similar to: Call-limit field

With some carriers the initial Audio (2-4 secs) seems to get cut off when using a Auto Attendant or Conf Meetme. Is there any known remedies for that. Just want to know if others have seen that esp. with Level 3. If Auto Attendant says - "Welcome to ABC bank" Caller only hears "Bank" -------------- next part -------------- An HTML attachment was scrubbed... URL:

We are running asteriskNow 1.4.18 and after a few days it becomes unresponsive and inbound INVITEs timeout. We just reboot the box to resolve it. But it seems to be occurring more regularly now. I am hesitant to move to latest version, but will do if needed. Any guidance or troubleshooting modes I may use will be helpful. -------------- next part -------------- An HTML attachment was

Sam, You might want to give glusterfs-3.12.1 a try instead. On Fri, Sep 15, 2017 at 6:42 AM, Sam McLeod <mailinglists at smcleod.net> wrote: > Howdy, > > I'm setting up several gluster 3.12 clusters running on CentOS 7 and have > having issues with glusterd.log and glustershd.log both being filled with > errors relating to null client errors and client-callback

Howdy, I'm setting up several gluster 3.12 clusters running on CentOS 7 and have having issues with glusterd.log and glustershd.log both being filled with errors relating to null client errors and client-callback functions. They seem to be related to high CPU usage across the nodes although I don't have a way of confirming that (suggestions welcomed!). in

FYI - I've been testing the Gluster 3.12.1 packages with the help of the SIG maintainer and I can confirm that the logs are no longer being filled with NFS or null client errors after the upgrade. -- Sam McLeod @s_mcleod https://smcleod.net > On 18 Sep 2017, at 10:14 pm, Sam McLeod <mailinglists at smcleod.net> wrote: > > Thanks Milind, > > Yes I?m hanging out for

Thanks Milind, Yes I?m hanging out for CentOS?s Storage / Gluster SIG to release the packages for 3.12.1, I can see the packages were built a week ago but they?re still not on the repo :( -- Sam > On 18 Sep 2017, at 9:57 pm, Milind Changire <mchangir at redhat.com> wrote: > > Sam, > You might want to give glusterfs-3.12.1 a try instead. > > > >> On Fri, Sep

Hi. I am trying to pass a variable from one Asterisk PBX to another. I'm using DUNDi with IAX2. Is there a way to do it? I tried the following but it fails. On peer1: [dundi-outgoing] switch => DUNDI/priv exten => s,1,Set(CDR(userfield)=test) exten => s,2,Set(DUNDIVAR=${ARG1}#TEST) exten => s,3,NoOp(Passing ${DUNDIVAR} to DUNDi peer.) exten => s,4,Goto(${DUNDIVAR},1) On

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not

Hi, I too have found this matching to be frustrating. I would like it to behave as you describe. Doug -- Doug Meredith 506-854-7997 ext. 801 ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Damon Estep Sent: Thursday, January 04, 2007 1:50 AM To: Asterisk Users Mailing List - Non-Commercial

I have considered opening a bug report on this, but wanted to get some feedback and make sure I am not missing something in the way of a simple work around. What is the scenario in which this impacts your implementation? Ours is the desire to use the same realtime SIP database for many asterisk servers, and route the call based on a "home server" value in the realtime database. The

Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38

Take an example where there is two sip users defined in sip.conf as follows; [peer1] Host=192.168.1.1 ... [peer2] Host=dynamic Secret=password ... [Peer3] Config not relevant ... The intention is to accept calls from peer1 without authentication (ip address authentication only), but require authentication from peer2 If by chance a SIP invite comes "From"

We have multiple entries like the one below in our users.conf file... where the username. Contact and secret changes for different customers and we register on their behalf to the Service Provider. For the trunk below: when the call is placed out, Asterisk (1.4.18) sends the username of "abc.com" in the MD5 Auth .....which obviously does not match the trunk setup for this Customer with

How can we set the CDR Master file to rollover at say 30 Meg and create a new one -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/2e98385f/attachment.htm

Hi, all. Is there any chance to set individual CALLERID(num) for channels SIP/peer1, SIP/peer2 in a call Dial(SIP/peer1&SIP/peer2). There is an option to use Dial(SIP/peer1&SIP/peer2,,M(set_callerid)), but the macro will be launched after the channel answered. Not really want to use local channel because of not quite usable cdr. Thanks.

-- Attempting native bridge of IAX2/peer1-iax-7 and IAX2/peer2-21 I want everything to stay in the VoIP server rather then briding. I have notransfer=yes on, but it still seems to bridge the call natively.. can I keep the RTP stream on the asterisk server some how?

hi all! I have the following situation: 1 -------- 2 ? ? ? ? 3----------4 ? ? ? ? 5----------6 where 1 ... 6 are nodes and every direct neighbor is specified as a dundi peer (in *). When I start a dundi request, every queried node is mentioned in the dpdiscover. For example 1 sends a discover to 2 and 3, so 2 sees in the EID or

Hello I have two peers setup and working with x2 bricks each. They have been working via tcp for the last 4-5 months. I just got two Infiniband cards and put the on the peers. I want to change the transport type to rdma instead of tcp but I don't see an easy way to do this. Can you please help me with proper instructions. Best Regards Ivan Dimitrov

Setup is: Asterisk --->NAT--> SIP Proxy I have following entry for SIP Proxy in sip.conf [Proxy] type=peer host=Static IP (NAT Firewalls public IP) username=xxxx secret=xxxxx nat=yes???????????????? canreinvite=no???????? qualify=yes Proxy sends a call and I get this error Found no matching peer or user for <NAT's Public IP:70001 NAT is using 70001 as the source port in the

Hi, After Googling, I found information on how you can read the status of an outgoing call but I didn't find anything on tunning reply to incoming calls. My question is : I've got a system receiving SIP calls from different callers. I would like to end some calls with a "480 Temporarily Unavailable (Call limit)" reply Is it possible ? Regards -------------- next part