similar to: I figured it out!!

Hello, Several people report that outgoing phone calls to our clients sound muffled, like they are talking underwater. Reported for both the Snom 870, and the polycom ip650. Incoming calls sound ok. Could this be a codec problem? My dialplan looks like: [general] port = 5060 bindaddr = 0.0.0.0 srvlookup = no tos_sip = cs7 tos_audio = ef registertimeout = 1 relaxdtmf = yes context =

All: I've starting building an asterisk system for our company, which has about 60 users. I am new to asterisk, so thank you for your patience. I've stripped the sip.conf and the extensions.conf down to the bare minimum: Here is my extensions.conf file [globals] [general] autofallthrough=no [default] [fromprovider] exten => YYYYYYYYYY,1,Dial(SIP/151,20) [phones] exten =>

Hi, I'm stuck on a silly thing. I need to get the "billsec" CDR value after a call. But I'm finding its always 0. Here's my test code: exten => *244*,1,Dial(Local/test@custom-tests/n,,g) exten => *244*,n,Noop(after dial duration is ${CDR(duration)} billsec is ${CDR(billsec)}) exten => *244*,n,Hangup [custom-tests] exten => test,1,Answer exten =>

Hello i use dovecot 2.0.14 , with exim 4.76 using dovecot-lda. We have the following problem: when I receive mail from the site http://www.transfer.ro, which is a file transfer site, most emails appear to be empty. Empty rows appear in email body slipped through the existing, and this makemy mail client to show me an empty mail. You can see the content only by viewing the message source.

Hello again dear list. Could you please help with this? Thank you for all support, you are great, and i am now at a late stage in the setup and tweaking this server, So I hope you can help me again. I Can't make include the context nighttime. Just to demonstrate if it works, I have a playback function there. But CLI reports: CLI [Jun 18 14:20:22] WARNING[2287]: pbx.c:9542

All: RKG needs an asterisk consultant to help us track down issues we are having with our system. Mainly dropouts and dropped calls. If you have experience in troubleshooting these issues, please contact me at email attached to this messages. Regards, Eddie -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emikell at rimmkaufman.com --

All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure

Hi All, Have completely moved off the old ESI system, and things have been going pretty good with the new server. I have one issue, which has been reported by several of our customers. I've tested it, and it does indeed seem to be a problem. When the customer is asked to dial in the first three letters of the person they are trying to reach, they will be routed to the wrong extension.

"sip show channels" shows some info about active sip channels, the current codec included. What does it say? jg" jg, sip show channels reports the Format as being ulaw for 17 active calls. Holds - no Peer User/ANR Call ID Format Hold Last Message Expiry Peer xxxxxxxxxx kbrown xxxxxxxx (ulaw) No Rx:

All, I'm looking for sip phones that support something other than openvpn. There are a lot of vendors in China (mainly Alibaba) that sell l2tp VPN phones. Are there any American vendors that support l2tp? Thanks, -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emikell at rimmkaufman.com -- <http://www.rimmkaufman.com>

All: Yesterday I discovered something interesting. I dialed 1800ANCESTRY from the asterisk system I am testing and got the number doesn't exist message. I then dialed the same number from our old system and it went through. I realized that the "Y" in ancestry made the number too long, and went back to my dialplan. How do I ignore numbers that are too long? Obviously,

Signed-off-by: Gerd Hoffmann <kraxel at redhat.com> --- drivers/gpu/drm/virtio/virtgpu_plane.c | 31 ++++++++++++++------------ 1 file changed, 17 insertions(+), 14 deletions(-) diff --git a/drivers/gpu/drm/virtio/virtgpu_plane.c b/drivers/gpu/drm/virtio/virtgpu_plane.c index bc4bc4475a8c..a0f91658c2bc 100644 --- a/drivers/gpu/drm/virtio/virtgpu_plane.c +++

There are many messages about samba usershares in the logs but no usershares are defined: # grep '\[2012/03/20' /var/log/samba/log.smbd | grep process_usershare_file | wc -l 1518 # grep --ignore-case usershare /etc/samba/smb.conf.source /etc/samba /smb.conf [no output] # ls -l /var/lib/samba/usershares/ total 0 The messages are typically like: [2012/03/20 10:05:05.002947, 0]

Hi :-) This is cross-posted from Linux Questions where it has not been answered. This Samba configuration does not put anything in /var/log/samba/log.audit as expected. The messages are triplicated into /var/log/{messages,syslog,user.log}. Here is the global section of smb.conf created by testparm smb.conf.source > smb.conf (no error messages) ===== smb.conf begins ===== [global]

All: My company has an existing ESI IVX E-class system with 45 phones. I can add one more card, to expand it another 6 phones, but it's $8000, and then the system will have to be replaced. I have the Asterisk server up and running, with 2 sip lines from the local phone service. (Thanks to you guys, it is working great!). I'm pretty sure this is the way the company will move, and

Hi all. I've been reading through Wi-Ki and at the extensions.conf file description (http://www.voip-info.org/wiki-Asterisk+config+extensions.conf) The author says this: "One day, someone is going to write a proper scripting language for Asterisk that can understand a simpler, easier (and more traditional) scripting syntax. All it would need to do is translate the "high

v2: also check src rect (Chia-I Wu). Signed-off-by: Gerd Hoffmann <kraxel at redhat.com> --- drivers/gpu/drm/virtio/virtgpu_plane.c | 35 +++++++++++++++----------- 1 file changed, 21 insertions(+), 14 deletions(-) diff --git a/drivers/gpu/drm/virtio/virtgpu_plane.c b/drivers/gpu/drm/virtio/virtgpu_plane.c index bc4bc4475a8c..59bf76d4a333 100644 ---

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Anyone have the telemarketer torture prompts? I would seriously like to revive this..... - -- James Finstrom -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH3I8qdloC7YyaIOoRAlAjAJ9Hp+3SS2Z8179HecWIETp4RVDzWQCeMizp fW2JPZdYl/uxG1ziUwYnHGo= =QPbv -----END PGP

I had to dig through the source and do a lot of debug printing, but I figured out that because I was using NavigableString objects generated from RubyfulSoup parsing lib, instead of String, the drb server was not properly unmarshalling the object. I fixed it by calling to_s on all my strings. ActiveRecord and the local ferret server handled this fine, but the remote one chokes. The error was

a while back i reported an issue where /dev/null was getting set to 600 perms after a system update. i finally figured out what it is. i don't care about failed logins and have limited space on some servers, so i symlinked /var/log/btmp to /dev/null. the initscripts package does a chmod 600 /var/log/btmp, so voila /dev/null gets changed. so now i know why it happened just to me. i now