similar to: how can i config a Cisco IAD 2430 config as a sip client

I repost a message I put a week ago: I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? Also I recently

I'm considering switching my incoming phones lines from standard analog to a T-1 service from XO communications. They propose to bring in an "IAD" which has 12 lines of voice and 768k of internet bandwidth as part of a package deal. Since I want to keep the voice traffic in the digital domain the equipment they're proposing is a "Lucent Digital Vina Integrator" IAD

Hi, For MGCP users. Is there any success stories with any MGCP IAD vendor. I?m trying to find an IAD which works with Asterisk. I?ve tried the Cisco IAD 2430 without success; but SIP on this IAD works but it?s limited (no authentication, no notify messages, etc) and with higher density IAD (16 or more ports) it?s nice to control using MGCP. Any information will be apreciated ! Thanks. --

As a follow-up to my previous post where I stated the IAD would be a Vina model, after some more prodding to XO, they have told me it will either be an Adtran TA-600 or a CAC Adit 600. These products are covered pretty well on the web and I have manuals on both. So, if those knowledgeable folks had to use one of these to attach to an Asterisk box, what interface would be best or at least

All, I need to run this by everyone and see if someone has any idea's. I have a asterisk server setup and currently am receiving the inbound calling number where the name should be. My setup is.... One pri terminating into a Cisco 2431 router Sip messages from the Cisco get sent to a asterisk server linksys ata's a each remote end. I can receive the calling name if the call originates

To Members, I currently looking for someone who can configure Cisco IAD's Ie.. IAD 2431 And yes we are more than willing to pay for the service. If interested please drop me an email mdm at openaccessinc.com<mailto:mdm at openaccessinc.com> Michael DiMartino | Director of IT | Open Access, Inc. 115 Bi County Blvd | Farmingdale, NY 11735 631.227.1034| 631.694.6730 FAX |631.988.6060

I figured I should post this because before I went to do it I searched the web for caveats etc and didn''t really find much. So anyway, I set up bonding + vlan for a project. Computer is dell powerdge 350 with 2 intel 82559 10/100 nic running stock 2.4.20. I tried both the eepro100 and the e100 drivers for the nics, both work 802.1q and bonding are modules. Switch is cisco 2924xl-en

On 04/11/14 19:18, ray klassen wrote: > > Back when I was first exploring samba4, the debian packages were > expressly without the active directory component. From your question, > I assume that has changed? > > > The bind9 thing was to enable dlopen which the stock debian bind > didn't come with. Again, are you saying that the backports version has > that? >

This is mildly off topic since it has more to do with TCP/IP networking than with CentOS per se. However, at the risk of ruffling a few feathers I would like some advice on how to resolve the following routing problem: Given: dual homed host running CentOS-5.2 with eth0 = 2xx.yyy.zzz.23 eth1 = 192.168.219.1 and a Cisco gateway at 2xx.1xx.y7y.1 with the following networks: interface

Hi All, I tried doing SIP from Asterisk to the old Cisco 2400 series IAD's but could not get signalling reliable or DTMF either. I do SIP to Cisco routers with PRI cards in quite a bit, but I guess the old IAD SIP stack is not a robust at the router sip stack so I just could not get it working. I'm wondering if anyone has tried the new Cisco 2430 series IAD's and have been

Hi, I facing a problem with flash hook. When ever I do a flash hook to place an extsing call on hold, the call gets disconnected. The debugs on Asterisk shows that 'on hook event detected' when I press the flash button on the phone. The setup is like this Asterisk box with T1 cards and FXS cards. The T1 card is connected to an IAD and configured for ISDN PRI lines. Analog phones come

Hi Stephen, Thanks again for getting back to me, Ivan Krylov responded also and suggested windows binaries and I must confess I was only familiar with installing from files via the package sources (apart from the conventional install.packages method), so the solution was as simple as installing via the binaries. Thanks again, best wishes, James From: stephen sefick <ssefick at gmail.com>

All, I am posting this here to announce I have finally managed to get my Cisco IAD2421 to speak MGCP with Asterisk. Due to an acute lack of reading on the subject as searched on Google, I'm putting this out with the hope that it helps whomever should need to do this in the future. This should also apply to the IAD2420 and the other models in the line, but as I do not have access to those,

Hello Alice, On Wed, 2016-10-19 at 14:22 -0700, Alice Wonder wrote: > I formerly used secp521r1 but suddenly Google with no warning stopped > supporting it in chrome. That company is too powerful. Actually this is something the NSA insists on:

Hello, after upgrading from Samba 3.5.6 to 3.6.5 I encounter some strange NT_STATUS_ACCESS_DENIED trouble when trying to access group readable directories from Windows 7 or Windows 2008 Terminal Server. Very strange stuff is that it works fine using smbclient from a Linux machine using the same userid and Kerberos authentication. The Server is fully integrated into active directory and uses

P.H.B. is insisting on having the ability to create a transparant SIP tunnel between old style ISDN telephony PBX with overlap dialing: PBX - ISDN - IAD - SIP - * - DAHDI - PRI The idea is that dialed numbers a the PBX are transmitted to the PRI as they are typed, whenever the PRI gets the signal that the number is complete the dialer instantly gets a ringing. This behavior is described in RFC

Hi, I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch the hangup. I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like it such that I call from * via E1/PRI to the cisco, and call out via R2 to

Thank you Joshua. I tried setting the from_domain for the endpoint, but it still sends the internal ip address for the INVITE's From field [acl1] type = acl deny = 0.0.0.0/0.0.0.0 permit = variousaddress permit = bluipaddress [transport1] type = transport bind = 0.0.0.0 protocol = udp [BLUIPIN] type = aor remove_existing = yes contact = sip:bluipaddress [auth7] type = auth username =

I'm having difficulty with an Asterisk system. The external party has very good call quality, but the internal party hears clipping and drop outs. The WAN comes in from the Cisco IAD and into a LAN switch (DLink DGS-1005D w/ 802.1p) where the two public IPs are switched to different devices. One device is a FireBox device controlling a separate LAN with VPNs. The other device is eth0

These changes apparently broke something: > 2006-04-24 09:17 Timo Sirainen <timo.sirainen at movial.fi> > > * src/lib-index/mail-index-transaction.c: Fixed crash while > updating keywords (broken by recent changes). > > 2006-04-20 14:10 Timo Sirainen <tss at iki.fi> > > * src/lib-index/mail-index-transaction.c: If we sorted appended > messages, we