similar to: CISCO 1760 with 1 BRI

Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like

I am starting to work on a similar solution, but with full call manager rather than CME. I am going to use Asterisk to accept POTS calls through PCI FXO ports (winmodems) and then forward the calls through to call manager via SIP. I don't have my FXO cards yet (waiting for UPS man!!) but I have * talking to the CM through SIP just fine. I am testing with the Cisco softphone, connected as a

I have the following configuration and am obviously missing something small that is causing * not to work as expected. I have the following defined in sip.conf [ccme-in] type=peer host=10.0.9.1 context=devel_in disallow=all allow=alaw nat=no canreinvite=yes qualify=yes and [devel_in] is defined in extentions.conf However when I try to call via the dial peer I have configured on the cisco

Hi, Did anyone manage to connect either Digium or Sangoma T1 card to any other PBX/gateway using T1 E&M Wink/Wink signaling? I'm trying to connect Avaya Definity to an Asterisk box with T100P and so far no luck. (I know I can do so with ISDN PRI, but need an additional ISDN processor card for Definity.) I tried to connect Definity to Cisco 3640 CCME (call manager express) to test the link

Hi, I'm trying to get a remote Cisco Call Manager Express (CME) system behind a dynamic IP address routing both inbound and outbound calls via SIP to my local asterisk server. I've got a local CME system working fine on the LAN, where the CME is at a static IP (host=10.5.7.130 in sip.conf), but I can't figure out how to get it working with host=dynamic, even locally on a test

Hello to all! I apologize in advance, for the following could be too simple, but I can't figure it out. I configured a vps using centos 8, all is working fine but I'm not able to send email from this server. I tried $ telnet smtp.mailtrap.io 25 but it doesn't work yet $ telnet www.google.ca 80 does work and particularly $ telnet smtp.mailtrap.io 2525 does work too. I temporarily

I apparently have a dead FXS module. Is there any kind of test I can run on it (on a live system) to determine if its good or bad? The telephone has no dialtone, gets no calls. It has been working for several months, and just quit yesterday. Thanks! Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO

Hello everybody, It's my first post on this mailing list, and my english is not very good. I hope this is not a problem for you. I have successfully install a lot of tind under windows XP, but a friend made one installation (I help with mail) and he have this error when he try to run tincd : --- system call 'listen' failed: l'operation tent? n'est pas prise en charge

Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, the call comes in from ISDN in the Patton gw, patton sends it to asterisk, asterisk run a macro to make a tif file using rxfax, the tif file is correctly created but with a 0 size the call

thanks to everyone offering sussgestions and pointers in answer to my questions about SME VoiP/* in the UK; i'll update you on my progress as and when this ends up as an implemented * solution (and i totally take on board the recommendation to look at 2nd hand CCME -- i should have said at the start that i've just co-ordinated an NBX install for a much larger project and that part of

Good day everyone. I have Cisco 1760 routers that do site to site voip. Each router has 2 fxs ports that connect to the local pbx and use sip to connect to other routers over the WAN. I am thinking of putting in an asterisk box at the hub site for interconnectivity with our global office voip provider. This provider runs asterisk. Question is - can Cisco 1760 routers make/receive calls to/fro

hi: Just starting out with *, and I'm planning to heed the advice to start simple and small, but the goal i'm aiming for eventually is: *-based pbx for 10-20 seat small business, based in the UK. Users will have PoE SIP hardphones. So far so good, but two questions, both UK-specific, relating to connection to the outside world (PSTN or VoIP): - are there any UK-based VoIP providers

Hi i am search a sample config (for asterisk and for cisco) for connect a cisco 1760 with a FXO card to my asterisk. Thanks for your help Jerome

https://bugzilla.mindrot.org/show_bug.cgi?id=1760 Summary: Timestamp offset using softflowd with nfdump Product: softflowd Version: -current Platform: ix86 OS/Version: Linux Status: NEW Severity: major Priority: P2 Component: softflowd AssignedTo: djm at mindrot.org ReportedBy: stephen at

Hello- I have a system with one D410P and one B200P (both OpenVox). All is well with the D410P, inbound and outbound, and I can initiate calls on the B200P BRI span, but there may be something wrong with my inbound BRI setup: there is no indication of an inbound call when I dial in to it from the PSTN. When I run "pri intense debug" and make a call to the BRI span, I can see a

Hi All, I've a BT Versatility PBX that I want to connect to my asterisk 11.9.0 box via BRI port in NT mode but actually I wasn't able to get it working. I've another standalone PBX, it's a Panasonic model, which works fine connected to the same port. The BT connected to a UK BT BRI line works quite fine after roughly a minute once connected. The PBX can be seen at

Hi, I'd like to understand what should i do to use G729 codec in a legal way, how do I order licences ? to whom ? how do I install them on asterisk etc ? thanks in advance , jl

I have a cisco 1760 phone running sip and I need to configure for our receptionist so that she can answer calls on more then one extension. What is the easiest way to configure this so that incomming calls go to the next availble extension? Does each extension on the phone need to be set seperately in the sip.conf file (see below for my example)? sip.conf file =================

Hi all, I have shorewall up and running on my system. (GNU-Linux Mandrake 9) When I tested my firewall at grc.com, Shields-Up informs me that ports 113 and 135 are closed and not ''stealthed'' When reading the faq on the Shorewall site I saw that shorewall rejects rather than denys connection requests on ''TCP ports 113, 135, 137 and 139 as well as UDP ports

Hi, I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1 and DAHDI 2.6.1. Am I supposed to use DAHDI for this card and ISDN BRI for my country (Belgium)? thx, BC