similar to: How Separate a few channels from the others on a PRI

Okay, here is the background. I have a PRI with 15 active channels on it. I originally setup all of them in group=1 and all outgoing and incoming calls used this group. The phone number that I have associated with these channels ends with 750 and that is how I direct the calls. i.e. In my extensions.conf I have: exten => 750,1,Dial(SIP/120,20) All this works fine. Now I have the need

Okay, here is the background. I have a PRI with 15 active channels on it. I originally setup all of them in group=1 and all outgoing and incoming calls used this group. The phone number that I have associated with these channels ends with 750 and that is how I direct the calls. i.e. In my extensions.conf I have: exten => 750,1,Dial(SIP/120,20) All this works fine. Now I have the need

I have an extension setup in my extensions.conf for hold music. ext. 600. If I pick up a phone (polycom soundpoint 300 sip) and dial extension 600 I hear the hold music playing. If I call another extension and pick it up and put the call on hold with the hold button on the phone I hear nothing at all. Does anyone have any experience with these phones and getting the hold button to work?

I've read as much as I can on this error and still can't seem to figure out what's causing this error: Jun 23 20:54:58 NOTICE[7483]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 23 20:55:03 NOTICE[7483]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 23 20:55:03 NOTICE[7483]:

I have a single PRI to our local telco that is shared by multiple companies. I would like to find a way to get our long distance carrier to provide separate billing for these companies. I am currently calling Authenticate() to collect an account code which is written to the call detail record, then I estimate the charges for each company by parsing the call detail records. This doesn't work

Ok so I'm thinking about migrating to a Flex T-1 with 1.024Mb Data and 6 Hard Incoming/Outgoing Phone Lines. I can either break them out via channel bank and use a FXO to convert them but I'm told it will add a dely of at least one ring. And you have another analog/digital conversion to the mix. The 2nd option is to bring them in over PRI's Question is the solution provider will

"Google: Evil or Not?" is my first Rails webapp and I''ve just made it public at http://evilornot.info Do you still believe the ?Do no evil? Google mantra? Do you think Google Book Search, the AOL deal, and Larry and Sergey?s 767 point to Google losing it?s pristine morality and turning over to the dark side? Now you can discover what the world thinks and contribute your

Dear all, I have got a PRI line with E1 20 channels, my question is: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 If this is my zaptel config for an E1 PRI line, Which would be the config for a reduced PRI line for 15 channels? and for 20 channels? Thanks in advance. VoIPcrazy -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I got a problem with PRI that I'm not sure how to solve. Asterisk sits between PABX and PRI. PRI is span 1 and PABX is span 2. After every single call (no matter in what direction) I get "pri_fixup_principle: Call specified, but not found?" and "pri_dchannel: Hangup on bad channel" messages and the channel in question is restarted. As far as I can see, all

Hi all, i have observed, that sometimes all BChannels on my Zaptel Pri device (E400P) will be restarted. The E400P is connected to another pri switch. In the traces from the other side (pri switch) i can see that libpri request for the channelid is 255. Is this a bug or a feature ...? Or, can it be a bug on the other side (terminator switch) ? Have anyone an idea ? Thanks, Thomas.

Is there a simple way to enable/disable on-the-fly a pri channel? I want to control the number of incoming lines based on the number of agents I have. I took a look at libpri and zaptel and found some hints about a "SERVICE" message within the pri protocol, but coudn't find out much more about it. Best, PHM

Hi bkw Yep, which is going to be a huge problem since it's only taking a line and not doing any transmittal until after you get a line out, the line of course is being rejected before I can even get there :( Of course I can't even establish connectivity to the telco whilst having it peered to the PBX too due to the D channel issue :( Lee From: "brian" <brian@bkw.org>

Hi, I've a busy server where one E1/PRI span has been behaving oddly. Here's the span: Primary D-channel: 16 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 Looks good, eh? But every

Hey Folks....I am on the 1.2 branch with the latest from Subversion. I've been having a rough go for the last several months integrating asterisk with out Altigen system. I can get calls inward just fine. I have zero missed interrupts on the digium 110p card. I have zero frame slips according to both sides. Outgoing calls sometimes work, but more often than not I get the following: --

Hi All, We are using MAX TNT to for some T1 PRI interconnects. I'm seeing the voice channels connect at 56K. Does anyone have the DS0 channels connecting at 64K for voice, if so what is the parameter to select 56k or 64k channels? I'm not having any issues that I know of, just wanted to bounce this off the group for a sanity check. Thanks. JR -- JR Richardson Engineering for the

Is it possible to start MOH when calling to DAHDI Channel that has ISDN E1 connected with it. When the called party press hold on his phone then asterisk start MOH?? -- Regards, Shariq Khan 0333-3501125 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/ca6cfc50/attachment.htm>

Hi, I'm running a test system with Ast 11.7 and DAHDI 2.9.0. I loaded a TE205P card. Then I enter asterisk console, and once I do a "pri show channels", the console no longer works correctly. There's no output for any command after that. If I type "pri show" and hit tab for completion, console freezes there, I need to kill session. Any hints? new bug? known issue?

Hi all, I have a very strange situation, which I can''t explain. On a Core 2Quad server with 8GB RAM, we have 10 xen domU''s, as follows: vps.animalsaregreat.info login: root@zaxen01:[~]$ xm list Name ID Mem(MiB) VCPUs State Time(s) Domain-0 0 512 4 r----- 232.2 actionco.vm

Hi. I still hace problems getting my line to work. When I start asterisk, I can see: == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling ... -- Registered channel 31, PRI Signalling signalling == Starting D-Channel on span 1 == Registered channel type 'Zap' (Zapata Telephony

Hi Guys.. Just wondering if someone could help me with a few questions please. were currently using the ulaw codec with our grandstream/iconnect/asterisk setup and its working pretty good except for the fact it downloads heaps. Does anyone know a good site to get referances to how much each codec downloads/quality etc etc ? Ive tried using that g723 codec but i have have problems as soon as a i