Re: [asterisk-users] Forbidden call
On Thu, 11 Jun 2020, Jerry Geis wrote: I have a call from a call file: This looks a lot more like an AMI event than a call file. In any case, it doesn't matter. Action: Originate Async: yes Channel: SIP/2012 Codecs: ulaw,alaw,gsm Context: dialout Exten: callprogress Priority: 1 Timeout: 2 Variable: SIPADDHEADER="Alert-Info: Ring Answer" ActionID: 100014 CallerID: Axis < 525 > The SIP/2012 is a IP Speaker on the computer. The error is: [Jun 11 15:44:45] WARNING[8132]: chan_sip.c:24191 handle_response_invite: Received response: "Forbidden" Why am I getting "Forbidden" ? Its a call file on my server It's not a call file permissions thing. That would be a different error and reported by something before chan_sip. the speaker is directly connected to my server. How is an IP speaker 'directly connected?' Do you mean directly from the Ethernet on the speaker to a NIC on the computer? It doesn't matter, just curious :) The only thing that will tell you what is going on is the packets. Crank up 'sip set debug on' and see if that yields a clue. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- 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] Forbidden call
I have a call from a call file: Action: Originate Async: yes Channel: SIP/2012 Codecs: ulaw,alaw,gsm Context: dialout Exten: callprogress Priority: 1 Timeout: 2 Variable: SIPADDHEADER="Alert-Info: Ring Answer" ActionID: 100014 CallerID: Axis < 525 > The SIP/2012 is a IP Speaker on the computer. The error is: [Jun 11 15:44:45] WARNING[8132]: chan_sip.c:24191 handle_response_invite: Received response: "Forbidden" Why am I getting "Forbidden" ? Its a call file on my server and the speaker is directly connected to my server. Thanks Jerry -- _ -- 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] x-ast-orig-host - How is this IP taken ?
On Wed, Jun 10, 2020 at 3:25 PM Administrator wrote: > Hi list, > > We have a strange behavior: a customer Snom300 behind a public FW has > contact like > > contact : > sip:user@x.y.39.147:2048;x-ast-orig-host=169.254.252.1:2048 x-ast-orig-host is a header we add to incoming requests when rewrite_contact is on AND the host we get the request from is different from the host in the contact URI. We do this so we can restore the original contact URI when we send responses. Here's the scenario... A client behind a firewall sends Asterisk a REGISTER request. The contact URI is probably going to be a non-routable ip address like 192.168.0.1 but the host the packet comes from will be the public ip address of the firewall. In order to properly route responses and subsequent requests, the "rewrite_contact" option can be used to force Asterisk to substitute the private ip address in the contact header with the public ip address we actually got the packet from. This way we send responses and new requests to the public ip address. This all works well except for 1 scenario... When a client sends a REGISTER request, they can use the IP address in the contact header of the response to match it to the request. If we've rewritten the contact header, they won't be able to match it. So we save off the contact host into that x-ast-orig-host header and when we send responses back to the client, we still send it to the public ip address git we reset the contact host back to what was in the original request. We then strip all x-ast* headers before we actually send the packets. > > The phone can place calls but not receive any. Also, qualify give > unreachable which seems correct when looking the x-ast-orig-host IP. > Problem is that the local IP of this phone is 192.168.1.75 > Well, 169.254.x.x addresses are Automatic Assigned Ip Addresses assigned by the device itself when it can't get a dhcp ip address. It's highly unlikely that things are going to function normally if the device doesn't have a real ip address. > > Question: how asterisk sets this IP ? It looks for us like a FW issue as > we have other customers with approaching local network organisation and > which are not facing this problem. > See above. You should also check that the router the phone is connected to does NOT have SIP ALG turned on because that will mess with the SIP headers. > > Thanks for any hint. > > -- > Daniel > > -- > _ > -- 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 -- George Joseph Asterisk Software Developer direct/fax +1 256 428 6012 Check us out at www.sangoma.com and www.asterisk.org [image: image.png] -- _ -- 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] Attempting to get BLF working with linphone
