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Did Linksys really use Asterisk for the SPA9000 software? -- Andres Technical Support http://www.telesip.net

...g g729 + Asterisk)? I would like to buy a couple for a SIP test lab but I have not found any documentation on wether it works for SIP UAs or not. The Digium page only mentions: "The G.729 codec works with all Digium cards." Can somebody tell me please? Thanks, Ricardo Villa http://www.telesip.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030727/bb7659f9/attachment.htm

...do not show "late packets". I have gone back and forth installing each version again with the same results. Bandwidth is not an issue as the ATA and * are on the same LAN. Is there something that can be fixed on the CVS version to prevent this problem? Thanks, Ricardo VIlla http://www.telesip.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030726/1de895a5/attachment.htm

...k to find them. Outgoing CNAM is a different beast however. They can't take it via IE. You need to get it into their database (they have two) and have them push it to all the other telco's. -larry > Message: 10 > Date: Mon, 10 Apr 2006 22:42:35 -0400 > From: Andres <andres@telesip.net> > Subject: Re: [Asterisk-Users] callerid name inboune from PRI > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <443B179B.3030109@telesip.net> > Content-Type: text/plain; charset=ISO-8859-1; format=fl...

...Fi router and make the call. It appears this VoIP offering is tightly integrated since you use the same phone number to make and receive calls over WiFi or Cell. Does anybody know if its SIP? I wanted to get some packet captures but he was in a hurry. -- Andres Technical Support http://www.telesip.net

DSL works by using the frequencies above 4k that were unused in POTS loops of yesterday. Load Coils, Bridge Taps, and DC taps are all devices added to lines to increase their reach and stability, unfortunately, they are DEADLY for DSL. Other problems can effect DSL service, and cause it to be 'flaky'. 1 Temperature, in Florida the large black cables are constantly beaten down by the sun,

This was asked many years ago but I thought I would check to see if things have changed. Is it possible to take over a call in progress - using a replacement Asterisk server? In other words, if 2 user agents are connected through an Asterisk PBX, and I tracked the call ID, IP of each UA (and anything else needed), could I remove the PBX and put a new one in its place (at the same IP

...t> wrote: > And even worst. > There are some kind of NAT that STUN does not work. > You can check the mailing list i think some people call it "crap nat". > Regards, > > Ricardo Martinez.- > > > -----Mensaje original----- > > De: Andres [mailto:andres@telesip.net] > > Enviado el: Martes, 05 de Julio de 2005 17:17 > > Para: Giovanni Balasso > > CC: serusers@iptel.org > > Asunto: Re: [Serusers] NAT considerations... > > > > > > Giovanni Balasso wrote: > > > > >Just some thoughts based on my experien...

Hello, I have had several instances over the last month of Asterisk freezing, sometimes after 12 hours, sometimes after 8 days. The common elements are that: - all Zap channels lock[hangups don't register and no new calls in or out] - no new in/outbound calls can be made on Zap or SIP channels - people who are still connected to calls can continue to talk - in the CLI interface, you can

I have found a site that list the following (no date in the post, so it may be old): "since all transcoding and calls still go through one core in asterisk, it doesn't make sense to buy a multi-core or hyperthreaded system that will only slow you down" Does that still applies in asterisk 1.2.14/1.4.x ? Or do we have to tweak source code to balance loads (transcoding,etc) between

Can someone share pointers to Asterisk's Dialogic support? Which boards are supported, driver status, and etc. Thnx

The subject says it all. Is it possible, code wise, configuration wise, at all - to reverse the order in which the available zap channels are used for *outgoing* calls? Code wise, I looked at the channel structure and it appears as though there is only a next pointer, not a previous pointer, so to 'easily' to this in the code would require a change to the code that reads in zapata.conf?

I found a thread [1] last month about the poor/crappy g729 quality on Sipura units. Anyone noticed an improvement or the quality is still poor? If the Sipura firmware/g729 offers no quality yet, who else is offering a dual channel g729 ATA? I heard about Uniden, but I have no "reports" about their ATA... [1] Sipura g729 call quality to PSTN

Google has given me too many responses, so I'll ask the list: Where in the Asterisk rtp source code can I find the default millisecond frame size? I've looked around for obvious pointers, but it's not clear. I'd like to "force" my Asterisk server to use a certain frame size all the time. (Of course, ideally I'd like to prefer or even force that frame size in a

...-------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/c9db8975/attachment-0001.htm > > ------------------------------ > > Message: 5 > Date: Thu, 26 Aug 2010 15:23:44 -0400 > From: Andres <andres at telesip.net> > Subject: Re: [asterisk-users] double DTMF digits > To: asterisk-users at lists.digium.com > Message-ID: <4C76BF40.20204 at telesip.net> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > On 8/26/2010 2:55 PM, M S wrote: >> Hi, >> >>...

I've just had the most appalling performance from * ever. Dialling: Cisco 7960 => asterisk => IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 => asterisk => IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be

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Symptom: Phone after about 15mins will stop functioning Problem: DHCP lease renewed but default route dropped Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is released It turn's out that these phones have a few issue in 1.0.3.81 firmware. The phone may stop transmitting packets if configured with DHCP, if DHCP is being provided by certain devices. Netopia

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will