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Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

...e to the newer astshape script. It classifies traffic using iptables, which is much more flexible. It also has beta support for the HFSC qdisc: http://astlinux.svn.sourceforge.net/viewvc/astlinux/trunk/package/iproute2/a stshape -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/ast...

...age options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I thought I'd get my order in early. Of course one of my first tasks will be to get Asterisk running on it... ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

Hi, I'm new to this list I'm developing asterisk application where users can call and control volume up and down in music player. Problem I'm getting is if users press 222228 in fast speed, system will process all those 2s and then process 8, so there is few seconds ( around 4-5) processing key press 8 , therefore users will feel unresponsiveness in system.(in other words users will

Happy New Year in advance by a few ticks for the northern hemisphere. Here's the first topic and guest for 2009: In any voice path there are several potential sources of quality problems, ranging from echo to voice dropouts and everything in between. With VoIP systems the potential for quality problems increases dramatically, often times making it very difficult to identify the source of

I'm trying to figure out how to handle our fax line when we switch to our asterisk for voice. After a lot of reading and poking about I have concluded, as have many others it would seem, that the best thing to do is either to have a separate pstn fax line or use some sort of internet faxing service rather than try and make faxing work in a way it's not meant to over voip lines.

I am running Asterisk and would like to add SER to register my (sip) DID and connect it to asterisk; but I'm not sure if this is the correct forum. I have as DID, sip account with one VoIP provider; currently I"m using just stand alone SIP phone and register with the VoIP provider via: stun.fwdnet.net Is it possible to use SER to register with the provider and forward the call

...option to disable CDP in the setup menu on the phone. Because CDP discovery is the first thing these phones do, there isn't a way (at least not one that's practical) to disable it in a config file.* * Classic chicken or the egg... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

...t; use G711ULAW w/ INBAND DTMF to get around the issue. Looks like an issue on > the SONUS side. > > Anyone else have this issue? > Welcome to the club! ;) I'll be blogging about this later today. Look out for that post... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

...Cisco admin nor myself can find any documentation on how to disable this feature (3pcc). Does anyone happen to know how to disable 3pcc on Cisco Unified Communications Manager 7.0? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

We're creating a SIP gateway for a client that will take one leg of a call in via SIP, and out the other side via H.323. To minimize load on the gateway, we would like to have the RTP stream bypass the gatewayy altogether (directrtp/reinvite). Is this possible with these to protocols? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I want to be able to bridge two sip channels using direct RTP between my endpoints (Audio IP : not local) but without using reinvites. So I set up my asterisk sip endpoints as follows: [test1] type=friend host=dynamic username=test1 dtmfmode=info context=test_rtp allow=all canreinvite=no directrtpsetup=yes [test2] type=friend host=dynamic username=test2 dtmfmode=info context=test_rtp

...em, don't you think? ;) While I don't have any better alternative responses I'm just bothered by the "global" nature of 6xx failures in the first place. Any thoughts? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented with GSM sound files. The problem is I have IP phones Utopix HyperPhone 202 which support only G.729a/u and G.723.1 high/low, but not GSM. If I choose G.729A the "pass-throu" calls among users are OK, but Asterisk can't transcode GSM to G.729A in voicemail calls. My company doesn'y want to pay for a G.729

Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. -- Your

After getting some ERRORS like this: [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports

Hello, all. I've come across a nasty problem just as we are ready to put our first system into production. During our final testing, we were plagued with several "invalid extension" or "password incorrect" messages when we knew the information entered was correct. Upon investigation, we saw that DTMF signals were occasionally but not consistently duplicated. We might

Hi, Would like to run the software to monitor the quality of the bandwidth. Suggestions welcome? Thank you. Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081211/85bd0069/attachment.htm

Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500

Hi I am trying to connect asterisk with a Cisco GW 2600 with E1 pri using a Digium, Inc. Wildcard TE210P dual-span T1/E1/J1 card 3.3V (rev 02), Errors: [Jan 28 17:32:33] VERBOSE[6182] logger.c: == Primary D-Channel on span 1 up [Jan 28 17:32:33] WARNING[6182] chan_dahdi.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jan 28 17:32:34] NOTICE[6182]