similar to: Upgrading to 11.4.0 and ast_channel_make_compatible_helper: No path to translate

Hi After upgrading from 1.4 to 11.4.0, I am able to receive calls and direct them to extensions via defined trunks. However, when making outgoing calls I receive the following error: -- Executing [000441111111 at default:4] Dial("SIP/fixedline-00000004", "SIP/mydevice/00441111111,60,w") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/mydevice/00441111111

Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: [Jun 10 16:18:22] WARNING[4090][C-0000000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng-00000000 to Motif/+12025551212 at voice.google.com-da3c [Jun 10 16:18:22] WARNING[4090][C-0000000a]: app_dial.c:3032 dial_exec_full: Had to drop call because I couldn't make

I'm using R embedded in PostgreSQL (via PL/R), and would like to use it to create images. It works fine, except that I have to create every image in a file (owned by and only readable by the PostgreSQL server), and then use PostgreSQL to read from that file and return it to the client. It would be much nicer if I could plot images into an R variable (for instance, a matrix), and return that

Hi All I just installed a 2 node ,10 r2 rac on centos Linux 4.6 My rac install was done by following Jeff Hunters document (excellent work!) http://www.idevelopment.info/ For storage I used openfiler with the iscsi protocol http://www.openfiler.com/ everything went perfect... to the letter, I gracefully shutdown the 2 nodes , then the openfiler appliance, when I went to start it back up,

Hello, using the LLVM API, I've build one very simple function that adds two ConstantInts and returns the result. I noticed that, when I emit IR code, it is optimized to a simple "ret i16 42" when I add 40 and 2. I'd like to see the operations that are necessary to compute the result, though. Can I somehow disable this optimization in the pass, leading to more verbose IR code?

IRBuilder is a templated class, and one of the template arguments is the constant folder to use. By default it uses the ConstantFolder class which does target-independant constant folding. If you want to disable constant folding you can specify the NoFolder class instead, i.e. declare the builder as follows: IRBuilder<true, llvm::NoFolder> builder(body) On 26 Nov 2013, at 19:23, Daniel

Thanks Jud, I was able to re-encode my entire FLAC archive into 1.1.4, although on my slow computer (with all other apps closed) it took over 3 days to complete! Only one song returned an error and that was quite easy to fix. I did run into one issue though. I have a FLAC album in which the songs are all 24/96 and REFLAC fails when it runs. I want to re-encode them from compression level 5

I have begun testing dovecot as (primarly) a POP replacement. So far the only lacking bit is the logging. It would be great if pop3 could log a line at client exit with the following info: (wishlist follows :) o nr deleted mails + total bytes deletes o nr mails left in spool + bytes left o time spent o indication if TLS was used or not If you would lika generic logging API I have one you can

The Asterisk Development Team has announced the release of Asterisk 11.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.4.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Hi, I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan: [macro-ccdev2-rec] exten => s,1,MixMonitor(${ARG1},b) [outgoing-originate] exten => _X.,1,NoOp(Will send call to ${EXTEN}) exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z) [outgoing-originate-rec] exten => h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})

Hi, I was attempting to play with LUA in Asterisk, but couldn't compile 11.4.0 stable with LUA on my test box running CentOS 5.5 64bit. I've download lua-5.1 from lua.org, and have used the following as ./configure: ./configure --prefix=/usr/local/asterisk-11.4.0 --with-lua=/usr/src/asterisk/lua-5.1/src This runs well, and creates this output: http://pastebin.com/iKW707Bt Now, when I

Hello, I'm working on building Asterisk 11.4.0-rc1 with pjproject 2.1 instead of 2.0 due to a crashing issue resulting from ICE. https://issues.asterisk.org/jira/browse/ASTERISK-21696 Currently, I'm systematically going through each Makefile in every directory in pjproject and changing the paths that exist in the pjproject 2.0 included with Asterisk, so that I can successfully build

Hi I'm using asterisk 1.8 on CentOS 5 I'm initiating call recordings with MixMonitor and trying to pause them with the features.conf. Whenever I try to pause the recording the call dies. Is PauseMonitor incompatible with MixMonitor? Here are some key log excerpts features reload == Parsing '/etc/asterisk/features.conf': == Found == Registered Feature

Hey All, I followed the procedure in the XenIntro document to add a new device and trigger the frontend and backend drivers over xenbus. The issue that I am facing now is when the backend switches state, the frontend driver does NOT pick it up. I suspect that the frontend permissions to read part of the backend xenstore are not set properly. I did do a xenstore-chmod in the frontend trigger

Hi list! We currently have a PRI gateway composed by a box with two Digium quad-span PRI cards (a TE420 and a ). One of the cards is filled with TELCO1, while the other has first two slots filled with TELCO2, and 3rd slot with TELCO3. I am currently having (timer ?) issues on TELCO3 (span 7) D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing on-going calls to terminate.

I have 11.4.0 on an Amazon EC2 instance. SIP works fine, but I can't get iax to work. I've opened 4569 in the EC2 Security Group. I'm using the zoiper client. Using tcpdump I can see the zoiper packets coming in on 4569, but nothing shows on the asterisk cli. Frame 33: 79 bytes on wire (632 bits), 79 bytes captured (632 bits) on interface 0 0000 12 31 3b 12 40 84 fe ff ff ff

Hi, I have a difficulty on my Asterisk's database.How can I get the info about list of ringing agents on my queue In console : -- Started music on hold, class 'default', on DAHDI/77-1 *-- SIP/6002-00cc0f90 is ringing -- SIP/6004-00c23270 is ringing -- SIP/6005-00be6220 is ringing* -- SIP/6004-00c23270 answered DAHDI/77-1 -- Stopped music on hold on DAHDI/77-1

So, if I remove the hostdev element related to the USB device, running `lsusb` and `usb-devices` inside the container lists the proper device, but I do not see the appropriate character device bus for the Android device under /dev/bus/usb. We are not entirely sure that `lsusb` and `usb-devices` queries the appropriate location - it somehow gets information from the host, not the guest container,

Can you see the USB device in the lxc? My answering assumes you cannot see the USB device in the container. If you can see it, please ignore my answer. According to my understanding, “hostdev mode='subsystem' “ is used with KVM only. We use this if we want to passthrough a device to a KVM virtual machine. After doing that, the host machine will lose control of the device. If container is

Hi, I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now when a caller is placed into Queue and gets connected with Member, I want to record the call. It does record the call when I use MixMonitor() before placing the caller into Queue, but not when MixMonitor() is used in macro which is called upon Member answering the call. Following is my dialplan... [mixmonitortest]