similar to: sip.conf - register and peer groups

sip.conf.sample: ;register => 2345:password at sip_proxy/1234 ; ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider ; connect to local extension 1234 in extensions.conf, default context, ; unless you configure a [sip_proxy] section below, and configure a ; context. sip.conf: [general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no

Hi, to register my Asterisk with a SIP provider I use the following syntax, as shown in the default sip.conf: register => 2345:password@sip_proxy where [sip_proxy] type=peer context=from-messagenet host=sip.messagenet.it port=5061 <------------- please note this one!!! 5061 is provider's port I have to register to. This also would work for me: register =>

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

I am new to asterisk and need to dig up some info on how to set it all up. It looks a bit daunting especially all the options available in the .conf files. I have 2 SIP phones, GXP2000 and a budgettone 100. phone1 - 192.168.0.160/24 extension 1000 phone2 - 192.168.0.161/24 extension 1001 Server - 192.168.0.57 I get the following all the time, but can make calls between the 2 extensions,

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on

This may be a newbie SIP/NAT question. If so I am sorry. But any help would be appreciated. My Asterisk server is behind an ipchains box and I am trying to connect to Broadvoice. All works fine without the NAT. I have a global nat=yes prior to my register, but the sip debug allows shows "no nat)". Is this "label" issue, and am I barking up the wrong tree? Sip.conf....

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

I am running asterisk 1.0.1 with OH323 compiled in. We have a 323 trunk to CallManager with a mgcp controlled pri router. When using sip phones (directly registered with asterisk) to call out the 323 trunnk to PSTN, calls timeout after 3 rings. If I answer b4 3 rings - no problem, otherwise I get "no one is available to answer at this time" on the consoel and it redirects to an

HELP! Ok, so I have the following SIP.CONF: [general] context=default port=5060 bindaddr=10.1.1.200 externip = XX.XXX.XX.XX localnet=10.0.0.0/255.0.0.0 disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw register => ##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234 [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromdomain=sip.broadvoice.com

Hey, Im running Asterisk 1.2.2 and im having problems with the audio when bridging calls between the zap interfaces and sip. zap to zap work fine, as do sip to sip (but asterisk isnt in the media stream, as it doesnt need to be) and terminating the call and playing a test message via either sip or zap work fine. Basically, the only time I see this problem is trying to bridge between sip and

I want to setup a blind transer for auto forwarding to SIP peer. I have context forwarding looks like this in extension.conf [forwarding] ... exten => 511,1,Dial(SIP/sip_proxy-out) ... This will do the re-invite, which is attendance transfer maybe. But I want a blind transfer by REFER method. How can I do that? I know that the transfer() function may be able to do that. But I don't know

Hi, I have been trying to configure my Asterisk to use a Sip provider for out and incoming calls. I only have one user and password for connect to my sip provider. My sip.conf is: [general] ;disallow=gsm ;allow=ulaw port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls callerid=No

In a 1.2 release of asterisk, I've had no problem connecting to a Broadvoice SIP peer, to allow routing outgoing calls from Asterisk to Broadvoice. Now, with the same SIP configuration, I cannot establish the peer. I've enclosed a SIP log in the hope that someone can help me analyze this failure. I'd guess the issue is NAT related and wondering if someone can spot a problem in the

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

> > I've tried setting up my sip.conf in two ways: > > > ------------------------------------------------------ > register => [240xxxxxxx]:[my_password]@sip.broadvoice.com > > > [Broadvoice] > type=peer > username=[240xxxxxxx] > fromuser=[240xxxxxxx] > secret=[my_password] > host=sip.broadvoice.com > context=incoming >

Hi All Total noob on the list so all help appreciated.... I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones.... The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......

I am having trouble setting this system up and wonder if some one help me. Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other. I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2 extensions 5060/5061, this is on the lan side of my gateway/router WRT54G 192.168.1.1 BusyBox v1.00 (2006.11.07-01:40+0000)

Hi all, We seem to be having an interesting issue maintaining our registrations with Broadvoice. It seems to be related to the fact that Asterisk does not normally include authentication with its first REGISTER attempt. Normally, Broadvoice doesn't care, responds with a 401 Proxy Auth Required, and asterisk answers accordingly. However, if the account is already registered, Broadvoice

Where does the date/time stamp from Caller ID come from? On my extensions ATA188 and IAX2 soft phone the caller id date / time is 12/30 12:00AM. The Linux time is correct. SayUnixTime return the correct time. Any Ideas? Does this work? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm using Asterisk 1.8.6.0 on my router talking to my ISP's Taqua 7000 (?) switch. I'm using a config that looks like: [sip_proxy-out] type=peer authuser=208nnnnnnnn remotesecret=xyzzy qualify=100 host=n.n.n.n call-limit=5 nat=no ; sendrpid=yes insecure=no But the Taqua responds to outbound INVITES with 403 Forbidden (oddly, not 401 or 407). Also, from what I can tell, the