similar to: VoicePulse 1-800 numbers sound problem

Hi all, Does anyone know how to get the channel ID on the other side of the call? For example: When SIP/50 calls SIP/21, and the call is answered by SIP/21 I get: SIP/21-6735 answered SIP/50-b456 ${CHANNEL} will show me SIP/50-b456. Is there a parameter or a workaround to get the SIP/21-6735 part? Thanks. Assaf Benharoosh -------------- next part -------------- An HTML attachment was

Hi All, Did anyone came across a 411 XML service I can feed to the "service" button with XML? Some other feed I can manipulate to XML query? Assaf Benharoosh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041004/dbf552ac/attachment.htm

For some reason I can't get the Dial command 'C' flag to work. The calls are recorded in the CDR with the 'C' on. Does anyone have an idea? extensions.conf: exten => 114,1,Dial(SIP/114,,C) It shows in the lastapp: cdr: | 2004-10-05 13:16:02 | "112" | | 114 | intern | SIP/112-3fb6 | SIP/114-0e7a | Dial | SIP/114||c | 6 |

I found a message from you to the asterisk users mailing list from 2001. I was wondering if you got (or still have) an asterisk system working with the linejack? If so, would you be willing to assist me with mine? I seem to have things working, and * says that caller ID is coming in, but I can't get * to actually answer the call. Thanks, Greg -- NetIO.org

Hi Everyone - I have webvmail up and running, I can see the messages, forward them, pretty much everything but listen to them. Here is what I see in my logs: 192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] "GET /vmail/vmail.cgi?action=audio&folder=INBOX&mailbox=2377&context=default &password=000012&msgid=0000&format=gsm&dontcasheme=4624.gsm HTTP/1.1" 200 9438

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I'm not sure if BV will support multiple lines. Any

Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl

I can ping it just fine. I am on gw5.voicepulse.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040615/054c83f4/attachment.htm

atlast after working of 7 hours i got voange soft account working on asterisk.

There were a number of messages about using a digium card as a data t1 (router) kind of thing. Is there a howto out there or a status that I can see whether this is feasible or not? Or is it feasible but very difficult? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

I'm running a Samba 3.0.0 server in production in security = ADS mode against a W2k ADS server. Works just fine, thanks! We're sort of under pressure to regrade to a 2003 AD server, which sent me trying stuff out a bit. Meager results. The 3.0.0 I have (linked with MIT krb5-1.2.8) refuses to verify incoming tickets: [2003/10/28 16:27:36, 3]

Sorry for the frantic nature of this message, but we need to allow pings on our firewall so our ISP can test things. I''ve done this, and it still doesn''t work: (I am now at v.2.0.10) rules: AllowPing net fw AllowPing sls fw show indicates some matches, so where are they? Chain AllowPing (4 references) pkts bytes target prot opt in out source

Rupert Gallagher via rsync <rsync at lists.samba.org> wrote: > On 7 Aug 2020, 23:44, Wayne Davison < wayne at opencoder.net> wrote: > > >> Also, I have 12GB of cache in ecc ram that rsync is not using. > > >It uses whatever memory it needs plus whatever filesystem caching > >your OS provides. > > Hmmm... bad day today... > > No, it is not

Hello everyone, I upgraded my Samba server on my RHL 7.2 box from 2.2.3a to 2.2.7. Almost immediately afterwards, all of my admin users suddenly were unable to access the printers that were being shared by Samba. Any idea of what is wrong and how to fix it? The messages I'm getting are that I do not have enough rights. I am using a Windows 2000 Server PDC and a lot of Windows XP Pro, and

Hello, I'm willing to swear that with the 0.99.10.4 Debian package in my rather extensive tests with mailbox format dovecot created not only any missing parental directories but also the INBOX file, but the later I can't prove/test now as I upgraded all respective boxes. However 0.99.10.5 will still create (if needed and possible) missing directories for mbox storage, just not an

Hi all, I have an interesting project that I am working on. It is a large volume file download service that is in need of a new box. There current systems are not able to handle the load because for various reasons they have become very I/O limited. We currently run on Debian Linux with 3ware hardware RAID5. I am not sure of the exact disk config as it is a leased system that I have never

I am currently experiencing intermittent silences with my asterisk system. The symptoms are as follows: * Both for incoming and outgoing calls, I (and other users) occasionally experience a brief period of silence. * The silence lasts anywhere from 3 to 10 seconds. * It is not due to silence suppression, because the silences generally occur in the middle of sentences. * Silences occur at

On Thu, Aug 6, 2020 at 10:46 PM Rupert Gallagher wrote: > I noted that rsync writes a gmon file on the source path and leaves it > there when it terminates. Nope, it doesn't. You'll need to figure out what's going on with your setup. Also, I have 12GB of cache in ecc ram that rsync is not using. > It uses whatever memory it needs plus whatever filesystem caching your OS

After rebooting my asteriks server, e-mail notifications are no longer being sent after a voice-mail is left. I can see the messages in /var/spool/asterisk/vm. has anybody had the same experience? how was it resolved? Uri -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040410/fc494bb4/attachment.htm

Hello, I am very new to asterisk and voip in general and so far have managed to get the FXO card and a few sip phones working fine. My question is where does the Sipura SPA 2000 come in the picture? Can it be used as an extension (i.e FXS) ? Or is it to be used as a line (i.e FXO)? Or it can be used as both? My understanding is that its just like another ATA186. Is that true? I guess what I