similar to: Problems with length of voicemail

Could anyone who has successfully configured Asterisk to use g729 to conference 10-20 people please post their configs. I purchased and successfully installed 2 g729 licenses and but when I dial into my conference number on the Asterisk box from a SPA-2000 set to allow all codecs, it always appears to connect using ULAW. My iax.conf file includes the following under the general section

I understand there has been some issues sending DTMF tone through Broadvoice. Can some provide me with symptoms? --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004

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.

Since Thursday evening my asterisk box has been failing to register with broadvoice. I haven't changed any of my config files in the last week. Can anyone suggest anything? Asterisk is reporting: *CLI> Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout: Registration for '703XXXXXXX@147.135.8.129' timed out, trying again -- Got SIP response 404 "Not found"

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

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"

Can anyone tell me how I can implement the features added in the following link for call transfer? The authors seem to feel they are finished but it doesn't appear to have been integrated into what everyone can download. It is referred to as a patch but I don't understand how it could be applied. Here is the link: http://bugs.digium.com/bug_view_page.php?bug_id=0002010 I guess I just

> 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 >

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

Hey there, I don't know who else has suffered broadvoices terrible service, but I am about to my end with them. The lack of a LBR codec, the outages, the changing of servers without notifying subscribers haspushed me to my end. Now most incoming calls are abbruptly cut off within a minute of the call starting. Anyone know of any other * friendly providers that have DID, besides Voicepulse,

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

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

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:

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

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

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, When I listen to a voicemail message, the recorded message is played back at extremely low volume. All the supporting prompts are at the correct volume, it's just the incoming recorded message that is played back almost inaudibly quiet. There's no problem with the volume during normal converstaions so I'm thinking this must be specific to the Voicemail application. I've

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, Is it at all possible to have the "maxmessage" setting on per user/mailbox value? We have a requirement whereby we want the global maxmessage setting to be 180 seconds per mail box, however, we would like to have certain users to be able to store longer voice mail messages. Is this at all doable in asterisk? Thanks Bruce

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