similar to: SIP.Conf - bindaddr per peer?

Sorry but what does the ACL mean and its relation to the bindaddr? Regards Bilal > > 30 jan 2009 kl. 16.59 skrev Mike: > > > 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

Hi to everybody, My IAX is not working, When I type reload IAX it returns me: AsteriskSlave*CLI> iax2 reload == Parsing '/etc/asterisk/iax.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Aug 26 10:05:04] NOTICE[18078]: chan_iax2.c:13546 set_config: Ignoring bindport on reload [Aug 26 10:05:04] NOTICE[18078]: chan_iax2.c:13610 set_config: Ignoring bindaddr on

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'm running CVS HEAD. The last time I updated was January 7th, at which time everything was fine. Having updated again today, January 17th, I'm having problems with IAX2. I use the "bindaddr" directive for both SIP and IAX2, and while SIP respects it, IAX2 doesn't. It listens on every interface, and uses every one of them for outgoing source addresses. This breaks IAX2

So ... I've finally tried to sort that out for powerpc and I can't find a way to make that work that isn't a complete pile of stinking shit. I'm very tempted to go back to my original idea: virtio itself should indicate it's "bypassing ability" via the virtio config space or some other bit (like the ProgIf of the PCI header etc...). I don't see how I can make

So ... I've finally tried to sort that out for powerpc and I can't find a way to make that work that isn't a complete pile of stinking shit. I'm very tempted to go back to my original idea: virtio itself should indicate it's "bypassing ability" via the virtio config space or some other bit (like the ProgIf of the PCI header etc...). I don't see how I can make

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

Hello, I've found a little documentation on voip-info and on the asterisk- users list, although I was hoping for an example of a tried-and-true failover setup between PRI and SIP. We are an outgoing call center that uses asterisk 1.4 connected to 2 PRIs from the local telephone company in one group (g1) and a SIP trunk from bandwidth.com. The PRIs are the primary outgoing service,

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

I have a configuration where RTP traffic is going out interface pub0, and coming back into through pub1. I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows: udp 0 788 0.0.0.0:5060 0.0.0.0:* which means that Asterisk is listening on all addresses (on all interfaces?). Anyway, when the RTP traffic comes back in on interface pub0, Asterisk does nothing with it. A

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

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

Hi all, is it possible to bind SIP protokol not to all but to more that one interfaces. I did try use bindaddr, but i don't know right syntax. Could anyone help me. Thanks, Bob.

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

I've got a couple of PRIs. When I call out on them from internal SIP phones, I will get ringing if the dialed number is ringing, but if the dialed number is busy I'll get dead air. Can anyone suggest ways to trouble shoot this? Don't seem to having any other problems with the PRIs. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at

I'm having trouble setting the outbound caller ID on calls I make from my PRI trunk group. The PRIs are served out of a 5ESS. Telco has set the PRIs up for user provided caller id information, so I believe I just don't have it set up right in my dialplan or something. I can't seem to find an example of setting the outbound caller ID specifically for a 5ESS. Does anyone have an

Hello, I have a crossover PRI(Asterisk server to PBX) and a regular telco PRI T1 line and currently have echocancel=yes and echocancelwhenbridged=yes on those spans in zapata.conf. I was discussing CPU load with another Asterisk user and he mentioned that PRIs don't need echo cancelation and that turning it off will reduced CPU load on the server. I checked many sample configs and the

I did this: nb <- naiveBayes(users, platform) pl <- predict(nb,users) nrow(users) ==> 314781 ncol(users) ==> 109 1. naiveBayes() was quite fast (~20 seconds), while predict() was slow (tens of minutes). why? 2. the predict results were completely off the mark (quite the opposite of the expected overfitting). suffice it to show the tables: pl: android blackberry ipad