Re: [asterisk-users] Asterisk realtime peer registration
On 10 June 2014 05:27, ortei...@tiscali.it wrote: Hello there I'd like to use sip users and peers realtime. I think I done all I need to get asterisk works fine in realtime: res_odbc.conf configuration. extconfig.conf sippeers = odbc,asterisk,sipclient sipusers = odbc,asterisk,sipclient sip.conf [general] rtcachefriends=yes The sipclient table as suggest in this article: SIP Realtime, MySQL table structure (https://wiki.asterisk.org/wiki/display/ ... +structure https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure ) The user registered on asterisk works fine, but not the peer. I'd like to use my voipdiscount account as a peer to do external call. Name/username Host Dyn Forcerport ACL Port Status Realtime 2000/2000 xxx.xxx.xxx.xxx D N 65476 OK (117 ms) Cached RT Mysql entry on sipclient table is below: 3 sip.voipdiscount.com 5060 \N XX \N \N \N \N sip.voipdiscount.com peer default \N \N XXX \N \N rfc2833 yes no \N \N \N \N \N port,invite \N \N \N \N \N \N 01234556678 \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N sip.voipdiscount.com X yes \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N XX \N voipdiscount_out \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N I enabled also sip debug, but I don't see any attempt towards sip.voipaccount.com What am I doing wrong? Someone can help me? Thanks in advance Pietro Try changing the type from peer to friend. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Mixing res_mysql and res_odbc
Hi Is there any harm in using res_mysql for some things and res_odbc for others? We already use res_mysql for ARA but could do with having CEL logged to MySQL. Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] R: Asterisk realtime peer registration
I had already done this, but nothing is changed Thanks Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Ishfaq Malik Inviato: martedì 10 giugno 2014 10:05 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] Asterisk realtime peer registration On 10 June 2014 05:27, ortei...@tiscali.it mailto:ortei...@tiscali.it wrote: Hello there I'd like to use sip users and peers realtime. I think I done all I need to get asterisk works fine in realtime: res_odbc.conf configuration. extconfig.conf sippeers = odbc,asterisk,sipclient sipusers = odbc,asterisk,sipclient sip.conf [general] rtcachefriends=yes The sipclient table as suggest in this article: SIP Realtime, MySQL table structure ( https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure https://wiki.asterisk.org/wiki/display/ ... +structure) The user registered on asterisk works fine, but not the peer. I'd like to use my voipdiscount account as a peer to do external call. Name/username Host Dyn Forcerport ACL Port Status Realtime 2000/2000 xxx.xxx.xxx.xxx D N 65476 OK (117 ms) Cached RT Mysql entry on sipclient table is below: 3 sip.voipdiscount.com http://sip.voipdiscount.com 5060 \N XX \N \N \N \N sip.voipdiscount.com http://sip.voipdiscount.com peer default \N \N XXX \N \N rfc2833 yes no \N \N \N \N \N port,invite \N \N \N \N \N \N 01234556678 \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N sip.voipdiscount.com http://sip.voipdiscount.com X yes \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N XX \N voipdiscount_out \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N I enabled also sip debug, but I don't see any attempt towards sip.voipaccount.com http://sip.voipaccount.com What am I doing wrong? Someone can help me? Thanks in advance Pietro Try changing the type from peer to friend. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk mailto:i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] R: Mixing res_mysql and res_odbc
Ok Ish, I will try with res_mysql. Still thanks Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Ishfaq Malik Inviato: martedì 10 giugno 2014 12:05 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: [asterisk-users] Mixing res_mysql and res_odbc Hi Is there any harm in using res_mysql for some things and res_odbc for others? We already use res_mysql for ARA but could do with having CEL logged to MySQL. Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk mailto:i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] R: Mixing res_mysql and res_odbc
Hi Pietro That wasn't a response to you but a genuine question for myself out to the users list! Regards Ish On 10 June 2014 13:13, ortei...@tiscali.it wrote: Ok Ish, I will try with res_mysql. Still thanks *Da:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *Per conto di *Ishfaq Malik *Inviato:* martedì 10 giugno 2014 12:05 *A:* Asterisk Users Mailing List - Non-Commercial Discussion *Oggetto:* [asterisk-users] Mixing res_mysql and res_odbc Hi Is there any harm in using res_mysql for some things and res_odbc for others? We already use res_mysql for ARA but could do with having CEL logged to MySQL. Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] CDR custom variable on second call leg - via originate or .call file
Hi
We have the following test .call file and test dialplan:
I can't set a custom CDR var to a value on one channel leg, and another
value on the connected channel leg?
