similar to: Call forwarding (from PHONE configuration) with PRI

running asterisk 1.4.13... I've noticed with SOME email clients that the timestamp reported from voicemail is 5 hours off (difference of EST vs UTC). That is, a voicemail received at 15:17:59 EST is sent via email with a Date: header of "10:17:59 - 0500". An email sent through normal means (mail client) from the asterisk server reports the correct time, so it's definitely

Has anyone else encountered this? I have a fairly large installation (~50 phones, almost all Polycom 501's and a handful of 601's. We're running into a number of phones on which the outbound voice (Polycom phone user doesn't hear any problems, but the other end does) is breaking up occasionally -- enough to be noticeable and make you say "what?". In each case, rebooting

How do you transfer or re-park a call that's been picked up from a parking lot? I don't see any options for specifying the transfer options on the parked call, so that you could transfer or repark it. -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Folks, having an issue with outbound calls through a VOIP provider. Calls get sent out with the CallerID(number), but where does callerID(name) come from? Apparently not from provider, as we are seeing different (sometime missing) names on inbound calls, different than what we have configured. Apparently this comes from some telco database somewhere? Numbers were ported from a wired-telco.

I've tried to find other threads with this same topic, but haven't found any... Apologies if this already being discussed.... Running asterisk 1.4.13 (upgraded from 1.4.9) and zaptel 1.4.4. Having an issue with (I think) parked calls. We tend to park calls, but we're often not able to pick them back up, or the other party says they get dropped, etc. There doesn't seem to be a

Does anyone know how much space the appliance has for voicemail and/or logs? Doesn't have an embedded disk from what I can see, and only a 1G flash card? -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: exten => s,1,SetCIDName(Test) exten => s,2,SetCallerID(1234561234)

Hello Everyone, I am looking to getting converged with the local ILEC here in Canada (Bell or Telus), and was wondering if I can get some more information about typical setups. DIDs and channel offerings from third party clecs does not fit our business model and that's why we are looking to purchasing phone numbers directly from the ILEC, and manging our own PRIs for a convergence of sorts.

Hello, Quick Diagram: Telco-PRI -> Asterisk <- Norstar PRI -> Norstar PBX (DMS100) (TE405P) (DMS100) | | V Cisco 7960G (SIP) I'm trying to change the Origination Number on my outgoing PRI, and running into a weird problem. If I make a call from a SIP extension off asterisk using the following context:

When receiving multiple calling numbers via a PRI for a call setup, I cannot find the ability to select between either first or last. Is there a way to do this currently? If not, can anyone help me in getting this to work? Here is the dump of the PRI in intensive debugging. Please note that I want the first calling number, not the last. Any help would be much appreciated. Below the PRI trace

Hi all, using latest asterisk-svn I want to reflect an video call incoming via an PRI EuroISDN channel to another outgoing PRI channel, and I want the the outgoing channel to have the exact same bearer capability < Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) < Ext: 1 Trans mode/rate:

This indicate that the connection with the local provider PTSN it is ok? : -- Attempting call on Zap/10 for s@inbound:1 (Retry 2) -- Making new call for cr 32781 > Protocol Discriminator: Q.931 (8) len=28 > Call Ref: len= 2 (reference 13/0xD) (Originator) > Message type: SETUP (5) > Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) >

hi all Attachd is a PRI DEBUG dumped while dialling out to a busy number among with zap(ata|tel).conf. asterisk did not flag busy, and I got a busy indicator going meeeeep-meep-meeeeep-meep-meeeeep-meep (never heard this before) Can someone help me out here? thanks roy -------------- next part -------------- A non-text attachment was scrubbed... Name: zapata.conf Type:

Hi all, I'm debugging a PRI problem, i can see the calling number but i get a busy all the time. From the output below, I guess asterisk hangs up immediately. Can anyone point out what the problem is? Thanks in advance. *CLI> < Protocol Discriminator: Q.931 (8) len=32 < Call Ref: len= 2 (reference 4865/0x1301) (Originator) < Message type: SETUP (5) < [04 03 80 90 a3] <

Below is a snipped debug on our PRI. We are getting number only for the CallerID but the telco says they are sending us Name and Number. We are getting the Name in a second frame but Asterisk is not passing it to the device it rings. The message below says "Presenation allowed of network provided number" which leads me to believe Asterisk thinks it should not be displaying it. Can anyone

Hi all I wonder if anyone can give a little insight into this; asterisk@home 2.2, HP Proliant gl3, sangoma A101,Cable and Wireless, ISDN30. 10 zactive channels Incoming calls work fine no problems tested 43 DIDs all working. zaptel.conf # Global data loadzone = uk defaultzone = uk span=1,1,0,ccs,hdb3,crc4 bchan=1-10 dchan=16 zapata.conf [channels] language=en context=from-pstn

hi! I have a question of ISDN PRI "calling number" setting for outgoing call. I need the receiver to see the CLI that I have set. Eventough I've set the CLI as below, the receiver keep on getting a fixed number( used with the PRI to receive incoming calls). My ISDN PRI E1 provider says he has not done any restricions on the custom CLI side.

After sometime I got my E1 PRI configured correctly in /etc/zaptel.conf and I now don't see any alarms on the E1, but I can't still dialout correctly, I enable every debug that I could though of and this is what I see: At the end the D-channel is down and I cannot even try to connect another call because it tell me that the zap channel is unavailable ,the other this that I notice is that I

Yes its a T3 split into seven trunk groups with one D channel and NFAS on each. Can you explain the cut function or point me somewhere please? Thanks, Steve -----Original Message----- From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] Sent: Mon 4/10/2006 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] ANI and DNIS Seperation on

I haven't seen anything about this is the archives, so here we go. Sorry, its a long one. My setup: Dual Xeon 2.4 GHz. One TE410P card. Span 1 populated with a PRI. Span's 2-4 empty. zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us I'm currently having a problem where one of our PRI channels seems to keep a call even after it is hung up. We have