Asterisk 1.4.9 on a box with (standard) Debian Etch. There is no /etc/asound.conf or ./asoundrc. It is only one audio controller available. Any help appreciated -- Stefan Tichy ( asterisk at pi4tel dot de ) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_pppd - Could not read send data
Hi, did anyone managed to get chan-capi and app_pppd to work? Incoming call is accepted, pppd started, but no data transfered to pppd. I used app_pppd-060822.tgz, chan-capi 0.7.1, asterisk 1.2.13. Error messages: chan_capi.c:918 local_queue_frame: Could not write to pipe for ISDN1#0 DEBUG[3364] app_pppd.c: Could not read send data: Input/output error DEBUG[3364] app_pppd.c: Cancelling threads DEBUG[3364] app_pppd.c: pthread_join(info-thread_run, NULL) returned 0 -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: sending fax with chan-capi
On Wed, Oct 11, 2006 at 11:32:57AM -0400, Jens Vagelpohl wrote: The call file created by the outgoing script file2fax.py specifies 3 retries in case of failure. Fax may fail even if the phone call was successfull. This just retries it within Asterisk, I don't know if I could have chan_capi do that. chan_capi 0.7 does set some variables which can / should be used in the dialplan (FAXSTATUS, ) -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: How to manipulate a plus in a phone number
On Tue, Sep 05, 2006 at 11:21:43AM +0100, Tim Panton wrote:
exten = 1,n,set(INNAT=${REGEX(^\+,${ATELNO})})
exten = 1,n,Set(PLUS=\\+)
exten = 1,n,set(INNAT=${REGEX(^${PLUS} ${ATELNO})})
If it is an extension, this should work too
exten = _+.,1,Goto(011${EXTEN:1},1)
--
Stefan Tichy [EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Executing a Function from AGI
On Fri, Jun 16, 2006 at 09:20:18AM -0600, Douglas Garstang wrote: Oh... Thanks... This doesn't seem to be documented anywhere. Where did you find out about this? The output of dump agihtml gives some hint Understands complex variable names and builtin variables, unlike GET VARIABLE. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: ISDN BRI NetJet
On Sat, Jun 17, 2006 at 02:33:26PM -0300, Hermann Wecke wrote: I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1 It could work with the deprecated chan_modem. Don't wast your time. Anyone was able to use this card with asterisk? I couldn't find much information about it. Any help? Replace it by some Cologne Chip Card. The single port card is cheap. Than you can use bristuff, chan_misdn or visdn. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Executing a Function from AGI
On Thu, Jun 15, 2006 at 03:21:32PM -0600, Douglas Garstang wrote:
I've tried this:
EXEC Set DIALPATH=${DUNDILOOKUP(2944093|180net)}
and also:
SET VARIABLE DIALPATH ${DUNDILOOKUP(2944093|180net)}
in both cases, DIALPATH is set to a literal ${DUNDILOOKUP2944093|180net}
get full variable evaluates a channel expression, but set
variable cannot be used this way.
Use GET FULL VARIABLE to get the value and then use SET VARIABLE
to store this value in the DIALPATH variable.
--
Stefan Tichy [EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk perms in manager.conf
On Wed, Mar 22, 2006 at 05:54:27AM -0500, David Hajek wrote: [public] secret = private deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/255.255.0.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Lets say I want some users to use dial through manager interface. But don't want to allow them to run asterisk commands? read = write = call That is sufficient, but if you use AstTapi to dial from outlook authorization for additional classes is necessary. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 190 occasionally NR, SIP 401
Snom 190 phone (snom190-SIP 3.60k) occasionally gets SIP 401 response from Asterisk 1.2.1 server. A few minutes later is registered again. It happend at least two times since Asterisk version 1.2.1 is used at the server, but I am not shure if the problem already existed before this update. Has anyone encountered a similar problem? -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Eicon DIVA Server V-BRI questions
On Mon, Dec 26, 2005 at 02:51:39PM +0100, Jens Vagelpohl wrote: My second question remains: Do I need BRIStuff? I guess I don't No, you don't. You have to choose one (and only one) asterisk channel module ( chan_modem / chan_capi / chan_misdn / chan_zap(britstuff) / chan_visdn / chan_sirrix ) Since you bought a Eicon Diva Server card you have to use chan_capi. IMHO you should use current CVS source from http://sourceforge.net/projects/chan-capi/ (or wait for chan_capi-cm-0.6.2) seeing how defining a simple extensions context with just Answer() and Echo() works through chan_capi when I tell Asterisk not to load any of the chan_modem* and chan_zap, and the zaptel module is unloaded... :) Just put noload statements in modules.conf. ztdummy and zaptel kernel modules are necessary if you want to use conferencing or iax trunking. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Eicon DIVA Server V-BRI questions
On Mon, Dec 26, 2005 at 03:47:48PM +0100, Jens Vagelpohl wrote: I downloaded version 0.6.1 and in my extremely limited testing this seemed to work OK. I can switch over to the current CVS HEAD if you think 0.6.1 has issues. Are there any? deadlock in faxreceive was the problem that forced me to update, but since you bought a V-Bri this should not be an issue in your situation. If you don't have problems using 0.6.1 there is no need to update, but cvs log chan_capi.c does list several modifications and it will be easier to update than to check each of them. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan-capi avm b1 and capi.conf problems
On Thu, Dec 15, 2005 at 02:09:59PM +0100, Ricardo wrote: The main problem is that can load asterisk (asterisk -vvvcg) but i get just one error message and, off course isdn do not work: Dec 15 11:13:25 ERROR[4887]: chan_capi.c:4835 load_module: Unable to load config capi.conf, CAPI disabled You need /etc/isdn/capi.conf and /etc/asterisk/capi.conf. The second is used by the asterisk module chan_capi. Check file permissions and content of /etc/asterisk/capi.conf. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk fax
On Sat, Nov 26, 2005 at 10:37:44AM -0500, Tom Rymes wrote: More specifically, you can make it work using an ATA or a TDM400P card with an fxs port, but it is not likely to be reliable. If you TDM400P (FXS) and some ISDN quad bri Card works fine for approximately 10 months in a small company. No VoIP involved. Threre is no guaranty, but it seems to be reliable enough. If the ISDN card has one port in NT mode you could / should use some a/b converter connected to this port. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.2.0-beta1 and Hints (Re: CVS HEAD and Hints)
Hi, On Thu, Oct 06, 2005 at 08:07:07AM +0330, Paradise Dove wrote: i was just wondering that is there anybody who has any success with hints on CVS HEAD? a sample configuration of sip.conf and extensions.conf is pleased. I am using asterisk-1.2.0-beta1. If the monitored phone makes a call this is notified to a snom 190 and the light goes on and off. If the monitored phone is ringing and/or someone picks up the phone nothing happens. Not shure if it is a bug in the code or in my configuration. According to the Update Aug. 2005 (for Asterisk 1.2.0) section in http://www.voip-info.org/tiki-index.php?page=Asterisk+standard+extensions incominglimit should be configured. According to http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+incominglimit incominglimit and outgoinglimit are deprecated. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: 1.2.0-beta1 and Hints (Re: CVS HEAD and Hints)
On Thu, Oct 06, 2005 at 10:35:46PM +0200, Olle E. Johansson wrote: incominglimit is replaced by call-limit. Please read sip.conf.sample. Outgoinglimit has not worked for ages, so we removed it. One limit works for both incoming and outgoing calls now. sip.conf.sample available in 1.2.0-beta1 lists incominglimit and outgoinglimit, but it is different in the current CVS head. I appreciate that hint, but it does not help me in getting dialplan hint working. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Best drivers for HFC-S ISDN cards
On Mon, Sep 26, 2005 at 02:23:04PM +0100, Chris Bagnall wrote: It seems that HFC-S cards can be connected with asterisk in a few different ways - isdn4linux, mISDN, chan_capi or zaphfc. mISDN (kernel modules and user lib) is used by chan_misdn and chan_capi. vISDN might be another option. I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). There are some Audio Quality Tuning Options for zap channels. What's the recommended way to hook up these ISDN cards? Is switching to capi or mISDN likely to remove the echo problem completely, or is this one of those things one has to accept? chan_misdn, chan_capi (echosquelch) and vISDN include a very basic echo cancellation or non echo cancellation. IMHO bristuff is the best (least worse) choice. chan_modem/isdn4linux will cause additional delay and will itensify the echo problem. Do not use reply if you want to start a new thread. The header In-Reply-To: is used by thread-aware mail clients even if the subject has been changed. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SMS using a PRI channel
Hi, On Wed, Sep 14, 2005 at 04:53:54PM +0200, Roger Schreiter wrote: I have some experience in sending SMSs using smsclient. I call the german Vodafone SMSC (01722278020), and smsclient takes approx 20 secs to send a SMS. The hardware is an Sedlbauer ISDN card. smsclient seems to be similar to yaps. http://www.smsclient.org http://freshmeat.net/projects/yaps/ Now, I want to do the same using asterisk and a digium PRI card. It is different. Asterisk sms application is designed for a protocol described in ETSI ES 201 912. (German: Festnetz SMS) I dialed using the manager with: action: originate channel: Zap/g4/01722278020 ... I assumed, the call will fail, because the remote end will become signalled a voice call, and imho the SMSC wouldn't answer a voice call, but expects data calls. It is a voice call and it has to be a voice call. FSK (Frequency Shift Keying) is used for modulation. Did anybody try sending SMS to german Vodafone or other SMSC mentioned in the smsclient package? Send the SMS using Telekom 0193010 gateway. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP audio port usage
