similar to: Audiocodes MP124-FXS replying BUSY when line is not.

Just a question, has anyone knows how to blank or factory reset an AudioCodes MP124-C/FSX/AC/SIP unit (it's a 24 FSX to SIP unit). I purchased them second-handed with no manuals (thank god for the internet!!) but i guess the pdf manual I have does not have the section of factory-reset. Also, any sucess stories with: AudioCodes MP124-C/FSX/AC/SIP

Jerry and Michael, many many thanks for your posts. Erick. On 5/24/06, The VoIP Connection <asterisk-biz@thevoipconnection.com> wrote: > Here are the step by step instructions for setting up a brand new Audiocodes > FXS gateway for use with an Asterisk server: > > Connect the gateway to a network switch and connect a computer to the same > switch. Then configure the IP

Hi, I just had my production box deadlocked - no calls could go trough, CLI didn't load. Last lines in log were: [Jan 11 09:15:43] VERBOSE[7265] logger.c: -- Executing [28901 at local_dial:40] GotoIf("SIP/204.11.200.152-c0070ed0", "1?41:57") in new stack [Jan 11 09:15:43] VERBOSE[7265] logger.c: -- Goto (local_dial,28901,41) [Jan 11 09:15:43] VERBOSE[7265]

hello list, last night i upgraded my asterisk box from 1.0.5 to 1.0.6 and my music on hold did not work anymore. my setup is ISDN (wct1xxp)->SIP (Audiocodes mp124) and reverse. the system refuses to activate music on hold resource... i returned to 1.0.5 and it works fine again... i'm i missing something? i can really make use of 1.0.6 bug-fixes and i'm sorry i can't use it :((

All, Thanks for all your help. Now, I can compile the latest source code base on RedHat 5.5 after update nasm(to 2.09) and binutil(2.17). And *make spotless* before *make* under core/ directory. But with the new pxelinux.0, the PXEClient can not bootup. The error info, No valid file system found! And stuck in there. I think maybe the gcc cause the problem. My gcc version is 4.1.2. Thanks aaron

Hello Everyone, Unlike going through DIDx, DIDLogic etc.., we have an option of getting DIDs directly from local telco Bell Canada. Currently our SIP Trunk provider assigned a DID to us, and as you know, they just redirect requests it to our PBX. However, when dealing directly with a telco, what equipment will we need? Basically giving us the same capability as a DID provider. If someone can

Asterisk doesn't seem to be relaying 183, Session Progress SIP messages received from an upstream host back to the phone. Anyone know why? Here's the SIP message that Asterisk receives, and it does nothing with it. It doesn't pass it back to the phone. <-- SIP read from xxx.yyy.142.234:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP

Here is what I have in my extensions.conf file now. Trustrcid and sendrcid are turned to "yes" in the conf file. I'm not fully sure how the SIPCalledRPID works though. The example I found seems to try and provide the stuff automatically (id and name), but does the SIPPEER stuff even exist? I think this is probably the right track though. Any insight would be much appreciated.

Hello Stephen, I may have tracked down some unexpected behaviour from a common bridge setup, and would like to incite expert oppinion on my observations. The issue relates to both 2.6 and 2.4 kernel series bridging code, and as far as I can see might have been present in all releases hitherto. Consider this setup: - two ethernet devices in a simple bridge configuration - bridge-interface

Sorry Gene, I got the version of binutils is 2.17. I download the latest Binutils to make the *objdump* and *objcopy*. with both of these utilities to create new pxelinux.raw. then, the error message shows me that, objcopy -O binary pxelinux.elf pxelinux.raw ../lzo/prepcore pxelinux.raw pxelinux.bin ../lzo/prepcore: pxelinux.raw: output too big (30165, max 0) make[1]: *** [pxelinux.bin] Error 1

Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

Hello, I have a problem getting the domain 0 (dom0) of xen running with SuSE 10. I have a SuSE 10 installation, which runs. When I switch to the xen kernel, the network does not work any more. Also the serial console does not work any more. If I login to the PC through (a very lousy) KVM, I can do local stuff, but the network does not work. (Ok, I can ping my own IP Address.) e.g. nslookup

Like many users I get my voicemails emailed to me, AND left on the asterisk server, so that I can retrieve them by phone or by email. However, I was frustrated that after I deleted a message in outlook that I still had to delete it from asterisk manually. So, I wrote a script that runs on the asterisk box that allows the user to simply reply to an email (from the asterisk pbx) - which causes

We apparently have a spam bot subscribed to the list and replying *directly* to anyone who posts on the list. The E-mails, generally use the name Alyssa or a Katie in the mail and have images attached. They come from a variety of addresses that so far don't appear subscribed to the list. However spammers don't typically subscribe to lists at the addresses they send from or appear to send

We must consume the payload of any option received, whether or not we can answer the option successfully, in order to be in sync for reading the next option. This is particularly true once clients start using NBD_OPT_GO, as long as we don't happen to service that option. Signed-off-by: Eric Blake <eblake@redhat.com> --- src/connections.c | 8 ++++++++ 1 file changed, 8 insertions(+)

I have a naive question here, but only because I've managed to screw this up twice in the last week. What is the correct way to reply to a bug report? Should r-bugs be in the To: or Cc: fields? I originally thought that hitting "Reply" and stripping of the r-devel email address was sufficient but apparently not. -roger

Sorry for the non-R question, but how do I reply through the Nabble interface and have my reply emailed to the person I'm replying to (in case they don't use Nabble), with cc to the mailing list? If I choose the option to reply to the person by email, I don't see an option to cc to the mailing list. If I reply to the list, there's an option to email the post to someone, but I

Hello, I am trying to send OPTIONS message asterisk in order to find out that whether it is alive or not. everything is going fine except for the port it is sending the reply to. The problem is that it is not replying to the port specied in the VIA header, and is replying on the port from which it has recieved the request. How can I be able to send the reply on the Port specified in the VIA

Hi, When I reply to a post, it generally appears as a separate thread, rather than branching off the original post. Anyone know what I'm doing wrong? Noting I am currently receiving a couple of the lists in digest form. kind regards -- Charlotte Maia http://sites.google.com/site/maiagx/home

See the following SIP trace. Where in the world does Asterisk get port 1025 to respond to? This is asterisk 1.6.x. Thanks. -- James <--- SIP read from zzz.zzz.zzz.44:9363 ---> NOTIFY sip:pbx1.mydomain.com SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3^M From: "xxx-xxx-xxxx" <sip:xxxxxxxxxx at pbx1.mydomain.com>;tag=467525dd6fac949do0^M To: