similar to: CISCO 2600 - VWIC 1MFT-E1

Anybody on the list who implemented Cisco ATA + * + Cisco 2600? I cannot get my calls from ATA to terminate to the Cisco gateway via *. I am not sure if it is my hardware problem. I'm getting the following "codec negotiation problem" from Cisco. 23:39:08: Unexpected VoIPCodec Type :g729br8 23:39:08: Unexpected VoIPCodec Type :gsmefr I appreciate any help I can get. Thanks.

Hi list, I'm trying, as usual, to set up voicemail. It works, but signaling to phones, doesn't. Into XLite logs, I have: -- Messages-Waiting: yes Message-Account: sip:voicemail@mydomain.com Voice-Message: 1/0 (0/0) -- but nothing appear on the XLite screen. So, I understand that I'm able to send the right signal, but something is still wrong. Ideas? Thanks in advance -- .:FaberK:.

Hi folks, we use meetme application with pin so when a customer joins he's prompted for his name. Then the voice say:"press one to accept the recording..." My question is, is it possible to cut off that request to"press one"? Thanks to all -- .:FaberK:.

I've done some reviewing of the archives for G729 and H323 experiences. The landscape of that query isn't pretty - lots of pleas for help, and nor do I see too many "answers." I have a pending bid that requires some data before I can implement * on this particular solution. My question is perhaps a slightly differently worded one than has been asked before, but it may be

Hello ¿How could I add more disk space and another partition to my Windows 2k3 domu? I dont want to resize the C: partition a need a newone. kind regards -----Mensaje original----- De: John Leach [mailto:john@johnleach.co.uk] Enviado el: Viernes, 15 de Agosto de 2008 12:25 Para: Xen-users@lists.xensource.com Asunto: Re: [Xen-users] Live migration - Networking take 2 minutes or so to return

Hi all, suppose I would like to change the initial value of a global. Is the supposed way to change the initializer via const_global_iterator I I->setInitializer(newOne) ? afaik, newOne is a constant - right? Is the original Initializer automatically purged or do I need to delete it? thanks Alex

Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated

Hi folks, I've just updated my * and I noticed that from the update the uniqueid field into mysql, is not written and ASTPP do not charge the calls. I got an eye to cdr_mysql.c and I found that at line 212, into one insert query, uniqueid is missing. But I can be wrong. In any case, somebody got same problem? Any suggestions? Thanks to all. -- .:FaberK:. -------------- next part

Hello Guys, first of all, I'm very new with asterisk. I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7 Now I'm trying with asterisk-oh323 I've already installed pwlib, oh323 and I've already set the variables. Now, when I try to "make" asterisk-oh323 I receive this error messagge: for x in wrapper asterisk-driver; do make -C $x all ||

hi new to rails ... wat does it means... uninitialized constant BooksController::Books /usr/lib/ruby/gems/1.8/gems/activesupport-2.2.2/lib/active_support/dependencies.rb:102:in `const_missing'' app/controllers/books_controller.rb:5:in `index'' -- Posted via http://www.ruby-forum.com/. --~--~---------~--~----~------------~-------~--~----~ You received this message because you are

Hi to All, I've an Asterisk CVS Head working with Mysql. My problem is that instead of ANSWERED or something like, into the CDR database records, I find only numbers. This is also a problem to let ASTPP works, infact I receive an error: ERROR - ERROR - ERROR - ERROR - ERROR DISPOSITION NOT MATCHED and the call has no cost. Any suggestions? Thanks -- .:FaberK:.

Hi, I'm using voipbuster at work, and I've got 2 questions: 1) Is it possible to send faxes using voipbuster connex? 2) Is it possible to cut off or cover the voice that say the charge per minute?(I've payed the '5' euro, and from that moment I've got it!). Of course I understand that is to let me know how much I'm going to spend, but I do not like it, expecially when

Hi all, I've got an Asterisk box with 1 Sangoma A102 and 1 Ericsson PBX. I need to use Asterisk as E1 line for the Ericsson PBX. How do I have to connect them? I'm trying to connect the Sangoma to the Ericsson, but RED alarms remain. Any suggestions? Thanks -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi to all, I've just started to see that Asterisk-gui from Digium. Does anybody know, when the first official-realese will be released? Thanks to all -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071012/09197260/attachment.htm

Hi guys, I'm using Zyxel Prestige 2602R, as router/SIP-ua with my architecture SER+Asterisk. Normally, everything is fine. In these days I'm experiencing some problems: some guests said me that, if he call everything is right, but if is called, he cannot hear the caller. Immediately, I though into an RTP-Proxy problem, but is not. Then I saw that message appear on the Asterisk CLI, during

Hello list, I have a problem with an NFS server for my network. It has ran kernels 2.4.18-ac4 - 2.4.21-ac1, all with problems. The -ac patches are used to provide the new style quota support. The system seems to have gotten even less stable with the new kernel versions. This morning around 5 am, I got a page the system was unresponding to NFS requests. I ssh'd in, and found the loadavg

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

Hi to all, I got an Asterisk with 2 BRI(7 pstn numbers and 4 concurrent calls) and 15 SIP extensions. The receptionist has a SNOM-360. How many SIP accounts would you configure on that phone? Only one would be enough? One SIP account, has a limit on concurrent calls? I saw that the SNOM-360 can handle up to eleven SIP accounts. Thanks -- .:FaberK:.

Hi guys, maybe you?ve got the answer...! When a caller(not internal, but from PSTN) call *, I need to let him hear a message, before * answer and the bill start running. If is not clear, just let me know. caller->telco(telco bill to the caller as soon as * answer)->asterisk Thanks in advance. -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi folks, I'm trying to install asterisk with radius cdr support. I got freeradius up and running, so following radius instructions inside asterisk source package, I've installed radiusclient-ng and relative headers. But when I start configure(asterisk 1.4.18.1) I got: checking for rc_read_config in -lradiusclient-ng... no If I type: ./configure --with-radius=/usr/share/radiusclient-ng the