similar to: Sample cisco config for cisco 7206

I am running an Asterisk server through a Cisco 7206 PSTN gateway. I am able to make outgoing SIP calls without a problem, though incoming calls have been somewhat of a problem. I am not sure exactly how sip.conf should look in such a scenario. I believe most Cisco gateways are just managed through ACL's, with no authentication, so I think I have the outgoing "peer" statement

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

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

Hi, We are having an issue here. It seems that whenever we initialize Asterisk on our network, the router that the Asterisk server is connected to (Cisco 7200) crashes and loses it configuration. This has happended five times and each time we have tested it, it is always when Asterisk starts up. Has anyone else seen this problem? It is very odd because this is a very good router and we

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

Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I already have t38pt_udptl=yes in sip.conf Thank you.

'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

Hi, I'm using Grandstream IP phone GXP2000, with Asterisk 1.4.15 I'm using g729 codec and want to use only this codec for the calls. My normal calls are going fine. But issue is coming when I'm using the conference from the Line1 and Line2 Option. When I'm initiating the conference at that time, IP phone is sending the G711ulaw for the conference call, while in my phone I've

What codec should be used to connect a * box to a cisco router which has a t1 with 24 trunks coming in? My router voip dial plan looks like this: dial-peer voice 2 voip destination-pattern [1,2,,3,5,8].. session protocol sipv2 session target ipv4:10.x.x.x dtmf-relay cisco-rtp codec g711ulaw no vad ! The problem I have is when more than one call is on it, sometimes the quality gets very

Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

# 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,

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.

Does anyone in here know how to setup auto negotiation between g729 and g711ulaw on a cisco 5400? I would imagine it would be the same on a 3660. The problem I am having is natively the call is setup for g729 however when the call is transferred to voicemail it uses ULAW so when the cisco tries to connect to the voice mail I get a SIP error that the codec couldn't be negotiated. I need

Ok here is a question that has gotten me stumped. I have an Asterisk system up and running. I need to connect it via the Internet to a Sip Cisco system. This is what they have. I have there IP address's and login. X-lite is able to connect to them and make a call! So I have the name right! CISCO router model: 2621 VoIP module: NM-HDA-4FXS I have done Google lookup and at the Wiki about

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

I have the following configuration and am obviously missing something small that is causing * not to work as expected. I have the following defined in sip.conf [ccme-in] type=peer host=10.0.9.1 context=devel_in disallow=all allow=alaw nat=no canreinvite=yes qualify=yes and [devel_in] is defined in extentions.conf However when I try to call via the dial peer I have configured on the cisco

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

Can anyone explain why this is happening? I have a server attached to a phone line that will play a .wav file, then play all the dtmf digits (after it answers the call). If I place a call from a SIP device (like a Cisco 7960 phone) through Asterisk and on to the test server, via PSTN, the .wav file sounds fine, but the DTMF digits are distorted ----->------------->--------------------audio