Lately, I read [1]. So it seems both Jitsi desktop and Linphone are on par, on this ;-))) [1] https://community.jitsi.org/t/busy-lamp-field-bug/15931 Le ven. 5 juin 2020 à 13:34, John Hughes a écrit : > On 26/05/2020 15:33, Olivier wrote: > > Hi John, > > 1. Could you get any further, in your quest for working BLF with linphone ? > > The patches to get linphone-3.12 BLF working with Asterisk are here: > > http://perso.calvaedi.com/~john/linphone-3/ > > They're pretty damnned trivial: > > 1. add the "Accept" header to the SUBSCRIBE message so asterisk doesn't > reject it. > > 2. don't trash the SIP dialog if the SUBSCRIBE refresh is rejected because > of a stale nonce. > > 3. If asterisk says the user is on the phone set the status to on the > phone. > > All except the 3rd one are compatible with linphone-4. Implementing the > same feature with linphone-4 is left as an exercise for the reader. > > > 2. Have you tried with a different Linphone version (4.12 is pending on > Linux, packaged as an AppImage, or 4.11 exists on iOS/Android/Win10) ? > > > Version 4 of linphone is, frankly, rubbish. I have managed to hack it to > the point where presence shows green for connected contact and grey for > disconnected. However this requires setting the "send subscribe" flag in > the linphone contacts db and linphone 4 has no UI for setting this flag, > you have to do it using sqlite3 directly (or setting up your contacts in > linphone 3). > > -- > _ > -- 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 -- _ -- 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 17.5.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 17.5.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: --- * ASTERISK-28940 - /channels/create doesn't get any parameters from the body (Reported by sungtae kim) * ASTERISK-28932 - res_pjsip_logger writing too big packets (Reported by nappsoft) * ASTERISK-28921 - Wrong return value check for fwrite when writing to pcap file (Reported by nappsoft) * ASTERISK-28794 - res_pjsip: Crash when escaping during URI printing (Reported by nappsoft) * ASTERISK-28884 - x-ast-orig-host not filtered out from request URI and To header (Reported by nappsoft) * ASTERISK-28871 - res_pjsip_session: Unnecessary re-Invite on call answer (Reported by Alexei Gradinari) * ASTERISK-28903 - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES. (Reported by Alexander Traud) * ASTERISK-28898 - bridge_softmix: Conference bridge not passing silent rtp packets (Reported by Jonathan Hunter) * ASTERISK-28892 - res_musiconhold: Module res_musiconhold throws false warning (Reported by Nicholas John Koch) * ASTERISK-28904 - RTP ICE leaks the memory (Reported by sungtae kim) * ASTERISK-26780 - res_pjsip: PJSIP Registration Fails when transport=transport-udp6 (Reported by Peter Sokolov) * ASTERISK-28854 - SIGSEGV when pjsip show history encounters IPV6 address (Reported by Roger James) * ASTERISK-28797 - [patch] tcptls: Fix notice when TLS is enabled but not configured. (Reported by Alexander Traud) * ASTERISK-28804 - [patch] app_osplookup.c: Avoid a format truncation. (Reported by Alexander Traud) * ASTERISK-28776 - Non async-signal-safe syscalls used after fork before exec (Reported by nappsoft) * ASTERISK-28870 - streams: One memory leak and one issue cloning streams (Reported by George Joseph) * ASTERISK-28829 - app_queue: leaking stasis subscription when Redirecting call (Reported by lvl) * ASTERISK-25844 - app_queue: Ghost channels in "core show channels" output (Reported by Etienne Lessard) * ASTERISK-28859 - pjsip: Increase maximum candidate count (Reported by Joshua C. Colp) * ASTERISK-22920 - Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling (Reported by Shlomi Gutman) * ASTERISK-28852 - Unprotected access to nochecksums variable, causes build failures (Reported by Guido Falsi) * ASTERISK-28848 - app_fax: Compile. (Reported by Alexander Traud) Improvements made in this release: --- * ASTERISK-28895 - res_pjsip_logger: Add tons'o'functionality (Reported by Joshua C. Colp) * ASTERISK-28896 - ari: Add support for specifying variables on channel create (Reported by Joshua C. Colp) * ASTERISK-28879 - pjproject has race conditions in it's build system (Reported by Guido Falsi) * ASTERISK-28866 - third-party/pjproject/configure.m4 contains bashisms (Reported by Guido Falsi) * ASTERISK-28853 - Missing include on FreeBSD (Reported by Guido Falsi) * ASTERISK-28832 - chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio (Reported by Peter Turczak) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0 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 16.11.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 16.11.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.11.