similar to: Outbound call through a Single Asterisk Server

Hi, Where can I find doc related to IMAP storage. Usually, config options can be found either in voicemail.conf or voip-info.org but almost none relates to IMAP configuration. At the moment, I'm looking for data related to imapflags possible values. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

When I have voip conversations over asterisk through my computer the voice quality is nice and loud and quite clear. When I go through my Sipura 2K then the conversations are typically very muted and my responses sound somewhat delayed. I've tried fiddling with settings under asterisk and the 2K but haven't had any luck. Can someone point me in the right direction to get this

Hi. I'm using Debian's 1.6.2.0 version of asterisk and trying to get an understanding of bridging two lines together. I've seen the documentian on the bridge command but a good example would probably help me more. At the moment we have a number of SIP lines, one DAHDI line, and an IAX2 line that incoming calls can originate from. All the local phones are SIP. What I would like

Hi. I've got a working dial plan on my home system but there are problems with it and I was hoping someone more comfortable with dial plans might be able to help. In a nutshell here's what I'm currently doing on an incoming outside phone call [default] Set(TIMEOUT(digit)=3 Set(TIMEOUT(response)=60 exten => s,1,NoOp(Answering in default context) exten =>

Hi, I am unable to log into samba anymore since I upgraded to 2.0.3. I've attached my smb.conf below and I've even gone to trying the CVS releases to see if one of them will help. First of all, I get soemthing like {33}: smbclient -L vilnius -U mharrell Added interface ip=207.97.105.146 bcast=207.97.105.255 nmask=255.255.255.128 Password: session setup failed: ERRSRV

Hello R, To analyze multi-level data, I started learning and using lmer. So far so wonderful. I then found some useful functions in package languageR. But then the following problem ocurred: Whenever I load and use the languageR package, then save the workspace - or quit R with saving the workspace - I am unable to reload that workspace in a later session. That is, R doesn't start at all

Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work

I'm getting down to the last of my * issues ... After calling a bank, or cc processing center; you have to enter your social security number, or the cc number - followed by the # key. The lovely * voice responds "transfering" I'm sorry that was an invalade selection. Sometimes the IVR on the other end still gets the digits and proceeds; other times it breaks the IVR on the

Hello, We have an Asterisk server with two public IP addresses, let's say 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call dialled from Asterisk to an external destination. The external destination sees the SIP packet as coming from 1.1.1.1 and the media address in the SDP is 1.1.1.1, which is great. However if we receive a call in to 2.2.2.2 then the call

Hi George, Thank you for the response. I'm a little unclear on what you mean by a transport. We're using chan_sip, not pjsip. Do you mean a device in sip.conf, using bindaddr to set the address to bind for that device? We've only used bindaddr in the [general] section before, but if it will work in a device that could be the answer. On Fri, 23 Oct 2020 at 00:13, George Joseph

OK, thank you George. On Sat, 24 Oct 2020 at 03:16, George Joseph <gjoseph at digium.com> wrote: > > > On Thu, Oct 22, 2020 at 4:13 PM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hi George, >> >> Thank you for the response. I'm a little unclear on what you mean by a >> transport. We're using chan_sip, not pjsip.

Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio On Thu, Oct 29, 2020 at 20:44 David Cunningham <dcunningham at voisonics.com> wrote: > Hello, > > Does anyone know a way with chan_sip to tell Asterisk to use a specific IP > address for its end of the communication for a specific

Hi Folks, I want to setup a follow me routine so that asterisk can call me on the multiple numbers. I tried some of the samples at voip-info but there is a problem with those examples. I dont have coverage in my home area and my cell phone answering machine picks up the phone right away so my home phone never rings. I also want the caller to be able to leave a voicemail and the cell phone

On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hello, > > We have an Asterisk server with two public IP addresses, let's say 1.1.1.1 > and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call > dialled from Asterisk to an external destination. The external destination > sees the SIP packet as coming from

On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi George, > > Thank you for the response. I'm a little unclear on what you mean by a > transport. We're using chan_sip, not pjsip. > > Do you mean a device in sip.conf, using bindaddr to set the address to > bind for that device? We've only used bindaddr in the

I am having problems with silent data corruption on (some) drives connected to an MCP55 SATA controller. I have two servers, both running RELENG_7_0/amd64. One has the 570 Ultra chipset, the other has 570 SLI. Both chipsets have the MCP55 SATA controller. The server with 570 Ultra chipset has a bunch of older 250GB SATA-150 drives hooked up to the MCP55 controller and it is working just fine.

Hello, Does anyone know a way with chan_sip to tell Asterisk to use a specific IP address for its end of the communication for a specific device? Something like: [device] type = friend host = 11.22.11.22 ouraddress = 33.44.33.44 This is for use on a server with multiple IP addresses. There is the "extenip" setting, but it's really designed for NAT, and can only appear in the

Hi Dovid, We can change the SDP in Kamailio, but Asterisk will still send its RTP from its default address. The remote end is strict about accepting RTP from the specified source and won't accept it. Have you any suggestions to solve that problem? Thank you. On Fri, 30 Oct 2020 at 14:49, Dovid Bender <dovid at telecurve.com> wrote: > Why not use OpenSips/Kamailoo in between?

Hi, I'm trying to get a remote Cisco Call Manager Express (CME) system behind a dynamic IP address routing both inbound and outbound calls via SIP to my local asterisk server. I've got a local CME system working fine on the LAN, where the CME is at a static IP (host=10.5.7.130 in sip.conf), but I can't figure out how to get it working with host=dynamic, even locally on a test

Hello all, Came across my first real "problem" in all the years I've been using Linux today and I'm stumped for an answer. I've recently built a back-up server on top of Centos 5.3 using a 3Ware 9650SE 4 port SATA card and 4 x 1.5TB drives in a RAID 5 array. I went for the 3Ware card as it has driver support already so any kernel updates won't require me to rebuild the