similar to: Sip registration period

Hi all, I've got something strange, that got me searching for quite awhile. Configuration as followed: Linphone on a laptop, that is connected via openvpn to a proxy. That proxy is connected with iax to another asterisk. On the second one i have several hard and softphones. Behaviour at first glance: >From the softphone i can allways set up a connection, But the otherway round fails 9

I was hoping someone can help with a problem with my GrandStream Budgetone "hanging" after a while. Problem seems related to the SIP registration - I am using dynamic registration (host=dynamic in sip.conf). Static IP is not an option in my case. I start Asterisk and all is groovy, phone works fine and can dial around. Then, at the time specified on the phone in the "Register

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

Hi, Does anybody know how often does IAX2 registration happen ? Also I'm getting a feeling that there is no way of changing it through an iax.conf file ? Thanks, Ivan

On Thu, 17 Feb 2005 15:04:50 +0100 Stefan Gofferje <stefan@gofferje.homelinux.org> wrote: > Hi folks, > > I'm registered with sipgate, a German SIP provider. >Configs works fine so far. Trouble is, after a while, it >seems, my registration is dropped by sipgate. How do I >tell * the interval for * registering with a provider? I >suppose, the re-registration

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

Thanks for the info so far. I am still trying to asterisk'ize my ML9.2 firewall box and can't get the external SIP registration to work. If I hook up my Sipura directly to the WAN it registers OK. This is the message I get from asterisk: Mar 8 21:03:07 NOTICE[196621]: chan_sip.c:3140 sip_reg_timeout: Registration for '263872@192.246.69.223' timed out, trying again If tried

Hi everybody, I'm trying to SIP register between two asterisk, each one have a Public IP. Asterisk told me that Unathorizae In * one sip.conf register =>usuario1:pass1@<public_ip_2> In * two sip.conf [usuario1] type=friend username=usuario1 secret=pass1 host=<public_ip_1> dtmfmode=inband Logs in * are the followings In * one logs: Sip

We have an Asterisk v1.2.16 box registering with an ITSP using SIP. The registration succeeds, and is confirmed with SIP SHOW REGISTER. However, we frequently (every few minutes) see this on our console: REGISTER attempt 1 to 999@pbx.itsp.com REGISTER attempt 2 to 999@pbx.itsp.com Any ideas what is going on? In particular 1. What causes the two register attempt messages above? 2. Why

I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to be exact). I get the X on the display sometimes for loosing registration. I have the config file for the 7912's SipRegInterval: 60 and asterisk is the default. ; maxexpirey=3600 ;defaultexpirey=120 I've not changed them. How can I keep these phones online and stop loosing registration? Thanks, Jerry

ekiga registration fails. I've set nat = yes ( also blank ) and i've set externip. Anybody have a sip.conf that works? Here's the sip debug: Reliably Transmitting (NAT) to 86.64.162.35:5060: REGISTER sip:ekiga.net SIP/2.0 Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport Max-Forwards: 70 From: <sip:test at ekiga.net>;tag=as64618445 To: <sip:test at

Hello, Don't know if this is related but I just got a segmentation fault today while trying to register my new SNOM200 phone: *CLI> *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '<sip:mattf2@10.10.10.15>' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from

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

Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports opened and 5060 forwarded to Asterisk (192.168.2.2) Can someone see why SIP-registration fails ?? register => 092779077:XXXX at 85.119.188.3 [3starsnet] type=peer host=85.119.188.3 username=092779077 secret=XXXX fromuser=092779077 fromdomain=sip.3starsnet.com dtmfmode=rfc2833 canreinvite=no insecure=port,invite qualify=yes

Hello All, I am trying to get some ATA 188 units to register with my Asterisk box over SIP. I continue to get the same "401 Unauthorized" Error when they try to register. If I turn Sip registration off, I can use the phones without any problems with a static IP assigned in my sip.conf file, but I can't get the second phone port working. I've set up two separate logins both

I'm having problems getting Asterisk SIP to register with an Entice softswitch SIP Gateway. My provider tells me that all thats needed is a user name, password and the IP address and to register and it needs to be using MD5 authentication. I continualy get a "603 Decline" message. The provider of the gateway says they are not receiving any authentication information. Registration

Hello list, I am new to this list and have been using Asterisk as part of my research project for about 2 weeks now. I would like to get your thoughts on a scenario that I am attempting at the moment. I haven't had luck until now. In this scenario, I am trying to register a SIP user configured on the zoiper client installed on a laptop, which is on the same Local Area Network, with the

I'm trying to build a test suite so that I can run "calls" through and verify the call results. I've made a cross over cable and linked my 2 ISDN30 ports together. So now I can dial out on span 1 , and to receive the call on span 2. in the context for span 2, I have the following: <snip> ; #1 "answer" a call and play music 000XXX : ring for a random period,

I have a Hitachi WirelessIP5000 phone (the original 1st gen hardware). It's running the latest firmware (v2.2.6) and my Asterisk server is running 1.2.10. This setup has been working great for me for a long time, and then last week I started having a problem where the Hitachi phone loses registration. It registers fine when I first power it on, but at some point later the display shows

Hi, We are having major problems with a remote site that links to the head office via a VPN tunnel. The phones will register fine and work for a few minutes to hours but then will drop their connection and will no register to asterisk even with a restart of the phone. We have 2 other remote sites that work exactly same and they are not having any issues so i believe it has to be be something