similar to: Odd one way RTP on SIP to SIP calls

Hi guys, I am wondering (and have been unable to find out thus far) whether Asterisk sets some special channel variables or something when a call is transfered with the REFER method. Basically, I'm trying to figure out if it is possible to somehow get a transferred call back to the transferrer (as it is done with the built-in atxfer) after X seconds (or an unsuccessful attempt). Using a

Ok so the install has been seamless on a few machines and I decided to try it on my laptop (a Dell D620). I had everything working great and kicked it around a bit and then figured, if I'm going to keep this on my laptop I'll need to reformat everything, enable some encrypted filesystems etc ... not a big deal ... kind of Here's the problem, the laptop has a resolution of 1440x900 and

I've got a Dell Latitude D620, dell docking station and a Dell 2007WFP (20" widescreen lcd). With XP I had the laptop screen and the lcd working as dual screens. But for the life of me I cannot get CentOS5 to work with dual screens. Anyone have this working? I would greatly appreciate any assistance. -matt

Hello. I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold functionality has changed (or is bugged?). I have Aastra 6757i and Aastra 6731i phones, and now when i press the MusicOnHold button / change lines on the phone, MOH no longer starts. It did this in v 1.6.0.9. The invites received are exactly the same, only 1.6.1.10 doesn't ever start MOH. Is there some

Hello, When I bridge an incoming and outgoing call (attempting to simulate call-forwarding) I'm only getting one CDR -- that of the outgoing call. A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone on PSTN) and bridges the call. The only CDR created is from B to C. I have even tried using Answer() and ForkCDR() to get two CDRs, but to no avail. I am starting to

Hello to all, I have a problem with an asterisk qsig. I have three machines: Nortel CS1000 --- Card Sangoma PRI ---> Asterisk QSIG ---SIP Trunk---> Asterisk I use Snom phones on Asterisk. If I call from Asterisk to Nortel, Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone, but if I call from Nortel to Asterisk, the QSIG does not send

Hi, Just tried upgrading to 1.8 and ran into two problem immediately; 1. Caller-ID behavior is different -- now when I set the caller-id name to something with special characters (?, for example), the SIP INVITE now has %C3%96 instead of the ? character. I've tried doing Set(CALLERID(name-charset)=utf8) as well as iso8859-1, but it's always the same behavior. 2. My AMI scripts have

Well I am entering into a realm that I don't know. 3 sites with Asterisk 1 site with Nortel Asterisk/Sip calls working fine between the 3 sites. Asterisk to Nortel set calls working fine. (call comes from asterisk to nortel and rings telephone, people answer and talk happens, hangup call clears) Nortel to Asterisk. Set on Nortel gets a busy signal. Any suggestions on what to look

Dear List, Need to know if anyone on this list has had any experience with using the Nortel BCM 50 for Call Account Reporting using an IP connection to a Linux / Asterisk interface? Presently, I have a BCM 50 installed that uses a local Lenova Small Form Factor PC with a windows XP / os that quit reporting because of a up-grade to the Nortel reporting software. Nortel support is now telling

I have gone through some archive about Nortel MICS (Meridian ?)+ Asterisk Integration but I'm not sure whether same as my case . 70 telephone sets | | Nortel MICS 4.1 --------- Asterisk | PSTN I have read the David Gomillion's Guide and got the idea . However, my plan is slightly different from what he did , I need to use Nortel MICS to connect to PSTN (I

Has anyone had any luck getting an asterisk box to talk to a Nortel C15K softswitch? Or any Nortel "sip" products? I've been playing with it for several days and can't seem to pass calls either direction. I know that whike the Nortel says the C15K speaks SIP, it really speaks nortel's implementation of SIP, but I thought I could get it to at least pass simple calls back

Hey All, I have an Asterisk box connected to a Nortel Option 11C via a T1. In the Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI card. The Nortel is also hooked to the PSTN via a T1 on a different NTAK09 PRI card. I've included the Zapata.conf and zaptel.conf files below. Our issue is that when a call is sent over the tie line between the two systems, the audio on the

Hi, Did anyone manage to get 4.4 working on a Dell D620 laptop? The thing has a 1440x900 LCD screen, but I don't get X the work with that resolution. (lspci: Intel Corporation Mobile 945GM/GMS/940GML Express Integrated Graphics Controller (rev 03)) Google suggested to use '915resolution', which I installed (915resolution-0.5.2-1.el4.rf), but that doesn't help. Regards, --

I am trying to integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only

I am using asterisk 1.4.32 and wish to connect using SIP to a nortel 1000 switch with the ability to have 90 calls at a one time outgoing or incoming. the nortel reseller is asking me what to do. I dont know nortel at all. I thought I just needed a "SIP trunk and IP address of the their server and an account name, and provide her my IP address". They didn't know what to do with

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

I need help on my zaptel.conf and Zapata.conf for a TE207P I'd like Span 1 to receive a PRI from the phone company(US PRI). I'd like Span 2 to interface with a Nortel Phone system as a PRI(acting as the phone company) Essentially my asterisk box is a man in the middle intercepting calls from the PRI passing certain DID to the Nortel, also intercepting calls from the Nortel passing them

Hi, I have obtained some results with factanal that seem to support a hypothesis I already had, and I'd like to verify that I can indeed conclude this from this new analysis. We had subjects reproduce perceived trajectories with a device that allowed us to measure spatial position (the path) and the device's orientation at any of those positions. From this, we calculated the rotation of

Hi, Some time ago, there was a discussion about the inability of nortel meridian pbx to dial analog tones thru an meridian ATA, and the work arround was to enable 808 function that makes the dtmf tones long for the current call. The nortel meridian is connected via a nortel ATA to a TDM400 to a FXO port. Anyone can say me who to actually use that function (you dial something or is pbx

Hi all, I've seen a couple of posts recently from people who are doing something with Asterisk and a T100P and a Nortel PBX. However it's not clear exactly what they are doing. Does anyone know if it's possible to use a T100P or T400P as a DSX-1 interface to connect to a T1/PRI CO module in a Nortel PBX? We have a Nortel Norstar Modular ICS PBX and I'd like to plug its CO