similar to: Can I do this ...

Hello all, I was wondering if anybody knows where one might obtain a PCI ISDN card supporting a single BRI for use with Asterisk in Australia (and using something like chan_capi). Because of the Isdn4Linux DTMF issue, I don't want one of those cards. I've already spent too much time messing about with my current card. I'm after something like the AVM Fritz! cards. I found one place

I'm having the following problem: I call into my Asterisk box (RedHat Linux 9.0, 1 Digium X100P card) to access voicemail. After dialing the appropriate extension I get voicemail, am presented with the user's unavailable message, and can leave a message normally. The problem comes when I press "#" to end the recording, at which point I am told "Your message has been

I've noticed that a few people here are from Australia. I'm wondering where you all get your hardware from. I'll probably order some of the X100P cards from Digium soon (and possibly some FXO cards), but for other things such as ATAs, etc. However - these obviously aren't ACA compliant. Does anybody know of compatible hardware (for POTS lines) that is ACA compliant? The cheapest

Hi to all, I've just received my Cisco 7905G ipphone. I want to connect it to asterisk server but it looks that it has been preloaded with sccp protocol, so I guess I need H.323 or SIP firmware image of some kind. I have a working tftp server on my asterisk box also....What do I need to do now to get things wokring? Thanx in advance, Victor...

Hi All, Just a heads up - I was looking around the Cisco FTP a little while ago and noticed that the SIP 7.2 images for Cisco IP Phone 7940/7960 were released yesterday (16th August). No new features - all bug fixes according to the release notes. I've already started using it. I thought those of you running the Cisco phones and the appropriate access who didn't yet know would like to

Hi all, This is my first post here - I started with Asterisk a few days ago and have "fallen in love" - fantastic product. I've only got softphones connected at the moment - I'll probably order the FXO/FXS cards in about a month (and then think about getting some hardware SIP phones). Our current phone system is quite a few years old and isn't growing with us (when a single

Hello all, I know this isn't strictly Asterisk, but I'm sure that there are more people here using the Cisco 7960 w/ SIP, so I thought I'd post here. I've just bought a Cisco 7960 phone to use with Asterisk. It came with the CallManager image on it. I've got the 4.4 SIP images (P0S3-04-4-00). If I put "P0S3-04-4-00" in the OS79XX.TXT file, the phone downloads this

Good afternoon all, I was wondering if anybody knows of a way to limit the number of calls going out over an interface (or respond with some sort of 'circuits-busy' message?) The reason I ask is my outgoing bandwidth is only 128kbit and if there are any more than 2 calls going over the internet interface the QoS is reduced dramatically for all calls. Failing that, does anybody know if

The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c #define CALLERID_UNKNOWN "Asterisk" I've changed mine to: #define CALLERID_UNKNOWN "Unknown" -----Original Message----- From: Shaun Ewing [mailto:sewing@gmail.com] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List

I have a Cisco 7940 when you call in from outside and dial the Cisco phone extension I get this Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[18451]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3)

What headsets have people found work well with the Cisco 7940 and 7960 phones? To date, I have tried a couple of the headsets within the Plantronics H series (H41-N), and noticed that the volume of my speaking is lower over the headset than on the regular handset. I am currently looking for headsets that are known to work well. I do know that Cisco lists the H-91 and H-101 as certified to

Hi. I have sjphone installed on windows and working except for dtmf. I read the docs for sjphone and it uses inband dtmf. I configired dtmfmode=inband but it still does not recognize it. Someone on the lists said that inband only works using alaw or ulaw but i tried only allowing that too but still no go. Hmm.. any other ideas? I can't get any other client to work on windows :-/ I

Some of you may remember back in May the thread on using Asterisk as a voicemail server for a Cisco Callmanager system. My own Callmanager system is integrated into an Asterisk server for voicemail (and other things). Back in May I was using H323 for integration, but since I've upgraded to CCM 4.1 I have switched over to SIP. The integration with H323 required using Call forwarding to send

Has anybody successfully (or I guess unsuccessfully for that matter) implemented Cisco Call Manager and used an * box for voicemail? I checked the wiki and google and I see some references to Call Manager Express and *, but CME is completely different than CM. If anybody has done this or has any insight, it would be appeciated. We are trying to migrate ~ 300 users off of Cisco Unity and

Hello all, I currently have an Eicon Diva Client isdn card using i4l. Outbound dtmf doesn't work (and never has), but there has been an annoying problem with false dtmf detection in calls (that could be triggered easily by blowing into the receiver on the remote end). I looked through the list and found two patches that need to be applied - 1 to isdn_tty.c in the kernel, and another to

Typo in your OS79XX.TXT P00 ? instead of P0S !? -----Original Message----- From: Michael L?jtnant [mailto:ml@zyxel.dk] Sent: 17 August 2004 13:31 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released Hi Shaun, Saw you post, and rushed to their ftp-server and downloaded it :-) But, I can't make my phone (7940) upgrade, so maybe you

As per my problem yesterday with the Cisco 7960 and getting it talking to Asterisk on a different subnet, I gave up trying and just put the Asterisk box back on the internal subnet. However, I made two changes: - the external IP address is set on an ethernet alias eth0:0 - the main Linux router will change outgoing requests from 10.1.1.2 to the external IP (rather than the default behaviour of

Hi! When I call a colleague of mine from my Cisco (via Asterisk), they get on their display: From Evert asterisk How do I remove/change the 'asterisk' part? Regards, Evert

Hi All, This isn't really a question, but it's an issue I experienced that was driving me crazy for a few days, so I thought it might be good for the archives. Basically what was happening was everytime a particular customer called (long distance), the line would disconnect immediately after answering. I thought it might have been the phone, so I swapped the phone with another - still

Hi, Today I was assigned a block of IPs by my ISP and decided to move the Asterisk box onto a public IP. I am also using 10.1.0.0/16 on my LAN, with the IP phones under 10.1.4.0/24 There is clear routing between the public and private IP ranges (no NAT). Private can see public and vice-versa. I changed the proxy config in the phone to point to the new IP address. Previously the 7960 worked