similar to: Good by asterisk 1.4? Please not.

Hello, short question: is there a possibility to use asterisk as an outbound proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly workarounds, everything. What is want to build is: SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy -> via UDP/RTP -> VoIP-Provider So Asterisk should just forward any incoming SIP messages (INVITE, REGISTER) to the

Hello, what about: "This feature means you can restart Asterisk after a failure (or asterisk restart itself with safe_asterisk), and keep existing calls up with only a few seconds of audio dropped." That would be a feature! there is a "other" pbx that has this feature... Anyone else would like to see that feature? just want to start some brainstorming, Kristijan

hi all, any one know, about echo cancellation with digium card, is it actually needed or it okay if we dont purchase because it increase price which half of new card, regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090827/8d6c680a/attachment.htm

Hi, I would appreciate some advice on the following: how does one use a single BRI port HFC chipset based ISDN cards with the latest DAHDI, libpri and Asterisk 1.8? For Asterisk 1.4 I would first install mISDN & mISDNuser and then build Asterisk 1.4. Now I noticed that Digium's HFC chipset based B410P is supported by the latest DAHDI & libpri but reading the source of the

Hello, i just had some fax abortions because of some packet loss. so i startet to examine in the pcap recording from the res_fax_digium, if the T.38 EC mode redundancy was really used. So i watched into it, and compared it with a t.38 pcap from spandsp (same asterisk setup, but with app_fax) and i see differences in t38.error_recovery (error-recovery: secondary-ifp-packets) With spandsp here are

Hello, i just made the reproducible watching: I send a Fax from asterisk (trunk) with spandsp (latest snapshot) via T.38 -> Audiocodes Mediant 2000 (FW 5.60.43.5) -> PSTN Fax With Internal timing Enabled, the Fax break after the first quarter from the first page is transfered. With Internal timing Disabled, the fax is transferred flawless. Both test with pthread timing module on a QEMU

Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting differences between them. So I prefer to double check ask for recommendations. Regards

Morning, Have some questions regarding receiving and sending faxes... 1:st example: exten => 101,1,Answer() exten => 101,2,Wait(3) exten => 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten => 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf) exten => 101,5,System(mutt -s 'New FAX for you sir' -a

Hi, i have just install debian wheezy and xen-hypervisor-4.1-amd64 helped with http://wiki.debian.org/Xen http://wiki.xenproject.org/wiki/Xen_Beginners_Guide#Installing_Debian_Squeeze created test.cfg http://pastebin.com/D5Wsx6eB xl create test.conf Parsing config file test.conf xc: info: VIRTUAL MEMORY ARRANGEMENT: Loader: 0000000000100000->0000000000174170 TOTAL:

Hi, trying to compile but have problems: + /root/xen-unstable.hg-rev/tools/../scripts/git-checkout.sh git:// xenbits.xensource.com/qemu-xen-unstable.git82db8de16530f016809264d3179823999d702849 qemu-xen-traditional-dir Cloning into qemu-xen-traditional-dir-remote.tmp... fatal: read error: Connection reset by peer git down? Best Regards, Kristijan Lecnik

Hi, Having problems with getting either RxFax or FaxReceive to compile. Running Asterisk 1.4 on CentOS 5. Does anyone have the free/open source executables that you could send me? Thanks for your help! P. S.: TxFax and FaxSend would also be appreciated.

Greetings, this is less of a bug report or a help request, but we would like to know if someone can explain the following: Environment: Centos 7 with Dovecot 2.3.4-2 default_pass_scheme = BLF-CRYPT password hash in database : BLF-CRYPT login = works default_pass_scheme = SHA512 or SHA256-CRYPT password hash in database : BLF-CRYPT login = also works default_pass_scheme = BLF-CRYPT password

We recently upgraded from dovecot 2.2 to 2.3.7.1-1 Not many, but some users are experiencing difficulties. The dovecot directors log: Aug 21 14:28:49 director01 dovecot: pop3-login: Disconnected (no auth attempts in 0 secs): user=<>, rip=redacted, lip=10.0.0.120, TLS handshaking: SSL_accept() failed: error:1408A0C1:SSL routines:ssl3_get_client_hello:no shared cipher,

> You could configure default scheme as CRYPT. It covers these all. Otherwise > you need to make sure passwords have {SCHEME} prefix when it differs from > default or oddities occur. --- Thank you for the tip with CRYPT. Is there any explanation for this behaviour though? Why are BCRYPT hashes accepted when default_pass_scheme is set to SHA512-CRYPT and not vice versa? Is this

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.14 and 1.6.1.5. Asterisk 1.6.0.14 and 1.6.1.5 are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Asterisk 1.6.0.14 is the first full, non-security release since 1.6.0.10. The release candidate 1.6.0.11-rc1 was redone as 1.6.0.14-rc1 (which this release has been created

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.14 and 1.6.1.5. Asterisk 1.6.0.14 and 1.6.1.5 are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Asterisk 1.6.0.14 is the first full, non-security release since 1.6.0.10. The release candidate 1.6.0.11-rc1 was redone as 1.6.0.14-rc1 (which this release has been created

I hope that someone can help me here. Software: CentOS 7.6.1810, dovecot-2.3.5.1, dovecot-pigeonhole-2.3.5.1, exim 4.91-1 Example filter: # rule:[test] if header :contains "subject" "meow" { redirect :copy "ks at ratiokontakt.de"; } Mail log: Apr 16 11:29:02 frontend-17 dovecot: lmtp(testi at mrhoang.de)<25201><QPgZHF6gtVxxYgAARS1pjg>:

Hello I'd like to write a script that would make it easier for people to call in, listen to the IVR, and make an appointment (eg. "When? ASAP? A given day?" -> "Morning? Afternon", etc.) I assume I'm not the first one to try and write this type of IVR, so would appreciate any feedback on writing this. Thank you.

The Asterisk Development Team would like to announce the release of Asterisk 13.19.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.19.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following issue is resolved in this release: Bugs

The Asterisk Development Team would like to announce the release of Asterisk 15.2.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.2.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following issue is resolved in this release: Bugs fixed