similar to: Asterisk bugs make a right mess of RTP

I am stuck in channel PJSIP trying to see the real flow of SIP messages, what in regular sip we used to type "sip set debug on" Also, is there an automated way to convert sip.conf options to pjsip.conf? Philip

I can not connect to my linux box from my Mac. I get a 'Could not connect to the server because the name or password is not correct' error. I'm not sure where to check to even see what username/pwd samba is using. If I open up my smbpasswd file, it looks like this: drrl:501:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx:[UD]:LCT-402FFFE4: Is there

Hi, I have run into the following strange behavior and wanted to ask for some advice. For the C program below, function sum() gets inlined in foo() but the code generated looks very suboptimal (the code is an extract from a larger program). Below I show the 32-bit x86 assembly as produced by the demo page on the llvm home page ("Output A"). As you can see from the assembly, after

I don't know much about this, but maybe -mllvm -unroll-count=1 can be used as a workaround? /Patrik Hägglund -----Original Message----- From: llvmdev-bounces at cs.uiuc.edu [mailto:llvmdev-bounces at cs.uiuc.edu] On Behalf Of Brent Walker Sent: den 28 mars 2012 03:18 To: llvmdev Subject: [LLVMdev] Suboptimal code due to excessive spilling Hi, I have run into the following strange behavior

On Fri, Sep 1, 2017 at 9:13 AM, Joshua Colp <jcolp at digium.com> wrote: > On Fri, Sep 1, 2017, at 09:01 AM, Dave Topping wrote: > > http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/ > > This specific issue exists in a lot of different implementations and > devices. Unfortunately there's nothing within SDP that guarantees or > provides what the source of

Hi everybody, after many weeks of searching/trying, I still can't manage to get vista to use/save roaming profiles with samba as PDC... It has apparently worked a few times in the beginning since the profiles directories are populated. Could it be related to a Vista update or SP1? Here's what I did so far: [HKEY_LOCAL_MACHINE\SOFTWARE\Policies\Microsoft\Windows\System]

Hello using pjproject 2.5.5 using asterisk-certified-13.8-cert1 Compiled pjproject 2.5.5 with : ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound --disable-opencore-amr Compiled Asterisk 13 with ./configure --libdir=/usr/lib64 All pjproject modules are selectable in menuselect, so here no problem.

Today I faced a problem. Please help to solve this problem. Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware v2.06(AAGJ.9)C1 Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk). Call using early media (183 Session in progress) and rtp_timeout=10. After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] res_pjsip_sdp_rtp.c: Disconnecting channel

I am trying to rename column names in a dataframe within a function. I am seeing an error (listed below) that I don't understand. Would be grateful of an explanation of what I am doing wrong and how I should rewrite the function to allow me to be able to rename my variables. Thanks. # Test Function myfunc <-function(var){ d = c(1,2,3,4,5) dts =

Oh god it works ! to switch cisco to upd I used config: <transportLayerProtocol>2</transportLayerProtocol> with udp it works well, thanks for your help :) > On 24 Feb 2015, at 17:02, Joshua Colp <jcolp at digium.com> wrote: > > If you use UDP with force_rport=no it'll work. > If you use TCP then set rewrite_contact=yes so it'll reuse the established TCP

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:

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:

I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001 at 192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes.

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:

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:

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:

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:

Hi, my dovecot installation works since months and clients authenticate using CRAM-MD5. But today I got the first chance to test an client that supports DIGEST-MD5 - and it doesn't work. Because of lack of other supporting clients and servers I'm now at the point I don't know which side is to blame. The error I get after the client answers the servers challenge is "-ERR

Hi, I need to create something like this: <?xml version="1.0" encoding="UTF-8"?> <Container> <id>aQlfVHX+qPM</id> <lifetime>2009-09-19T08:14:55Z</lifetime> </Container> The response should contain the next headers: Content-Type=`application/vnd.3gpp+xml`

Again rc, just in case I broke something. Or maybe I should just start creating nightly CVS snapshots. Anyway, fixes the few bugs people have reported lately: - Mails with nested MIME parts might have caused incorrect BODY and BODYSTRUCTURE fetches and sometimes might have crashed dovecot (assert at imap-bodystructure.c). If client had already successfully done the BODY fetching a