Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

Thank you for your help! Centos 7 firewall was enable. systemctl stop firewalld issue fixed. Thanks, On Thu, Oct 13, 2016 at 3:54 PM, Victor Villarreal wrote: > Ok. > > Please, note that 192.168.1.37 (I suspect) is the internal LAN address Of > the Polycom hardphone.

[asterisk-users] SIP on multiple ports

I have SIP (asterisk 11.23.0) running on port 5060 just fine. udp. I have another SIP trunk thats wants to run on port 5068 (long story). I have enabled tcpenable=yes in sip.conf and defined port=5068 in my trunk definition. It does not seem that anything is listening on 5068? How can I run SIP

Re: [asterisk-users] iptables for SIP talk to other port

> > Your correct. I forgot to mention that the other end IS using tcp. So I have in my SIP trunk. transport=tcp So correct my iptables line was specifying "-p tcp" I also set tcpenable=yes in sip.conf Thanks. Jerry -- _ --

[asterisk-users] How can I "lock" a device or extension state to only specific states?

How can I lock a device state so it can only publish AVAILABE, BUSY, or RINGNING? (Eg, if the device is not BUSY or RINGNING, its AVAILABLE) I have a hint published for a fixed phone and a mobile phone. But if the mobile phone is out of coverage, off or similar, the queue application will

Re: [asterisk-users] Feature access codes

On Sat, 15 Oct 2016, tux john wrote: Hi. Kinda new to the area and I would like some help please. I have asterisk 11 in my system and I have 10 users and 12 DIDs. One did routed to each user and 2 DIDs for faxing. Everything works fine but I do not have call transfer between extensions and

[asterisk-users] Feature access codes

Hi. Kinda new to the area and I would like some help please. I have asterisk 11 in my system and I have 10 users and 12 DIDs. One did routed to each user and 2 DIDs for faxing. Everything works fine but I do not have call transfer between extensions and feature access codes. I have read somewhere

Re: [asterisk-users] iptables for SIP talk to other port

You're redirecting tcp, sip defaults to udp. -- Sent from my cellphone. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016

Re: [asterisk-users] iptables for SIP talk to other port

Le 15/10/2016 à 18:17, Jerry Geis a écrit : I have a host 192.168.1.3 that wants to run SIP on 5068 (long story). My host is 192.168.10.201. My host needs to stay on 5060 because of all the other devices I have connected. I tried putting port=5068 in my SIP extension definition but that did not

[asterisk-users] iptables for SIP talk to other port

I have a host 192.168.1.3 that wants to run SIP on 5068 (long story). My host is 192.168.10.201. My host needs to stay on 5060 because of all the other devices I have connected. I tried putting port=5068 in my SIP extension definition but that did not work. So I thought about using iptables to

Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

ok, now it is getting weird... actually i don't see any firewall issues, but i am not able to get a call from outside to my sipgate account. in asterisk nothing is visible, core set verbose is activated. sngrep (on my asterisk server) shows me indeed the invite from sipgate!? What I see via

Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

Have you tried setting keepalive(20 seconds) on your sip.conf and on your phones ? From: Andre Gronwald To: asterisk-users@lists.digium.com Sent: Saturday, October 15, 2016 9:17 AM Subject: Re: [asterisk-users] Registered successfully, but after a minute

Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

ok, solved the firewall issue. A first test call worked fine. Another one now still gets disconnected after 32s. But in FW there are no blocked packets anymore?! And I don't understand why the registration to the same IP and same Port is working, but not later transmission of further SIP

Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

Hi, I don’t see any SIP ACK’s in your trace. Is the SIP 200 OK reaching the originating caller, or being blocked on the way through? Asterisk will tear down the call after ~30secs of audio playing in both directions if it doesn't receive the SIP ACK. Regards, Ian On 15/10/2016 12:05,

Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

hi, let me explain in detail, what i have configured and what is happening now: 1st router w724v (Deutsche Telekom AG): - port forwarding, everything to destination port 51000-55999 to device with ip 192.168.2.50 (interface of 2nd router) 2nd router Bintec RS353j): - configured NAT,

Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

Thanks Jonathan for your support. I would like to avoid TLS at the moment (in general I am a fan of secured communication!) because the other provider is not supporting TLS. And sipgate is just used for testing. However I can see the following which is quite interesting: [2016-10-15

Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

Hmmm, sorry, I can't think of anything except... why do you need the STUN server? And are you sure that all the ports in your router definitely match the ones Asterisk thinks it's using? Then there is always the SIP-ALG problem with some routers, which some people have been able to overcome by

Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

ping times are fine as well: [root@freepbx asterisk]# ping sipgate.de PING sipgate.de (217.10.79.9) 56(84) bytes of data. 64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms 64 bytes from sipgate.de

Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

All other things aside, this stands out immediately: RTT: 434.393 msec That's almost half a second round trip for a packet. I'm amazed anything works at all. For SIP connections, mine are usually about 26ms max, anything above about 35 is bad. Looks like a serious config issue. Try pinging and

Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

Very interesting: I have another provider configured, that was not reachable as well. I disabled the STUN-server (external STUN server), and now the second registration works fine, BUT with the same "error" messages (unreachable etc) as the other provider. But in contrast the number is always

[asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

Hi all, I have an issue with asterisk 13 and pjsip. I guess it is somehow Firewall related, but I'm unsure. A registration to Sipgate is established successfully: ==