similar to: Is H323 dying?

i have had some problems with the H323 channel ... Other party not anwsering SIP 2 H323 bridge. the chan_oh323 solves the problem. Use it. (Even though it is quite complicated to install but READ the README file) Nahuel that should solve it!! Kido -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi all, i need an asterisk board that can support up to 30 FXS ports. i found the following on digium: TDM400P. It only has 4 port, does it mean that i have to buy 8 of those to have my 30 FXS line? Or is there any channel bank solution that digium or other providers can provide us with in order to do that. Experienced asterisk user, how do you manage to make PBXs for 30 to 100 regular phones?

Hello all, I found a LSI161SCREV2 Dialogic board in one of my drawers, and i was wondering if by any luck, i could make some magic happen with asterisk ... If asterisk does not support it, is there any PSTN to H323 or PSTN to SIP gateway that support this dialogic card and that can be connected to an Asterisk Box? Digium, I PROMISE that I will buy my cardf rom you once my tests are conclusive

I can't seem to get music on hold working, it tries to work, but I just hear strange noises on the extension.. Here is some debug info. Looks like mpg123 starts fine, but I hear nothing. I'm on todays CVS build. -- Executing Answer("SIP/2203-062c", "") in new stack -- Executing MusicOnHold("SIP/2203-062c", "default") in new stack --

Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is

Hello, Digium support kindly proposed to ship a TE120P card to help resolve the issue. I plugged in the card, and introduced the loopback plug. I cleared the red alarm for a while and then i started seeing alarm switching from Yel/Recovering to Blue/Rec with a lot of IRQ Misses. I call Digium that assisted us, and we noticed IRQ sharing with the VGA adapter and the Ethernet port. I changed

Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to

Hello all, Has anyone use the Supermicro PDSME+ in combination with the TE110P successfully? My experience so far is not very good. I am running trixbox 2.0 but: 1) with zttool I am getting IRQ Misses. Don't seem to have IRQ conflict, but I am now running my SATA HD in DMA. And I am not able to set it in DMA(HIO... operation not permitted) 2) With zttool the Alarm is RED 3) When I do the

sorry - hope this question is not tooo silly, but i needed to "autologin" to a remote machine found out that this works fine for me: sftp -opassword=PASSWORD USER at 192.168.1.1 << EOF cd ANYDIRECTORY get ANYFILE bye EOF why isn't this (sftp -opassword=SECRET USER at 192.168.1.1), setting the password with -opassword=PASSWORD, documentated anywhere? bug or feature? kind

Hello, I'm planning a server migration and being able to mount xfs file systems with the live cd would be a cruical feature. So before I download and try ... can anyone tell me whether the xfs is included in the 5.2 live cd? Later on I'm planning to install a new system with xen, 3ware 9550SX-4LP and xfs. The xen domains are of course located on xfs partitions. Do these features come

Hi, I'm trying to find out what is the minimum release I need to use to have this field 'state' available ? <channel type='unix'> <source mode='bind' path='//var/lib/libvirt/qemu/dummy_agent'/> <target type='virtio' name=dummy' state='connected'/> <alias name='channel0'/> <address

Hi, we are using SAMBA 3.6.1-1 (updating this archlinux machine is tooo ugly) and 3.6.6-1 on archlinux with the LDAP (Server version is 2.4.26-3) backend and manage the users, groups and computer by using the smbldap-tools. Currently we are experiencing the following problems: 1. changing the passwords takes longer than 30 seconds <- That's bad because we are using a gigabit ethernet

As I am working my way to understand Asterisk, I have a couple of questions that hopefully someone will answer. I know there are differences between a PBX switch and a CO switch. Can Asterisk completely replace and act as a CO switch? Are there any telecoms out there using Asterisk as a CO switch? If so, how well does it work? If not, why not?

Okay, I download login generator and I used it on my new app but, it''s not working correctly. If the user logins on the same session as he created he will login if it isn''t the same session it fails. What''s going on? -- Posted via http://www.ruby-forum.com/.

I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and they all are able to register and make calls with no problem . My voip carrier supports gsm as well as ilbc .. Server takes calls from sip phones , does call recording in between and forwards to voip carrier . My problem is that half of my softphones use ilbc and rest use gsm and my provider supports both gsm as well as

Hello, I'm experiencing some problems with running IAX2 protocol on quite reliable link with G729A codec. My customer has 2mb FR link to the Internet used in about 20%. Ping statistics: 50 packets transmitted, 49 received, 2% packet loss, time 49496ms rtt min/avg/max/mdev = 9.308/13.126/33.307/4.851 ms Everything would be great, but the quality isn't good enough. I have 2mb/512kb DSL

Hello I like to make calls to an h.323 device. I'm using Nuphone h323. Compiled everything okay "I Guess" When I make a connection * SIP > h323 device, the phone is ringing and then * tells me "No one available....." and disconnect Thinking this is a codec problem and check in CLI h.323 show codecs and * shows nothing. I try many combination in the h323.conf like.

> On Apr 24, 2017, at 5:30 PM, Joerg Sonnenberger via llvm-dev <llvm-dev at lists.llvm.org> wrote: > > On Mon, Apr 24, 2017 at 11:06:36AM -0700, Matthias Braun via llvm-dev wrote: >> Would be cool to create a suite of extreme inputs, maybe a special llvm >> test-suite module. This module would contain scripts that produce >> extreme inputs (long basic blocks,

I am new to Puppet and very eager to apply it to a project. But I am somewhat stymied by the learning curve. So far I''ve found many very simple examples of how to modify a file or add a user and a very complex example http://reductivelabs.com/trac/puppet/wiki/CompleteConfiguration . I have not been able to find any other examples of a total configuration tree (ie /etc/puppet/*)

hi all, After having some discussions in the IRC, I am trying here to come up with a proposal for GSoC 2008 for improving the llvm nightly tester[1].Following are the ideas and suggestions that came up in the discussion, if you have any comment or any other suggestion please add them to the list. I have some doubts in some places. 1. Improvements to the perl script which manage actual testing