similar to: callerid= being ignored

Hi my Digium FXO card isn't picking up the callerid I get from the PSTN. I have verified with a deskphone that can display the callerid that the facility works. So, it's definitely the FXO card not picking it up. As I am in Japan, I guess that NTT uses a different method to provide the callerid and so I guess that it is just a matter of configuring the FXO card so that it uses the

Hi since "callerid=" in sip.conf doesn't set the Caller ID, I suppose it must be there for some other reason. Is this a not-yet-working feature for future releases of Asterisk? If not, what does it actually do? thanks regards bk

Dear list, I have a data frame like: > log2.ratios[1:3,1:4] Clone a1 a2 a3 1 GS1-232B23 -0.0207500 0.17553833 0.21939333 2 RP11-82D16 -0.1896667 0.02645167 -0.03112333 3 RP11-62M23 -0.1761700 0.08214500 -0.04877000 how to make it to look like: > log2.ratios[1:3,1:4] a1 a2

Hi, I'm having trouble explicitly forcing a codec between sip devices. Am I missing something or is this not really possible? I have a grandstream registering to asterisk, named sip0. Sip0 registers, via sip, to another asterisk box, sip1. When I place a call from the grandstream, it will travel through sip0 to sip1, where it is then placed to the PSTN. Nothing can reinvite, this path is

Asterisk 11.1.0 One queue with strategy=leastrecent. (Full queues.conf below.) Occasionally (several times today), a caller will get "stuck" in the queue - there are operators available to take the call, but the caller stays in the queue for a long time. Any idea what might cause this, or where I can start looking to debug it? I'm going to start digging through the queue log

> Dear list, > I have a data frame like: > > > log2.ratios[1:3,1:4] > ID a1 a2 a3 > 1 GS1-232B23 -0.0207500 0.17553833 0.21939333 > 2 RP11-82D16 -0.1896667 0.02645167 -0.03112333 > 3 RP11-62M23 -0.1761700 0.08214500 -0.04877000 > 4 RP11-62M23 0.2761700 -0.15214500 -0.05877000 > the 3rd and

Hello list, my case is as follows: SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729. When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the keypad on the phone. As suggested by you, I need to configure the SIP1 with out band dtmf mode , what is about the sip.conf, should I specify the SIP1 with demfmode=rfc2238 ? do I also need to make same kind

Hello, I've found GlusterFS to be an interesting project. Not so much experience of it (although from similar usecases with DRBD+NFS setups) so I setup some testcase to try out failover and recovery. For this I have a setup with two glusterfs servers (each is a VM) and one client (also a VM). I'm using GlusterFS 3.4 btw. The servers manages a gluster volume created as: gluster volume

Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ / \ User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB / \ / User3 --

Hi I have just been struggling for four days to get SIP working and now as I created a voicemail box, Asterisk has become very unstable and it can't bridge SIP phone to SIP provider calls anymore. Calling internally from one SIP phone to another works fine. Calling internally from a SIP phone to an analog phone on a Zap channel and vice versa works fine. Incoming PSTN calls delivered to

I'm using the TSA package (along with all prerequisites) to do some GARCH work and for some reason, something which used to work for me has decided to up and stop. The code is as follows, after loading the package: " gs <- garch.sim(alpha=c(1.9,0.1), beta=c(0.700001, -0.0800003, -0.016),rnd = rnorm, n = 400, ntrans=500) gs1 <- garch.sim(alpha=c(1.9,0.1), beta=c(0.7, -0.08,

Hi Samba, I hope you are doing well. I run a cifs / nfs CTDB clustered NAS solution, and I find that when I reboot any of the nodes in the cluster, I must re-export the nfs mounts so they show up properly. Perhaps this is a general linux nfs bug and I am barking up the wrong tree, but I haven't found any problem / solution mentioning this as of yet besides my own known workaround

hello, i have a FreeBSD 4.8-PRERELEASE #0 that i use as a gateway / nat box for my home. It also acts as a dns / mail server to the outside world. I'm using ipf and basically filter for bogus networks on the way in and out. I allow everything out keeping state, and allow this in: pass in proto icmp from any to any icmp-type squench group 200 pass in proto icmp from any to any icmp-type timex

On Tue, Jul 9, 2019 at 6:05 AM Joshua C. Colp <jcolp at digium.com> wrote: > On Tue, Jul 9, 2019, at 7:00 AM, Dovid Bender wrote: > > Hi, > > > > Looking at http://the-asterisk-book.com/1.6/applikationen-dial.html you > > should be able to dial with SIP credentials in the DP. Is this still > > possible in recent versions of Asterisk either with chan_sip or

Hey, You know, the Digium guys said both are good. They said the the DNS method is better because you dont have the extra point of failure (SER) but said the SER method is better in that it gives you more exact control in the handling of the calls and registration. They did acknowledge there would be a possible downtime only for incoming calls to users with dynamic IPs if the

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem

Hi I thought I share this one, just in case this is an indication of some bug ... When I was trying to use music on hold at first, I didn't bother to copy any music into /var/lib/asterisk/mohmp3 since there was a sample- hold.mp3 in there which played just fine in a standalone MP3 player. But after uncommenting one of the lines in musiconhold.conf and doing reload on the console, there

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The

On a server Centos 7.2 ?I insert my 1Tb usb disk and run sudo mount LABEL=bk-benvet-01 /mnt/bk-benvet-01 the command seem to work but nothing is mounted Into log I see this issue: > apr 01 13:49:06 s-virt.dom.loc kernel: XFS (sdb1): Mounting V4 Filesystem > apr 01 13:49:06 s-virt.dom.loc kernel: XFS (sdb1): Ending clean mount > apr 01 13:49:06 s-virt.dom.loc systemd[1]: Unit

Hi please excuse if this seems obvious, but I am new to this and the SIP section in the Asterisk handbook do not give any clues nor do the SIP examples in there seem to represent real-world situations. I am using Nikotel as a VoIP provider (for now) and I would like to configure Asterisk to sign on with Nikotel so that I can use the telephones connected to Asterisk to make calls using the