Re: [Asterisk-Users] Re: Codec Problem
Have you paid the $30k+ (US dollars) that it costs just to put the G723 codec on your Asterisk server? Due to cost issues, Asterisk does not come the G723.1 codec. Simularly, it does not come with the G729 codec, but you can download and license that one for a very reasonable $10/connection (a one-time cost)... much more reasonable than $30k just to have the codec + some amount for each and every connection. Asterisk can pass G723 calls off to other devices that handle G723, but your implementation is trying to transcode from G723 to some other format - and that just isn't going to happen unless you have the codec loaded. Good Luck! On Thu, 2005-12-01 at 12:29, Code Lover wrote: > Hi, > > My IP Phone is using well G.723.1 because when i am testing it with > another SIP GK, working well with G.723.1. > > But the problem is only accuring in Asterisk, my sip.conf is already > having the configuration of this codec. > > [123456] > > > disallow=all > allow=g723 > > > > > -- > Thank You, > Code Lover > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Two Phones - Same extension?
Lots of options here. You could turn her DID into a hunt group that rings both phones. You could write a small webapp that lets her toggle which phone to use (the web app would edit the extensions.conf file and then reload the configs in Asterisk)... or the easiest way is to simply let her take the phone home and plug it into her home network. If you don't allow public access then simply get a Linksys firewall/router with VPN access and set that up for her so that it creates a local VPN between her home and your office. It works fairly well. Good Luck! On Thu, 2005-12-01 at 17:14, Mike McMullen wrote: > Hi All, > > I have an employee who works mostly in our office but > maybe once or twice a week has to work from home to help > care for her special needs child. > > As background we have AAH 2.0 running with 8 analog lines > connected to two digium t400P cards. We have 10 sipura-841s > as handsets in the office. > > I would like the employee to be able to make and take calls > from her house when the she has to work from home. I'm leaning > towards just installing s/w on her laptop with a headset for that > setup. > > My question is how to handle setting her up so that she only has one > extension shared between the office phone and her laptop. For this > to work, do I need to unplug her phone from power/network in > the office when she is at home or, hopefully, is there some other > magic that can happen? > > TIA, > > Mike > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec Problem
On Fri, 2005-12-02 at 05:59, Code Lover wrote: > Hi, > > Do you know from where i can buy g723 codec. for g729 i can buy it > from digium.com. But Please let me know from where i can get g723 > codec. > > And the codecs purchasing can solved my problem? > > > -- > Thank You, > Code Lover Become the fisherman... http://www.voip-info.org/wiki/index.php?page=Asterisk+G.723.1+Licensing ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 problems
On Fri, 2005-12-02 at 14:22, [EMAIL PROTECTED] wrote: > Help! I've encountered some problems with Asterisk that Im unable to solve. > We have been running Asterisk version 1.0.9 for many months using a few local > network connected Cisco 7960 phones as SIP clients. All our phones are > currently internal so there is no NAT involved. We were not having any > problems until last week when some strange issues started to crop up. I > started experiencing calls that I initially believed were being dropped, but > discovered that only one side of the conversation had dropped. The other > party could hear me but I couldn't hear them. This seems to happen more often > on longer calls but is not consistent. I am also seeing issues where > incoming or local extension calls that are hung up by the originator before > being answered will continue to ring the SIP phone. At the time the errors > occur, the Asterisk console displays a variety of "...retrans_pkt: Maximum > retries exceeded on call.." messages. I scoured the forums for an answer, > found many refere nce > s to these errors, tried every suggested fix that I could find, but none > have resolved these problems. After working on the problem for several days, > I finally built a new box and installed Asterisk 1.2 on it. Using this new > 1.2 box I no longer see the "Maximum retries exceeded on call" warnings on > the console but still experience the strange behavior. Unfortunately, the > errors occur randomly so I am unable to reproduce the error on demand. I > turned on SIP debugging and set console logging to debug and captured an > instance of the problem with the hang up not being recognized. The details > are below: > > I dial in from my cell phone. My Cisco phone begins to ring. I then hang up > my cell phone. Asterisk acknowledges the hang up, but the Cisco phone > continues to ring. After a minute or so, or if I pickup the phone, Asterisk > display the following message "That's odd... Got a response on a call we > dont know about. Cseq 102 Cmd SIP/2.0" I've included a copy of the console > output when this occurs that shows both the SIP message and the Asterisk > debug output. Odds are you have local network congestion -- Dropped packets or delayed packets. Try moving your phone and asterisk server to an isolated network switch - no other traffic (certainly no computers) - then test. If the problems go away, then update your virus scanners and check your computers. Good Luck Jon Carnes ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Context confict question??
