similar to: Friend/user/peer in plain English?

Hello I'm having a difficult time finding precisely what to put in sip.conf and extensions.conf (and possibly other files) to get a working configuration to connect an Asterisk (1.4) server to a VoIP provider with the Asterisk server + SIP clients located in a private LAN behind a NAT router: http://img560.imageshack.us/img560/3749/asterisknat.png Would someone have a full, direct (ie.

Hello At this point, I have an Asterisk 1.4 + PC running XLite behind a NAT set up with a VOSP trunk that I can use to make/receive calls to/from the PSTN. Now, I'd like to be able to call any number on the Net that is advertised as "sip:user at domain.com", such as those: www.voip-info.org/wiki/view/Phone+Numbers Do I need to register a second trunk (FWD, etc.) through which

Hello I don't have a landine and use a VOSP to provide access to the telephone network. In case a call comes in and I'm not home, I'd like Asterisk to log the call, and then send an SIP message to my VOSP so the call is forwarded to my cellphone and is thus charged to the caller, without Asterisk having to dial out to my cellphone through my VOSP at my expense and bridge the two

Hello I've never used Asterisk for a three-person call, and would like to check that MeetMe is the way to do this. The ADSL modem provided by my ISP offers free calls to landlines/cellphones when using a handset connected to an RJ11 port on the modem. A three-person call can be set up by using the standard PBX sequence: 1. Using the handset, call party #1 2. Hit "R" key on

Hi guys, I configure one Fedora Core Linux 5 for use with asterisk as gateway using Digium TE110P interconected in Alcantel 4100 I've set up it to register 100 voip numbers on my provider. All calls on Alcatel is send to asterisk. In some periods of day i receive this messages on asterisk console: Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer: Peer

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

On 11.9, trying to set up a callcentric peer: sip debug: > <--- SIP read from UDP:204.11.192.161:5060 ---> > INVITE sip:1777<myccid>@10.10.11.180:5060 SIP/2.0 > v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46aaaaef4249814d16a2ffb990d > f: <sip:<calling number>@66.193.176.35>;tag=3606475083-968127 > t:

Hello This is a newbie question : With a simple Asterisk server on a private LAN, an FXO port to handle the PSTN, and an ADSL connection to the Net, ie. with no VOSP in the mix... how should I configure Asterisk so that SIP clients can dial SIP numbers on the Net, such as those below to perform an echo test? www.voip-info.org/wiki/view/Phone+Numbers I tried pasting numbers in XLite, but nothing

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Hello, Is there anyway to encrypt or scramble a bit the secret used to register with a provider? Im talking about the register => fromuser at fromdomain:secret at host directive in sip.conf<http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf> This clever dude modified the code back in 1.4: http://www.oneharding.com/voip/asterisk_md5_register.html I imagine that so many years

Hi, I have a few accounts with sipgate.co.uk to get some different DiD numbers. However, when an incoming call comes in, it seems to pick the wrong peer from sip.conf which sends the call into the wrong context and it fails because there is no extension in that context to match the register. Using the config's below, if I dial the DiD on account 2222222, it works fine - picks peer 2222222

Hello I was wondering if someone knew of an application that could check that the user has a firewall and a broadband connection that will work OK with Asterisk and VoIP. The app would first perform some bandwith + jitter tests, and will then call a STUN server to check that the firewall isn't symetric. BTW, is Asterisk now STUN-capable, or is it still to map ports manually on the firewall

I have an odd problem. I have just installed asterisk on an ubuntu box, and migrated the previous configuration of asterisk (on another ubuntu box) to this new server (scp -pr xxx at oldserver:/etc/asterisk/* /etc/asterisk/) Asterisk worked fine on the old server, but on this server my SIP trunk peer does not login after initial server startup. "sip show peers" shows my phones

Hi all, I'm running Asterisk SVN-trunk-r8643M and face following problem: I'm trying to get incoming call from a provider and calls ended with a 404 error. On the INVITE I get "Found no matching peer or user for <IP address>:5060" and then "Looking for <UserName> in <SIP default context> (domain xxx.xxx.xxx.xxx)". My question is why asterisk

Hi list, I try to connect to a GW which have one domain eg sip.mydomain.com and have few IPs related to this domain. I register * to this domain with host=sip.mydomain.com and type=user. So DNS will decide on which IP of my domain I will register (or redirection on the GW side). If an incoming call arrive, I would guess that, as type=user, it will not try to match the IP from INVITE as I

Hi All, I need to limit outgoing calls in my sip peers... I tried to use "call-limit=1" in these peers in the sip.conf, but it didn't work... Here is my peer configuration in the sip.conf: [sip.broadvoice.com] accountcode=broadvoice type=peer dynamic=yes username=MYUSERNAME fromuser=MYUSERNAME authname=MYUSERNAME user=MYUSERNAME secret=xxxxxxxx host=sip.broadvoice.com

Turn on sip debugging for this peer and watch for the options sending and response. If you are getting a response to your options asterisk shouldn't be marking the peer as unavailable. is your asterisk behind a firewall? On 20 March 2015 at 13:42, thufir <hawat.thufir at gmail.com> wrote: > I wasn't able to get much out of babytel, beyond the fact that I was, > apparently,

Hi, I'm trying to create a link between two PBXs. One is Asterisk 1.4.15, the other is an unknown 3rd party PBX. In my internal testing, beween two A*k servers, I found that if I created two sip accounts from the same IP, one as peer and one as user (intending to give an -IN and -OUT setup), then inbound calls always seemed to route via the -OUT account and failed. My fix was to use

I wasn't able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why I'm not getting 200 OK. How can I, generally speaking, ping/telnet or otherwise test the connection to get more data? A connection to this peer directly from a softphone, Jitsi, works fine. linux-k7qk*CLI> linux-k7qk*CLI> sip show peer testcarrier * Name :

Hi Everyone, I was hoping someone might know why I am experiencing a problem with Asterisk logging the event: [May 3 12:07:41] WARNING[30371] chan_sip.c: Maximum retries exceeded on transmission 03f007af2b15cd0b54b0c368265d97be@sip.externalprovider.com for seqno 669371069 (Critical Response) This is happening after: - call is setup, 2 way audio - call can function correctly for up to 5