On Mon, Sep 19, 2005 at 12:22:35PM -0400, Adrien Laurent wrote: I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? RTP protocol is used for audio. Port range is defined in /etc/asterisk/rtp.conf -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: ztdummy configuration help
On Mon, Sep 19, 2005 at 01:53:16PM -0400, kurt x wrote: Upon setting up and configuring the my extension.conf, meetme.conf and following the instruction outlined at this web page: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy I get the following errors when calling the meetme number. Executing Wait(SIP/216.53.118.2-f41196e0, 1) in new stack -- Executing MeetMe(SIP/216.53.118.2-f41196e0, |sicp) in new stack -- Playing 'conf-getconfno' (language 'en') == Parsing '/etc/asterisk/meetme.conf': Found Sep 19 13:51:22 WARNING[14066]: chan_zap.c:757 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory The device /dev/zap/pseudo (and probably ctl, timer, channel) is missing. Asterisk process needs write permissions for this device. If you are using kernel 2.6 and udev check README.udev (as described in the wiki mentioned in your mail) -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk connected to Concept XI520
Hi Asterisk 1.0.9-BRIstuffed-0.2.0-RC8m is connected to a T-Concept XI520 System. Phone calls on both directions do work, but transfers are not possible. Asterisk recognizes that some sip phone requests a transfer. Is it possible to forward this transfer request to the XI520? Users of analog and ISDN phones have to use the R key. Any hints? -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Strange problem with Bristuff
On Fri, Sep 02, 2005 at 08:46:10AM +0200, Thomas Petersen wrote: [EMAIL PROTECTED] wrote: The caller digit 123456 (at the fourth digit connect get the ring back) exten = _1234,1,dial,sip/25 exten = _123456,dial.sip/30 in this way the other hand get the first estension and does not pass the call to the extension Try switching the two lines around. When dialing 123456, the first line might also match that number. By the way it looks like you are missing the priority in the second line. No, this is not a solution. Check Controlling Sort Order at http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting This solution will work if the complete number is submitted in one block. IMHO there is no solution if it is ISDN PTP (in germany called Anlagenanschluss). Omit 1234 if you want to use _1234. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Strange problem with Bristuff
On Tue, Aug 30, 2005 at 07:45:50PM +0200, [EMAIL PROTECTED] wrote: I have a strange problem with a quadbri card and my asterisk box with installed verson 1.0.7 of asterisk Bristuffed. I have connected to the card 3 isdn in ptp mode configured in selection passing (I don't know if is exact the english traduction but I have 3 isdn with 99 numbers and asterisk forward the extensions) The problem is this: if I call from a cellular to asterisk all is Ok but when I try to call from a fixed line the extension (the last part of the number) is not sent but only the first part. It might be a pattern matching problem in your dial plan. The digits used as extension (0,..,9,00,...,99) might be transmitted one by one or as block. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: gotoiftime
On Sat, Aug 27, 2005 at 07:41:55PM -0600, Damon Estep wrote: Does anyone know if gotoiftime can take any subset of 7 days for the days of the week or only a contiguous range? According to voip-info.org it has to be one value, a range or '*'. It is not possible to use a list of values. Each of the elements may be specified either as '*' (for always) or as a range. http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+GotoIfTime -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: DIALSTATUS for Originate Command
On Sun, Aug 28, 2005 at 10:45:18AM -, saket setu wrote:
I am trying to use the originate Command from the Asterisk manager on both
SIP and ZAP.
The command works successfully but does not return any DIALSTATUS...
Response: Success
Message: Originate successfully queued
Indeed this response to a originate manager command is not what you
may have expected. You can listen to the events provided by the
manager interface and wait for something like this:
Event: Newstate
Channel: SIP/201-
State: Up
I have also tried to write a dial plan in a manner such that i originate a
call
to my internal extension and jump to a context in the dial plan and execute
the Dial command and trap all the statuses but this also does not work and it
straight away bridges my internal extension to the external call without
returning any dial status.
No dial plan application will be executed until someone picks up at
the sip phone. If the dial plan gets involved the sip channel has
been answered.
Here is the example of what i did:
1. Originate:
Action: Originate
Channel: SIP/201 (Internal extension)
Context: Airtel
Extension: 26191341(External PSTN Number)
Priority: 1
2. Dial Plan :
[AIRTEL]
exten = _XX.,1,Dial(ZAP/${EXTEN},15,t)
exten = _XX.,2,NoOp(${DIALSTATUS})
exten = _XX.,3,Goto(_XX.-${DIALSTATUS},1)
The target extension looks strange. But do not use
Goto(${EXTEN}-${DIALSTATUS}) here. _XX. will match this and you
get an endless loop.
