similar to: PRI failover to SIP trunk

Hello! I have been struggling with Asterisk 1.6 and DAHDI for the past few weeks. We are an outgoing call center with 30 internal analog phones hooked up to 2 Rhino CB24 channel banks. The banks are connected to a Rhino R4T1 card in a Dell 2950 server with 8 gigs of RAM. The 2 other ports on the R4T1 go to our 2 PRIs. In this configuration, we have trouble maintaining stability. It may be fine

Hello, The company I work for recently purchased 2 Rhino CB24s and a Rhino PCI-E R4T1. The channel banks are plugged into the R4T1, as well as 2 PRIs from our telco. The CB24s are for all internal analog phones. Most of the phones are setup in "batphone mode", which is "immediate=on" in the DAHDI config. They are set up this way because we are an outgoing call

Hello, We use MS Access 2000 (I know, we're migrating away from it) as an application to automatically dial phone numbers. The old phone system we have allowed the call representative using the application to take their phone off hook, push a button in the app, and the app would send the phone number to the phone system and dial the number. We are moving to Asterisk for our main phone

Hello, We have finally upgraded to Asterisk 1.4, however we've run into two issues that weren't occurring before the upgrade. Issue #1: We're an outgoing call center and need to record all calls. We use the uniqueid field in the CDR to match with the recording, which we labeled with {UNIQUEID} in MixMonitor. For some reason, the uniqueid is not correct in the CDR. Here is the

Hello, I've been looking at some SIP packet dumps captured with tcpdump on the PBX itself, and analyzed with Wireshark 0.99.4. I'm noticing something strange, at least to me. All of the SIP packets going out from our Asterisk PBX to either of our 2 VoIP providers are consistently 50% out of order. In addition, if I use Wireshark's voip call player, the outgoing side of the call

Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY

Hello, I may have found a serious issue with 1.6.1.6. I just compiled it yesterday on our server. When anyone tries to access the name directory through the Directory app, the asterisk process completely dies. Our extensions are in a realtime MySQL table, and the directory has worked fine with previous versions of asterisk. Jason Martin Metrix Matrix, Inc. 785 Elmgrove Rd, Bldg 1

Hello! I am migrating an old Red Hat Samba 3.0.9 server to a new Debian Etch Samba 3.0.23d with an OpenLDAP backend. I've got almost everything working with the new server except the roaming profiles. When a user logs off, Windows complains that the permissions are not correct and the profile can't be saved. I would LOVE to get rid of roaming profiles but that isn't an option

Hello, I have a strange problem with one of my Zap channels. A user told me that he was in a voicemail system during a call, hit the Flash button, and the call hung up. Now we get no dialtone on the phone hooked up to the channel. Here's the status of the channel: jmartin at rogue:~$ sudo asterisk -r -x "zap show channel 7" Parsing /etc/asterisk/extconfig.conf Channel: 7 File

Hello, We've purchased a 2nd Wildcard TDM2400P for our asterisk server. I've looked around but I can't find documentation on how to properly setup 2 or more telephony boards in asterisk. In particular, I want to ensure that the boards are assigned the same device channels on boot (for example, if you have two NICs in a Linux box and you want to make sure one card is always eth0

Hello, I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out when a user gets a voicemail don't have the timezone set in the header, so they're appearing in the user's email clients at the wrong time. Has anyone else seen this? I didn't find any bug reports or other info with Google. -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road,

Anybody have suggestions on having a 56K dialpool and VOIP connections with an Asterisk box over the same set of PRIs? We've done the PM3 with PRIs for just dialup, but are looking for a way to integrate our Asterisk box and move our voice calls onto the same PRIs. Ian -- Ian White Victoria Free-Net Association email: iwhite@victoria.tc.ca http://victoria.tc.ca/

I currently have a single PRI however we are getting a second PRI, and the provider (qwest) wants to know if our PBX supports GSAS (they say its a redundant d-channel technology but searching on google for GSAS reveals less than nothing). I've set something similar up before on a cisco 5350, where if one of the PRIs fails, all of the calls destined for either PRI will be routed down the one

With multiple inbound PRIs (and hunting across them) coming to multiple [asterisk] servers it is important to be able to do administration, i.e. control which PRIs in the same hunt group take (and which don't take) calls from telco at any given period of time. Our pre-asterisk platform uses SERVICE commands for this purpose to put B-channels into 'out-of-service'/'maintenance'

I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go

Hello, I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like to be able to direct an inbound fax call into my TNT, have it answer the fax and send the image file over to Asterisk, or some other system to deliver to an e-mail address(s). I'm not sure if I need Asterisk to any of the call control or not. I'd also like to setup a print queue and have outbound

hI, Trying to understand how to setup two PRIs in sip.conf. Using Asterisk 1.4.23. I have a provider giving me two PRI (different rate centers) through SIP. Both PRI comes in from the same IP on the provider side, but go to two different IPs (both on the same box) on my side. How can I setup two different SIP peer, one for each of the PRIs I get, if all I can use to differenciate them

I am currently working on an Asterisk test system, and will be presenting a demo to the Board of Directors tomorrow night. I want to make sure I have all of my ducks in a row. The Asterisk system will be used to replace a Norstar MICS. The location has two PRI's coming in, with a few hundred DIDs. I know how to make * use the DIDs incoming, and I know how Nortel uses the DIDs. Now for the

Does anyone have any recommendations on how to connect 160 analogue phones to an asterisk PBX? Background information: A client wishes to replace their current PBX with a new VoIP system. Currently they have 2 PRIs. I intent to set up 2 asterisk PBXs with Debian GNU/Linux on raided drives. These drives will be mounted only read-only to recover gracefully from power-cycles. I am considering 2

We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header