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FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip communication between endpoints and * does not support it, I need to change codecs when a user wa...

I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would expect the RTP session should ideally be between the two end points of the call, in my case the AS5300 and the 7940 which are connected on the same VLAN as the Asterisk server. When I sniff the packets on the VLAN I find that all RTP packets are being relayed by the Asterisk server causing increased load on the

Is there a way to reduce the number of rings if there is a message on the mailbox. That is I set the Wait app to 10 secs but then want it to pick up a call right away after someone leaves a message (ie I am not at home, office) How can i do this? Thanks in advance Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

All of a sudden I am getting the following warning "Ring/off-hook in strange state 6 on channel1" from chan_zap.c and I cannot answer calls. I can place calls out without a problem though. Any ideas what can be the problem. I have checked zapata.conf and zaptel.conf and they both seem fine. Thanks in advance. Dan -------------- next part -------------- An HTML attachment was

I am looking into supporting around 20 SIP clients (ATAs, IP softphones, etc) distributed in around 10 different end points (in South America). For the most part they all have narrow band connections 64kpbs, 128 at most and I?d like to use g729 all around (don?t have too many alternatives) To start with, I will have one * with no gateway to the PSTN and eventually a few * boxes with termination

Hi, Do the g729 codec licenses for Asterisk work on a SIP environment (only SIP UAs running 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

Folks, I cannot seem to be able to place a call from a dialup connection (this is the first time I try to do this)

I have two X100P cards that were working just fine for months. When I run modprobe wcfxo today I got the following error: ZT_CHANCONFIG failed on channel 2: No such device or address (6) /lib/modules/2.4.19/misc/wcfxo.o: post-install wcfxo failed /lib/modules/2.4.19/misc/wcfxo.o: insmod wcfxo failed The output of dmesg said that wcfxo:Out of space to write register 06 with e0. What?s

I would like to know how to calculate the amount of bandwith I would need to host X number of calls. For example, if user A in San Francisco with an ATA 186 calls user B in New York with an ATA 186 and Asterisk is being hosted in a PC in Miami. How much bandwith do I need to have in Miami? Do I just need bandwith for the setup of the call (ie the SIP part) or are there any instances where the

* is being hosted on location A and on location B, connected to A through a 64kbps ADSL connection,I have two SIP clients. Shouldn?t the RTP stream between the two endpoints in location B be direct, without going through * on location A? Are there any instances in which the RTP stream always goes through *? (codec translation, maybe?) Please advise. I would like to host a box with * for several

Is there a way in iax to have to endpoints talk to each other directly (after the call is setup by *) without going through *. In sip, with * you can do it by configuring sip.conf with canreinvite = yes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030728/0c711d05/attachment.htm

I am having problems using the manager even though I am following the instructions from the Manager.rtf doc. In manager.conf I have the following [general] enabled=yes port=5038 [fred] username=fred secret=fred read=system,call,log,verbose,command,agent write=system,call,log,verbose,command,agent I do the following: System prompt # telnet localhost 5038 Trying 127.0.0.1... Connected

I have an X100P connected to an extension of a Panasonic PBX. When a call from the PSTN comes in, it is routed directly to the extension where the x100p is . I want * to answer the call, play a message and then transfer the call to another extension via the Zap channel where the call was received (I need to flash the zap channel) . If this extension doesn't answer I want then to dial an IAX

I cannot get Festival to work with asterisk. I have the following: exten => 555,1,Answer exten => 555,2,Festival(mary has a little lamb) exten => 555,3,Hangup I get the following from asterisk: "Festival returned ER" and the festival logs shows the following: client(1) Fri Jul 16 15:35:54 2004 : disconnected client(2) Fri Jul 16 15:40:26 2004 : accepted from localhost

I have 4 sipuras 3000 in a small callcenter connected to the PSTN receiving calls and forwarding them to Asterisk and viceversa. In addtiion I have some x100s, linksys FXSs, etc Strange things are happening with the Sipura and Asterisk which I cannot seem to figure out. During off hours at the callcenter, when no one is placing calls, if I place or receive a call with any of the Sipura,

Folks! I discovered some serious problem with several Sipuras 3000 but I don't know if the problem is with them or Asterisk. Basically, if I call a Sipura PSTN line, when there is a call already in progress, generally I get a 503 Sevice Unavailable, but if I try hard enough, I am able to get through and connect to dialed number. The other call gets disconnected but the originator of the

I have an X100P and when I place calls to the PSTN which are not answered, the Billsec field of the CDR still logs the seconds that the phone rang. Can someone please confirm that this has to do with the ringcadance of the indications.conf file? Is there anything else I need to check ? Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL:

I just updated to the new CVS and now I am getting the following error from chan_zap (modprobe wcfxo works fine): WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable to specify channel 1: Device or resource busy ERROR[16384]: File chan_zap.c, Line 4781 (mkintf): Unable to open channel 1: Device or resource busy here = 0, tmp->channel = 0, channel = 1 ERROR[16384]: File

I would like to run an agi script (to calculate the cost of a long distance or international call) right after I execute a Dial app. Can this be configured in extensions.conf? It seems the entries right after a Dial app get executed only if the Dial app was executed unsucessfully. Would I have to execute the dial app from the agi script? Thanks in advance. Dan -------------- next part

I am having some problems with g729 with SIP and ZAP channels. 1) I have two g729 licences. Very frequetnly (I don?t know what triggers the error) I get the following warnings and error when I try to place a call via SIP to my X100P. The only way to get out of this is through a restart of *. When the error ocurrs there are no other calls in place. Any ideas? Error Opening channel:2 not