similar to: SIP to IAX to SIP

Just an FYI to anyone out there who will be attending Astricon and who would like to play around with embedded Asterisk on the Netgear WGT634U platform. If you want to "bring your own" to the show, I'll be bringing all the appropriate stuff to flash them there with my latest openWGT/Asterisk build. They are available from www.justdeals.com, refurbs, for $44.95 delivered.

Hi, Does anyone have any advice on customising firmware of an SPA921 so that it can be locked to a sip provider and display logos on the config pages. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091019/f6aa2510/attachment.htm

Hello, I would like to know how Asterisk deal in this case: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk

Hello everybody, I have a very strange issue with a Linksys SPA-3000 (1 fxo + 1 fxs) used as PSTN gateway to asterisk in a small office. Everything works just fine, except that sometimes, and it seems that only for long incoming calls, the IVR menu appears on the middle of the call(like a three way call, call goes on with prompts playing over the parties). Dialing an extension at the prompt at

I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. Thanks Cory Andrews ++++++++++++++++++ VoIPSupply.com PBXSelect.com ++++++++++++++++++ 454 Sonwil Drive Buffalo, NY 14225 voice direct - 716.250.3402 fax -

Hello. I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound. What can cause that problem? Thanks in

Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/af1f9656/attachment.htm

Is it possible to have 2000 windows machines reconize Domain Users under the local Power Users group? Right now I'm using samba 3beta3. Do I need kerberos support compiled in? Thanks for your help Jason

Hello, I'm working on a call recording solution. I would like recordings to either be automatically uploaded via FTP, or posted to a URL for processing by our main server. Is Asterisk capable of doing this or will I have to create a separate application that monitors a temp directory for new recordings? I ask because I don't have any experience in Linux programming, so I

Hello, At one time I installed an alpha of samba-3.0 and it had a smbgroupedit command that mapped unix and windows groups via webmin. This seems to be missing in beta3, has it been depreciated? Also I can't seem to get the group map to take effect unless the unix group is the users primary group. Either that or I don't understand something. I do not have winbindd running or any idmap

When trying to browser users or groups on the server I see these messages in the log file. [2004/09/02 10:40:15, 0] lib/smbldap.c:smbldap_search_suffix(1101) smbldap_search_suffix: Problem during the LDAP search: (Size limit exceeded) [2004/09/02 10:40:15, 0] passdb/pdb_ldap.c:ldapsam_setsampwent(1173) ldapsam_setsampwent: LDAP search failed: Size limit exceeded We are unable to browse

I am using asterisk. I also have an access to VOIPSwitch ver 2 where I can see live calls. Many times I have seen that my calls are getting strucked and then it gets disconneected after 59 mins ( as settings are done accordingly in VOIPSwitch) What could be the reason ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello I dont want to disgrace any company but i want to know from your(user)experience which one is good in case of g.729 (performace etc) is it Howler(http://www.howlertech.com/products/howlets) OR its Digium ( http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC&main_category_id=5 ) plz note i dont want to degrade any company... But to know what experience you

Hi people, I want to make a meetme between 2 numbers. First I enter the number1 into the meetme. It is waiting for the other number, but the other number never entered, so, how can I finish the meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick all the users? Thanks, Anahi Ludue?a _________________________________________________________________ Descubre

I have a user but I need to give that user only kill and disable all connection cut calls what is the command in the CLIC -- Bayardo S?nchez Garc?a Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxy Support - Linux Server E-mail: bayardo.sanchez at gmail.com Linux User: #418392 America Central - Managua, NI (505) 2249-2853 - 84886876 IM msn messenger:

Asterisk to Asterisk voicemail not working (accessing voicemail from another asterisk). PSTN to Asterisk is working, but not between two asterisk :-( I've tried setting my asterisk dtmf to rfc2833, inband it is not working. The other Asterisk Linksys is set dtmf = auto -- Joseph

Is there a ready add-on to asterisk that will display the ACD/ASR per channel, source & destination? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/48076c6b/attachment.htm

Hi, As you may know by now, yesterday on the Astricon the City of Amsterdam presented their large scale asterisk deployment of 20000 phones. Because they do not allow brand names to be used within the city, they call it 'IP Business Manager', but the software they use is in fact the Astium PBX, by NeoNova. Since we are very proud of this project, we have made the Astium available for

Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzzzzzzzzzzzzzzzzzzzzzzzzzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? Regards Bilal

Is there a way to disable ringing while dialing? Example, external users come into our IVR, and if they dial certain IVR options, these are sent off to a remote server for call handling ( Dial(SIP/extension at remoteserver) for example). It rings once, then the remote system picks up. I would like it to be more transparent to the users.