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Hi, Can anyone tell me what should be included in h323.conf to get asterisk to talk to a Cisco 2600 gateway? Any statement examples for extensions.conf would also be appreciated. Thanks. Will chan_h323 use the Cisco as a gateway anyway? Regards, Steven Thomas

...Vonage uses for their VoIP technologies. I stumbled across this article (although it's from 2002, I think) that suggests strongly that they use Cisco. There is no telling what they might use in conjunction with this but this should clear some of the conjecture. http://www.cisco.com/en/US/tech/tk652/tk701/technologies_case_study09186a008 00b559e.shtml Curt

...fails the download, and keeps repeatedly trying. This scenario is covered in the Cisco FAQ for converting a CallManager 7960 to SIP. It essentially is a bug in the firmware on the phone which requires upgrading to an intermediary, older SIP firmware first. URL: http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a 0080094584.shtml I don't have this, and my reseller doesn't have it handy either, though they promise to get it (some day?). At any rate, I've opened a Cisco tech support incident with hopes that they'll be able to provide me the files quick an...

Hi, I can't upgrade an old 7960 from SCCP 3.0 to SIP 7.5. Could you help ? From http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2, I got the following: 1. Copy the desired binary image from Cisco.com to the root directory of the TFTP server. 2. Specify the image in the configuration file image parameter for the protocol to which you are conver...

...9;s FXO to one of the SIP phones, there is no echo problem. Sometimes when we dial from SIP --> Voip provider --> PSTN --> destination it is okay, but other times the echo is horrible. In trying to figure this out, I found this article at Cisco's site: http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a00800d6b68.shtml#1041385 It claims that echo always comes from the far end of the connection. So if I hear echo, then the origin of the echo is in the equipment on the end of the line near the person to whom I'm talking. The description seems to make sense...

According to CarrierAccess, the Adit 600 uses CAS for voice signalling. What is this? This should not be a problem for Asterisk? Does the Asterisk server need to reencode CAS into aLaw when going to Euro ISDN? BR Daniel Nystr?m

...> I poured over my logs most of the morning. I'm fairly convinced at > this point the disconnect is coming from the Norstar 10 seconds after > the call was initiated. This points to the 'T310' timer, similar to > what is described here: > http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_ > note09186a0080094487.shtml > for the Cisco CallManager. > When comparing the log from a successful call to a failed call, I > noticed Asterisk was not passing back call-progress from the PSTN-span > to the PBX-span. http://bugs.digium.com/view.php?id=4468...

Is there a guy somewhere on how much bandwidth each codec uses, along with the advantages and disadvantages of each one? Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050923/3bee2776/attachment.htm

There's an excellent tutorial on Cisco's web page at http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a00800d6b68.shtml It will tell you just about everything you wanted to know about echo and more :) The short answer to your question, however, is that echo is comprised of two components: volume and delay. Increase either one and the problem gets worse. In th...

Hi! Can anybody help me with convert Cisco 7940 CallManager Phone to a SIP Phone? I have continious error in tftp log: connect from 192.168.1.111 Mar 18 12:12:30 AKrasavin utftpd[10081]: peer requests OS79XX.TXT, conversion octet Mar 18 12:12:30 AKrasavin utftpd[10081]: unterminated option value in init packet Mar 18 15:12:30 AKrasavin xinetd[10068]: START: tftp pid=10080 from=192.168.1.111 Mar

Is there anyone that could explain to me the phenomenon of Echo or at least point me where I can learn more? Why is this affecting the VoIP world so much and not the regular PSTN analog world? What does the PSTN industry have that they can handle such high volume of calls and there is "no" echo problem? Thanks, Daniel

...; ----- Original Message ----- > From: "Brian West" <brian@bkw.org> > To: <asterisk-users@lists.digium.com> > Sent: Wednesday, September 24, 2003 2:25 PM > Subject: RE: [Asterisk-Users] Cisco 2600 and ASTERISK > > >> > http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080093f62.shtml >> >> That covers the thridparty h323 stuff with * >> >> bkw >> >> On Wed, 24 Sep 2003, Sean Figgins wrote: >> >> > >> > That is about what I have been seing for help. Has anyone any clue...

I am an Asterisk newby, and I cannot seem to get Caller ID information from our T1 line. When calls appear at the phones, they say the call came from "asterisk" and unknown number. I know how Caller ID information is passed on an analog phone line (between the rings) but with a T1 line, I don't know technically how it is done. I don't see the caller's number in the

Hello all. This is my very first message to the list. I have a TDM400P card, It has 2 FXO channels which are connected to extensions of my PBX (Ericsson BP250), so I can dial from any SIP softphone directly to physical (analog and digital) extensions on my company. My PBX is configured so when I dial 8 on any extension, it will redirect to the first free FXO channel on my TDM400P card.

Does anyone know the default E&M Wink timings for Nortel DID ports? The default settings on Asterisk are: ; prewink: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ;

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

I suppose this is rather an informative e-mail than a question. However if people had similar experiences or could comment what the differences are in other countries or with business analog lines, it would be interesting. It took me a week until a BT engineer was sent to my home home, since BT tech support was unable to provide information about the problem. Problem: Calling party controls how