similar to: SIP Telephone Quality/Price

Here is a SIP phone I haven't seen before. Does anyone have any experience with this one? -----Original Message----- From: George Richardson [mailto:georger@netxusa.com] Sent: Wednesday, April 02, 2003 4:56 PM To: clay@ctitec.com Subject: ipDialog Ethernet SIP Phone $199 pad <http://us.st1.yimg.com/store1.yimg.com/Img/trans_1x1.gif>

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

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

Hi! I am trying to know well asterisk. For that I would like to know the exact role for each config file. Can someone tell me what is the role of the next 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 -

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

Dear Asterisk Group. I have two Asterisk servers serving two data/help desk centers, both centers have a near identical setup. However, when connected to one of my data centers, I call a user, I can see on the CLI that the phone is ringing, but I hear no ringing on my SIP soft phone? Has anyone had a similar scenario? How as it resolved. Warm Regards Shad Mortazavi

Hi all! I am using ohphone (well, I am trying to) to make calls. I will make an H.323 - SIP Gateway but I don't understand the architecture of all this. What is the exact role of asterisk? It can be used as gateway, that I know, but what else can he do? Is it necessary to have ohphone to make calls or asterisk can also do that? So when the gateway it is going to be implemented how is it

Hi All, * is working very well for us now. But I have an issue that I cannot find the answer to - enter guru's!! When our receptionist does a blind call forward I receive the Caller ID, however I do not know if the call is fresh (i.e. ringing in) or forwarded. What I would like to do is to have * prefix the CID External (so that I can tell that it is a fresh call) or Internal (to tell me

Hi all! One easy question... I hope someone will answer me. I've installed asterisk with 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

Hi! I am in a H.323 network with a gatekeeper and some terminals. Asterisk is a gateway between this network and the SIP network. Now I can do calls from de foreign network (SIP) to the locla (H.323) but I don't know how to do the inverse. The H.323 terminals use NetMeeting, and when I try to make a call, it says that the number dialed must be registered in the gatekeeper. How can I register

Has anyone used the Swissvoice IP 10S (www.swissvoice.net) VoIP Phone with *? Adrian -- Adrian Walker adrian@digitaltraffic.co.uk ======================================================================= This email has been scanned for Virus infection by MessageLabs For more information please contact messagelabs@atomwide.com

Hi all! Please I need someone that have already done 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

When I try to make a call, I have this error: dial 06302@gatekeeper -- Executing Dial("OSS/dsp", "OH323/06302|20|tT") in new stack *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Called 06302 WARNING[1272234688]: File chan_oss.c, Line 624 (oss_read): Error reading from sound device (If you're running 'artsd' then kill it):

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Dear List, Can anyone recommend a sip phone for residential use? (asterisk home pbx) Thanks!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040819/7b107ebc/attachment.htm

Hello, I would like to buy a SIP IP phone, but I don't know wich one to choose... Can you tell me wich IP Phone is known to work with Asterisk please. I've seen the Cisco 7940, but I don't know if it works, and how expensive is it ? I'm french, so if you know some french resellers, tell me. Thanks a lot, ---------------------- Fabrice Tereszkiewicz Sawadka.org

Hi All, * is running a dream now, however we have an odd problem that I am sure some guru will be able to sort out for me in no time!! When receiving or making a call about 60 seconds or so into the call we develop choppy/stutter audio problems. It then seems to clear itself only to return again, and so the pattern carries on! This has got me stumped! Our equipment is SipTone II handsets, AVM

Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike

Anybody used the above phone with asterisk I have one working ok for calls, but having a problem with voice mail. Using either the 'Voice mail function key' or dialing 88 (for my system) just gets me to Call Terminated Asterisk CLI shows the error message 'unable to get User name' My Grandstream works ok, asking for User name, then Password Any ideas ? -- Clive Email :

I just got 2 IpDialog phones for use with my Asterisk system. I have been able to get the phones to just dial local extensions but it is not able to register with my system correctly. I would like to know if someone has set these phones up before and how they did it! Is there any examples for use with Asterisk? They seem simple enough to config with there web interface. Thanks