similar to: Slightly Off Topic: Cisco & Premisys Slimline

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

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

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.

I need to connect a bunch of analog telephone sets. Does anyone have any comments about this channel bank? Disconnect supervision? Echo? ADSI problems? The price is right @ $995 new and $695 refurbished. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

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 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

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

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

I've encountered an odd situation with Asterisk 1.4.19 that I can't figure out. If I dial an extension via a Cisco AS5400 with the "g" option to come back, when I then Dial another extension after that, we don't get audio from the caller. There are no firewalls, no routers, no anything but a network switch between. The calls come in as SIP from the Cisco and

I've been searching to a solution to this for a while and can't figure it out, perhaps someone has done something similar. I have a Cisco AS5400 sending SIP traffic via PCMU / ulaw directly to my Asterisk (1.4.19-rc2) box. Jitter and latency are incredibly low on my lightly loaded switched gigabit ethernet network. One Asterisk uses Zaptel and a Digium card, and DTMF recognition

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

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

Hello - I have a Premisys Slimline Channel Bank connected to a Digium TE110P. I am not able to call the FXS extensions or get dialtone on them. The channel bank is connected via a T1 crossover to the cable and lights show green. I really need to get this functioning by end of day. If anyone can help me out I would be greatly appreciative. Thanks, Dave zaptel.conf loadzone =

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 /

How do you specify in a tablegen pattern that all destination registers are also source registers? I know I could just duplicate them, but I was wondering if there was a way it could be done without doing this. Basically an inplace operation. So for example, take a hypothetical swap w/ sqrt. if I had swap $ds0, $ds1 which swaps ds1 and ds0 and applies sqrt on the registers afterwards. Micah

I've been using pxelinux for automated installs, and it works well. I'm using it on SUSE Linux 8.1, and it has been going well until recently. The TFTP, NFS, and DHCP servers are all on the same system. It uses PXELINUX 1.69 (part of SYSLINUX 1.62 I believe). It installed systems to my slimline Compaq Deskpro Pentium IIIs just fine. Recently, I ran it on a Compaq Deskpro Pentiums

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Hi Micah, On Tue, Jun 25, 2013 at 03:47:14PM +0000, Micah Villmow wrote: > How do you specify in a tablegen pattern that all destination registers are also source registers? I know I could just duplicate them, but I was wondering if there was a way it could be done without doing this. Basically an inplace operation. > I'm not sure if there is a good way to do this. The

Hey All, I'm just daydreaming here.. but what's the status of SIP-T in Asterisk? I haven't been able to find a whole lot of info on SIP-T but seems like just an extension of SIP. Right? Now if I had a PSTN Gateway (that is a SS7 gateway) that supported SIP-T, could I signal * with SIP-T from it and have asterisk utilize MGCP to sieze a particular DS0 on a remote DS1? Hmm.. What am

While I was tying down a supernet to the discard interface, the box crashed on me. Its a STABLE box from June 4th. I was in zebra at the time and thought I would route a /24 to ds0 instead of to the IP on ds0 (which I had done for a number of other aggregate routes). The only other "strange" thing about the box is that ds0 is loaded via kld. I will see if I can reproduce it on a