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Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system to work with *? I'm not getting inbound or outbound calls to work. (Inbound caller gets line busy tone.) SETTINGS FROM MITEL: I built a Crossover cable and connected it like this: PSTN--T1--ASTRISK--T1--OLD MITEL -Crossover Cable Pin-out: 1 - 4 2 - 5

Hi, I am experiencing one-way audio from: SIP Device ----> Asterisk -----> Cisco 7200 The Cisco 7200 has a VXE+ card that will allow you to do SIP. I can pass audio from SIP Device to Asterisk through the Cisco 7200 to the other end, but the Cisco 7200 does not return any audio back to the SIP Device or Asterisk, it seems. I have tried upgrading to 12.3T IOS version, but no

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.

If you have a PRI terminated in a Cisco router talking SIP to * and would be willing to share your Cisco config, please respond. Also, I would be interested in knowing what version of IOS you are using. We are using an NM-HDV in a 3640. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

I have an adit 600 with one fxo card connected to a Digium single span T1 card. CallerID, disconnect supervision work perfect, however the users complain that they have some sound quality issues, after testing it I realized that whenever one is in a phone call they get like silence between the sounds coming from the other party, almost like a cell phone, in other words if there is no sound coming

Heh guys, I just received a T400 card, I've been using a T100 for a little while, and it works fine when using a raw channelized T1. I'm relocating my asterisk machine, and PRI's will only be available, haven't found any good config info for PRI's, can someone point me to PRI config info, or let me know what changes I need to make in order to bring them up, I imagine,

Hi ! I am connecting to Adit 600 thru a T100P card I have configured 1-16 FXS channels and 17-24 FXO. Everything looks fine on Asterisk side I get a tone on all FXS channels, but when I try to dialout thru one of the FXO channels 17-24 it doesn't connect to the POTS line and echoes back my voice. I use fxsls and fxols for the T1 channels and ls on Adit side. Whats wrong here ? here is my Adit

Hello all, I have a new Carrier Access Adit 600 and am having a problem dialing out and receiving incoming calls. After the initial ring, the calling party will here nothing. There is a silence, but the SIP phones that is supposed to ring will ring. The calling party will not hear the sound of the ring that * sends out on the line. Additionally, if someone picks up the receiver there is no

Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk

I would think I could do this but for some reason I am stymied. I have a PRI from RCN connected to a cisco 3640 (in my day "cisco" is all lower case :-)). My config looks something like this on the cisco... --------------------------------------------------------- voice-card 3 dsp services dspfarm ! ip cef ! isdn switch-type primary-5ess ! controller T1 3/0 framing esf linecode

Has anyone got a Premisys Slimline channel bank working with a Cisco AS5400 or similar? I'm not sure if my unit is bad, or what. I'm using FXS Loop Start. Calling the port connects immediately without ringing the attached phone. If I pick up the phone, it's connected and I can talk to the caller. Hanging up has no effect. I can see the bit transitions (0101 to 1111 when I go

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

hi All I need your help , for used Digium Card TE405P, for setting this Board AS E1 ISDN PRI. 1 .Current for make sure my config its rights or no I inform my configurations in Board Jumper T1/E1 is Closed is that rights or no ? for E1 i closed the Jumper. 2. I Want To seeting E1 in ASterisk/PC Back To Back To Cisco E1 AS5300 Use ISDN Signaling, my configutration :

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 /

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Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec

Hello, I have an Adit 600 (3 FXS cards) hooked up to a digium T1 card in my asterisk box. I 'connected' the slots to the a:1 T1 interfaces via the command line. The slots (3 fxs) are configured with 'ls' signaling. I configured the T1 card with the same line settings as the T1 interfaces on the adit and I get green lights on both the T1 card and the T1 interface on the adit (so

Hi All, I'm having problems trying to get a TE120P operational in Canada. I keep getting a congestion error when I try to make a call. I'm not sure if my switching, parity, etc is correct. I'm hoping that someone will be able to verify my config. The Telco is SaskTel, with a 10 channel 50 DDI service. Zap show channels show and ztcfg -vv looks ok and the zttool show

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

List users, It's been a while since I've posted here, but I've been hard at work pushing toward our large scale Asterisk goal and keeping up with this list can be a full time job by itself (I have19,543 unread list messages!!). This Friday, September 16th 2005, my team will be at the MCI Development Lab in Richardson, Texas testing our setup. We have a three server system