similar to: SIP KEEPALIVES, with QUALIFY and without making UNREACHABLE

Qualify=yes will send a SIP OPTIONS periodically and keep the NAT open, if you use 1 to 1 NAT (versus PAT where it is "many to one NAT") it will work because port 5060 on the private address will still be port 5060 on the public address. With PAT the port could be anything over 1024, but usually much higher, and the originator will send to port 5060, which your NAT router will drop.

I have logs full with this messages... I must have qualify turned on, because phone is behind firewall, main problem si, that phone is each hour about one hour unavailable! :'( I tried to modify minexpiry/maxexpiry sip.conf timeouts, but nothing help me. I'm using latest firmware 8.4 in phone, will be better to downgrade? to what version? (latest asterisk 1.4branch) [Dec 7 00:36:56]

Hi guys, I configure one Fedora Core Linux 5 for use with asterisk as gateway using Digium TE110P interconected in Alcantel 4100 I've set up it to register 100 voip numbers on my provider. All calls on Alcatel is send to asterisk. In some periods of day i receive this messages on asterisk console: Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer: Peer

Hi, if I change Iaxphone settings to qualify=yes it says it's unreachable. Can iax2 clients be monitored with qualify option ? Is this problem related to iaxphone ? Anyone sucessfully using iax qualify feature ? Regards, Rob.

I would like to ask, if someone observe also problem with peer qualify problems, my asterisk log is full with UNREACHABLE/REACHABLE messages, even when two asterisks are in LAN environment, please take a look into this debug, I can't find any problem with packet loss, all qualify requests are replied and acknowledged, I will submit bug report, if you will also not find any problems here...

Hey, I just started trying to use the qualify=yes option on my Cisco 7960 SIP phones. Of the 13 I have, 2 of them seem to loose their registration with asterisk on a regular basis. I see lots of these lines: -- Registered SIP '3030' at 62.74.107.1 port 58825 expires 60 in my console. But I only see them for 2 extensions. Never see them for the other 11. All 13 phones have the exact same

Hi folks, I have a couple of observations about NetBIOS keepalive handling in Samba, in case anyone is interested. I came across a few oddities when investigating why some of our Samba servers were logging: lib/util_sock.c:write_socket_data(540) write_socket_data: write failure. Error = Broken pipe type errors on a very frequent and regular basis. We have a number of Solaris 2.6 machines,

I have an asterisk system behind NAT, and need to connect to public PSTN originators via SIP or IAX2, but don't have the option of forwarding any ports (4569, 5060, etc) to the asterisk system. However, the NAT system does properly establish transient UDP forwarding on the basis of outgoing connections, so is it possible to configure asterisk to send frequent keepalive UDP packets (say every

I have attached a patch which adds null packet keepalive functionality to the client. This patch is made against the current CVS tree as of 3/14/01. Please consider this patch for inclusion in the OpenSSH main tree. This patch is based upon and includes code from the Chris Lightfoot (chris at ex-parrot.com) patch posted 2/23. The original patch from Chris is at:

Hi, I think I encountered a bug in the qualify timers for IAX on asterisk 1.8 but I'd like to check if I'm not messing up in my config somewhere before reporting a bug. In my IAX peer configuration I have this: [remote-host] type=friend host=172.16.6.45 username=remote-host secret=test notransfer=yes qualify=16000 qualifyfreqnotok=30000 disallow=all allow=alaw allow=ulaw allow=ilbc

I'll begin working on full cross-version support (Asterisk 1.2, 1.4, and 1.6) in early May for nvfaxdetect and a handful of other modules. Justin Newman > >Hi, > >we are using the app_nvfaxdetect from Newman Telecom with Asterisk 1.4 >and tried to build the trunk/next release 1.6 with this application, but >it failed (We are using fax stuff with iaxmodem/Hylafax). >

Hi, I've got a problem with rtp keepalives. I'm using basically the same config on 2 hosts, but one of them sends rtp comfort noise when it's on hold, the other doesn't. The only difference I can think of now is that one of the machines is multihomed, but that might be unrelated. rtpkeepalive is set to 2 and I can confirm is by doing `sip show settings`. I've tried all

Hi, We have a few Aastra 480ci phones and we've noticed that in order to get the phone to receive a call, qualify must be = no. Apparently the Aastras do not respond to the qualify message (or respond in a way Asterisk doesn't understand) and Asterisk thinks the phone is unreachable. However, this now prevents MWI from working properly on the phones. Does anyone know how to get MWI

Name/username Host Dyn Nat ACL Port Status 2011/2011 10.1.1.10 5071 UNREACHABLE 2010/2010 10.1.1.10 5070 UNREACHABLE 2009/2009 10.1.1.10 5069 UNREACHABLE 2008/2008 10.1.1.10 5068 UNREACHABLE 2007/2007

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

When qualify is set to yes in sip.conf for a "friend" and the OPTIONS packet gets returned with an ICMP port unreachable message, what is the behavior of Asterisk? It looks to me like Asterisk tries sending the OPTION request again right away (well within a second or two). Some of our devices are being Linux firewalls that make use of iptables to do portforwarding. This generates

Hi! I know that is not SER discuss, but probably some of you faced with the same problem: to detect trunk status (ok/unreachable) in *, it's a must, to set qualify=yes as * connecting to SER, it's not replying to qualify messages, so even i can use it well without qualify, with qualify it's says unreachable immediately. What i have to set in SER to reply? Thanks! --

How frequent do the qualify packets go out? I've got 5 phones at a remote location with ADSL and their qualify is set to 5000. I've noticed that they will go "UNREACHABLE" and then immediatly go "REACHABLE" within a second or two. We have this theory that the ADSL link (SBC) "caches" its route to our network (where Asterisk is) and so we get bad ping

I am using Realtime on Asterisk 11.5 with a SQLite3 backend. While everything seems to be working fine I keep getting this error on my log files: [2014-02-17 19:55:18] WARNING[20569] res_config_sqlite3.c: Could not execute 'UPDATE "sip_buddies" SET "ipaddr" = '192.168.2.23', "port" = '5060', "regseconds" = '1392692118',

We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at nt_jnewman at yahoo.com. Justin