similar to: authentication time for asterisk server

Hi All, I have one asterisks server and 3 client (i'm using voip sip client for my handphone). I've configured sip.conf and extension.conf with 3 user different. And nothing wrong with them, i could make them to make a call too. what i want to ask is, i was try to use 1 user (ex:1001) in 2 different client. example: client 1 (1001) make a call to client 2 (1002) --> ok then in client

Hi all, I want to make initial VoIP authentication process from asterisk server to be based on EAP-SIM authentication of Freeradius server (so it will be not necessary to insert account datas in the asterisk database). Is there any way of doing that from Freeradius and Asterisk? Or at least, is there any way to sync the EAP-SIM data on Freeradius to asterisk server? thank you -------------- next

I am trying to connect two asterisk servers using the information from: http://www.voip-info.org/tiki-index.php ... +2+servers<http://www.voip-info.org/tiki-index.php?page=Asterisk+Connect+2+servers> <http://www.voip-info.org/tiki-index.php?page=Asterisk+Connect+2+servers> It works fine with Method 1. If I use method 3, I get errors: on sending server: Registration of

This week starts with the exciting news: We're getting close to Asterisk 1.0 again. After the failed attempt earlier this year, we've been able to remove a lot of the MAJOR/CRASH bugs from the bug tracker and Mark feel's it's time to target 1.0 again. At this point, the community needs to work as a community, spending extra time on finding bugs, solving issues, improving

Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16

Hi, I'm using Asterisk 1.2.9.1. I have big problem with SIP VoIP providers registrations: Asterisk freezes when it cannot (re-)register with VoIP provider (registration timeout). The problem is related to DNS names resolution: if DNS server is very slow to respond Asterisk stops every activity (no zap or restart commands on CLI). The bad news is VoIP providers usually do not give their IP

hi, i want use asterisk+pjsip as voip client with multiple registrations (perf testing) i'm using this example configuration for one account [308] type=registration outbound_auth=308 server_uri=sip:308 at example.com:5060 client_uri=sip:308 at example.com:5060 [308](auth-userpass) username=308 password=pass [308](aor-single-reg) contact=sip:example.com:5060 [308](endpoint-basic)

Sunday news is today published on a monday. Yesterday was fourth of july, and I used that as an excuse for being off line yesterday. (Sweden's national day is June 6th - and it's not yet a public holiday, btw). Most of my Asterisk time lately have been used for producing the registration site for Astricon and tracking down speakers that haven't sent in their material for the conference

I have to register asterisk/sip with a sipproxy that does not support authentication, I do not know how to tell Asterisk not to send authentication request? # sip.conf [general] insecure=very permit=207.148.115.10/255.255.255.0 [myproxy] type=friend host=217.118.115.10 context=from-sip # Logging: <--- Reliably Transmitting (NAT) to 207.148.115.10:5060 ---> SIP/2.0 407 Proxy

I saw several threads requesting an Asterisk forum to complement the email list. i.e. http://lists.digium.com/pipermail/asterisk-dev/2004-February/003103.html I recently created an Asterisk forum within TMC's popular VoIP forums for everyone to use. http://voip-forum.tmcnet.com/voip-forum/forum/forum_topics.asp?FID=15

During debugging of H323 trunk side (using Jeremy Macnamara's H323 driver in ~/channels/h323) a couple of things come don't quite work as advertised... 1/ the following line in extensions.conf explicitly sets the outgoing caller ID (required in my case for downstream GK processing..) exten => _1NX.,1,SetCallerID,6400047602100 exten => _1NX.,2,Dial,H323/${EXTEN:1} what

Calling all SIP gurus-- I'm trying to register my asterisk to an ISP's SIP gateway. I'm getting authentification errors. Here's the results of SIP DEBUG against it's IP. [I've tweaked all confidential fields so as to protect the innocent (namely, me).] --- (9 headers 0 lines)--- Responding to challenge, registration to domain/host name myfavoriteisp 12 headers, 0

Hi, Is it possible asterisk could provide voip client have auto authentication ability? so in the voip client we don't have to input user and password and what we have to input in voip client just ip gateway? Just in the voip client, because in asterisk we have to input user and password in the extension file. have anyone tried like that before? or at least there's any way to do like

Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The

Hi, I managed to set up a Samba server that accepts Kerberos 5 TGTs via SPNEGO/GSSAPI for login. However, when I don't have a TGT it fails for Unix clients. It asks for username/password for Windows clients and then fails trying to do NTLMv2 authentication. How can I set up a Samba server that asks for username/password and then uses a Unix Kerberos KDC (Heimdal v. 1.2 in my case) for

Hello All, I've been trying for some time to get Asterisk to register with a remote SIP gateway. I?ve recently managed to configure an SJ Phone to work with W2000 so know the configuration parameters work correctly. Asterisk doesn't authenticate properly and I notice that the authentication request appears different to SJPhone's. Do any tools exist to enable me to check these

Hi everybody, I've got a very strange problem: As far as I remember I didn't change anything on my Asterisk side. I have 2 SIP providers to which I can place outbound calls. Today I noticed that outbound calls to provider "inode" fail (and inbound from this provider too). On the CLI I get every 20 seconds following messages: Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422

Hi, I am trying to setup asterisk so that anyone from any IP can call using any callerid as long they have an account - also no registration is required. However, it seems like asterisk tries to find peer based on either the IP address or from header. What I really want is asterisk to find account/peer based on username passed as part of the authentication and NOT from the IP address or the

Hi, I have Asterisk 1.4.10 under LMCE (upgrade is not an option) and have this strange error appearing in full log : [Apr 16 14:35:48] NOTICE[10802] chan_sip.c: -- Registration for 'NUMBER at voip.siol' timed out, trying again (Attempt #22) [Apr 16 14:35:48] WARNING[10802] chan_sip.c: No such host: voip.siol [Apr 16 14:35:48] WARNING[10802] chan_sip.c: Probably a DNS error for

Hello, We need Asterisk to register with a variable and changing number (hundreds) of VoIP providers, is there a way to do this in a database and without reloading the entire sip config? Where Asterisk needs to register is determined by downstream users, so we need to do it real time and with minimal impact on the server. If Asterisk can't do this, is anyone using anything else to