similar to: Asterisk 1.8 - SIP losing registration

On Thursday 16 Jun 2005 09:25, Mark Brown wrote: > Hi Everyone, > > I'm using Asterisk, actually A@H 1.1 with all Grandstream 102 phones. > NAT is not an issue as all including the server have public IP's > > The problem is that the phones keep losing registration with the server. > I have not timed this exactly to see if they drop off with exactly the > same

Hi Everyone, I'm using Asterisk, actually A@H 1.1 with all Grandstream 102 phones. NAT is not an issue as all including the server have public IP's The problem is that the phones keep losing registration with the server. I have not timed this exactly to see if they drop off with exactly the same frequency. The SIP TRUNK connection to my provider SIPGATE does not lose registration, and

For all thos Asterisk users not on the FWD list, it works for me!: -----Original Message----- From: Free World Dialup - The Future of Dialing [mailto:FWD@LISTSERV.PULVER.COM] On Behalf Of Leonidas Piagkos Sent: donderdag 12 juni 2003 0:58 To: FWD@LISTSERV.PULVER.COM Subject: Re: [FWD] FWD losing Registration Hi Don, All you have to do with your ATA is to set the following parameters as :

All, I'm having a problem with an Innomedia 6328-2Re (old Sunrocket Gizmo). It keeps losing registration after a period of time ranging from a few minutes to a few hours. It seems that right before it loses registration, it fails to send a second register (after the 401 unauthorized). Here's a transcript from wireshark (at the end). The last message is all that's received and

Now that I have most everything actually working I've noticed that about every 3-4 days on average..... and at worse... Once a day my asterisk box seems to lose it's registered state with our sip provider and no longer will take any incoming calls. The caller simply hears a fast busy (reorder) If I do a reload at the command prompt all is well for another few days..... What I'm

How can I monitor asterisk if all lines are registered etc? I have an asterisk on a remote location and sometime they reporting problems that phone is not ringing, they can not dial out etc. Usually I just restart asterisk and it solves the problem. Is there an application that will email me if case any line looses registration with with asterisk? Or any better solution! -- Joseph

Hello, I have a local Asterisk server that keep loosing its registration to main Asterisk server. The local asterisk server is on the local subnet, it acts as a client with extension 808. Local server Sip.conf register => 808:password at as2.xxxxx.com registertimeout=20 registerattempts=10 Main Asterisk Server sip.conf [808] type=friend context=sip-phones call-limit=99

On 02/15/2018 04:08 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:03 PM, thelma at sys-concept.com wrote: >> On 02/15/2018 03:44 PM, Joshua Colp wrote: >>> On Thu, Feb 15, 2018, at 6:43 PM, thelma at sys-concept.com wrote: >>>> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports >>>> >>>> IN audocodes setting I

When I call my internal extension the phone rings only once and goes to voicemail. It suppose to ring for 30sec. before calls transfer to voicemail; -- Executing [315 at internal:1] Dial("SIP/11-00780380", "SIP/218|20|rw") in new stack -- Called 218 -- SIP/218-00786910 is ringing -- Nobody picked up in 20000 ms -- Executing [315 at internal:2]

Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple

On 02/15/2018 03:44 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 6:43 PM, thelma at sys-concept.com wrote: >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports >> >> IN audocodes setting I have: >> "EndPoint Phone Number" >> >> Channel: 3 phone number: pstn-4444 >> Channel: 4 phone number: pstn-9998

Hi folks, Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9 to 1.8.13, my server is no longer able to register a connection to a SIP account at my ISP (XS4ALL in the Netherlands). At the same time, it is still able to register a different account with another SIP provider, so it must be that they no longer have the same basic requirements. The relevant part of my

On Sun, Mar 15, 2015 at 1:33 PM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > Yes, I think the dial does get executed (sonny calling outbound > 202-555-1212): > > core set verbose 3 > Console verbose was OFF and is now 3. > -- Executing [912025551212 at from-internal:1] > Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from

Hi, Thank for your answer. 22.04.2019 23:47, Joshua C. Colp пишет: > On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: >> Hi, >> >> Got problems with incoming SIP calls. >> >> Scenario: >> >> Server1: 3cx or any other server >> >> Server2: Asterisk 16.2.1 . PJPROJECT 2.8 >> >> Server2 registers on Server1 with SIP ID 1121.

On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > That was the issue, thanks. I now am able to get the caller ringing on an > outbound call, but an external phone number (E164) I am dialing does not > ring. > Any error messages? If you set 'core set verbose 3' and try it, does the Dial get executed? > > On Sun, Mar

I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports IN audocodes setting I have: "EndPoint Phone Number" Channel: 3 phone number: pstn-4444 Channel: 4 phone number: pstn-9998 When I am calling " pstn-4444" the port number "Channel:3" lights up but asterisk is showing that the call is coming on "pstn-9998" -- Executing .....

I have one Digium adapter S101I on a local network and I'm losing the connection periodically. I'm using Asterisk-1.2.21.1 There is no pattern, sometimes it stays connected for a weak sometimes longer. In comparison I have few Sipura adapters and they stay connected for months (never lost single connection). -- #Joseph

Hi all, I have installed X-lite client on a windowsXP machine and asterisk on an enterprise linux m/c. The client is sending a registration message to asterisk server. It is able to process the message and sends 200 OK back. But later it says "Scheduling destruction of sip dialog xxxx ". Then it says "Really destroying sip dialog xxxx". What to do for this problem??? I

That was the issue, thanks. I now am able to get the caller ringing on an outbound call, but an external phone number (E164) I am dialing does not ring. On Sun, Mar 15, 2015 at 12:19 PM, George Joseph <george.joseph at fairview5.com > wrote: > > > On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> I have setup my

Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I have an intertex ix66 which up until the CVS update allowed me to register my * server with the ix66 for my local domain (eg sip.mydomain.com). Now it appears that asterisk gets totally confused and tries to register with itself! Anyone got any