similar to: Problem with conferences, Vlada, Pancevo

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Hi, my situation: One internet connection 256 kbps uplink/downlink, pretty stable speed, a Linux router with three NICs (one to ISP, one to DMZ and one with VLANs enabled to our LAN). Each of three VLANs (1, 10, 11) is a region for me; id 1 gets guaranteed 128 kbps, id 10 and 11 both get guaranteed 64 kbps (both uplink and downlink). Simple configuration. After I read relevant parts of lartc

http://bugzilla.mindrot.org/show_bug.cgi?id=885 Summary: List of preferred address families ? Product: Portable OpenSSH Version: 3.8p1 Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P2 Component: Miscellaneous AssignedTo: openssh-bugs at mindrot.org ReportedBy:

Hello, I've got a deployment with the SBC in between the clients and Asterisk (11.17.1 version). When the UAC tries to subscribe for "dialog" event package, the NOTIFY request sent by Asterisk fails. The SBC uses a different Contact (user part) for the 1st and the 2nd SUBSCRIBE (with Auth.). The issue is that Asterisk sends the NOTIFY with R-URI of the first SUBSCRIBE's Contact,

Hi, I cannot use '#' to initiate transfers. I have tried on different phones (7960, ATA, X-Lite). When I press '#' during a call, nothing happen. I have both T and t switches in Dial application. The transfer function works with Flash key on ATA, but in a very strange way....The final destination is hunged up and then automatically called by the initial caller... This behavior

Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty

I'm having trouble setting the outbound caller ID on calls I make from my PRI trunk group. The PRIs are served out of a 5ESS. Telco has set the PRIs up for user provided caller id information, so I believe I just don't have it set up right in my dialplan or something. I can't seem to find an example of setting the outbound caller ID specifically for a 5ESS. Does anyone have an

Hi, I'm using 2-factor authentication (pubkey+googe_authenticator) and have an issue with rsync. It's configured to use pubkey to authenticate to server so when google_authentication is bypassed by not creating .google_authenticator file for particular user (thanks to nullok option in PAM) it still sends to stderr "Authenticated with partial success." message although it

Thanks to this thread, we got it working too... but have a question... Once this is setup... does it stream forever, or does the stream only start when someone goes on hold/into a queue/etc? If it streams forever, at 24k... it looks like over 7GB/month in bandwidth... so we're not going to want to do that if a) it streams constantly and b) my math is correct. Thanks, Doug >

Hi, there. I am running CentOS 4.4 on HVM Domain. And I tried Para-virtual driver for HVM Domain. Hunged up then "modprobe xenbus". My Domain-0 information: OS: CentOS 4.4 xen: 3.0.4.1 kernel: 2.6.16.33-xen_3.0.4.1 My HVM Domain information: OS: CentOS 4.4 Kernel: 2.6.9-42.0.10.EL PV Drv: xen-unstable.hg I traced the source code. I think problem at line:132 in xenbus_xs.c. It stopped

Does anyone have an example of hoe to use these two commands? I have read he documentation, and I am still unclear on where this command goes, as part of extensions.conf or where? If someone could post a working example it would be most helpful. Regards to all, Joe

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

I've had my * setup installed with an X100P card for a couple of weeks and it's great fun! I'm even giving a demo to the local Linux group in a couple of days. But I have a snag. I have the X100P on a shared line, and configured to wait for 20 seconds before answering and doing the auto-attendant/voicemail dance. My problem is I can't find an application command to cancel the

Hi nut-upsdev, I have just got OMRON BX35F. Only object files and usage information is included. Where should I start to help you to get it supported? Any interest in supporting it at all?

Hi, I would like to know what is "Asterisk Dynamic Loader". Let me explain what I'm about to ask. I have three Asterisk servers running my in-house built app_xyz.so application. Now what I do to save time is compile application on one server and scp app_xyz.so on rest of servers. All servers have same OS, H/W specs. Today I checked the logs and observed that at the time when

Hello! I Have some problems with running Heroes IV. It's run by wine 0.9.12. But If I play it 10-15 minutes, it's hunged up. What's wrong? Wine 0.9.12, Slackware Linux 10.2, Kernel 2.6.16.9

Hi. I've noticed a problem with the expression parsing in Asterisk. If the variable is not defined, I will get a parse error. Yeah, there are ways around it, but I would think that it should return false if 0, null, or undefined. I would change it, but I have no idea about bison and I only have very basic C skills. There was a bug opened on this, and there was a valid work-around posted,

There are often times that I want to read a DB value from the dialplan, and if this family/key pair does not exist, set it to some default value. for example: 1234,1 => Set(EMAILADDR=${DB(x/y)} 1234,2 => GotoIf($["${EMAILADDR}" = ""]?3:4) 1234,3 => Set(EMAILADDR=Someone@test.com) 1234,4 => NoOp(${EMAILADDR}) 1234,5 => Hangup() I have modified the db function

Can someone help me what went wrong with this call? This call was initiated from dev/ttyI0 device on my asterisk server to mediatrix unit. Mediatrix unit user received the call and call started. I can hear them OK but they can not hear me correctly (cut-off sound, noise). Call was finally hunged up. Can anyone point out if there was something wrong? -*CLI> sip debug SIP Debugging Enabled

As an employee in the technical operations of a CLEC this information is easily obtainable by anyone that has access to the Class 5 switch servicing that PRI... A Q.931 trace in the Class 5 Switch will tell the whole story.... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rich Adamson Sent: Tuesday, June