similar to: Choppy Sound On Bridging From SIP->IAX

I understand that the Asterisk SLA implementation is somewhat different from most key systems and PBX systems. I also understand that in Asterisk, one does not put an SLA line on hold since it is just a MeetMe conference. However, is there any way to make the BLF flash when the answering party on the Asterisk system presses the hold key on their set and leaves the calling party alone in the

Does anybody know of any commercial providers of IAX termination with DIDs in the Seattle, WA area? I believe the area codes are: 425, 206, 253 Failing any commercial providers, is there anybody in the seattle area running Asterisk with a PRI coming in who might be willing to sell me an IAX trunk with a DID in Seattle? -- ____________________________________________________________ Muiz

I am using IAX soft clients (firefly, IAXComm, IAXPhone) from a Win2K machine on a NATed private LAN to connect to my IAXTel account. All of my calls to outside numbers(1-800, 1-877, etc) seem to be disconnected after 9 seconds. Has this limit been placed on IAXTel? My call to the echo test number at 1-700-999-9613 seemed to run for as long as I wanted. --

I am running a LEAF Bering firewall with a DMZ at one of my client''s sites. The client has a Windows IIS server set up in the DMZ. The DMZ rules block all outgoing connections on all ports. The client wants to be able to run Windows Update on the server, which uses ports 80 and 443. However, opening all port 80 and 443 connections from the server to the net zone is not an option,

Does anybody know what happened to the opencall.org website? I can't get into the home page or the ftp site.

Hi, How many user's you have assigned on that directory? --Regards Ashishkumar S. Yadav On Tue, Mar 24, 2015 at 5:30 AM, muiz <muiz at 163.com> wrote: > Hi, James, > The file system is xfs (/dev/mapper/centos-home xfs 1891292900 > 1556745108 334547792 83% /home) > > > Thanks and best regards, > Muiz > > At 2015-03-23 19:36:02, "James

Dear Concerns, I would like to share a very strange problem. I am from Pakistan/Islamabad. Last month i was on trainning from Askari Bank Limited (Juniper). Here in Askari i m running NMS--> MRTGs using CentOs 4.4. On trainning i recieved a call from collique saying "when i "su -l" NMS says "root user doesnot exist". and also MRTGs not working. well i was amazed how

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This message was incorrectly discarded by Mailman -- I''m still looking for the cause. - -Tom |On Mon, 08 Nov 2004 14:17:17 -0800 |> "Muiz Motani" <muiz@i-dist.com> wrote: |> | |>> > Has anybody found an elegant way to solve this problem? I am sure that I |>> > am not the only one who has run into

On 20.03.2015 02:14, muiz wrote: > Dear all, > > > I have a Centos 6 (64bit) server used for samba and ACL control. > There has a folder which a lot of user can access, when I add a new user's ACL, but failed, do you know how to resolve this problem? use xargs like in: |find . -name "*.pdf" -print0 | xargs -0 RUNACOMMAND robert | > > > [root at s1

muiz wrote: > Dear all, > > > I have a Centos 6 (64bit) server used for samba and ACL control. > There has a folder which a lot of user can access, when I add a new user's ACL, but failed, do you know how to resolve this problem? > > > [root at s1 abc]# getfacl /abc > # file: abc > # owner: root > # group: root > user::rwx > user:user01:--- >

On 3/19/2015 6:14 PM, muiz wrote: > I have a Centos 6 (64bit) server used for samba and ACL control. > There has a folder which a lot of user can access, when I add a new user's ACL, but failed, do you know how to resolve this problem? put all those users in a group, and delete all the user ACLs on that directory, just add the single group ACL -- john, recycling bits in santa

Dear all, Dear support and users: Sorry to trouble you! I configure the shorewall firewall to forward ftp and ssh port to another server, but failed. Can you help me check? I cannot login both SSH 2222 and ftp! Below is my environment: (attachment is shorewall dump) 1. Gateway (FC6) 1.1) eth0: lan static IP: 192.168.1.20 1.2) eth1: external public static IP:

Hello, everyone I'm relatively new to Asterisk (and VOIP in general), but I have a project that it will really help with. So, I setup a test system on an ancient 400MHz P3 we had lying around. It worked great. I had a test dialplan working, and had no trouble connecting to it with SIP using 3CX SoftPhone over our LAN (and over the Net through our NAT). So, we went ahead and bought a

muiz wrote: > Hi, James, > The file system is xfs (/dev/mapper/centos-home xfs 1891292900 1556745108 334547792 83% /home) XFS has a low limit on the number of entries in an ACL (about 20) However, I don't know why you are getting the error 'Argument list too long' - I get 'Operation not permitted' when I try to and an entry that goes over the limit ... I

Dear Concerns, I have backup script which TAR the data, makes one copy on local hardisk and then FTP to another machine. Last few days i m facing problem with it. The cron job only makes the copy automatically on local disk but doesnot automatically FTP to another machine. I ran the script manually, the script runs successfully. For trouble shooting i displayed the output in some file and shown

Hi there, We're having some complaints of choppy audio from our SIP customers. Asterisk is showing no errors, but I'm getting a lot of these in my syslog: Feb 17 13:34:31 ntop[2863]: **WARNING** packet truncated (14654->8232) The first number varies, but the last number is always 8232. I've read that this is a common MTU size, but none of our interfaces have an MTU of 8232.

Hi folks, I'm experiencing some problems with sound through phpAGI ... What I'm trying to do is a menu, doing some database lookups and so ... But sometimes the sound become too choppy ... just sometimes .. like 1 of 5 calls ... but is a big percentage ... And I have my current menu on the dialplan that I have no problems with it ... I'm using .gsm for both but different

Hi all, I have my asterisk setup as following: IP 2 x E1 x-lite <-------> Asterisk -------> PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data

Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw => ulaw is choppy, ulaw => alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with asterisk 1.2.13? -Benoit-

Hi Guys I'm conecting 2 astersk servers using this arquitecture (Ext softphone)<==sip==>(asterisk 1)<====iax2 trunk====>(asterisk 2) <===alaw==>(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call from Ext to the pstn, i can hear perfect but they tell me that sound really choppy, i tried using several codecs (same problem) but i