similar to: Incoming SIP from Cisco 7206

Newbie to Asterisk I've been looking around for a little while, can't seem to find some sample configs for using a Cisco 7206 as a gateway. The below link is an initial plan of an Asterisk solution that may replace our Cisco Call Manager 3.1/ IPCC / IVR setup. We currently have all of the hardware below. Just take a peak and see if there is anything that is off base. I don't know

I see a lot of comments but no actual show runs. Can someone post a 7206 config. I am having a dickens of a time getting calls to pass. I currently have the following loaded. Cisco IOS Software, 7200 Software (C7200-IK9O3S-M), Version 12.3(8)T6, RELEASE SOFTWARE (fc2) Thanks !!! Jerry -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 /

All, I know you can use, say, a 2620 w/2 port FXO card as a SIP gateway. Clearly you can use the 5300, 5800, and MGX8850 too. Does anyone know which cards, if any, exist for a 7206VXR to act in a similar capacity, either as a T1/PRI, DS3, or POTS FXO/FXS? What other Cisco routers can act as SIP gateways today? Thanks, Dave

Hello, hope this isn't too far offtopic here but being a troller for a long time here I've realized there is a great knowledge base so I wanted to at least see if i could get some tips. I help run a small colocation company in California and I am in the middle of recommending a new 'core router' platform for our network. We offer mainly colo and dedicated servers, and several of

'morning everybody, Here is the setup: 5126800422 called 3035 (3035 is a Cisco 7960). The call is g729. 3035 presses 'Conference' on her phone and calls 8327549222. This call is ulaw. (65.72.107.2 is our Cisco 7206 SIP->PRI gateway.) asterisk*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 65.72.107.2 8327549222 1758081f67e

I would like to ude the function "qdagum", to compite quantiles of dagum distribution, but if I use the form: >qdagum(0.1, 4, 7206, 0.648) Error in qdagum(0.1, 4, 7206, 0.648) : object 'Scale' not found Error will apper, Could you help me? Thank you very much in advance, Alena -- View this message in context:

Can anyone point me to some sample Cisco QoS configurations suitable for IAX2? I've looked through Cisco's site, and get overwhelmed with the level of documentation (too much of a good thing). My PSTN gateway and PBX (both *) are connected via 2xT1 (per-packet load balancing) between a Cisco 7206 and a 3640. When the total bandwidth pushes much past 50%, I start getting some crazy

# I would like to outline the squares in the legend with a black line. Does anyone know how to do this? x.t <- structure(c(5987.387, 4354.516, 3685.789, 6478.592, 5924.315, NA, 8386, 5559.468, NA, 4651.273, 3967.5, NA, 4339.167, 5053.56, NA, 4631.978, 4808.694, NA, 5217.306, 4017.632, NA, 5846.903, 3969.883, NA, 3867.825, 3910.236, NA, 3886.434, 3782.094, NA, 3959.668, 3961.286, NA, 3848.853,

Hello, I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323. I tested both oh323 from inaccessnetwork and JerJers chan_h323. I used 1.12.2 version of oh323 and 1.5.2 version of pwlib. After latest changes from JerJer chan_h323.c works ok when receiving traffic from ciscos. I havnt found any audio problems although I didnt send much traffic. Latest oh323 has some

Hello All, I know that this should be a trivial issue, but I''m stuck. I''m totally new to Shorewall and although I''ve read all about the zones, they''re still a bit confusing for me. What I''m attempting to do is run an FTP server on an internal machine. I''ve read the example guide and troubleshooting guide, but I can''t figure it

Here is the setup: Dell 6250, Quad Proc P3 500Mhz. Digium Single Span T1 card. System has 71 sip peers/users. All calls are G729; we have 10 licenses. All calls follow this path: UA -> Asterisk -> Digium PRI -> Class4/5 switch. The switch dictates if it should go out local PRI for local termination or out PRI to our Cisco AS5300 for LD termination. Our biggest problem is echo.

> Message: 5 > Date: Fri, 12 Sep 2003 13:24:13 -0400 (EDT) > From: "David C. Troy" <dave@toad.net> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] 7206 as SIP->PSTN Gateway? > All, > > I know you can use, say, a 2620 w/2 port FXO card as > a SIP gateway. Clearly you can use the 5300, 5800, and > MGX8850 too. > >

Hi All, I have an R2 installation still undergoing testings, during the test I notice that the Unicall always respond B6 to a II-1 (from a forward switch). Except, for a DNIS that can't be found in the dial plan, in this case it respond with B5. My real problem is, the call will be terminate on a Cisco 7206 with ISDN/PRI thru SIP. If the Called number is busy or the Cisco 7206 is busy or

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

================================================================= "Why don't we save time and you just tell me what I want?" Mr. Big to Carie, Sex and the City ================================================================= Release Announcements ===================== This is the latest stable release of Samba 3.5. Major enhancements in Samba 3.5.2 include: o Fix

================================================================= "Why don't we save time and you just tell me what I want?" Mr. Big to Carie, Sex and the City ================================================================= Release Announcements ===================== This is the latest stable release of Samba 3.5. Major enhancements in Samba 3.5.2 include: o Fix

Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A)----> Cisco AS5xxx ----> ASterisk---->PSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains

Dear Sir, We would lik to subscribe the support for OpenSSH for a tender project. Please kindly advise the annual subscription plan (with details of SLA) & annual fee. For the price, is it determined by number of server? or number of CPU? or number of core? Shall we get quotation/support from you? or please advise the reseller contact. Since the tender will be closed next week, may we

I am using Dev-C++ under WINE and Ubuntu as I am not yet familiar with Ubuntu enough to study C without a simple compiler I am used to. It installed ok, and compiled .exe files ok, but when it came to run those programs I had a problem. The first was that, normally, when I would run an executable written to use the windows command prompt, the command prompt wouldn't appear. Now, I make sure