--
Stefan Tichy [EMAIL PROTECTED]
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Automatic start with SuSe linux
On Wed, Aug 17, 2005 at 02:11:09PM +0100, Angus Comber wrote: You could just add the line asterisk to /etc/init.d/boot.local Excerpt from /etc/init.d/boot.local # Here you should add things, that should happen directly after # booting # before we're going to the first run level. Do not attempt to start asterisk here. There is some SuSE asterisk rpm available for SuSE 9.3. It is asterisk 1.0.6, but you can extract the boot script without installing anything else. rpm2cpio asterisk-1.0.6-4.i586.rpm | cpio -i -d -v './etc/init.d/asterisk' Modify the script and copy it to /etc/init.d/. SuSE program insserv can be used to add the symlinks to /etc/init.d/rc3.d/ directory. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: snom mass deployment - settings via DHCP
I have to adjust my last statement. If Setting URL field of advanced.htm webinterface is empty the value of tftp-server-name is used. option tftp-server-name http://192.168.100.1;; On reboot the phone sends two requests to the specified IP: GET /download/snom190.htm GET /download/snom190-mac.htm bootfile-name does indeed seem to be ineffective. I tried using snom 190 with 3.60b firmware and a dhcp-3.0.1 server. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: snom mass deployment - settings via DHCP
Hello, On Fri, May 06, 2005 at 02:13:02PM +0200, Nils Ohlmeier wrote: Regarding the real topic: did you already read our white-papers about mass-deployment and setting up snom phones? http://www.snom.com/white_papers.html My Snom 190 gets several options via dhcp, but defining the setting server URL does not work (in my environment) .. can be set ... automatically via DHCP (options 66 and 67) ... /etc/dhcp/dhcpd.conf 66: tftp-server-name 67: bootfile-name Is this correct? If not, do you have a working example? -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_capi crashes asterisk
On Fri, Apr 29, 2005 at 01:44:24AM +0200, Sebastian Voitzsch wrote: I can?t get chan_capi to work with any version of asterisk. I tried several versions, all with the same effect: the phone rings, as soon as the call gets answerd, asterisk crashes. Certainly it is chan_capi 0.3.5, but which kernel and libcapi (capi4linux) versions are used? -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Outgoing faxes with chan_capi?
On Tue, Apr 05, 2005 at 01:46:34PM +0800, Andrew Furey wrote: The wiki comments mention CapiSuite, but that appears to be a completely separate package to Asterisk, so I'd (I think) need to fully shut down Asterisk, send the fax, then start it up again? Not No, this is not neccessary. If you don't want to install CapiSuite you may use c2faxsend from the capi4hylafax package (currently have no Debian up and running - just checked on a SuSE 9.1). If a tiff file is used it has to be in an appropriate format. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: mISDN + chan_misdn and DTMF
On Mon, Apr 04, 2005 at 07:46:49PM +0200, Alex wrote: Asterisk CVS-HEAD 30/3/05 + mISDN + chan_misdn = everything works fine but outgoing DTMFs are not sent to the called party when placing calls from SIP clients to PSTN (ISDN). On the contrary, incoming calls from PSTN can send DTMF to Asterisk. I've already tried every combinations of dtmfmode (inband, rfc2833, info) either in sip.conf, either in clients options, with all codecs. I've also tried setting SIPDtmfMode() before dialing, but had no success. Turn on inband DTMF on your SIP device, this works. (copied from http://www.beronet.com/bugs/ bug_id 10 ) Is the problem specific to CVS-HEAD 30/3/05 ? If you think it is a chan_misdn problem, please use beronets bug tracker. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Compilation problem chan_capi and Eicon Diva 4Bri
On Thu, Mar 17, 2005 at 12:02:43PM +0100, Kib Eki wrote: I found the chan_capi for asterisk from www.junghanns.net. Also loaded the patch and applied to the chan_capi source tree. The patch - is it this one? http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 I changed the Makefile to include the capi20.h from eicon: INCLUDE=-I$(ASTERISK_HEADER_DIR) -I/usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon It should compile if /usr/include/capi20.h (capi4linux package) is used. Why do you want to use the kernel header file? Unfortunately it is german system so also the compiler errors are in german. Unset LANG and try again. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Compilation problem chan_capi and Eicon Diva 4Bri
On Fri, Mar 18, 2005 at 12:14:14AM +0800, Craig Guy wrote: Upgrade to kernel 2.6.9, there are supposed to be significant bugfixes for CAPI support in 2.6.9. All of my CAPI systems use FC2, 2.6.9. I tried to go 2.6.10 but had problems. What kind of problems. Could you give us some hint? Thanks in advance -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: possible bug in chan_capi concerning context handling
Hello, On Sun, Mar 13, 2005 at 12:21:42PM +0200, Dimitris Kounalakis wrote: I am trying to configure asterisk 1.0.7pre to get incoming calls from an ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is that the context is not recognised in the /etc/asterisk/capi.conf Is this problem specific to asterisk 1.0.7pre? The statement order in /etc/asterisk/capi.conf may be relevant. Could you send the complete section of capi.conf. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zyxel P2000W - CallerId
Caller Name set using SetCIDName or SetCallerID is not displayed by Zyxel P2000W (Firmware VWJ000F). The same problem has been mentionend before, but I did not find any solution or hint. http://lists.digium.com/pipermail/asterisk-users/2005-January/082801.html -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi - fax patch - crash
WARNING[pid]: CAPI[contr3/123456]/178 already has PBX structure?? WARNING[pid]: CAPI[contr3/123456]/178 already has a call record?? WARNING[pid]: CDR on channel 'CAPI[contr3/12345]/177' already started WARNING[pid]: Thread 1109916592 Blocking 'CAPI[contr3/123456]/178', already blocked by thread 1116277680 in procedure ast_waitfor_nandfds WARNING[pid]: Stack is not at expected value WARNING[pid]: Stack returned to an unexpected place! WARNING[pid]: Stack is not at expected value WARNING[pid]: Stack returned to an unexpected place! It is Asterisk 1.0.5, chan_capi 0.3.5, http://200.59.203.76/pub/chan_capi-0.3.5-patch.stable.diff 123456 is the extension of some SIP phone. Unfortunately there is no further information available and I do not know how to reproduce the crash. Any ideas? -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: I need very fast quick info how to setup ISDN card
On Tue, Dec 07, 2004 at 08:31:27PM +, Corvin wrote: I've compiled chan_capi - but I can't force it to work. Problem description? Error Messages? If you want to use chan_capi, it has to be TE mode. NT mode is not possible. Hfc based ISDN cards will generate lots of interrupts. Interrupt sharing will cause problems. A solution would be to use a quadBRI card instead of the 4 HFC cards and bristuff instead of misdn/chan_capi. Thank you again, but what to do if it's imposible? If you need NT mode you cannot use chan_capi (for this card/channel), but you can use bristuff. http://www.junghanns.net/asterisk/downloads/ Using 4 Hfc based ISDN cards in one box is not impossible, but you should expect problems. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: BLOCKING incoming FAXES on voice line.
