[asterisk-users] Tranfer the called number in 3 way call
Hey there, I have a question i want a dialplan to send the called number of the client instead of my callerID when making a 3way call or when transfering to an extension from a bridge to another pbx. The problem i add a variable and using thw two underscores but i still see the my calledID , i am using both asterisk 1.8.29 and other is asterisk 11.4 here is the dialplan inherit which i do : exten=> _6XX,n,set(__var=${EXTEN}) exten=> _6XX,n,Dial(IAX2/bridge/${EXTEN},,tTor) I want to recieve the client number that were called from my first pvx with 5XX extensions to be shown on my second pbx with 6XX while making a transfer to the bridge with 3 way call or blind transfer , I know i am missing something here , can you guys help me. Sent from my LG Mobile -- _ -- 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] IMPORT from bridged Local channels not importing.
On Tue, 15 Nov 2016 12:21:31 -0200 "Ethy H. Brito"wrote: > > Hi All > > I have some users that can access outside world telephone number. > They have external numbers to be reached as well. > > Due to internal policy restrictions, they are not allowed to dial > each other internal numbers. I Can't change that. > > When an internal user dials the external number for another internal > user, I Dial(Local/...) the second user. > > So I end up with two channels: > SIP/origin to Local/dest_ext_num;1 > and > Local/dest_ext_num;2 to SIP/destination > > When the call is hung up (h extension), I need to grab the stats of both > legs (SIP/origin and SIP/destination) of the call, so I use: > > ${RTPAUDIOQOS} > > to grab the origin leg stats and > > Set(MyDESTCH=${CUT(CDR(dstchannel),\;,1)}\;2) > Set(DESTCH=${IMPORT(${MyDESTCH},BRIDGEPEER)}) > Set(STATS=${IMPORT(${DESTCH},CHANNEL(rtpqos,audio,all))}) > > to grab the stats for destination leg. > > MyDESTCH is correctly set to "Local/dest_ext_num;2" > DESTCH receives "SIP/destination" > But STATS is "" > > What am I missing here? > Is there a smarter way for grabbing these? > > Another questions: when the call is hung, in which context is "h > extension" run? Always originator? Always destination? Depends on what? > What about in this scenario I describe (four contexts involved)? > > Environment is: asterisk -V > > Asterisk 11.7.0~dfsg-1ubuntu1 running on Ubuntu 14.04.5 LTS > > > Thanx for your time. > Cheers > > Ethy Hi All No suggestions at all?? Cheers Ethy -- _ -- 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] Change Media IP in SDP [SOLVED]
Hello Max, Thank you for your detailed reply. Indeed all my nat-related settings are configured properly and I make use of two NIC cards to access the various networks I'm connected to. When you mentioned 'externip' I assume you meant 'externaddr' because there is no 'externip' in sip.conf... I've made some tests and found out that this setting has no effect on the [peer] level and can only be used on the [general] section of sip.conf. Your suggestion of using the 'localnet' together with the 'externaddr' did the trick for me. Even though the remote non-public networks are by no way 'local' I have defined them as such thus disabling any IP replacements with the 'externaddr' which is the wrong IP for these networks. For the benefit of others who may come across this thread I would mention that this solution would work on the following two conditions: 1. Only one external network may be NATed to the Asterisk. This limitation is due to the fact that only one 'externaddr' may be configured per-box. 2. All other end-points in other networks are within a well-known IP range which can be defined in 'localnet' settings. It will be nice to have the 'externaddr' work on the [peer] level thus providing more valuable tools to solve NAT issues. Harel Original Message * Hi, normally, Asterisk handles RTP IP addresses in SDP correctly, if you have specified - that NAT traversal is enabled for all peers (e.g. nat=force_rport,comedia) - your local network with "localnet=yournetwork/networkmask" - e.g. "localnet=192.168.1.0/255.255.255.0" - directmedia, canreinvite, directrtpsetup is deacitivated In this case, your Asterisk will always stay in the RTP stream and signalling only it's own IP address to other peers which is configured for the interface which Asterisk uses to reach the peer. But: If you have only one interface configured on your Asterisk server and an external firewall/router is managing your separated networks, this might not help. In this case you can use "externip" on a per peer basis in your SIP configuration to specify the IP address Asterisk uses in the SDP. Maybe, a global configuration of "externip" and "localnet" is all you need to help Asterisk setting