Is there a way I can woraround this issue?
## test call file
Channel: Local/queue@TiagoGeada
CallerID: teste-geada:0:210332450:
MaxRetries: 0
RetryTime: 1
WaitTime: 8640
Account: teste-geada
Context: TiagoGeada
Extension: outbound
Archive: Yes
## dialplan
queue = {
Set(CDR(remoteUid)=foo);
Queue(TiagoGeada,t,,,100);
Hangup();
}
outbound = {
//NoCDR();
//ForkCDR(vdD);
//ResetCDR(v);
Set(CDR(remoteUid,r)=bar);
Dial(Local/932485457@outbound,,gT);
Hangup();
}
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Re: [asterisk-users] CDR custom variable on second call leg - via originate or .call file
As far as I know, only way to set variables on another channel would be:
asterisk -rx core show help dialplan set chanvar
Usage: dialplan set chanvar channel varname value
Set channel variable varname to value
On 10 June 2014 16:39, Tiago Geada tiago.ge...@gmail.com wrote:
Hi
We have the following test .call file and test dialplan:
I can't set a custom CDR var to a value on one channel leg, and another
value on the connected channel leg?
Is there a way I can woraround this issue?
## test call file
Channel: Local/queue@TiagoGeada
CallerID: teste-geada:0:210332450:
MaxRetries: 0
RetryTime: 1
WaitTime: 8640
Account: teste-geada
Context: TiagoGeada
Extension: outbound
Archive: Yes
## dialplan
queue = {
Set(CDR(remoteUid)=foo);
Queue(TiagoGeada,t,,,100);
Hangup();
}
outbound = {
//NoCDR();
//ForkCDR(vdD);
//ResetCDR(v);
Set(CDR(remoteUid,r)=bar);
Dial(Local/932485457@outbound,,gT);
Hangup();
}
--
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Re: [asterisk-users] CDR custom variable on second call leg - via originate or .call file
Using Set(MASTER_CHANNEL(CDR(remoteUid))=foo); might do what you want
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mikael Fredin
Sent: Tuesday, June 10, 2014 11:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR custom variable on second call leg - via
originate or .call file
As far as I know, only way to set variables on another channel would be:
asterisk -rx core show help dialplan set chanvar
Usage: dialplan set chanvar channel varname value
Set channel variable varname to value
On 10 June 2014 16:39, Tiago Geada
tiago.ge...@gmail.commailto:tiago.ge...@gmail.com wrote:
Hi
We have the following test .call file and test dialplan:
I can't set a custom CDR var to a value on one channel leg, and another value
on the connected channel leg?
Is there a way I can woraround this issue?
## test call file
Channel: Local/queue@TiagoGeada
CallerID: teste-geada:0:210332450:
MaxRetries: 0
RetryTime: 1
WaitTime: 8640
Account: teste-geada
Context: TiagoGeada
Extension: outbound
Archive: Yes
## dialplan
queue = {
Set(CDR(remoteUid)=foo);
Queue(TiagoGeada,t,,,100);
Hangup();
}
outbound = {
//NoCDR();
//ForkCDR(vdD);
//ResetCDR(v);
Set(CDR(remoteUid,r)=bar);
Dial(Local/932485457@outbound,,gT);
Hangup();
}
--
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Re: [asterisk-users] SIP call control via RTCP
Hello, I have found here http://www.voip-info.org/wiki/view/Asterisk+RTCP that there has been a patch for RTCP of Asterisk 1.4. Does this mean that starting with Asterisk version 1.6 RTCP call control is working correctly? Kind regards Jan Gaida On Mon, May 12, 2014 at 2:40 PM, Jan Gaida jan.ga...@grupoamper.com wrote: Thank you. Yes, that should work. But if I understand it correctly, only if there's no silence detection activated. Otherwise, when silence is detected no RTP would be send, so that rtptimeout would hang up a still active call. I there no option to use RTCP? Not even in Asterisk 11? Regards On Mon, May 12, 2014 at 2:12 PM, Matt Behrens m...@zigg.com wrote: On May 12, 2014, at 5:02 AM, Jan Gaida jan.ga...@grupoamper.com wrote: We are using Asterisk 1.4 as call distribution system with simple queues for SIP calls. With high load (4000 calls/hour) some calls remain in queue forever (until queue's max wait time) in spite of being hung up already by the caller. It seems that when a BYE is lost, Asterisk has no mechanism to check whether a call is still active. Is there a way to activate a RTCP call control, e.g. Asterisk should hang up when he stops receiving RTCP messages? Have you looked at the rtptimeout and rtpholdtimeout options? -- _ -- 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 -- *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.ga...@grupoamper.com | www.grupoamper.com -- *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.ga...