similar to: Sip dynamic host question

It's my first post here, so I'll cut to the chase I have 2 Asterisk servers and want to connect them using sip on one and pjsip on the other one. One is running at home and another at a VPS. The first one will be the client (with dynamic ip) and the 2nd the server. The client uses sip and the server pjsip. This is the client's sip.conf [general] context = default allowguest = no

Hi All; I am using IAX Trunk and I used ddns (dyndns.org) with the host (host=xyz.dyndns.org), when I restart the router who has the hostname xyz.dyndns.org then its IP address change and updated, but at asterisk level still it keeps sending for the old IP address and sometimes this problem does not resolve until I restart asterisk. Any one faced this and has idea how to resolve it so Asterisk

Hello I have an asterisk 16.2.1 on an ubuntu on AWS, which is experiencing a drop in call. It does not have a certain time, it is random. The audio is flowing normally and the call is dropped. Has anyone ever experienced this? My settings changed below: allowoverlap = no udpbindaddr = 0.0.0.0 tcpenable = no tcpbindaddr = 0.0.0.0 transport = udp, ws, wss srvlookup = yes directmedia = no

how do i troubleshoot no ring tone. It was working and all i added was the lines below now it doesn't ring. Edit sip_nat.conf for proper NAT: localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here) externrefresh=10 fromdomain=DOMAIN.com (Set your external domain name here) nat=yes qualify=yes canreinvite=no Add extra codecs to

Hello. Thanks for the reply. Yes. In the traffic analyzed, the BYE is sent by the originator of the call, but there is no "human" hangup, but the asterisk one. BYE is sent, received and confirmed. I don't know how I could investigate the reason for this BYE. Em 21/09/2020 17:12, Dovid Bender escreveu: > Is there anything in the Asterisk logs? Which side sends the BYE? Were

My Asterisk server is behind a NAT and I have set: ---------------------------------------------------------------------------- externhost="my.server.address" externrefresh=180 localnet=192.168.0.0/255.255.0.0 localnet=10.0.0.0/255.0.0.0 localnet=172.16.0.0/12 nat=yes --------------------------------------------------------------------------- in [general] section of sip.conf. I can

On 4/1/15 10:48 AM, Daniel Heckl wrote: > John, > > thank you four your answer. I think you have misunderstood the > problem. It?s about a ip address change of the sip trunk, not of my > asterisk server. You would probably benefit by enabling the DNS Manager to allow for dynamic IP changes: # cat dnsmgr.conf [general] enable=yes ; enable creation of managed DNS

Hi list! Since the internal calls work as expected and I can register my Asterisk on an external provider, I'd like to add a new feature and allow my mobile phone to connect to my Asterisk and manage calls. Well, first of all, my Asterisk is NOT direct on Internet available, but behind a NAT. So I configured my sip.conf: localnet=192.168.200.0/24 externhost=myhost.noip.com externrefresh=180

Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13

I just jumped to asterisk-1.6.1.8 and I calls will not go through to my asterisk. Same setup with asterisk-1.4 and calls get accepted. sip show registry (asterisk-1.6): Host dnsmgr Username Refresh State sip.actio.pl:5060 N 4589835 105 Registered sip show registry (asterisk-1.4): Host Username Refresh State sip.actio.pl:5060 4589835

Hi, I am try to configure Asterisk as PBX system with two interfaces as shown below. One interface pointing to the local subnet with a SIP phone and another interface pointing to the external ISP SIP Sever. SJPhone(X.X.141.32)<--------->(Y.Y.47.149)local-intf-|Asterisk|external- intf(Z.Z.247.106)<-------->(w.w.158.26)ISP-SIP-Server----OutsideWorld I am able to setup a call from the

In a .Rnw file I want to insert the R command pairs(mydataframe) and achieve the following effects 1. the command itseld is echoed into the tex document generated by Sweave <<fig=TRUE,echo=TRUE>>= 2. The graphics generated appears in the tex document, with the graphics centred. 3. The R command > pairs(mydataframe) is not centered. Sweave-manual.pdf gives the following code chunk

Ok, I have tested dnsmgr. This is not a solution, the situation has not changed. With dnsmgr I can not place outbound calls. I do not know why and what dnsmgr really do. My current solution is as follows: Say allowguest=yes, configure the default context that there can not be placed outbound calls. Use iptables to DROP all at your SIP port and allow only your local phones and the sip trunk ip

Hi list! Yesterday Deutsche Telekom had a really big problem and Asterisk couldn't connect to the remote Server (by Telekom) until today about 7:30. Well, it could happen... What I find really annoying was that I needed to restart Asterisk as I checked with sipsak that the Telekom-Server works... I think, this should not be normal... Can someone explain me why it happens and what I have to

Hello everyone. I been working up with samba shared folders, but have some troubles with users/folder privileges. Users: ale jvillar I got a folder named "BACKUP" users ale and jvillar can read/write this folder inside "BACKUP" is another folder named "MAIL BACKUP" i want user ale to read/write this folder and user jvillar only read. Even though i tried everything i

Scott, I have changed the configuration as said it and will test it. I?m curious. Can you briefly explain what insecure=invite,port does? ;insecure=port ; Allow matching of peer by IP address without ; matching port number ;insecure=invite ; Do not require authentication of incoming INVITEs ;insecure=port,invite ; (both) Do I understand correctly that

What files at a bare minimum need to be in /etc/asterisk for an asterisk server that does sip only and voicemail. I'm setting up an asterisk server to provide service for a single SIP softphone extension with SIP origination and termination. The main purpose of using * is for voicemail and future expansion ability. I know I need sip.conf extensions.conf voicemail.conf but what else? do I

I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. -------------- next

[Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups. [Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net' changed from 64.2.142.17 to 64.2.142.29 [Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed Jitterbuf max 600 timeslots And the calls are dropped. I fixed this by turning off enable in dnsmgr.conf My question is: Do you attempt to

Hello! I have writen some subrutines in Free-form Fortran. I would like to includ them in a package, which I would like to build on WinXP. I have all suggested tools/programs for bulding R packages on Windows (except latex). What is the best way of using these subrutines? Does sombody mybe know any translation tools for converting Free-form to fixed-form Fortran? Thanks for any suggestions,