similar to: GSM codec

First of all sorry for the little offtopic post :-) For one of our customers i need to make a vpn between the Netherlands and Hungary. Over this vpn two * machines are gonna talk IAX and employees in Hungary are gonna use the Exchange server located in the Netherlands. So far no problem... The problem: I'm from in the Netherlands and don't understand the .hu websites :-( First question:

I have a Cisco 2600 which understands the "gsmfr" codec, which appears to be what Asterisk calls "gsm" -- at least it ends up using it. I also have a PSTN gateway which is speaking ulaw. When the 2600 calls through Asterisk to the PSTN, it negotiates the g711ulaw codec, but when the PSTN calls through Asterisk to the 2600, it seems that Asterisk is doing translation, and it

I am using Cucumber 0.2.3 and am having problems running a single feature. In particular, the cucumber Textmate bundle was not working so I traced it back and discovered that I could not run single files or features from the command line either. My setup has the following line in cucumber.yml default: -r features/support/env.rb -r features/support/plain.rb -r features/steps

Is there anyone who is currently using Asterisk as a production H323 gateway? And using which combination of asterisk and H323 (chan_h323, chan_oh323?) The main issue is interoperability with other H323 parties (Cisco AS53xx, Nextone, etc). Searching the mailing list it seems that both h323 and oh323 are not so stable, is it only an impression or using h323 is really not so advisable?

Dear all, Did anyone successfully test T38 fax pass thru to Cisco as53xx? We've tried 1.2.4 with latest patch and latest svn trunk and T38 patch but still not work. Reinvites from Cisco are correctly passed back to the originating gateway, but fax never able to connect. Cisco IOS 12.3.x configuration voice service voip fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback

Hello, everyone. Right now, we are trying launch our own PBX system based on Asterisk(Fedora) with Cisco 2611. Cisco has 2 port FXO card installed on it. For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx fine. (I'll call it F). Using softphone, I can dial in extension 1001 on asterisk, which should talk to cisco. After initial connection to Asterisk, I have try to

Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk

Hello, I'm trying to receive faxes with asterisk. My configuration is like this: PSTN fax -> ISDN -> Cisco router with VoIP module -> Asterisk When I try to send a fax from PSTN fax I got the standard fax signal, Asterisk starts rxfax application and then call ends and there is no tif anywhere. On the fax display there is still one message: Calling... Part of my extensions.conf:

Hi all I am looking at implementing asterisk at a company with two ISDN bricks (60 lines). I know that the VoIP will absorb at least on brick worth of lines but that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS ports and I've read on this list that at most two could coincide in a box simultaneously without causing an interupt

Aguardo-os amanhã. Confirmem presença aqui, por favor. ---------- Forwarded message ---------- From: Google Calendar <calendar-notification-hpIqsD4AKlfQT0dZR+AlfA@public.gmane.org> Date: 2012/10/9 Subject: Reminder: StartonRails na [:koshtech] @ Thu 2012-10-11 19:00 - 22:00 To: Fernando Kosh <fernando.kosh-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> more details

Hello, I have a cisco ATA 188 registering both of its lines to * I can place calls between then an to kphone an MSN messenger (both registering with * too), a few days ago a friend lend me a Cisco IAD 2430 and I was willing to do the same thing with it, since it has 24 ports I was willing to to use 24 analog phones with it however something really weird happens I can place calls from my ata,

I am thinking about setting up an asterisk PBX system for my company. But since I can't be at all the locations all the time I am setting up an automatic backup system where if the backup detects that the primay is down it takes over the IP so calls can be made once more. For this reason I want to setup a seperate HARDWARE PSTN Gateway. Are there any equiptment that can be plugged into

Does anyone have a config they'd like to share w/ the above hardware doing termination for asterisk? I've got one coming in tomorrow along w/ some DSP's and would like to not have to create the config from scratch to start testing. W. Kevin Hunt

Greetings all Grandstream released a new firmware and it seems like the speaker phone problem has been fixed. However we updated to firmware 1.0.1.12<http://1.0.1.12>to fix the echo problem but found other problems were now created. The worst of these new problems is that the whole phone starts degrading, the volume starts getting lower and lower. The ringing starts fading and the calls

I repost a message I put a week ago: I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? Also I recently

Has anyone used the Voip Billing System from http://advancedvoip.com/ ? They seem to also offer a billing solution for Interconnections. I'm curious if anyone has some experience using their software? Thanks, - Darren

Hi list, I'm evaluating a private telephony scenario of about 20 locations - 300 phones, 50 FAX machines. Initial overview points to the installation of asterisk at three locations connected to the PSTN via ISDN PRI. All other locations, small by themselves, would get SIP phones managed by asterisk, since there is good IP connectivity between all sites. Now on to the

Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A)----> Cisco AS5xxx ----> ASterisk---->PSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains

Greetings, I am PRIs connected to a Cisco 36xx gateway, which in turn connects to Asterisk via SIP. The problem I am having is that DTMF tones originated on the PSTN side are not heard on the SIP device. On the other hand, tones originating on the PSTN side are received by Asterisk when talking to voicemail or an autoattendant. >From the Cisco debug, I can see the Cisco sending NTE (RFC2833)

I'd really apreciate if someone could check the link supplied to see whether my Icecast server streams correctly, especially to Windows Media Player (it doesn't work for me). Thanks! Miguel -----Mensaje original----- De: icecast-bounces@xiph.org [mailto:icecast-bounces@xiph.org]En nombre de Zambra Enviado el: sabado, 30 de octubre de 2004 14:16 Para: icecast@xiph.org Asunto: [Icecast]