similar to: number 'register => ' in sip.conf

Hello list, scrambling different pieces of info together I've come with the following : I want to have my "register =>" statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:password at sip.provider.net In ext_config

What is the relationship between the peer definitions and the register command? In reviewing the sample sip.conf it seems that you can place the "sip_proxY" peer as the hostname. Is this correct? This question adds the the Broadvoice thread and where to place the dtmfmode variable. sip.conf --- (asterisk sample) -------------------------------- ;register =>

Hi All, I am trying to setup another sip trunk in addition to what I am already using. The sip provider we are using right now gives you your username as your email address. So IE. If my email is james@james.com.... that is my username . Now... When I put this in the sip.conf file I have found that Asterisk is not able to parse it correctly and instantly goes to the email server to authenticate

Hello, I recently upgraded from Asterisk 1.4.0 to 1.4.15... I am registering to a sip provider in my sip.conf as below [general] register=>user:password:authuser at serviceprovider/extension Later down in my sip.conf I have the definition for that service provider as follows [serviceprovider] type=peer host=x.x.x.x port=yyyy outboundproxy=t.t.t.t . . . With asterisk 1.4 it would know to

Asterisk cvs-head (up to date) keeps core dumping on me. I finally tracked it down to my register command for Vonage in the sip.conf file. If I remove the username and password from the register command, it won't core dump, but of course won't register either... This is odd. Any suggestions?

Is there any reason why we wouldn't want to add a setting for each REGISTER => line in sip.conf that allows us to set refresh time on a per-proxy basis? If not I thought I would try to add that functionality to chan_sip. Please tell me if there's some reason why we wouldn't want to do this, (e.g. it's coming in a future CVS or in chan_sip2 or it will horribly break the code

Mark has fixed the REGISTER issues to be more RFC compliant. I've created a new thread so that those of you who got bored with the old thread might read this new one. The feature that has just been added was added a while ago, but now it actually seems to _work_. :-) If you have a SIP server to which you are trying to REGISTER, and they demand valid domain (the part after the

Hi, the documentation of sip.conf is telling me this: ;register => 1234:password@mysipprovider.com ; ; This will pass incoming calls to the 's' extension In reality it jumps to the extension 1234 in the context and not to s So it is much more complicate to write an proper dialplan. Is this an bug or is the documentation not up to date? best regards Thomas

I'm registering 5 lines on my asterisk box from one voip provider. Lines; 4040.0000 4040.0001 4040.0002 4040.0003 4040.0004 All lines is registered in 5060 port so when someone call to 4040.0001 the call arrive on asterisk but arrive to last number registered 4040.0004becouse it is listening on same port as all others. How i make each number register in one different port, like

Hi all, My experiment scenario is like this: SIPp Uac -----------------------------> ASTERISK SERVER---------------------------------->SIPp uas 1. when i had registered bob with this command ./sipp -sf register_client.xml -inf register1.csv -i 192.168.1.6:5060 192.168.1.6 -p 5061 -m 10000 it has registered.... If i want to register another client alice with same command

I am talking about the SIP. Now the new mobiles (Nokia, Erecson, Panasonic, iPot, ... etc) all of them support SIP capability. They are able to register to any SIP server (by giving the IP address, username and password). Fring is one of the software that can be installed on the mobile devices and can register on the SIP servers. BUT, the new mobiles currently come with built in SIP (no need to

Hello everybody, I am having a problem with realtime SIP peers. On Asterisk 1.8, I had SIP peers for external SIP providers configured in database and additional register lines in sip.conf so they would register. Now I upgraded to Asterisk 11.3.0, partly because of the promised callbackextension feature for realtime peers (https://reviewboard.asterisk.org/r/1717/). Removed the 'register'

I can not register via sip to iptel or sipgate. i do not unterstand why.. but i am new to asterisk. Iam behind a susefirewall2 but asterisk even do not register if it shut down. No answer seems coming back. thx for help. nico here is my config if anybody can help: ----------------------------------------- [general] port = 5060?????????????????????; Port to bind to bindaddr =

Hi, I'm getting this error when registering with SIP server using Asterisk 1.4.10 and Freepbx... I'm getting this error no matter what I try to setup in sip.conf : - I'm getting confused whether options are maxexpirey=36000 or maxexpiry=36000 ? - Can I solve this with some settings in sip.conf or is this problem harder ? - I've read something about Asterisk's bug on this

hi I keep getting sound dropouts during SIP re-registration, and I can't find a remedy for it. I use SIP friends from MySQL. Below is SIP debug output for the re-registration Thanks in advance roy ------- *CLI> sip debug ip 80.202.161.221 SIP Debugging Enabled for IP: 80.202.161.221 *CLI> Sip read: REGISTER sip:sipgw1.briiz.no SIP/2.0 From:

I am seeing some very peculiar things in the routines that REGISTER my * server with several accounts. I saw this on my console: . . . NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout):

Some of you have unusual SIP configurations, and this SIP perl script may be useful to get remote devices registering with your Asterisk or other SIP server. Most Cisco routers, as an example, are too stupid to REGISTER, so this script would be required to dynamically register them with a remote server. This may not be 100% applicable to Asterisk, since static registrations are possible,

The SIP trunking service that I am trying to set up keeps saying that my registration from Asterisk is invalid. Asterisk registration: REGISTER sip:{registration_ip} SIP/2.0 Via: SIP/2.0/UDP {asterisk_ip}:5060;branch=z9hG4bK5c2eb10c;rport Max-Forwards: 70 From: <sip:{registration_user}@{registration_ip}>;tag=as5579cc0c To: <sip:{registration_user}@{registration_ip}> Call-ID:

Hi, when i include a sip configuration from another file in my sip.conf using #include /etc/asterisk/sip-sipgate.conf everything seems to be working. The peer is listed when i execute "sip show peers" and Status is "OK". But the peer is not listed using "sip show registry". I need to place the "register => ..." in the sip.conf to make it work. Is this

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mohammed Firdosh Nasim Sent: Tuesday, March 15, 2005 11:08 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] How to register two SIP phones ( e.g. WindowsMessenger) from different subnet to * On Sat, 2005-03-12 at 07:42, Luki wrote: >