On Sat, Dec 04, 2004 at 07:21:03PM -0700, Joseph wrote: Is there a way to block call / issue hangup command if the incoming call is a fax? Asterisk as a fax/voice switch documentation can be found at http://www.voip-info.org/wiki-Asterisk+fax -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Recomended ISDN for Asterisk ?
On Mon, Dec 06, 2004 at 12:11:45PM +0100, HBK wrote: Now I want to install my Billion PCI ISDN card (HFC based) in TE mode. I get a little confused with Isdn4Linux, ZapHFC HIAX and the need to install Capi ! You may fetch bristuff from junghanns.net and use the ZapHFC module. If you have a 2.6 kernel and have misdn installed you might use chan_capi or chan_misdn. chan_misdn can be used in TE and NT mode, but it has some bugs (or had some bugs two months ago). Since you want to use TE mode, chan_capi should be prefered. IMHO using Isdn4Linux / hisax / chan_modem_i4l is not a good idea. Please suggest best and easiest approach ? Best: bristuff Easier: misdn/chan_capi -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: how to start with ISDN
On Mon, Dec 06, 2004 at 08:48:58PM +, Corvin wrote: Ho to start with ISDN I am looking for howto: capi4 linux + isdn4linux + kernel 2.6 I hace 4 cheap passive cards, Would you recommend good howto for asterisk and above things? Start here: http://www.voip-info.org/wiki-ISDN First question: What kind of ISDN cards do you want to use? HFC chipset / AVM Fritz / other ? But in any case, using 4 passive cards in one box will cause problems. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: I need very fast quick info how to setup ISDN card
On Mon, Dec 06, 2004 at 10:11:56PM +, Corvin wrote: I am sitting on thos problem whole day I have never ISDN card in my life. Now I have to runn asterisk with 4 cards very quickly. I have installed chapi_chan and now I pached and recompiled kernel 2.8.1 for mISDN but I don't know how to check that cards are working and should look next step Thanks again for any help I'm Asterisk newbie. There is a Linux 2.6 and mISDN HowTo: http://rcum.uni-mb.si/~uvp00845b/ In this text the avmfritz module is used. In case of one hfc card it should be: modprobe hfcpci protocol=0x2 layermask=0x1f If you want to use chan_capi, it has to be TE mode. NT mode is not possible. Hfc based ISDN cards will generate lots of interrupts. Interrupt sharing will cause problems. A solution would be to use a quadBRI card instead of the 4 HFC cards and bristuff instead of misdn/chan_capi. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Snom 190 - dhcp - settings_server
On Sun, Nov 21, 2004 at 05:30:13PM +0200, Pertti Pikkarainen wrote: However I would use a more specific path for a web-server ;-) Something like: option tftp-server-name http://192.168.0.9/snom/snom200.htm But for the snom 190 tftp-server-name in dhcp config will set update_server. The field/variable setting_server remains empty. The documentation suggests that dhcp data can be used to define setting_server. Just a bug in the snom 190 firmware ? Best regards -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 190 - dhcp - settings_server
Hi,
in the Snom FAQ I found the following information:
After staring up, the phone tries the URL given in the Setting
URL of the phone. ... BTW this setting can also be set via DHCP.
option tftp-server-name http://192.168.0.9/snom200{mac}.htm;
The documents used:
FAQ-04-06-14-sf.pdf Setting up DHCP for snom phones
FAQ-04-03-24-sf.pdf How can I update a snom phone?
The phone used is a snom 190 (snom190-SIP 3.52e).
If I use the webinterface to insert the URL it works fine, but I am
not able to set this URL using dhcp.
--
Stefan Tichy [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sending SMS from ISDN to cellular
On Wed, Nov 10, 2004 at 11:02:14PM +0100, Elmar Haneke wrote:
how to configure * to send an SMS to an mobile phone (Germany, D2).
In the outgoing directory I do playe an call-file:
Channel: CAPI/[MYMSN]:0106301722270333
http://www.voip-info.org/wiki-Asterisk+cmd+Sms
SMS with T-Com (German Telekom)
Send outgoing messages to 0193010
You have to use the Telekom SMSC as gateway.
