search for: 00493512222222

....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 => 00493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=luca_incoming outboundproxy=17...

...t; > changes my ISDN to VoIP... > > Don't worry, Asterisk works very well with Deutsche Telekom and there > new ip-based connections ... That's a good news... Currenty I configured my sip.conf so: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register => 00493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYVERYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=luca_incoming outboundpro...

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in AsterikNOW **AND** Messagenet - 2 VoIP phones, logged into Ubuntu-PBX (my phone, my wife's...

...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 => 00493513333333:MYSECRET at pbxanika/00493513333333 > register => 4444444444:MYSECRET at messagenet/4444444444 > > [pbxluca] > type=peer > defaultuser=00493511111111 > secret= MYSECRET > dtmfmode=rfc2833 > host=172.16.3...

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

...s codec priority list -seems to be GSM on the first place. Luca Bertoncello <lucabert at lucabert.de> wrote: >Hi list! > >I noticed that when the phone of my wife calls the gsm codec will be used, >but if someone calls the phone, alaw will be used: > >00493511111111 calls 00493512222222: >OpenWrt*CLI> sip show channels >Peer User/ANR Call ID Format Hold Last Message Expiry Peer >192.168.200.11 00493512222222 5305ad0e07977dd 0x4 (ulaw) No Init: INVITE 0049351222 >192.168.200.10 0...

Hi list! I noticed that when the phone of my wife calls the gsm codec will be used, but if someone calls the phone, alaw will be used: 00493511111111 calls 00493512222222: OpenWrt*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 192.168.200.11 00493512222222 5305ad0e07977dd 0x4 (ulaw) No Init: INVITE 0049351222 192.168.200.10 00493511111111...

...(Unspecified) D 5060 UNKNOWN messagenet/1234567890 212.97.59.76 5061 Unmonitored pbxanika/00493511111111 172.16.34.132 5060 Unmonitored pbxfax/00493513333333 172.16.34.132 5060 Unmonitored pbxluca/00493512222222 172.16.34.132 5060 Unmonitored 8 sip peers [Monitored: 1 online, 3 offline Unmonitored: 4 online, 0 offline] Asterisk connects to another Test-VM with AsteriskNOW and to the italian provider Messagenet. Can someone suggest me, what can I do? I can send the configuration fil...

...moores.ca> schrieb: > I'd start by turning on sip debugging in asterisk > >sip set debug ip [your_phone_ip] Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172.16.34.133' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.11:5060: OPTIONS sip:00493512222222 at 192.168.200.11:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.34.133:5060;branch=z9hG4bK13db26f5;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk at 172.16.34.133>;tag=as1215345d To: <sip:00493512222222 at 192.168.200.11:5060> Contact: <sip:asterisk at 172.16.34.133> Call-...

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone

...'d start by turning on sip debugging in asterisk > > >sip set debug ip [your_phone_ip] > > Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172. > 16.34.133' Method: OPTIONS > Reliably Transmitting (no NAT) to 192.168.200.11:5060: > OPTIONS sip:00493512222222 at 192.168.200.11:5060 SIP/2.0 > Via: SIP/2.0/UDP 172.16.34.133:5060;branch=z9hG4bK13db26f5;rport > Max-Forwards: 70 > From: "asterisk" <sip:asterisk at 172.16.34.133>;tag=as1215345d > To: <sip:00493512222222 at 192.168.200.11:5060> > Contact: <sip:asterisk a...

...terisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html Thanks a lot, but it does not seems to work... Here my configuration: sip.conf: [general] allowsubscribe=yes subscribecontext = default notifyringing = yes notifycid = yes callcounter = yes extensions.conf: [anika_incoming] exten => _00493512222222,hint,SIP/00493511111111 exten => _00493512222222,1,Verbose(2,Call for Anika - [${CALLERID(num)}]) exten => _00493512222222,n,Dial(local/2222222 at anika_incoming) exten => _03512222222,hint,SIP/00493511111111 exten => _03512222222,1,Verbose(2,Call for Anika - [${CALLERID(num)}]) exten =...

...(default is *8) ;pickupsound = beep ; to indicate a successful pickup (default: no sound) ;pickupfailsound = beeperr ; to indicate that the pickup failed (default: no sound) my users.conf: [general] callgroup = 1 pickupgroup = 1 my extensions.conf: [anika_incoming] exten => _00493512222222,1,Verbose(2,Call for Anika - [${CALLERID(num)}]) exten => _00493512222222,Set(CHANNEL(pickupgroup)=1) exten => _00493512222222,n,Dial(local/2222222 at anika_incoming) exten => _03512222222,1,Verbose(2,Call for Anika - [${CALLERID(num)}]) exten => _03512222222,n,Dial(local/2222222 at ani...

...(Unspecified) D 5060 UNKNOWN > messagenet/1234567890 212.97.59.76 5061 Unmonitored > pbxanika/00493511111111 172.16.34.132 5060 Unmonitored > pbxfax/00493513333333 172.16.34.132 5060 Unmonitored > pbxluca/00493512222222 172.16.34.132 5060 Unmonitored > 8 sip peers [Monitored: 1 online, 3 offline Unmonitored: 4 online, 0 offline] > > Asterisk connects to another Test-VM with AsteriskNOW and to the italian > provider Messagenet. > > Can someone suggest me, what can I do? >...

...Unspecified) D 5060 UNKNOWN > messagenet/1234567890 212.97.59.76 5061 Unmonitored > pbxanika/00493511111111 172.16.34.132 5060 Unmonitored > pbxfax/00493513333333 172.16.34.132 5060 Unmonitored > pbxluca/00493512222222 172.16.34.132 5060 Unmonitored > 8 sip peers [Monitored: 1 online, 3 offline Unmonitored: 4 online, 0 offline] > > Asterisk connects to another Test-VM with AsteriskNOW and to the italian > provider Messagenet. > > Can someone suggest me, what can I do? >...

Hi again! With the "call pickup"-function I can now pickup a call directed to another phone in my Asterisk. Very nice. My problem, now, is that I can't see on my phone, that the other phone (in another room) rings. Is it possible to signal the incoming call on other extension? I use two phones "Thomson ST2022". Thanks a lot Luca Bertoncello (lucabert at lucabert.de)

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > What kind of phone are we talking about, both yours that works and your > wife's that does not? Right! > Can you ping the unreachable phone and does it respond to a ping? I can ping both phones from the VM > Many phones will have a network test function built in to them to help you > determine if the phone

Hi list! Right now I configured my Asterisk to forward the calls for the number X to both phones (mine and the phone of my wife). It works, of course, but I'm not enthusiast... I see what we have at office: if one phone rings, other phones in the same group can "catch the call", so that if a colleague is not present, another colleague can catch the call. I'd like to have the