search for: mireia

Hi! I am doing a research about the prices of SIP telephones. If someone can tell me which one are the cheapest and have an acceptable quality... it will be very kind. Best Regards, Mireia

Hi! When I try to make a call with ohphone, that is the message I get: Call to "06302" aborted, insufficient bandwidth Can anybody tell me a solution or a reason why this messages appears? Thanks a lot! Regards, Mireia

...one to make calls or asterisk can also do that? So when the gateway it is going to be implemented how is it going to work? - ohphone call gatekeeper and uses a gateway (asterisk) or - asterisk registers to a gatekeeper and it accepts and makes calls? or - None of both? Please help me. Regards, Mireia

Hi! I have been looking for a while for informatoin about how QoS is assured in Asterisk, but I haven't found a thing. Can someone give me some tips about that? Thanks, Best regards, Mireia

...ones or a web where I could find this information? That will be very helpful. - alsa.conf - enum.conf - modem.conf - modules.conf - oss.conf: what is oss? - parking.conf: what is parking? - rpt.conf: what is radio repeter? - queues.conf - skinny.conf - voip.conf Thanks a lot for all the aswers. Mireia

...ich to program my phone. 4.3.153.50 Load this into your phone's tftp area and reboot it. It'll go out to the net and check the firmware revision and change it if required. I've done this with 5 of my phones and 2 of my ATA's. Good luck. Mark On Fri, 2 Apr 2004 15:32:41 +0200, Mireia Munoz de jesus wrote: > > Hi, > > My gateway accepts G.711, but not my Grandstream 100 > series SIP phone, but I > thought that thanks to CapabilitySet process they will > agree to talk with any > other codec, so what is so important G.711 codec? > > Best Regards,...

...th the samples. Somewhere in my network I have an H.323 Gatekeeper. What must I do to make that the gatekeeper talk with Asterisk? And I another little question... with the samples installed asterisk works ok? What must I install to see how it works? I am lost!!!!!!!!!! Please help me! See you. Mireia

...registered in the gatekeeper. How can I register a SIP number in a H.323 gatekeeper? I know that with NetMeeting I can make calls peer-to-peer dialing an IP... but if the softtelephone of the other terminal is an SIP UA that is not going to work, is it? Please any help will be welcome. Regards, Mireia

Hi all! I am looking for some free software to monitoring all the calls that are being done in my network. Which telephone are connected, how long are the calls, quality of service, bandwidht,etc. If someone knows about a good one, plesea tell me. Regards, Mireia

...one an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia

...Connection ip$localhost/7530 terminated. WrapH323Connection::WrapH323Connection: WrapH323Connection deleted. ClearCallThread::ClearCallThread: Object deleted. Can Someone help me? In extension.conf I have written : exten => 06302,1,Dial,OH323/06302|20|tT Thanks for all your help. Regards, Mireia

...talled everything, asterisk, pwlib,openh323, chan_oh323. And now? I want to install ophone to talk, but I don't see what is the asterisk role. I mean, ophone lets us to talk with another phone,... why do we need asterisk? What does ophone do and what dows asterisk do? Thanks for all your help Mireia

...ad): Error reading from sound device (If you're running 'artsd' then kill it): Resource temporarily unavailable" ? I am using a VIA Souncard Model: VT82C686 AC97 Audio Controller. When I make lsof /dev/dsp the only process that is using it is asterisk. Please help me.. Regards, Mireia ___________________________________________________ Yahoo! Messenger - Nueva versi?n GRATIS Super Webcam, voz, caritas animadas, y m?s... http://messenger.yahoo.es

...so the Gatekeeper Asterisk. Asterisk then looks for if the user called is a SIP user or not. Is that possible to use Asterisk as SIP proxy and a SIP/H.323 gateway? In my network I have already a gatekeeper. If it is possible... how can I do that with asterisk? Thanks for all your help Regards, Mireia

...e's the variable CONSOLE who has Console/dsp value and it says that this is for demos. What are the other two types of Console (Zap/1 and Phone/phone0)? I have to change it to make asterisk operational? - How are prefixes used? What are they made for? Thanks for all your help. Best regards, Mireia

...ere are no so many phones connected, about four, but I would like to know if when the SIP network would be bigger, if there will be problems with the dimensions. 100 or more phones connected to the same Gateway and all these extensions will be passed to the Gatekeeper. Thanks a lot for your help. Mireia PS: If I haven't explain correctly or I haven't given all the information, please ask.

OK. So it would appear that my quest for FXO adapters unconvers more, and certainly more mature, H.323 based devices...not so many SIP devices. What would be the benefits of SIP over H.323 for a small office * server? All I need to do is bring 4 POTS lines into * with Caller ID, make outgoing local calls reliably without undo echo. FWIW, my * server is Fedora Core 1 on AMD XP2500 with 512 MB

hello all, I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record. Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault. the original version of cdr_mysql.so works fine but I need the start time and end

I have a test SIP account set up with WorldCom and I have been trying to have Asterisk register to the WorldCom server with no luck. It appears that the SIP headers are different coming from Asterisk. I have included a packet capture from a successful login with a Windows Messenger client for reference. I have also copied in the SIP packet I captured with sip debug turned on. In my sip.conf file,