Re: [Asterisk-Users] OT: MAX TNT and PRI calling name (CNAM) facility message

2005-06-27 Thread Kevin Blackham
provider cannot provide CNAM lookup results in the SETUP message, only locally configured names in the DMS-100 (Centrex style). On 6/23/05, Matt Fredrickson [EMAIL PROTECTED] wrote: On Thu, Jun 23, 2005 at 12:20:34AM -0600, Kevin Blackham wrote: Does anyone have a MAX/APX with working ingress PRI

[Asterisk-Users] OT: MAX TNT and PRI calling name (CNAM) facility message

2005-06-23 Thread Kevin Blackham
Does anyone have a MAX/APX with working ingress PRI calling name? I recently acquired a MAX TNT on the cheap and it's integrating fine except for one thing. In the 11.0.0 release notes, it is stated that ISDN calling name will, if present and permitted by presentation flags, be added to the

[Asterisk-Users] newlines in application data strings (e.g. userevent)

2005-01-30 Thread Kevin Blackham
exten = s,9,UserEvent(AgentMoreTime,Agent: ${agent}\r\nUntil: ${wrapupat}); Fragment \r\n parses into rn. \\r\\n turns into \r\n (uninterpreted). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Multiline / Console / Receptionist phone

2004-12-14 Thread Kevin Blackham
Have you configured the 'break' key in the web ui? Not exactly an intuitive option, but that's the 'xfer' key. On Tue, 14 Dec 2004 09:06:14 -0800, Tracy R Reed [EMAIL PROTECTED] wrote: The only thing that does not seem to work on the snom phone is transfers which is unfortunately a big

Re: [Asterisk-Users] New PRI with DID in US?

2004-12-12 Thread Kevin Blackham
On Fri, 10 Dec 2004 17:26:48 -0600, Rich Adamson [EMAIL PROTECTED] wrote: Just turned up a new PRI with DID's in the US. I'm receiving 5 digits of the DID numbers as I requested. Assuming I have 100 DID numbers but only define 50 of those in extensions.conf, is there an easy way to send the

[Asterisk-Users] -p real time priority and -U together

2004-12-10 Thread Kevin Blackham
When I start asterisk with 'asterisk -pU asterisk -G asterisk', I can get it up and running as non-root with priority. However, if I restart from CLI, it exits complaining about being unable to set prio. Clearly this is because it's non-root. Is this an impossible scenario?

Re: [Asterisk-Users] Two zaptel T1 cards: no clock from one

2004-12-06 Thread Kevin Blackham
On Sat, 4 Dec 2004 21:18:50 -0600, Rich Adamson [email protected] wrote: Help me understand what you mean by neither is providing clock. By definition, every single T1 provides clocking within the transmit side of a T1. Its embedded in the data stream and you can't turn it off. Are you

Re: [Asterisk-Users] Two zaptel T1 cards: no clock from one

2004-12-06 Thread Kevin Blackham
Oh, you got to be kidding me. :) I've removed zaptel, wct1xxp, wct4xxp completely, and reloaded in the same order as they are loaded, alternate different ways, etc. No help. After trying only wct1xxp with 12 channels and seeing alarms clear, something stray was left saying span 5 was

Re: [Asterisk-Users] Two zaptel T1 cards: no clock from one

2004-12-04 Thread Kevin Blackham
Yeah, proper crossover cable. I've eliminated all cabling issues with the T1 analyzer. I get a full and accurate pattern back when I test from the cable end where it would have been connected into the T100P, with the channel bank in loopback. The main symptom is that when I hook the analyzer

[Asterisk-Users] Two zaptel T1 cards: no clock from one

2004-12-03 Thread Kevin Blackham
List, I have a TE410P (T1 mode, all PRI) and a T100P (fxoks, for fxs channel bank). I cannot seem to get the T100P to send any clock to the channel bank. I prefer that it use the same clock source as the TE410P, but it doesn't matter if it's not in sync just as long as it's there. The TE410P

Re: [Asterisk-Users] Playing reveived message WAV file

2004-11-26 Thread Kevin Blackham
Look for the 'play' binary. It has never failed me in playing a wave (gsm, pcm, etc) from asterisk. It's an older OSS (/dev/dsp) binary, so you'll need OSS compatibility. On Thu, 25 Nov 2004 20:12:14 -0700, Joseph [EMAIL PROTECTED] wrote: After somebody records a message asterisk notifies me

Re: [Asterisk-Users] SIP Phones-Receptionist Setup

2004-11-22 Thread Kevin Blackham
I have a 200 and the hint() stuff works fine for indicating status of any channel (including Agent channels). The Snom subscribes to asterisk at whatever url you put in there, then * will send notify events when the dialog state changes. It's not quite a shared-line (at least the way I

Re: [Asterisk-Users] Polycom IP 300 PoE?

2004-11-18 Thread Kevin Blackham
List, How about circumventing this dongle? Switches or PoE midspan units that support forcing power on 4-5, 7-8 without detection? Found any 3rd party contraptions, like PoE splitters that tell the injector it's ok, which can simply have an end crimped on in the right way to hit the polycom or

Re: [Asterisk-Users] Music on Hold on Debian 2.6 help wanted

2004-11-18 Thread Kevin Blackham
Ensure the debian package 'mpg123' is installed, and that 'mpg321' is removed. It links mpg123 - mpg321 via /etc/alternatives. Beats cooking it up from source. On Thu, 18 Nov 2004 09:25:05 -0500, Peter Osborne [EMAIL PROTECTED] wrote: I had the same problem on Debian, the mpg123 in Debian is

[Asterisk-Users] ICD status

2004-11-04 Thread Kevin Blackham
I gave ICD a spin today, and had a lot of problems. It was yesterday's CVS checkout for both asterisk from digium, and ICD from orson.callenish.com. I was running into major problems with agent login/out, configuration, general loss of coherency in status, potential jumping of fenceposts

Re: [Asterisk-Users] Agents Log off

2004-09-11 Thread Kevin Blackham
I created a small patch to make this happen. You enter an l for the extension when you call AgentCallbackLogin. It sees that (ell) and sets the extension to as if the person on the phone simply hit #. This needs to be properly implemented as an AgentCallbackLogout() application, but this is

[Asterisk-Users] chan_agent and SIP UA transfers fail

2004-09-10 Thread Kevin Blackham
,,Kevin Blackham [sip.conf] [general] realm=xmission.com context=default port=5061 bindaddr=snip srvlookup=yes tos=0x18 maxexpirey=3600 defaultexpirey=120 canreinvite=yes relaxdtmf=yes ;my UAs register to a SER proxy, which forwards most non-REGISTER methods to asterisk ;asterisk talks back