Extension: [TARGET-PHONE-NO]
CallerID: Test Test Test
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: smsdial
Priority: 1
You may define some additional variable here and use it as argument
to the sms application:
Callfile SetVar: SmsText=Test Test Test
Extensions.conf SMS(${CALLERIDNUM},,${EXTEN},${SmsText})
New question:
Is it neccessary to register the local ISDN phone number if you want
to receive sms from D2 mobile phones? Short messages from D1
customers are transmitted as sms only if such a registration has
been made.
--
Stefan Tichy [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: can't compile chan_capi 0.3.5 under SuSE 9.0
On Mon, Sep 20, 2004 at 01:45:19PM +0200, Martin Mielke wrote: [EMAIL PROTECTED]:~/chan_capi-0.3.5 make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c chan_capi.c:23:31: asterisk/features.h: No such file or directory chan_capi.c:24:28: asterisk/utils.h: No such file or directory Have a look at the Makefile in the ./chan_capi-0.3.5/ directory: INSTALL_PREFIX= ASTERISK_HEADER_DIR=$(INSTALL_PREFIX)/usr/include INCLUDE=-I$(ASTERISK_HEADER_DIR) If asterisk has not already been installed, you have to adjust the Makefile. It may be easier first to install asterisk without chan_capi and to build chan_capi in a second step. Is there any chance to find the RPM for chan_capi 0.3.5?? :) It would be possible to build some asterisk rpm and a matching chan_capi rpm, but as far as I know there is no rpm available for the SuSE 9.0 distribution. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Assigning a higher irq to a digium card
On Wed, Sep 08, 2004 at 12:27:05PM +0200, Roger Schreiter wrote: Within the bios menu I can't find any appropriate mean. It depends on the board/bios. Thanks for any hints! Technik und Know How zu IRQs, Sharing, APIC und INT-Leitungen: http://www.hardtecs4u.com/reviews/2002/irq/ This might be usefull. The language is german, but since you are using a .de mail address it should be no problem. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: AVM B1, chan_capi, Kernel 2.6
On Tue, Aug 10, 2004 at 10:00:58AM +0200, Stefan Tichy wrote: Using active AVM cards in connection with kernel 2.6 seems to be a bad idea. http://listserv.isdn4linux.de/pipermail/i4ldeveloper/2004-August/000630.html This patch should be interesting if you are using AVM B1 cards and kernel 2.6. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk config and root
On Thu, Sep 02, 2004 at 01:30:05PM +0300, Tzafrir Cohen wrote: Another beginner's question: Can I gain root if I have write access to asterisk's config files? If the asterisk process has root priviledges only root should be allowed to modify its config files. But root priviledges are not mandatory for a running asterisk process. http://www.voip-info.org/wiki-Asterisk+non-root Asterisk can be started by root (init script) using a command like this: /usr/sbin/asterisk -p -Uasterisk -Gdialout (options are explained in the man page asterisk(8)) The user asterisk has been created as described in the wiki. Probably you have to choose a different group and it might even be necessary to change the permissions of some device file. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: where can I find spandsp?
On Sun, Aug 29, 2004 at 12:20:49AM -0600, Rich Adamson wrote: Seems the opencall.org site has basically been unavailable for days/weeks. Is there another location to obtain the current code? http://sremington.zapto.org/downloads/asterisk/spandsp/ Just one week ago Seth Remington did send this information to the list. Also, will spandsp install against the current * cvs? Yes, but make shure that libtiff version 3.5.7 or 3.6.0 is used. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: HFC-S in NT mode, wiring?
On Mon, Aug 23, 2004 at 04:15:25PM +0200, Simone Ricci wrote: I've got an old HFC-S card to play with, and I would like to use it in NT mode. I've a problem only: wiring. I can't fully understand the instructions I was able to find online. Someone can point me to a site which explains the whole procedure clearly (like with some schematics, even in ASCII)? This site may be usefull, although it is not an asterisk page ;-) http://home.foni.net/~jolly2/download/PBX4Linux-2.3.html -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SpanDSP/RxFax help...
On Thu, Aug 19, 2004 at 11:25:17AM -0500, Rob Fugina wrote: I'm using the latest Asterisk from CVS, SpanDSP 0.0.1k, and the latest app_rxfax.c (as mirrored by friendly list members recently), and libtiff 3.5.7. Asterisk is detecting the fax signal properly, and executing the fax extension in the dialplan. I am using the same software versions, but don't have Zap channels. The fax part of the dialplan is pretty simple. The incoming call is already answered by this point: exten = fax,1,RxFax(/tmp/fax.tif) exten = fax,2,Hangup This should be enough. I do get files in /tmp called fax-[tr]x-audio-*, but no tif... The console output follows. I don't really know what any of it means... Can anyone give me a hand getting this working? Files like /tmp/fax-[tr]x-audio-* are just audio logs for debug. LOG_FAX_AUDIO in ./src/t30.c is the trigger for this debugging. It is not related to your problem, but files used as argument to txfax() are left open. This is not the answer to your problem, I know. But maybe theese remarks are usefull anyway. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AVM B1, chan_capi, Kernel 2.6
On a SuSE 9.1 installation I have severe problems using asterisk with the active AVM B1. Just making some outgoing call works without problems, but if both isdn channels are used asterisk or the complete server may hang or start showing very strange behavior. It might not even be possible to reboot. Kernel (SuSE) 2.6.5-7.104 AVM B1 3.09 asterisk 04-08-08 chan_capi 0.3.4b Using active AVM cards in connection with kernel 2.6 seems to be a bad idea. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_alsa record problem
Some unsuccessfull attempts to make console calls working.