@grupoamper.com | www.grupoamper.com -- This message and any attachments are intended only for the use of the individual to whom they are addressed and it may contain information that is privileged or confidential. If you have received this communication by mistake, please notify us immediately by e-mail or telephone.The storage, recording, use or disclosure of this e-mail and its attachments by anyone other than the intended recipient is strictly prohibited. This message has been verified using antivirus software; however, the sender is not responsible for any damage to hardware or software resulting from the presence of any virus. Este mensaje y cualquier anexo son exclusivamente para la persona a quien van dirigidos y pueden contener información privilegiada o confidencial. Si usted ha recibido esta comunicación por error, le agradecemos notificarlo de inmediato por esta misma vía o por teléfono. Está prohibida su retención, grabación, utilización o divulgación con cualquier propósito. Este mensaje ha sido verificado con software antivirus; sin embargo, el remitente no se hace responsable en caso de que en éste o en los archivos adjuntos haya presencia de algún virus que pueda generar daños en los equipos o programas del destinatario. -- _ -- 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] SSL/TLS weakness impact on Asterisk authentication
After reading about the 2 major SSL (and TLS?) weaknesses discovered this year, I was wondering how it affects asterisk. Does the SIP authentication use TLS - or something that was recently broken? Is there a risk of exposing passwords? Thanks! -- _ -- 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] SSL/TLS weakness impact on Asterisk authentication
On Tue, Jun 10, 2014 at 4:44 PM, Michelle Dupuis mdup...@ocg.ca wrote: After reading about the 2 major SSL (and TLS?) weaknesses discovered this year, I was wondering how it affects asterisk. Asterisk uses OpenSSL for TLS. So, the answer is, it depends on the version of OpenSSL that was installed for your Asterisk server. See http://blogs.digium.com/2014/04/11/asterisk-heartbleed/ for more information. Does the SIP authentication use TLS - or something that was recently broken? Is there a risk of exposing passwords? SIP signalling - in both chan_sip and chan_pjsip - can use TLS as a transport. If your OpenSSL version is one of those affected by the various vulnerabilities, then yes, you are at risk. This also applies to all other modules in Asterisk that use TLS, including AMI, the HTTP server, and others. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Fwd: Asterisk 12.1.1 on Ubuntu 12.04 crashing
Hi,
I'm writing b/c I'm having a crash that happens a lot... it happened
without better_backtraces and then I went ahead and recompiled with it.
Then it stopped until today.
You can see the crash data at http://pastebin.com/1vbrWepr (bt, bt full
and thread apply all bt)
The extensions.conf is something like this
[incoming]
exten = usnumber,1,GotoIf($[${CALLERID(num)} =
1222333”]?validatepin,2)
exten = validatepin,2,DISA(,supercontext)
[supercontext]
exten = 9,1,Dial(PJSIP/line@somedevice)
The caller comes into incoming on the usnumber extension, the caller id
matches and then it does the DISA call validating the pin... I enter the
pin right, then dial 9 and that's when it crashes.
I'm about to install Asterisk 12.3 and see if it happens there too
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Re: [asterisk-users] SSL/TLS weakness impact on Asterisk authentication
On 10-06-14 23:44, Michelle Dupuis wrote: After reading about the 2 major SSL (and TLS?) weaknesses discovered this year, I was wondering how it affects asterisk. Does the SIP authentication use TLS - or something that was recently broken? Is there a risk of exposing passwords? Asterisk' SIP authentication uses a digest. See http://tools.ietf.org/html/rfc3261 for more info (20.6 and onwards). That does not mean that the recent OpenSSL issues have no impact on Asterisk. They do if you configure SIP to use TLS transport or enable TLS for other parts (for example AMI). So it's highly recommended to install the updated OpenSSL packages containing the fixes. My Asterisk packages link dynamically against the OpenSSL libraries. Assuming your packages do the same then, once you have updated the OpenSSL packages to the latest ones with the fixes and restart Asterisk, you should be good to go. While the recent OpenSSL issues don't directly expose your account passwords, the Heartbleed bug can expose (parts of) the private key used by TLS. Once the Men in Black have your private key its possible to setup a Man (in Black) in the Middle attack and sniff those passwords. See http://heartbleed.com/ Unless you want to mess around with the Men in Black and leave your system vulnerable to attack, you should install all security updates ASAP and then restart the services that rely upon them. HtH, Patrick -- _ -- 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