the SDP address correctly. Also, enabling ICE support can help you getting the correct IP address if the remote peer supports it. Greetings Max Am 07.12.2016 um 00:02 schrieb Harel: > Hello List, > I need your help with information going out on my SDP. > Is it possible to update the Media Address on a per-call basis or a > per-channel basis? > Reason: > My Asterisk is in a private network and needs to connect to UA on its > internal network and also few external networks. One network is public and > the others are not public. Between each other the external networks are not > routable. Signaling is flowing with no issues because SIP Registers and NAT > boxes maintain sessions correctly. The problem is with RTP. After making > traces on all possible nodes of this network I clearly found out that the RTP > fails because the Asterisk doesn't manage to communicate the correct address > to the UAs in the SDP. It will report its internal IP address and the remote > UA will try to send its RTP to this address which, of course, will fail > miserably. > Obviously I can't use externaddr or media_address in sip.conf because it will > only be good for one network while the other external networks will fail just > the same. Same applies for STUN, it will only be good for the network the > STUN requests are being sent from. > On all networks I have fix IP addresses on my side and I fully control a > professional security box. > Asterisk is 13.6.0 > I can't, and don't want to, touch user-side equipment which is normally some > kind of voip phone behind a standard home VDSL router. > > Any ideas how can I transmit the correct IP address in SDP to UAs on > different networks? > > Many thanks, > Harel > > -- _ -- 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] AST-2016-009:
Asterisk Project Security Advisory - ASTERISK-2016-009 ProductAsterisk Summary Nature of Advisory Authentication Bypass SusceptibilityRemote unauthenticated sessions Severity Minor Exploits KnownNo Reported On October 3, 2016 Reported By Walter Doekes Posted On Last Updated OnDecember 8, 2016 Advisory Contact Mmichelson AT digium DOT com CVE Name Description The chan_sip channel driver has a liberal definition for whitespace when attempting to strip the content between a SIP header name and a colon character. Rather than following RFC 3261 and stripping only spaces and horizontal tabs, Asterisk treats any non-printable ASCII character as if it were whitespace. This means that headers such as Contact\x01: will be seen as a valid Contact header. This mostly does not pose a problem until Asterisk is placed in tandem with an authenticating SIP proxy. In such a case, a crafty combination of valid and invalid To headers can cause a proxy to allow an INVITE request into Asterisk without authentication since it believes the request is an in-dialog request. However, because of the bug described above, the request will look like an out-of-dialog request to Asterisk. Asterisk will then process the request as a new call. The result is that Asterisk can process calls from unvetted sources without any authentication. If you do not use a proxy for authentication, then this issue does not affect you. If your proxy is dialog-aware (meaning that the proxy keeps track of what dialogs are currently valid), then this issue does not affect you. If you use chan_pjsip instead of chan_sip, then this issue does not affect you. Resolution chan_sip has been patched to only treat spaces and horizontal tabs as whitespace following a header name. This allows for Asterisk and authenticating proxies to view requests the same way Affected Versions Product Release Series Asterisk Open Source 11.xAll Releases Asterisk Open Source 13.xAll Releases Asterisk Open Source 14.xAll Releases Certified Asterisk 13.8All Releases Corrected In Product Release Asterisk Open Source 11.25.1, 13.13.1, 14.2.1 Certified Asterisk11.6-cert16, 13.8-cert4 Patches SVN URL Revision Links Asterisk Project Security Advisories are posted at http://www.asterisk.org/security This document may be superseded by later versions; if so, the latest version will be posted at
[asterisk-users] AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