If a sip phone is called, the other side will hear nothing.
If I try to record some sound the application will not finish. There
is a sound file, but it is empty (0 bytes). Record(${FILE}:gsm|10|30|skip)
is used in the dial plan. After hangup the following error messages
show up:
NOTICE[]: channel.c:1683 ast_set_read_format:
Unable to find a path from SLINR to UNKN
WARNING[]: app_record.c:287 record_exec:
Unable to restore read format on 'ALSA/default'
Kernel 2.6.5 and Asterisk CVS 04-97-23 are installed. Some test using
arecord and aplay shows that microphone and speakers are working.
Any help would be appreciated.
--
Stefan Tichy [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk Festival, not a happy couple
On Mon, Jun 28, 2004 at 04:48:36PM -0400, David Filion wrote: '/etc/asterisk/festival.conf': Found Jun 28 16:35:17 WARNING[360466]: app_festival.c:439 festival_exec: Festival returned ER In the console from which I started the Festival server, I sometimes get the following being displayed: SIOD ERROR: unbound variable : tts_textasterisk Check if the patch has been applied successfully. You will get this error messages if you are using the unpatched festival. I've confirmed the festival.conf file. It should work without any changes to the defaults. Sorry, I did overlook the question until today. There is a lot of traffic on this list. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Help needed for Seting Up Asterisk
On Wed, Jul 21, 2004 at 09:04:58PM +0200, Beierlein Moritz wrote: My idea was i set up a Asterisk-System and i will put in an ISDN Card where I can plug a ISDN Phone, I will have to use an ISDN card with the NT-Mode. HFC (Cologne Chip) cards can be used. I will use Asterisk behind a NAT Router. Did you read this: http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions And: The localnet/externip settings of sip.conf permit Asterisk to operate behind certain types of NATs. However, it requies that the external IP be known and set manually in the Asterisk configuration. More importantly, the localnet/externip settings do not solve the problem of Asterisk. behind a port translating (Full Cone) NAT. (copied from http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions ) If the Internetconnection interrupts the Asterisk has to wait 20 seconds, then has to register at the SIP-Provider. Asterisk does not detect this, but you can choose a small value for registration expiry. How is it possible to get the SIP Calls to the ISDN card? http://www.voip-info.org/wiki-Asterisk+config+extensions.conf -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: gnophone and asterisk
On Tue, Jul 13, 2004 at 09:11:46AM +0700, Isianto Istiadi wrote: On Mon, 12 Jul 2004 14:51:55 +0200 Stefan Tichy [EMAIL PROTECTED] wrote: On Mon, Jul 12, 2004 at 03:30:24PM +0700, Isianto Istiadi wrote: and then I do nmap -sU ip (I don't see port 4569 or 5036 available). I can't register gnophone with *, when I do ethereal, I can see that gnophone tried to connect to port 5036, but the * replied destination unreachable. Is there something wrong with my config? gnophone 0.2.4 uses iax only not iax2. -- Stefan Tichy [EMAIL PROTECTED] Is that means I can't use gnophone with cvs *? will version 0.2.5 works? The version 0.2.4 is more than 2 years old and it looks as if there is no further development for GnoPhone. November 13th, 2001 - 11:58am CST - GnoPhone 0.2.4 released http://www.gnophone.com/ If you really need a iax2 capable softphone, you may check this: http://www.holgerschurig.de/files/linux/qtiax-0.1.tar.bz2 -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Question about Asterisk Installation
On Sat, Jul 17, 2004 at 06:46:59PM +0200, Pisano Vincenzo wrote: when I try to execute the simbolic link ln -s /usr/lib/mysql/libmysqlclient.so /usr/lib/libmysqlclient.so this produce an error because the source object libmysqlclient.so is not present in any directory. So I can't continue with Asterisk compilation. May help me? thanks This appears to be a problem with your mysql installation. The system here has mysql-devel and mysql-shared packages containing symbolic links and the library: /usr/lib/libmysqlclient.so - libmysqlclient.so.12.0.0 /usr/lib/mysql/libmysqlclient.so - ../libmysqlclient.so.12.0.0 Which linux distribution is installed? -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: gnophone and asterisk
On Mon, Jul 12, 2004 at 03:30:24PM +0700, Isianto Istiadi wrote: and then I do nmap -sU ip (I don't see port 4569 or 5036 available). I can't register gnophone with *, when I do ethereal, I can see that gnophone tried to connect to port 5036, but the * replied destination unreachable. Is there something wrong with my config? gnophone 0.2.4 uses iax only not iax2. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: permission problem
On Mon, Jul 12, 2004 at 01:32:39PM +0200, Cyprien Simons wrote: I modified the permissions of /var/spool/asterisk and /var/log/asterisk and it seems that asterisk is launching now. But I still have messages at the beginning telling me that: Unable to open pid file '/var/run/asterisk.pid': Permission denied Unable to bind socket to /var/run/asterisk.ctl: Address already in use Create a directory /var/run/asterisk/, change its owner to asterisk (the non-root user) and set astrundir = /var/run/asterisk in /etc/asterisk/asterisk.conf. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Gogoif with variables acting funny?
