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Hi all, is it possible to bridge a call between a Junghanns quadBRI card and a TDM400 in the same server? It should be I think, -- I am trying this and when an incoming call comes in, it hangs both up at the moment the bridge is attempted (and a subsequent 'qozap: dropped audio' error is show in the /var/log/messages) Any thoughts appreciated -- I've seen posts, but no clear

I will be on vacation from <22/06/06> to <30/06/06>. I will not be reachable on my mobile. I will have limited access to mails, and please expect a delayed response. In my absence, please contact the following: Ray Richard or Safeer Mohammed Thanks H.Gireesh

Hi We are currently using Asterisk 1.2.4 with IAX and app_meetme for conferencing, but are looking to move to SIP because of issues with an IAX control we're using. The reason we moved from SIP to IAX in the first place was because of the poor NAT traversal with SIP. At that stage we were using Asterisk 1.0.*. How does Asterisk 1.2.4 handle NAT traversal and firewalls compared to the older

I am having major problems with the first run of Snom 360s that rolled out last month. I am working with the US vendor and they in turn are working with Snom but I wanted to see of anyone else was using these or having issues with them. Issues: Speakerphone/Hands Free volume spikes up and down during a call. You have to manually set the volume during every call. This makes it totally unusable.

Hi, is the following possible? I would like to transfer a call to my "personal" MeetMe conference room and get transferred there automatically as well. Currently I can transfer the call to the conference, have to hangup and then call the conference number myself. I would love to have this in one quick function. Moreover is there a way to disable the "You are currently the only

On Saturday, April 15, 2006 3:17 PM Remco Barende wrote: > I heard that Junghanns is working on such an interconnection. It is > already possible to connect their PRI cards, and they are working on > BRI<->PRI. Correct. The next driver generation is supposed to support this fully. > I ise their bristuff for an HFC-S BRI card and am not happy at all > with the way they

Hi friends ! I want to add stun functionality in asterisk. can anybody give me some hint that how can i start that. thanks in advance Deepak Dhiman

Hi, does anyone have test numbers in Germany that would allow me to tune my rxgain/txgain settings? I know there are numbers provided by other providers in UK e.g. but have yet failed to find a number in Germany (esp. by Deutsche Telekom). Kind regards, JP

On Samstag, 14. Januar 2006 1:47 tim panton wrote: > That is weird, you would expect IAX to do better than SIP (bandwidth > wise) My point exactly. > 1) are you sure IAX trunking is actually happening ? It shows (T) in iax2 show so I am pretty sure. Timestamps are enabled as well. > 2) what codecs are you using. Are the codecs the same for IAX as > for sip? G.711 alaw and

On Friday, March 31, 2006 3:52 PM Chris Earle wrote: > I'm wondering if I should be using zapHFC with my Junghanns card > instead of qozap? Why would you want to do that? Sorry if I missed the start of the problem but qozap is what you want to use with your Junghanns card (at least if you want to stay with zaptel/libpri). > Everyone always mentions zaphfc -- mostly I >

Hi, I would like to see if during a call a new voicemail was recorded. I want to send a SMS to mobile phones if someone recorded a message on our voicemail system. I can use VMCOUNT to see if there are new messages in the Inbox but this will result in new SMS being sent even if the caller hangs up during the Voicemailpromt, at least if there are still unread/unheard messages in the inbox. Is

Hi, this is the scenario: One * is placed in a central location with more than enough up/down bandwidth. One * is placed behind a DSL 3000/384. Both * are linked via IAX trunking. Everything is fine until the upload channel of the remote site is filled with a download, then heavy voice distortion starts. Well of course this is expected. So I fooled around with HFSC QoS scheduling on the remote