Asterisk Project Security Advisory - AST-2016-008 ProductAsterisk SummaryCrash on SDP offer or answer from endpoint using Opus Nature of Advisory Remote Crash SusceptibilityRemote unauthenticated sessions Severity Critical Exploits KnownNo Reported On November 11, 2016 Reported By jorgen Posted On Last Updated OnNovember 15, 2016 Advisory Contact jcolp AT digium DOT com CVE Name Description If an SDP offer or answer is received with the Opus codec and with the format parameters separated using a space the code responsible for parsing will recursively call itself until it crashes. This occurs as the code does not properly handle spaces separating the parameters. This does NOT require the endpoint to have Opus configured in Asterisk. This also does not require the endpoint to be authenticated. If guest is enabled for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still processed and the crash occurs. Resolution The code has been updated to properly handle spaces separating parameters in the fmtp line. Upgrade to a released version with the fix incorporated or apply patch. Affected Versions ProductRelease Series Asterisk Open Source 13.x13.12.0 and higher Asterisk Open Source 14.xAll Versions Corrected In Product Release Asterisk Open Source13.13.1, 14.2.1 Patches SVN URL Revision http://downloads.asterisk.org/pub/security/AST-2016-008-13.diff Asterisk 13 http://downloads.asterisk.org/pub/security/AST-2016-008-14.diff Asterisk 14 Links https://issues.asterisk.org/jira/browse/ASTERISK-26579 Asterisk Project Security Advisories are posted at http://www.asterisk.org/security This document may be superseded by later versions; if so, the latest version will be posted at http://downloads.digium.com/pub/security/AST-2016-008.pdf and http://downloads.digium.com/pub/security/AST-2016-008.html Revision History Date Editor Revisions Made November 15, 2016 Joshua Colp Initial draft of Advisory Asterisk Project Security Advisory - AST-2016-008 Copyright © 2016 Digium, Inc. All Rights Reserved. Permission is hereby granted to distribute and publish this advisory in its original, unaltered form. -- _ -- 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] Asterisk 11.25.1, 13.13.1, 14.2.1, 11.6-cert16, and 13.8-cert4 Now Available (Security Release)
The Asterisk Development Team has announced security releases for Asterisk 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16, and 13.8-cert4. These releases are available for immediate download at: http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ The release of versions 13.13.1 and 14.2.1 resolve the following security vulnerability: * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus If an SDP offer or answer is received with the Opus codec and with the format parameters separated using a space the code responsible for parsing will recursively call itself until it crashes. This occurs as the code does not properly handle spaces separating the parameters. This does NOT require the endpoint to have Opus configured in Asterisk. This also does not require the endpoint to be authenticated. If guest is enabled for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still processed and the crash occurs. The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4 resolve the following security vulnerability: * AST-2016-009: Remote unauthenticated sessions in chan_sip The chan_sip channel driver has a liberal definition for whitespace when attempting to strip the content between a SIP header name and a colon character. Rather than following RFC 3261 and stripping only spaces and horizontal tabs, Asterisk treats any non-printable ASCII character as if it were whitespace. This means that headers such as Contact\x01: will be seen as a valid Contact header. This mostly does not pose a problem until Asterisk is placed in tandem with an authenticating SIP proxy. In such a case, a crafty combination of valid and invalid To headers can cause a proxy to allow an INVITE request into Asterisk without authentication since it believes the request is an in-dialog request. However, because of the bug described above, the request will look like an out-of-dialog request to Asterisk. Asterisk will then process the request as a new call. The result is that Asterisk can process calls from unvetted sources without any authentication. If you do not use a proxy for authentication, then this issue does not affect you. If your proxy is dialog-aware (meaning that the proxy keeps track of what dialogs are currently valid), then this issue does not affect you. If you use chan_pjsip instead of chan_sip, then this issue does not affect you. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1 http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1 http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1 http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16 http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf Thank you for your continued support of Asterisk! -- _ -- 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] Asterisk 11.25.1, 13.13.1, 14.2.1, 11.6-cert16, and 13.8-cert4 Now Available (Security Release)
-- _ -- 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] What to do when changing from one asterisk version to another ?