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: --- * ASTERISK-28940 - /channels/create doesn't get any parameters from the body (Reported by sungtae kim) * ASTERISK-28932 - res_pjsip_logger writing too big packets (Reported by nappsoft) * ASTERISK-28921 - Wrong return value check for fwrite when writing to pcap file (Reported by nappsoft) * ASTERISK-28794 - res_pjsip: Crash when escaping during URI printing (Reported by nappsoft) * ASTERISK-28884 - x-ast-orig-host not filtered out from request URI and To header (Reported by nappsoft) * ASTERISK-28871 - res_pjsip_session: Unnecessary re-Invite on call answer (Reported by Alexei Gradinari) * ASTERISK-28903 - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES. (Reported by Alexander Traud) * ASTERISK-28898 - bridge_softmix: Conference bridge not passing silent rtp packets (Reported by Jonathan Hunter) * ASTERISK-28892 - res_musiconhold: Module res_musiconhold throws false warning (Reported by Nicholas John Koch) * ASTERISK-28904 - RTP ICE leaks the memory (Reported by sungtae kim) * ASTERISK-26780 - res_pjsip: PJSIP Registration Fails when transport=transport-udp6 (Reported by Peter Sokolov) * ASTERISK-28854 - SIGSEGV when pjsip show history encounters IPV6 address (Reported by Roger James) * ASTERISK-28804 - [patch] app_osplookup.c: Avoid a format truncation. (Reported by Alexander Traud) * ASTERISK-28797 - [patch] tcptls: Fix notice when TLS is enabled but not configured. (Reported by Alexander Traud) * ASTERISK-28776 - Non async-signal-safe syscalls used after fork before exec (Reported by nappsoft) * ASTERISK-28870 - streams: One memory leak and one issue cloning streams (Reported by George Joseph) * ASTERISK-28829 - app_queue: leaking stasis subscription when Redirecting call (Reported by lvl) * ASTERISK-25844 - app_queue: Ghost channels in "core show channels" output (Reported by Etienne Lessard) * ASTERISK-22920 - Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling (Reported by Shlomi Gutman) * ASTERISK-28859 - pjsip: Increase maximum candidate count (Reported by Joshua C. Colp) * ASTERISK-28852 - Unprotected access to nochecksums variable, causes build failures (Reported by Guido Falsi) * ASTERISK-28848 - app_fax: Compile. (Reported by Alexander Traud) Improvements made in this release: --- * ASTERISK-28895 - res_pjsip_logger: Add tons'o'functionality (Reported by Joshua C. Colp) * ASTERISK-28896 - ari: Add support for specifying variables on channel create (Reported by Joshua C. Colp) * ASTERISK-28879 - pjproject has race conditions in it's build system (Reported by Guido Falsi) * ASTERISK-28866 - third-party/pjproject/configure.m4 contains bashisms (Reported by Guido Falsi) * ASTERISK-28853 - Missing include on FreeBSD (Reported by Guido Falsi) * ASTERISK-28832 - chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio (Reported by Peter Turczak) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0 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 13.34.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 13.34.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.34.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: --- * ASTERISK-28932 - res_pjsip_logger writing too big packets (Reported by nappsoft) * ASTERISK-28921 - Wrong return value check for fwrite when writing to pcap file (Reported by nappsoft) * ASTERISK-28794 - res_pjsip: Crash when escaping during URI printing (Reported by nappsoft) * ASTERISK-28884 - x-ast-orig-host not filtered out from request URI and To header (Reported by nappsoft) * ASTERISK-28898 - bridge_softmix: Conference bridge not passing silent rtp packets (Reported by Jonathan Hunter) * ASTERISK-28904 - RTP ICE leaks the memory (Reported by sungtae kim) * ASTERISK-28854 - SIGSEGV when pjsip show history encounters IPV6 address (Reported by Roger James) * ASTERISK-28797 - [patch] tcptls: Fix notice when TLS is enabled but not configured. (Reported by Alexander Traud) * ASTERISK-28804 - [patch] app_osplookup.c: Avoid a format truncation. (Reported by Alexander Traud) * ASTERISK-28776 - Non async-signal-safe syscalls used after fork before exec (Reported by nappsoft) * ASTERISK-28829 - app_queue: leaking stasis subscription when Redirecting call (Reported by lvl) * ASTERISK-25844 - app_queue: Ghost channels in "core show channels" output (Reported by Etienne Lessard) * ASTERISK-22920 - Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling (Reported by Shlomi Gutman) * ASTERISK-28859 - pjsip: Increase maximum candidate count (Reported by Joshua C. Colp) * ASTERISK-28852 - Unprotected access to nochecksums variable, causes build failures (Reported by Guido Falsi) Improvements made in this release: --- * ASTERISK-28895 - res_pjsip_logger: Add tons'o'functionality (Reported by Joshua C. Colp) * ASTERISK-28879 - pjproject has race conditions in it's build system (Reported by Guido Falsi) * ASTERISK-28866 - third-party/pjproject/configure.m4 contains bashisms (Reported by Guido Falsi) * ASTERISK-28832 - chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio (Reported by Peter Turczak) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.34.0 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