On Mon, Jul 12, 2004 at 10:51:24AM -0400, Steve Woolley wrote:
exten = t,1,SetVar(counter=[${counter}+1])
exten = t,2,Gotoif([${counter}3]?s,7:h,1)
You need $2
Example:
SetVar(lala=$[1 + 2]);
GotoIf($[${CALLERIDNUM} = 303]?3:2)
http://www.voip-info.org/wiki-Asterisk+Expressions
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf
--
Stefan Tichy [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] rxfax - mISDN - missing logs
Hi,
using HFC cards, mISDN/chan_misdn and spandsp lib fax retrieval works,
but some log file entries are missing. There should be one of the lines:
Fax successfully received.
Fax receive not successful.
Dail Plan config used:
[fax]
exten = _.,1,SetVar(FAXFILE=.)
exten = _.,2,SetVar(LOCALSTATIONID=..)
exten = _.,3,rxfax(${FAXFILE})
exten = _.,4,NoOp,XYZ
exten = _.,5,Hangup
Log file:
-- Executing RxFAX(mISDN/2, ..) in new stack
Got hangup
Extension .., priority . returned normally even though call was hung up
cdr_mysql: inserting a CDR record.
The file ${FAXFILE} is available and seems to be complete, but isn't
this hangup to early?
Is this a rxfax/spandsp or a mISDN problem?
--
Stefan Tichy [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: i or s or whatever the invalid_exten is HELP !!!!!
You are using SIP phones and in your sip.conf there should be some
context definition for incoming calls. You may jump (goto) to some
other context later, but the context defined in sip.conf should
contain extensions catching all numbers you may dial from that phone.
On Thu, Jul 08, 2004 at 04:09:55PM +0100, Simon wrote:
Or can i test the first 2 digits of a number and send it to the right
context ?
Yes, that is what you have to do:
exten = i,1,goto(busy,1,1)
exten = _65.,1,goto(office1,${EXTEN},1)
exten = _66.,1,goto(office2,${EXTEN},1)
--
Stefan Tichy [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_misdn HFC-NT dialtone
How is it possible to get a dialtone using chan_misdn for a ISDN phone connected to a hfc nt-mode card? misdn.conf: [intern] ports=2 context=isdnIntern immediate=yes extensions.conf [isdnIntern] exten = s,1,DigitTimeout(5) I don't want to use answer here because the phone does not show the dialed digits in the display if the call has already been answered. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP Softphone
Hi, On Mon, Jun 28, 2004 at 11:00:43PM +0200, Arve Rasmussen wrote: What is the best SIP softphone to use with Asterisk? Really don't know what is the best SIP softphone but I am using linphone with alaw codec and dtmfmode rfc2833. Did not try low bandwidth codecs until now. http://www.linphone.org/ http://simon.morlat.free.fr/download/0.12.2/source/linphone-0.12.2.tar.gz -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] timestamp in the future (linphone)
Hi, a SIP softphone (linphone) and asterisk are installed on the same system. The softphone rings and can initiate outgoing calls (isdn/bri/chan_misdn), but there is no sound. There are error messages from the softphone : MediaStreamer-WARNING **: The remote sip-phone has send data with a future timestamp: 1762917720,resynchronising session. No problem with calls from one SIP phone to another. Using a gs bt 100 instead of the softphone does not make problems either. Is this a linphone problem or is it a asterisk problem ? -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Users in MySQL
On Mon, May 31, 2004 at 03:12:44PM +0200, Reto Stauss wrote: I've just compilied th latest CVS of * with USE_MYSQL_FRIENDS enabled (1). During startup * tells me that it connects to the db, so this should be fine. Nevertheless I don't see any users from the db when I run sip show users or iax2 show users although I configured some. In the sip case it is the consequence of the mode MYSQL_FRIENDS is implemented. Probably the same with iax2. It is also not possible to call them. When a sip phone registers, the current IP address and other parameters get updated in the database. This data will be used if there is a call to this phone. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Unable to create channel of type 'CAPI'
On Tue, May 25, 2004 at 07:40:51PM +0200, jo wrote: Since upgrading from stable to latest cvs I can't place CAPI calls (AVM Fritz/chan_capi-0.3.1) Did I miss something that has to be changed in configfiles? Also tried to recompile chan_capi which run into an error. Did you apply the patch from http://lists.digium.com/pipermail/asterisk-users/2004-April/044125.html -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