On Thu, Dec 8, 2016 at 11:38 AM, Olivierwrote: > > > 2016-12-08 18:23 GMT+01:00 Olivier : > >> Hello, >> >> I'm compiling Asterisk from source on Debian systems. >> >> I'm currently writing a script I'm planning to launch when upgrading from >> one Asterisk version to another one within the same class (from 13.4.0 to >> 13.12.0 or from 13.12.0 to 13.8.0, for instance). >> >> Reading [1], I thought the following would work: >> cd /usr/src/asterisk-13.4.0 >> ./configure >> make >> make install >> ... >> cd /usr/src/asterisk-13.4.0 >> make dist-clean >> >> After running above commands, /usr/sbin/asterisk and >> /usr/lib/asterisk/modules/*.so files still exist. >> I would expect both /usr/sbin/asterisk and /usr/lib/asterisk/modules/*.so >> filesto be removed so that if I newly installed asterisk instance wouldn't >> inherit uncontrolled files. >> >> I also tried with make clean and make uninstall with the same result but >> I may have missed some steps during my trials. >> > > Correcting myself, make uninstall seems to be what I was after for > Asterisk itself. > I'm still searching for the equivalent make target for pjproject. > pjproject has a make uninstall target as well. Since v13.8, Asterisk has a --with-pjproject-bundled option [1]. This will configure, build, and statically link with pjproject to give better integration with Asterisk. It also applies a few backported fixes to the pjproject version used. Richard [1] https://wiki.asterisk.org/wiki/display/AST/Building+and+Inst alling+Asterisk -- _ -- 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] What to do when changing from one asterisk version to another ?
2016-12-08 18:23 GMT+01:00 Olivier: > Hello, > > I'm compiling Asterisk from source on Debian systems. > > I'm currently writing a script I'm planning to launch when upgrading from > one Asterisk version to another one within the same class (from 13.4.0 to > 13.12.0 or from 13.12.0 to 13.8.0, for instance). > > Reading [1], I thought the following would work: > cd /usr/src/asterisk-13.4.0 > ./configure > make > make install > ... > cd /usr/src/asterisk-13.4.0 > make dist-clean > > After running above commands, /usr/sbin/asterisk and > /usr/lib/asterisk/modules/*.so files still exist. > I would expect both /usr/sbin/asterisk and /usr/lib/asterisk/modules/*.so > filesto be removed so that if I newly installed asterisk instance wouldn't > inherit uncontrolled files. > > I also tried with make clean and make uninstall with the same result but I > may have missed some steps during my trials. > Correcting myself, make uninstall seems to be what I was after for Asterisk itself. I'm still searching for the equivalent make target for pjproject. > > Before diving deeper, are my expectations correct ? > > Best regards > > [1] https://wiki.asterisk.org/wiki/display/AST/Building+and+ > Installing+Asterisk > -- _ -- 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] What to do when changing from one asterisk version to another ?
Hello, I'm compiling Asterisk from source on Debian systems. I'm currently writing a script I'm planning to launch when upgrading from one Asterisk version to another one within the same class (from 13.4.0 to 13.12.0 or from 13.12.0 to 13.8.0, for instance). Reading [1], I thought the following would work: cd /usr/src/asterisk-13.4.0 ./configure make make install ... cd /usr/src/asterisk-13.4.0 make dist-clean After running above commands, /usr/sbin/asterisk and /usr/lib/asterisk/modules/*.so files still exist. I would expect both /usr/sbin/asterisk and /usr/lib/asterisk/modules/*.so filesto be removed so that if I newly installed asterisk instance wouldn't inherit uncontrolled files. I also tried with make clean and make uninstall with the same result but I may have missed some steps during my trials. Before diving deeper, are my expectations correct ? Best regards [1] https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+Asterisk -- _ -- 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] how are channels numbers assigned
Hi All I have this system here where: # dahdi_hardware pci::07:04.0 wctdm24xxp+ d161:8005 Wildcard TDM410P pci::07:09.0 wcte11xp+e159:0001 Digium Wildcard TE110P T1/E1 Board channels are 0-31 for TE110P and 32-35 for TDM410P I want to insert a new TE110P. How will these new channels be assigned? What is the dahdi_genconf logic for these assignments? Is there a way to force these assignments to prevent they to change if I insert/remove any card from the system? chan_dahdi.conf is a bit confusing to me. What makes the configuration for one channel be finished and start the configuration for the next channel? Would you point me any docs on this matter? Asterisk is 11.7.0 Dahdi is 2.7.0 Ubuntu 14.04.5 LTS Cheers Ethy -- _ -- 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