similar to: Clarification - FAX on local network

Since upgrading to 7.2, I've noticed a random problem where I dial a number and hear all the correct tones in the handset, but the display won't show all the numbers I dialed. So you sit there waiting for the dialplan to kick the call off (b/c you heard the proper amount of tones played and think it's all good) but the phone is just sitting there b/c it somehow "missed"

I was looking at the sample "follow me" config (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me <http://www.voip-info.org/wiki-Asterisk+Tips+follow+me> ) which uses a dial modifier 'c' to enable Answer confirmation - "If the letter c follows, then "Answer Confirmation" is requested, in which the call is not considered answered until the called user

I haven't seen any status on the 7960 backlight project lately...I tried to email the original poster but his mailbox appears to be over quota. Does anyone have an update on this? Thanks, Marty Mastera M3 Resources marty@m3resources.com Phone: 303.680.1283 x200 FAX: 303.680.1283 IAXTel: 700.206.7507 FWD: 484162 -------------- next part -------------- An HTML attachment was

Can anyone help me to understand what the significance of this output is? May 17 10:50:23 DEBUG[2030]: Didn't get a frame from channel: SIP/105-1ae4 May 17 10:50:23 DEBUG[2030]: Bridge stops bridging channels SIP/105-1ae4 and SIP/outbound-7dc3 I searched for these phrases but am coming up short on what they really mean. I'm trying to investigate problems we are having with two

Does anyone know if the 7912G (which the wiki says can do either sccp or sip) uses the 7940/60 sip firmware? I ask this because the only firmware I can seem to find on TAC for the 7912G is sccp, no sip...if it takes it's own firmware and doesn't use 7940/60 firmware, can someone point me to the right location for it? Thanks, Marty Mastera M3 Resources marty@m3resources.com Phone:

Hello, I'm looking for any comments or user experiences from anyone who is using 7912G phones with SIP. Any installation issues? Usability problems? Do the features seem to work, etc...In short, I'm looking for your opinions on how suitable this phone is for an asterisk implementation for approx. 10 users. Next logical question: what other phones would you recommend for a situation

Hello, I am trying to come up with a good way to enforce a limit on the number of simultaneous calls that can occupy an IAX trunk at any given time. I have searched around and so far can't locate a config option that would directly label a IAX trunk with a specific number to obey (is there one?). Based on examples for the SetGroup and CheckGroup commands, I am thinking of using SetGroup

I need to purchase approx. 10 phones for a small office implementation. Nothing fancy is required besides a full-duplex speakphone, in the sub-$200 range. I am currently looking at the Polycom Soundpoint 500, Uniden UIP-200 and SwissVoice IP-10. I have searched around and found various posts regarding each phone's ability to work with asterisk (SIP, I probably should have mentioned), but

Has anyone got this to work? Under Idle Display Animation, the administrators guide says "For example, a company logo could be displayed".. In the ipmid.cfg file, I enabled 'ind.idleDisplay.enabled' (ie changed it to 1), and under the IP 500 section, I added an entry for the bitmap that I want to display: bitmap.IP_500.66.name ="arf" but from there I'm not sure

Hello everyone.... Searching the archives and google always comes up with entries regarding the "dyn" dns option in the 7960, but I can't find answers to my specific question.... My 7960 is connected via cable modem and is NAT'ed (everything is working fine). On the 7960 under SIP configuration\NAT Address I have the public IP of my cable connection. Comcast gives me a

Hello: I have read that music on hold requires a timing source (which I never had to worry about previously since the server had zaptel hardware in it)...now I'm configuring a server in a colo which has no zaptel hardware. If I use the dialplan to run MusicOnHold(), I do get the music upon dialling that extension, but if I try to use the hold button on either a 7960 or X-Lite I get

I wonder if anyone else's Asterisk box drops the connection to voicemail after 30 secs even when the maxmessage parameter is set to 180 (3 mins). Here is the general section of my voicemail: [general] ; format=wav49 maxmessage=180 attach=yes Even if it only gave the caller 30 sec to leave a message it would be nice to tell the caller that they have run out of time before ending the

> I too contacted CDW about the $9.37 Cisco support > contract. But because I did not buy my phone from them I was > not allowed to purchase it. The vendor I bought the phone > from does not provide them. What are the "magic words" to > get CDW to sell it to you? With all of this hassle I highly > doubt that I will buy more Cisco phones anyway. After >

We are having an issue with the SPA962/932 combo where the phone and the sidecar will reboot unexpectedly ? could be onhook, could be on a call, doesn?t seem to matter. I read that certain early firmware revisions could cause this so I?m running what was a week ago the newest available (5.1.18). A call to Linksys support suggested that I ensure that the phones are using a recent firmware version

Hello everyone, I playing around with ENUM and have configured * to query a few sources for testing purposes (fierymoon, e164.arpa, e164.org). I'd like to know if there is a way to query these servers manually (ie outside of asterisk via nslookup or equivalent) to find out if particular exchanges are listed with wildcards, so as to terminate calls to those prefixes (I'm not trying to

Any recommendations on an economical layer 3 switch? Preferably something that you have hands on experience with connecting to IP phones with attached PCs? Specifically I need the ability to set the VLAN in the phone to tag voice packets and to set a native VLAN on a per port basis on the switch to put the untagged packets from the attached PC into a separate VLAN. POE is not a requirement

Hello everyone, Having recently gotten Broadvoice inbound DTMF to work (thanks Greg)...I am now running into a frustrating problem...when a call comes in to the BV number via a cell phone (tested with 3 different cell phones; albeit all on T-Mobile) the beginning of the IVR welcome audio is cut off. A call placed via a landline phone over the PSTN to the BV number does not exhibit the problem.

Hello, I don't have any experience with channel banks and would appreciate any feedback on my theory outlined below: We have a single T1 entering the building with channels 1-12 being voice lines and 13-24 being a 768k internet connection. This T1 terminates to an Adit 600 (T1-1). Here's what I know. Channels 11-12 go out the Adit 600's 25-pair connector to a wiring block (and

Hi all, I've cross-posted this to the -users and -biz groups. Hope that's OK. I have a customer who REALLY needs to be able to send/receive faxes reliably. I could probably get hylafax configured, but I'm not sure how reliable it is. If it is considered reliable, would someone let me know? Otherwise, is there a product/service they can buy that will allow them to fax to/from

When are 'we' going to have this solution Steve? :) You keep talking about it, and we keep asking when it's going to come about. I know myself, SS7 will be a make or break for our continued use of Asterisk. Even if we had some price indications would be good, and/or a timeframe? Don't want to seem pushy, but it's been on the cards for quite some time now. Ben -----Original