similar to: Registered SIP:

Hello I can save you a lot of time, and tell you that it wont work. It does hold some registration information in the asterisk database, but most of the information is kept internally in Asterisk. Just FYI. Jon _____ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Shenen Shenen Sendt: 9. juni 2006 11:37 Til:

How can I copy all the contenent of the asterisk database to another machine? I want copy all the active sessions from one asterisk@home to another one and running on the second(this I can do using vrrp protocol, it isn't a problem), I want copy only all the active sessions and softphone registrations to another asterisk@home and then run on it. -------------- next part -------------- An HTML

Hi;I've a question: I use asterisk -R so I can see what's appening in my asterisk and the session of the calls: I use the vrrp protocol, I use 2 asterisk box;when the master falls down, the slave goes up, and I use X-lite,Phoner,3CXphone;some of this softphones are immediately registered to the slave, but sometimes this don't happen;I must close the softphone from my xp and restart

Hi:I 've a question: I'm using asterisk@home; I've seen the dialparties.agi , I want to do this; I've one softphone and I want register it in 2 different Proxy;only X-lite permitted this, all others no; I want have more proxy with others softphone;I run asterisk - R and I've seen when a softphone is registered(when I run it). My question is: Who is the file who makes the

I am attempting to establish SIP peering between Asterisk and an AltiGen soft PBX. This is my first experience with SIP peering. I can successfully make both inbound and outbound calls to/from a softphone on the AltiGen system (network access is provided by a PRI on the Asterisk system), but they are disconnected unexpectedly. The attachment is a redirect of the Asterisk CLI during a call that

Hi All, I have two SIP softphones(Windows Messenger) running on different subnet (Phone-1 on IP XXX.XXX.25.ABC & Phone-2 on IP XXX.XXX.15.XYZ) and my Asterisk Server is running on IP XXX.XXX.25.PQR.Because of some security issues both the subnets are completely isolated ( U cant even PING from one to other) and I want to connect Phone-1 & Phone-2 to the *. How can I proceed? Please

Hey folks, I am trying to get an E65 to connect to asterisk, and I would really appreciate a second set of eyes. The SIP dialog completes fine, but the phone subsequently says "Registration failed". I am in a network that has what seems to be a SIP-capable NAT gateway, but the asterisk is configured nat=yes anyway. Using a softphone (twinkle), I can connect just fine, SIP and RTP work.

Hi All, I have a new asterisk setup running at home and am very happy with it. One thing that I am trying to do is to take various actions in the dialplan *if* a particular phone is registered/authenticated/connected. For example, if someone dials *me* and is shows that I am connected via my softphone, to try it instead of my deskphone (and possibly notifiy the user in advance that it is

Hi Folks, I'm trying to register softphone(X-lite) but I'm not able to register softphone whenever I'm trying to register softphone I got below error [image: Inline image 1] Is there any document/guide line where I will get process to register softphone in asterisk(Which is installed in EC2 Cloud). Regards Akhilesh -------------- next part -------------- An HTML attachment was

Hi, We have set up our Asterisk server, our extension.conf and sip.conf according to http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=4 It's quite basic, and extension.conf is set up properly. The difficulty we are now encountering is in sip.conf, in trying to get any softphone to register at our own Asterisk server. We have searched the mailing list, and find bits and

Hi Guenther, What did you recommend to me, I did accordingly but there is no log showing in asterisk CLI. I'm getting same problem. Regards Akhilesh On Mon, Apr 20, 2015 at 6:05 PM, Guenther Boelter <gboelter at gmail.com> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 04/20/2015 12:31 PM, akhilesh chand wrote: > > Hi Folks, > > > >

Hi Guenther, When I executed nmap -p5060 <xx.xx.xx.xx> I got below output. [root at ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC Nmap scan report for ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com (xx.xx.xx.xx) Host is up (0.00080s latency). PORT STATE SERVICE 5060/tcp filtered sip Nmap done: 1 IP address (1 host

Hello. I tryto use Statis at my dialplan to run my app (a) When Statis is running from making call ( I dial from softphone some exten and run dialplan context where call Statis(MyApp)) Asterisk responsed: ERROR[61517][C-00000019]: res_stasis.c:852 stasis_app_exec: Stasis app 'MyApp' not registered How I must Register MyApp -------------- next part -------------- An HTML attachment was

Hi, Try as a first step a tcpdump capture to verify if the softphone is actually sending the register message to the server. For me it seems like the softphone is not able to reach the server ! Best Regards, On Fri, Apr 24, 2015 at 10:55 AM, Helvio Junior <helvio.listas at gmail.com> wrote: > Hi Akhilesh, > > SIP protocol use port 5060 (default) and many other ports to stablish

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

Hello, ------------- -------- --- -------- |Sip Softphone|-------|Internet|--------|F.W|-----|Asterisk| ------------- -------- --- -------- IP addresses: a.b.c.d q.w.e.r The SIP softphone(x-lite) is configured to register with the asterisk server through port 9090 (Domain q.w.e.r:9090).Firewall(F.W) is setup as the

I've noticed the following situation: In two softphones, I've configured the next codec order for each one softphone 1: 1 - PCMA 2 - GSM softphone 2: 1 - GSM 2 - PCMA and in Asterisk, the order is: disallow=all allow=gsm allow=alaw If I call from softphone 1 to softphone 2, I presume that Asterisk should do transcoding (canreinvite is set to no):

Hi James, Please let me know how could I implement for sip.I will appreciate your help. Regards Akhilesh On Mon, Apr 27, 2015 at 7:01 PM, James Cass <jcass78 at gmail.com> wrote: > Akhilesh, > I have implemented several ec2 instances with both sip and iax2 and have > no problems with xlite or hard phones. Have you already opened the ports in > the vpc security group on the

I have a software-only PBX set up. I can register various softphones and they will call each other -- but I've never succeeded in getting any voice routed from any of the softphones. Only the console will transmit audio. I am writing to ask if I have missed some obvious step in configuring the system. Conditions: (1) Softphones running on the same machine as the PBX: Only Kphone seems

Hi everyone, I'm wondering I would accomplish the following: I want to dial several SIP extensions simultaneously, HOWEVER, for different times (say ext 10 for 15 sec and ext 11 for 30 sec), and potentially with different headers (such as ALERT_INFO) and codecs for each extension. Obviously whoever picks up first gets the call. After the longest timeout expires (30 sec in this example) I want