similar to: SIP Registration issues

Hi Everyone, I'm new to * and I'm trying to setup a small configuration of SIP clients. Eventually when I get this working I plan on expanding with a Digium developers kit to add analog phones and PSTN access. My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both peers seem to register with * but I cannot call to one another. When I dial the associated extension, the

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

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Hi all I have 2 snom 100's and an ix66 (sip aware firewall) set up with asterisk. I needed to register a number of lines so what I've done is make asterisk register all the lines i need (attaching them to an extention eg 1000) and then register each phone with asterisk. so for example in sip.conf: register => andy@sip.mydomain1.org/1000 register => andy@sip.mydomain2.org/1000

Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details

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i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

Hi guies, I know I know this subject have been The most written subject about VoIP Right... but I just want to make clear, just one time ! If Asterisk is on a Public IP Address and a softphone behind the nat, sip.conf must contains for this phone: nat=yes .... Now if I want to configure my sipphone (X-Lite) placing behing the NAT, it must have in "Domain/Realm" the external IP

Hi! I'm trying to set up the possibility for users to call my Asterisk from the net. The Asterisk is behind a Intertex IX66 in which a have set "Static domains" so it forward all calls for my hostname and external IP to the * box. when somenone calls at lars@<external-ip> it all works, but if they call lars@<hostnmame> it wont work. At my Asterisk console I get the

I'm using X-Pro connected to an asterisk server (CVS-HEAD-01/27/05-23:17:07) and after about 15 minutes in a call I get a lot of noise in my end. I don't think the other part of the call hears it. After some 10 seconds or so everything is fine again. In my CLI I get NOTICE[32322]: RTP Transmission error to 85.xxx.xxx.xxx:35162: Operation not permitted. I get it on calls to the PSTN

This is a single aastra 9113i sip phone. asterisk 1.0.9 Why do I keep seeing this in the logs ? ------------------------------------------------------ Sep ?8 18:44:25 VERBOSE[18779]: Scheduling destruction of call '9157b7d5ef36e8ec556a68e446d4ad59@192.168.1.100' in 15000 ms Sep ?8 18:44:31 DEBUG[18779]: Setting NAT on RTP to 0 Sep ?8 18:44:31 VERBOSE[18779]: 11 headers, 2 lines Sep ?8

Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for

I''m trying to use the traffic control features of Shorewall. I share my network with some guys, and I have a tcclasses file like this: #Hunter Fuller eth0 100 1500kbit full 50 tos-maximize-throughput #(other guy''s name) eth0 110 1500kbit full 50 tos-minimize-delay etc., based on their personal preferences. Then

Hi, Has anyone got incoming IAX to work on the above router. I can call out, but incoming calls are not reaching the * box. Has anyone got this working? Could they give me some configuration hints. Thanks John

Monitoring a tcpdump between two systems, a FreeBSD 9.1 system has the following interface: msk0: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> metric 0 mtu 1500 options=c011b<RXCSUM,TXCSUM,VLAN_MTU,VLAN_HWTAGGING,TSO4,VLAN_HWTSO,LINKSTATE> ether 00:11:2f:2a:c7:03 inet 10.0.1.199 netmask 0xffffff00 broadcast 10.0.1.255 inet6 fe80::211:2fff:fe2a:c703%msk0 prefixlen 64 scopeid

Hi, I've recently upgrade to Samba 1.9.17p1 from 1.9.16p11 (for Solaris 2.5), but have now found that smbclient fails to connect to user shares on win95 machines. I've tried the same login account from a win95 account and am able to mount shares from other win 95 machines. I've also tried running the old smbclient binary with the new smbd & nmbd daemons, and that lets me log

Hi, I just purchased a 9331i phone. It's great! However, having a slight issue with line 2 and 3. line 1 works fine. But when I try to make a call with line 2 or 3 I get a fast busy from the phone. Asterisk shows this: Mar 23 15:19:12 NOTICE[18185] chan_sip.c: Failed to authenticate user No User <sip:No%20User@mysipserver.myhost.com:5060>;tag=34c1cd2bf55cb8d Can anyone with a

> > This sounds like a dns problem, are your windows machines using the DC > as their nameserver ? > yes they are using the server as dns. DNS is done through bind9 and implemented in samba via bind9_dlz. All looks fine: $ host -t A smb.wie smb.wie has address 192.168.1.50 $ host -t SRV _kerberos._udp.smb.wie _kerberos._udp.smb.wie has SRV record 0 100 88 srv.smb.wie. $ host -t SRV

Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is

Morning everyone. Just have a quick question regarding WINS on samba-3.0.3. Running FreeBSD 4.9, with samba-3.0.3 built from the ports tree. The only thing I needed from this particular server at this time, was it to function as a WINS server for our company. I went ahead and did the basics, as setting the netbios name as well as saying 'yes' to wins support. However, I cannot tell if