[asterisk-users] Certified Asterisk 16.8-cert3 Now Available
The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert3. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert3 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-28953 - res_pjsip_session: Preserve stream label (Reported by Joshua C. Colp) * ASTERISK-28938 - core_unreal / core_local: Add support for multistream and re-negotiation (Reported by Joshua C. Colp) * ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC (Reported by Joshua C. Colp) * ASTERISK-28944 - bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation (Reported by Joshua C. Colp) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-16.8-cert3 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
Re: [asterisk-users] CallerID fail with Voicetrading operator
Hi Antony Le 18/06/2020 à 20:19, Antony Stone a écrit : On Thursday 18 June 2020 at 19:57:03, Administrator wrote: does some people here use https://voicetrading.com which is a Dellmont service from Netherlands. At the high begining they were also known as Finarea (CH and DE mixed Co) Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or equal to CALLERID(num). We tried replacing + with 00, same problem. There support said they don't receive any callerID and we know that it's not correct as we don't have problem with other providers like PeopleFone, Sipgate or French Operators. Can you capture an INVITE packet from your system to theirs and show them the headers? Capture done and sended, thanks, -- 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
Re: [asterisk-users] Mail2Fax
On Wed, 2020-06-17 at 18:10 +0200, basti wrote: > txfax seem to be a port of spandsp. it is also old. > Is there a newer way to send fax via asterisk. I don't know if it's newer, but I use "sendfax" -- _ -- 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] Mail2Fax
On 6/19/20 4:23 AM, basti wrote: Fax is not send. No Sip stuff is show in log. I don't know what is wrong here. Best regards Basti, This really belongs on the iaxmodem mailing list or the HylaFAX+ Mailing list. Lee Howard is the author of both packages and very responsive. Doug -- _ -- 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] Mail2Fax
Hello, i try to setup asterisk with hylafax: the config is: egrep -v "(^#|^$)" /etc/hylafax/config.ttyIAX0 CountryCode:49 AreaCode: xxx FAXNumber: +49 LongDistancePrefix: 0 InternationalPrefix:00 DialStringRules:etc/dialrules ServerTracing: 1 SessionTracing: 11 RecvFileMode: 0600 LogFileMode:0600 DeviceMode: 0600 RingsBeforeAnswer: 1 SpeakerVolume: off GettyArgs: "-h %l dx_%s" LocalIdentifier:"example" TagLineFont:etc/lutRS18.pcf TagLineFormat: "From %%l|%c|Page %%P of %%T" MaxRecvPages: 25 ModemType: Class1 # use this to supply a hint Class1Cmd: AT+FCLASS=1.0 # command to enter class 1.0 Class1PPMWaitCmd: AT+FTS=7# command to stop and wait before PPM Class1TCFWaitCmd: AT+FTS=7# command to stop and wait before TCF Class1EOPWaitCmd: AT+FTS=9# command to stop and wait before EOP Class1SwitchingCmd: AT+FRS=7# command to stop and listen for silence Class1RecvAbortOK: 200 # wait 200ms for abort response Class1FrameOverhead:4 # 4 byte overhead in recvd HDLC frames Class1RecvIdentTimer: 4 # 35+5secs waiting for ident frames Class1TCFMaxNonZero:10 # max 10% of data may be non-zero Class1TCFMinRun:1000# min run is 2/3rds of TCF duration cat /etc/iaxmodem/ttyIAX0 device /dev/ttyIAX0 owner uucp:uucp mode 660 port 4570 refresh 60 server 127.0.0.1 peername 591 secret codec alaw cidname Fax cidnumber xxx nojitterbuffer cat /etc/asterisk/iax_additional.conf ;; ; Do NOT edit this file as it is auto-generated by FreePBX. ; ;; ; For information on adding additional paramaters to this file, please visit the ; ; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ; ; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ; ; is totally deliberate. ; ;; [591] deny=0.0.0.0/0.0.0.0 disallow=all secret= transfer=yes context=from-internal host=dynamic type=friend port=4570 qualify=yes allow=alaw dial=IAX2/591 accountcode= permit=0.0.0.0/0.0.0.0 requirecalltoken=no secret_origional= callerid=Hylafax <591> setvar=REALCALLERIDNUM= Asterisk log show only [19-06-2020 10:18:46] VERBOSE[3245] chan_iax2.c: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE [19-06-2020 10:18:46] VERBOSE[3245] chan_iax2.c:Timestamp: 00014ms SCall: 01337 DCall: 0 127.0.0.1:4570 [19-06-2020 10:18:46] VERBOSE[3245] chan_iax2.c: [19-06-2020 10:18:46] VERBOSE[3251] chan_iax2.c: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK [19-06-2020 10:18:46] VERBOSE[3251] chan_iax2.c:Timestamp: 00014ms SCall: 20234 DCall: 01337 127.0.0.1:4570 [19-06-2020 10:18:46] VERBOSE[3251] chan_iax2.c: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG [19-06-2020 10:18:46] VERBOSE[3251] chan_iax2.c:Timestamp: 00014ms SCall: 20234 DCall: 01337 127.0.0.1:4570 [19-06-2020 10:18:46] VERBOSE[3251] chan_iax2.c:RR_JITTER : 0 [19-06-2020 10:18:46] VERBOSE[3251] chan_iax2.c:RR_LOSS : 0 [19-06-2020 10:18:46] VERBOSE[3251] chan_iax2.c:RR_PKTS : 1 [19-06-2020 10:18:46] VERBOSE[3251] chan_iax2.c:RR_DELAY: 40 [19-06-2020 10:18:46] VERBOSE[3251] chan_iax2.c:RR_DROPPED : 0 [19-06-2020 10:18:46] VERBOSE[3251] chan_iax2.c:RR_OUTOFORDER : 0 [19-06-2020 10:18:46] VERBOSE[3251] chan_iax2.c: [19-06-2020 10:18:46] VERBOSE[3251] chan_iax2.c: Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK [19-06-2020 10:18:46] VERBOSE[3251] chan_iax2.c:Timestamp: 00014ms SCall: 01337 DCall: 20234 127.0.0.1:4570 Fax is not send. No Sip stuff is show in log. I don't know what is wrong here. Best regards -- _ -- 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
