similar to: Making digital/data calls through asterisk

Greetings to All, I hope someone has already gotten this working. I spent all day today trying to get ooh323 and gnugk to run on the same box. After a lot of tweaking to get everything compiled, I got both up and running. I can make calls IAX to H323, but cannot make calls in the reverse direction. I have tried many different configs on the GK, but always come up with the same error. It appears

Hi, Some time ago, there was a discussion about the inability of nortel meridian pbx to dial analog tones thru an meridian ATA, and the work arround was to enable 808 function that makes the dtmf tones long for the current call. The nortel meridian is connected via a nortel ATA to a TDM400 to a FXO port. Anyone can say me who to actually use that function (you dial something or is pbx

Did anybody know, Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? Thanks a lot for help. Nico Giefing -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060328/3b657058/attachment.htm

Hello again. I'm stll struggling trying to terminate calls from SIP through Asterisk and throught my H323 gateways... Basically the call is accepted by GnuGK but then dropped with *reason = unreachableDestination <<null>>* I did a *debug trc 10* on GnuGK and looked at the sessions... one from X-Lite through Asterisk... and one from OpenPhone... The one from OpenPhone works

Hi All, i'm going crazy trying to make asterisk work with the following hardware: 02:05.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) Subsystem: Cologne Chip Designs GmbH ISDN Board Flags: bus master, medium devsel, latency 16, IRQ 17 I/O ports at d000 [disabled] [size=8] Memory at ec000000 (32-bit, non-prefetchable)

Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware ?as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands

Well, I don't know how to tune it more, it connects at about that rate in a mediocre rural landline. ILBC uses samples of 30ms, so if you set the trunkfreq set to 20 you will be using more of the necesary scarce bandwidth AND dropping sample info in each frame, thus making audio choppy and unclear. Make shure to disallow all codecs and then allow only ILBC or lpc10 (search for it in

I'm new to VOIP. We are thinking of setting up a VOIP system between a couple remote offices. I've been lurking on this group for a while. What is the consensus on these phones: http://www.netvoice.ca/grandstream/budgetone101.htm I'm confused about the SIP protocol... can a SIP phone be located behind a NATing firewall ? When people use asterisk on a broadband connection used

After searching the archives, I came acrross a few people mentioning this, but I never saw anything about what became of it. Has anyone tried to make a virtual modem that could be directly handled by astrisk, I saw a while ago that someone was going to try and make one using the same DSP libaries that the WinModems use, but then nothing. Would do this even be possible, and if so, what kind of

Okay, here is the background. I have a PRI with 15 active channels on it. I originally setup all of them in group=1 and all outgoing and incoming calls used this group. The phone number that I have associated with these channels ends with 750 and that is how I direct the calls. i.e. In my extensions.conf I have: exten => 750,1,Dial(SIP/120,20) All this works fine. Now I have the need

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can

Hi all, I would like to call from SIP client to Asterisk then GnuGk, then Cisco 5300 to PSTN phone. Is this possible? I need simple config asterisk and gnugk.Can somebody help me? Ganbaa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040629/cfe09e1e/attachment.htm

Hello! I'm looking for TEXT sending over IAX2, more precisely sending text messages between IAX2 softphone and asterisk. Currently I'm trying iaxComm and I was able to send a text message from asterisk to iaxComm (with AGI SEND TEXT). After some searching I got some questions: 1. is it possible to send text messages during a call (established or during ringing)? 2. is it possible to send

class Trunk < ActiveRecord::Base has_many :call_type_trunks has_many :call_types, :through => :call_type_trunks end class CallType < ActiveRecord::Base has_many :call_type_trunks has_many :trunks, :through => :call_type_trunks end class CallTypeTrunk < ActiveRecord::Base belongs_to :call_type belongs_to :trunk end The associaton class has a column named price.

*spews coffee over keyboard* - FUNNIEST - THREAD - EVER - Also one of the most insightful. Teddy, your gmail invite is on the way.

*** If anyone has a better way of doing this, please post to the list. I hadn't seen anything on this list or in channel.c/chan_local.c - which prompted this email *** I'm not sure how many VoIP providers out there are using Asterisk as a service platform like we do, but I thought I'd share an experience with call looping that was racking my brain with the list. One of the

Hi- I have an upcoming application that requires use of PRI channels that are primarily used for high-volume incoming traffic, but that are to be used for outbound calling as well. Of course, one option is to have dedicated outbound channels reserved, but this is an inefficient use of channel resources. Normally PBX's are designed to have the CPE yield to an incoming call if a particular

Hi, I have built another Asterisk box using one ISDN HFC-S card and Bristuff-0.2.0-RC8p. But this time it behaves very strangely. Asterisk simply hangs and in logs I receive something like this: --NOTICE [1197]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 --NOTICE [1197]: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 And when I want to call a ZAP channel I get

Hi All I am working on setting up a * system to replace our current voicemail box. I may also end up using it for a few Voip calls. Anyway, I have heard some people complaining about the new Digium Fxo cards and having problems with them. I do not yet have the computer so if certain issues are caused by other hardware I could work around them. Does anyone trust these cards enough to use them?

I've read several other emails and pages on the wiki but none give any deffinate answers. if you have 20 asterisk servers each with 4 pri's, all running RealTime Extensions and RealTime SIPBuddies from the same MySQL server, what prevents you from putting all 20 servers behind a single load balancer? That way all of your UA's can use the same IP to register to; vs maintaining which