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I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,d4,ami fxsks=25 And in zapata.conf, I

I currently have Asterisk deployed in my office with a TE411P. On the first port of this card is a T1 from the telco setup for D4 AMI. Unfortunately, I'm not receiving caller ID on inbound calls from this line. The caller ID information is arriving in the form *ANI*DNIS*. In zapata.conf, I have signalling set to em_w. The DNIS always arrives correctly, but I'm never receiving the ANI

Hi listers, I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's. I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office. However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price. I would appreciate any input people have to offer. Thanks, James

I am getting ready to purchase my first Digium card to start experimenting with Asterisk. Before I make my purchase, I wanted to make sure I'm not going to have issues with these cards (need to see what the specs are on my box, 5V or 3.3V PCI ). I will be using Asterisk @ Home, so will be Asterisk v1.0.9. I took a quick poke at the lists, and it appears several people have been having

..., what is the maximum number of cards people are putting into boxes? If two cards is the limit, am I right in understanding that the preferred way to support a large number of users is to split out into multiple Asterisk servers, and then use IAX to connect them all together? Thanks, James Texter

Hello list, I have recently deployed Asterisk as the phone system for my office. So far, everything has been going really well, except for one little thorn in my side. I have a set of 6 analog lines that are connected to a TE411P via a Rhino FXO Channel bank. If I call the analog number, Asterisk answers the call, and routes it appropriately. The problem is, when I hangup, Asterisk never

I have a site running 1.4.17 with Zaptel. They want to add a TE420P for additional T1 capacity. I'm 99% sure this will work, anyone aware of a reason it wont? Thanks, James -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100806/b5f4bc0f/attachment.htm

I just put in a Audiocodes Mediant 1000, which seems to be working well except for one annoyance. I am using Polycom 501's and 601',s and if I do a supervised transfer of a PSTN call where I complete the transfer before the 3rd party has answered, the PSTN party hears dead air until the call is answered or goes to voicemail. I'm not really sure where to start my troubleshooting. Any

...xp.c that looks like the following: static int vpmdtmfsupport = 1; Change this to static int vpmdtmfsupport = 0; According to Digium support, this moves the playing of the DTMF's from hardware to software. Once that change was made, I have not had any other issues. Thanks, James James Texter wrote: So, I'm still having this problem with outbound calls not working when using a channel bank. I've purchased a Rhino FXO channel bank from VoIPSupply.com to make sure it wasn't an equipment problem. I am using a Digium TE411P card, and have simplified it down to just 1 port plu...

The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug.

So, I'm still having this problem with outbound calls not working when using a channel bank. I've purchased a Rhino FXO channel bank from VoIPSupply.com to make sure it wasn't an equipment problem. I am using a Digium TE411P card, and have simplified it down to just 1 port plugged into the channel bank, with just 1 analog line plugged in. If I place an inbound call on the line, it

I have a requirement to play different hold messages depending upon the extension that originated the call. I noticed a musicclass setting in sip.conf, but it appears this is global. I tried setting this on all of my individual extensions, but it didn't have any affect. Is there a way to achieve this, either through sip.conf or in the dial plan? Thanks, James

Hi everyone, I am running into wall today with simultaneous call limits. I have two Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a lot of sip calls from one machine to the other by issuing AMI Originate commands to one machine. The machine that makes calls plays a message (demo-intruct) upon the other machine answer. The machine receives the calls just waits for 40 seconds

Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this

I am doing some experimenting with Asterisk 1.0.9 and Polycom IP501's. I have the extensions setup, and everything is working well up to this point. Now, I want to setup my system so that a user at an extension can start a recording on demand. I have tried various Google searches, but am coming up empty. Could someone point me in the right direction, or have some sample Polycom/Asterisk

Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason a0 on CPU 0. Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely on the PCI bus. Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue Would my Digium TDM410P cause an NMI, or is my computer failing? - Mike

I've been noticing that my group of Polycom IP 501 phones seems to randomly reset themselves nearly every night (I guess it usually happens at night, since I've never seen it happen while I've been at work during the day).. When I say "reset", I mean, the hands free volume and ring volume are set to the default and the call logs (received calls, missed calls, placed calls)

Greetings list, I remember a discussion many months ago which ISTR concluded that asterisk didn't play nicely at all in multi-homed setups (e.g. SIP packets not being sent out through the same interface they were received on, etc.). Is this still the case, or are there versions which have resolved the issue? Even if it's still the case, is this only a problem for SIP, or does it affect

I sent this yesterday, but saw zero traffic, so I think it got lost in the ether, so I'm sending again. I'm having an issue using Asterisk 1.2.14 and an Audiocodes Mediant 1000 ISDN gateway. For the most part, everything is working except for attended transfers. When I do an attended transfer, and complete the transfer before the 3rd party answers, the PSTN side hears dead air until the

Hello listers, I went to pull some CDR's from my PBX, and noticed they were a bit light. I also noticed output on the console about CDR's not being posted. I am currently running 1.4.13, and in looking at the change log, this was a change in behavior as part of mantis 10659. Personally, I think the old behavior was more correct, but obviously at least one person disagrees. I think