On Fri, 2005-12-02 at 16:07, Chuck Bunn wrote: > Hi, > > If I have an extension in a context and I have another context with the > same extension and I include the second context in the first does this > cause a conflict or does Asterisk know that there is a 600 extension in > each context > > [big-business] > exten => 600,1,Dial(ZAP/1,20) > include => small-business > > [small-business] > exten => 600,1,Dial(ZAP/2,15) > > Thanks It's never caused any problems for me. Enjoy yourself. Jon Carnes ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 problems
On Fri, 2005-12-02 at 16:32, [EMAIL PROTECTED] wrote: > >> After working on the problem for > >> several days, I finally built a new box and installed Asterisk 1.2 on > >> it. Using this new 1.2 box I no longer see the "Maximum retries > >> exceeded on call" warnings on the console but still experience the > >> strange behavior. Unfortunately, the errors occur randomly so I am > >> unable to reproduce the error on demand. I turned on SIP debugging and > >> set console logging to debug and captured an instance of the problem > >> with the hang up not being recognized. The details are below: > >> > >> I dial in from my cell phone. My Cisco phone begins to ring. I then > >> hang up my cell phone. Asterisk acknowledges the hang up, but the Cisco > >> phone continues to ring. After a minute or so, or if I pickup the > >> phone, Asterisk display the following message "That's odd... Got a > >> response on a call we donÂt know about. Cseq 102 Cmd SIP/2.0" I've > >> included a copy of the console output when this occurs that shows both > >> the SIP message and the Asterisk debug output. > > > > Odds are you have local network congestion -- Dropped packets or delayed > > packets. Try moving your phone and asterisk server to an isolated network > > switch - no other traffic (certainly no computers) - then test. > > > > If the problems go away, then update your virus scanners and check your > > computers. > > > > Good Luck > > > > Jon Carnes > > > > Thanks for the feedback. We will definitely try this and let you know how it > goes. Do you know if there is a way for us to observe the loss of packets? > Because of the sporadic nature of the problem, we have thought several times > that the problem was solved only to have it crop back up hours later. If we > ran Ethereal on the pbx, would we be able to see that packets were being > dropped? > > Thanks, > Tim I do run Ethereal on mine when looking for real-time problems. It's great for helping you see what is going on at the packet level, but it is the wrong tool for measuring Latency/QOS problems. Shadow Ping works fairly well for measuring latencies. In earlier times I used to just run a quasi-flood ping to the offending phone (10 pings/second) and look for latency variations and dropped packets. On a clean network with no problems there should be NO dropped packets, and latency variations should be minimal. You'll find some interesting problems that accompany the use of cheap unmanaged switches (and please don't tell me you are trying to use hubs!). For our setups we use either Cisco 2900XL-EN or Cisco 3500 series switches. This come with built in VoIP detection at the port level *and* allow us to use VLANs to separate out Voice and Data. They are champion workhorses and using them lets us also run single wire to the desktop (running the PC off the pass-thru switch on the back of the phone). Good Luck, Jon Carnes FeatureTel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] route call based on codec? (g723 gets message, g729 goes to conf connection)
I have a rather curious integration problem. I need to direct a call connection based on the codec used for the connection. If my softswitch attaches to the Asterisk server using G729 I toss the connection into a requested conference - that works fine. On occasion my softswitch will attach to the Asterisk server using G723 (and request joining a conference that is using G729). When that happens I need to feed the connection a stock announcement (recorded in G723) and then hang up. Is there a way to direct a call based on the codec used to attach to the Asterisk server? More detail for those scratching their heads... I'm using Asterisk servers to augment my Vocal Data softswitch. One of the many things that Asterisk does for me is act as a conference bridge. This works just dandy except that my softswitch uses the conference bridge to transcode Voicemail announcements. My Softswitch automagically transcodes all announcements into G711, G723, and G729. Whenever someone records a voicemail announcement the VM server opens a conference using each of the codecs - plays the announcement in G729 (our default) and then records on the other connections. Obviously the G723 connection does not work since Asterisk won't transcode G723. That's cool. We don't *ever* use G723 - it's just built into the softswitch. The problem comes with the fact that the softswitch won't give up on doing the transcoding to G723. It continues to try and try and try and try... There is nothing dumber than a machine doing a task it can never finish. Unless its a machine opening hundereds of connections to my conferencing bridge trying to do a task it can never complete. I need to feed it something - anything - in a G723 format. I've got plenty of G723 audio files. If I can simply play one to the g723 connection then it will be happy and go away. ;-) Any help is appreciated. Thanks - Jon Carnes ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
