similar to: Qualify = UNKNOWN

Hello. I'm using Asterisk 1.4.22 with Postgres 8.3 in a Ubuntu 8.04 Server. I configured Asterisk to get sip from Postgres, and set qualify for all sips as yes, but the sip show peers command show the status of the peers as UNKNOWN srvcentral*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status Realtime 4900/4900 (Unspecified) D

[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'gerrie' Am I correct that when I turn on rtcachefriends in sip.conf, database-changes in my MySQL-DB will not be reflected untill a reload ?? Am I correct that when I turn off qualify in my realtime

Say, If bandwidth e.g. ADSL goes fuzzy, is there a way to force * to unregister the Peers? I noticed with qualify=200 for example, even if latency goes above and * shows Lagged and then UNREACHABLE The peer's calls are still accepted. Is there a way to automatically prevent this? Thanks Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm trying to run an ATA behind a NAT device, and am confused on exactly what the qualify config option does, other than send NOTIFY packets. Outbound calls work fine, but inbound calls do not go through. With qualify=yes and nat=yes, my show sip peers looks like: 7771111001/7771111001 10.0.0.10 D N 255.255.255.255 1222 OK (36 ms) So, it has established a

We have many IAXy devices in the field now. In all cases, in iax.conf, we have "qualify=yes", so that using "iax2 show peers", we can see whether or not the device is currently online. In some cases, the IAXy device and/or Asterisk are not communicating their qualification, because "iax2 show peers" shows the device as status UNKNOWN. However, when a user picks up

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

Although I'm convinced that Broadvoice doesn't have the most stable of ping times, it seems like I get ping results that are approximately the ping time +2000ms at times. Has anyone experienced this problem with qualify on a SIP connection before? So here, was the ping 20ms or 2020ms as reported? Aug 22 06:39:49 NOTICE[6964]: chan_sip.c:8481 handle_response_peerpoke: Peer

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

Hi there, I still have issues with the IAX connection between two servers (one static (server A), one dynamic (server B), none behind NAT): B registers with A, and "iax2 show registry" shows that everything is fine. However, after a while if I check on server A with "iax2 show peers" I see a status of UKNOWN (in iax.conf there is a qualify=yes statement for server B).

And after reload ALL your phones are unreachable for 2 minutes! Imagine you have several thousands devices unreachable for 2 minutes. How much calls will fail during that time? Regards, Mindaugas Kezys Kolmisoft UAB=20 VoIP Billing Solutions e-mail: info at kolmisoft.com URL: http://www.kolmisoft.com -----Original Message----- From: asterisk-users-bounces at lists.digium.com =

Hi, Yes, we're implementing the dialplan in realtime too. Here the contents of sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints aor=realtime,ps_aors contact=realtime,ps_contacts [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips Cheers, Francisco. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf

I'm using 1.8.3, and have 2 sip providers. Both are set with qualify=yes. Each of them sometimes have qualify times 10+ times the other. For instance, one will be at 10-15ms, the other at 200ms. Is there a way I can route an outgoing call to the provider with the lower qualify time? sean

Dears Friends, I currently have 16 Cisco SPA525g phones with a SPA500s (Attendant Console) connected to each phone. All of the 16 phones, have their Attendant Console configured the same way, where they are subscribing to each of the 16 phones. When I power on the switch, where all the phones are connected to, I then get 16 registers, and 256 subscriptions (16 * 16) happening at the same time.

Hello All, I manage to configure Asterisk RealTime and now it loads the SIP users/peers from MySQL DB. The table I am using is of A2Billing DB "cc_sip_buddies". Now the only problem I am facing is incoming calls are failing... The ATA which is assigned this DID number is behind NAT and according to Olle's explanations he said "*there's no support for NAT keep-alives

On 2016-08-09 10:06, Faheem Muhammad wrote: > trip time and Call Setup time of SIP Requests. > In case of GSM Network with high delay you need to set the T1 timer a > higher value like 1000ms (500 ms default). Similarly you can reduce the > Call setup time by configuring 'T2' upto you choice as per you telephony > network. Configure t1min, timert1 and timerb according to

Hello asterisk users, here is a little problem pulling out iax peers from real time database I have the following peer configured in my database mysql> select name,username,secret,type,context,host,disallow,allow,defaultip,deny,permit, ipaddr,port from iax_users where name='iaxtermination'; +----------------+----------+----------------------------------+------+-----

Hi, We have been migrating our PBX system from Asterisk 1.8 and chan_sip to Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have stumbled on a behaviour difference I don't like. With chan_pjsip when a phone went unexpectedly offline (Ethernet cable disconnected) Asterisk would detect this quickly (through the 'qualify' pings), mark the phone as 'Unavailable' and

We have a tenant who has been having issues with a congested connection and in trouble shooting it we've noticed that there seems to be a lot of SIP traffic even when none of the phones are doing anything. We've determined that this traffic is mostly INFO packets generated by setting qualify=2000. I understand that 2000 ms is the default value for the qualification parameter but what

In an earlier thread Andrew Kohlsmith enlightened me on the use of qualify in sip.conf to deal with a peer that is down. Since then I have been searching for information on how the behavior of qualify can be tuned. The wiki is vague on this; " Syntax: qualify=xxx|no|yes where XXX is the number of milliseconds used. If yes the default timeout is used, 2 seconds. If you turn on

Hi All, I'm connecting to my carrier which requires setting of outboundproxy. There has been few cases where the proxy server failed due to network issues and required us to use a secondary one. Is there a timeout or qualify setting for outboundproxy setting in sip.conf? I do appreciate if anyone can help please. Thank you -Abeed -------------- next part -------------- An HTML attachment