similar to: asterisk+nortel3904

[asterisk-users] Asterisk <-> Nortel Phone Switch Date: Thu, 29 Nov 2007 07:52:17 +0000 (GMT) X-Mailer: sendEmail-1.52 MIME-Version: 1.0 Content-Type: multipart/mixed; boundary="----MIME delimiter for sendEmail-20854.4017086787" This is a multi-part message in MIME format. To properly display this message you need a MIME-Version 1.0 compliant Email program. ------MIME delimiter

Still trying to make my Asterisk PBK talk to our Nortel Phone Switch (C15k). Nortel did an upgrade which changed a bunch of things today, so I thought I'd give it another shot. It looks like I'm much closer this time, but still no go. Can't do calling in either direction. Anyone have any ideas? Thanks! Shawn [nortel] host=10.0.0.10 insecure=very type=peer qualify=no

Still trying to make my Asterisk PBK talk to our Nortel Phone Switch (C15k). Nortel did an upgrade which changed a bunch of things today, so I thought I'd give it another shot. It looks like I'm much closer this time, but still no go. Can't do calling in either direction. Anyone have any ideas? Thanks! Shawn [nortel] host=10.0.0.10 insecure=very type=peer qualify=no

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

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

Skipped content of type multipart/alternative-------------- next part -------------- Nov 21 14:17:47 VERBOSE[32580] logger.c: <-- SIP read from 172.25.103.222:5060: INVITE sip:1715;phone-context=exp_net.ascom@ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From:

afternoon, i'm using call files to initiate a call between DISA and a telephone number in the field. caller in the field enters password then dials number to call. problem is that no CDR is generated for the call from asterisk to the telephone number in the field. I only get a cdr for the call from the telephone number to the number the caller dialed. any ideas on how I can get cdr's

Hi Everyone, I've been in contact with a few members off list about this, and I still can't figure out what to do (aside from buying a more current ADSI compatibile screenphone). The problem is that I have a Locked Nortel Vista-350 phone and I cannot update the ADSI scripts. I have used the reset time/date trick, and then was able to partially upload the ADSI information to the

Hello, I'm having this issue on my pbx, it appears that asterisk is refusing the codecs that my providers is proposing. My trunk configuration is: --- username=5x5x7x9x0x3 type=friend secret=CRcxn7sqwm qualify=yes port=5060 insecure=port,invite host=sip.txtxlxoxp.it fromuser=5x5x7x9x0x3 fromdomain=sip.txtxlxoxp.it disallow=all context=from-trunk allow=alaw --- A typical invite from my

I have a trixbox 2.2 and Nortel santral that are speak each other. I use digium TDM100M 2 fxs-2fxo. After I made yum update I had met with some problems when I want to make any call from extension of trixbox to extension of nortel. When I attend to log (/var/log/messages) I meet with these messages as you see below. When I try to make any call from trixbox extension the call seems established but

I listened to all the demos you showed. My ear discerns a little muffling and minor "slushiness" in the GSM files you sent, along with a much more narrow bandwidth, mainly on the high end side, and Allison either has a mild whistling s or slushy s sound in her voice or the producer didn't properly compress it to "de-ess" the recording. Or, I could just be rather tired.

Greetings to the list: this is my problen when I make a call from my asterisk towards a nortel PBX , the call is made but in my telephone sip I do not listen the dial tone or the busy tone but the call it is completed normally. sip-phone-g729-------------asterisk--------h323-g729--------------nortel-pbx thi is may configuration: RedHat 8 2.4.18-14 Asterisk 1.0.7 The NuFone

Hello - I am upgrading from asterisk v1.2 to v1.6 and I am seeing a problem with recording CDRs using MySQL. Unlike all of the other postings and web pages I have found on this issue, my installation successfully stores the -first- CDR, but nothing after that. As background info, I will note that I don't use CDRs for billing, but more in a logging fashion, to record how a given call

I use Asterisk with a HFC-S ISDN BRI card. This card needs bristuff patch from Junghanns.net. After upgrading to Asterisk 1.2.x, my CDR reports (located in /var/log/asterisk/cdr-csv/Master.csv*) are broken. Instead of telephone numbers, I get random characters like 'H? or $%. Sometimes, though, the telephone numbers are fine. The issue was also mentioned on Digium's asterisk

Hi For whatever reason we've getting 2 or 3 CDR lines logged for each call, often in different formats: as1:~# grep test-89-1e2c /var/log/asterisk/cdr-csv/*.csv

Hi All, Sorry for the double post, but I'm in a real bind. I have several * servers connected to T1 PRI's from various service providers in multiple locations the US. All the * servers use the same hardware with the same OS and * version CVS-v1-0-11/09/04-12:27:27. When connected to 5ESS Switches, using the NI2 (national) PRI protocol, the CID name and number come across fine and

Hello guys, I have this problem when a call is received in my PBX: (Caller) --> (Redirecting Service) --> (E1 PRI) --> (Asterisk PBX) --> (Internal Phone) Reception works fine, but when conversation finishes and the agent at internal phone hangs up, the call at caller's side is still alive for many seconds until it hangs up. The problem is that Telephone Company is billing me

*I'm trying to build an application that provides statistics of calls*>* and call recording. Someone told me this could be done out of band*>* with a SPAN (?) port that would replicate SIP and media packets to a*>* separate NIC without having to actually pass the real-calls thru*>* asterisk. It was explained that this SPAN port would in the SBC*>* would replicate data

We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call

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