similar to: canreinvite question

Hi William. Thanks a lot for your help. Im having some trouble recompiling my kernel after a installed the patch. Im running RH 7.3 with kernel 2.4.18-3. The patch I installed is routes-2.4.16-6.diff. I got no errors installing it. I added the multipath support, and recompiled it. The make dep and the make bzImage went fine. I got error during the make modules. These are the errors:

Hi, I'm using opensips as the registrar server for my users. I am redirecting calls going out to pstn to my asterisk server. call flow is basically: ua --> opensips server --> * server --> sip gateway provider if (uri=~"sip:00[0-9]*@sip\.myserver\.com") { xlog("L_INFO", "Call to PSTN\n"); #strip(2); #prefix("011");

Hi All, If i have this kind of setup, what do i need to make it's load balance. [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | --------------------------------------------------------------------- | mysql cluster |

Hi, I've been working on migrating my asterisk from zap to sip (due to compatibility issues between my TDM400P and my Hauppauge PVR500). I've purchased a Linksys SPA-3102 and a Siemens Gigaset SL75 WLAN (wireless SIP phone). I managed to get it all working with my asterisk 1.4.0 installation, but I'm seeing some interesting things with the canreinvite option that I can't explain,

Hi All, I have setup 2 trunks using 2 different voip providers using sip. the first one i have no problem calling inbound then redirected to an IVR, i can hear the IVR. the second one has issues, inbound works going to IVR as i can see it on the CLI, but i don't hear anything. i tried redirecting it to an extension not an IVR just to see if inbound really works, and it rings the

Hi, I have setup 2 asterisk talking? a single mysql cluster. I'm also using realtime db. I've setup sip peering between the two asterisk servers. [asterisk-1] insecure=port,invite type=peer host=201.202.203.204 context=from-asterisk-1 [asterisk-2] insecure=port,invite type=peer host=201.202.203.205 context=from-asterisk-2 scenario: ext 100 registers on Asterisk 1 ext 200

I have a number of internal SIP phones and a number of external SIP clients with the server running Asterisk on the boundary between the two. ie the server has two network cards with an internal private address and an external public address. For security reasons no routing is allowed between the two thus no internal phone can talk directly to external phone or visa versa. I am very happy with

current setup SIP phone 192.168.1.30 --> linksys wrt54g sveasoft -- INTERNET -- (xl0) Firewall (xl2:172.16.0.50)--> (em1:172.16.0.101) Asterisk problem is RTP stream not oging trouhg from * to sip and vice versa. #1 and asterusk is pushing 192.168.1.30 back to linksys with 172 as return address.... or #2 asterisk trying to get back to me as 192.168 on public internet.. got

I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer. 1) If I have SIP Provider ----> Asterisk -----> ATA and vice versa (ATA -----> Asterisk ----> SIP Provider) from what I understand if NO NAT is being used then asterisk just starts and stops the session however the RTP media stream will be passed directly from the SIP

Hi All, One of our user asked the question, when she tries to call another local extension but the other end is engaged she will keep on trying until she finally can get thru. So she asked would it be possible to request for an auto-callback from the user she's trying to call to once it's not engaged anymore. is this possible on asterisk? what is that feature called? i am using

I am trying to make 1 soft SIP UA behind NAT connect to a public hard CISCO UA via a public asterisk server. The CISCO UA can hear the voice from the SIP UA but not vice versa. I do set nat to yes for the soft phone. Any help would be greatly appreciated. Below is my sip.conf [general] port = 8060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

Hi, I've set up Tinc in switch-mode on the three nodes "gw", "rb493g" and "v900w", but the nodes "rb493g" and "w900v" do not connect to each other. On each node port 655 is opened with TCP and UDP. "gw": static IPv4- and IPv6 address listed in the hosts-file "gw" hosts-files: "gw",

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call

hi michael, here are the core dumps. only kphone works when 0.5.5 and * cvs. audiocodes and msn messenger all cause seg faults when calling ccm thru * (or vice-versa) ~kelvin [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper

On 30.08.2016 17:37, Guus Sliepen wrote: > On Tue, Aug 30, 2016 at 02:38:16PM +0200, Armin Schindler wrote: > >> we use a meshed VPN with TINC to connect 7 offices. >> Some office are in other countries and use other ISPs. The connection >> between some ISPs (peering partners) are not that good. This means we >> have packet loss between those direct connections.

I finally got the power supply for my 7960 and am having problems getting it working. What should be in sip.conf and the SIP(macaddr).cnf file? This is what I have in SIP0002FD3BA8F7.cnf # SIP Configuration Generic File # Line 1 appearance line1_name: Asterisk Test # Line 1 Registration Authentication line1_authname: "phone1" # Line 1 Registration Password line1_password:

Hello, This is not exactly an Asterisk question, but I was encouraged to seek advice here anyway. The kindness of the * open source community is legendary :) I am getting going with an Asterisk 1.2 box, and I'm having trouble getting good quality transmit sound using handsets with VoIP phones. I'm primarily trying to focus on SPA-941, but also experimenting with Aastra 9113i and Uniden

Until recently, I was using Broadvoice for my in/out calling thru Asterisk. I was extremely pleased to see that Broadvoice was actually passing the callerid info (number and text) that I had set up on each device in my SIP.CONF file. I had PSTN users tell me that they were actually seeing name and extension info when I called them from the Asterisk box. Last week, due to numerous user quality

Hello List, I tried to set up openvpn with the shorewall on my openwrt box but failed! I am not able to access the "loc"al Network from my vpn. I followed the roadwarrior setup. I define a vpn zone, that should be able to access the firewall and the local network: vpn fw ACCEPT info fw loc ACCEPT info vpn

So I have many Cisco 7960's that are running the latest 5.1 Cisco SIP code and I cannot get the phones to talk/RTP to each other. jtodd has had this problem in the past with the 186's. Just wondering if anyone has a reason why "Cisco sometimes poop on reinvite" is the Cisco code broke? if so we can push on Cisco to fix it. the U is a MAJOR Cisco shop so we have some puhs