similar to: Qualify sip users behind remote registrar

Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an

I currently do this for about 30 different cisco 79xx's connecting to some hosted Asterisk servers. Asterisk listens by default for any SIP connection on UDP port 5060. And will use RTP UDP port 10000 to 20000 The phones use UDP Port 5061 for incoming connections (from Asterisks or other SIP Devices) and use for RTP, UDP port 10000 to 20000. Now, if you are going to have the two remote

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

What is the best solution? I dont want to have modify firewall's at all or do port fowarding. Ideally I would like a solution that with either a softphone or wireless hardphone one could connect via friends, family, or hotspots without reconfiguring their devices. What are people using? STUN? SER? Thanks in advance! -blake -------------- next part -------------- An HTML attachment was

Hi list! I'm trying to make a SIP test call from Bria and/or 3CXPhone from a PC behind NAT. From Bria/3CXPhone I connect to an Asterisk 11.25.0 server on the internet at 100.100.94.210 with a SIP account "3333" created in sip.conf: [3333] type=friend secret=something host=dynamic nat=yes qualify=no disallow=all allow=alaw allow=ulaw canreinvite=no context=voipin I dial +1234

In an earlier thread Andrew Kohlsmith enlightened me on the use of qualify in sip.conf to deal with a peer that is down. Since then I have been searching for information on how the behavior of qualify can be tuned. The wiki is vague on this; " Syntax: qualify=xxx|no|yes where XXX is the number of milliseconds used. If yes the default timeout is used, 2 seconds. If you turn on

I'm using the latest SIP firmware from Avaya. The phone receives the 46xxsettings.txt OK, and then after entering extension and password it goes to the home screen saying 'Registering'. When I check options->ViewIPSettings->IPAddresses on the phone, the registrar and SIP Proxy fields are blank. I have both lines: SET SIPREGISTRAR "67.1XX.XX.XX" and SET

I have logs full with this messages... I must have qualify turned on, because phone is behind firewall, main problem si, that phone is each hour about one hour unavailable! :'( I tried to modify minexpiry/maxexpiry sip.conf timeouts, but nothing help me. I'm using latest firmware 8.4 in phone, will be better to downgrade? to what version? (latest asterisk 1.4branch) [Dec 7 00:36:56]

Hey, I just started trying to use the qualify=yes option on my Cisco 7960 SIP phones. Of the 13 I have, 2 of them seem to loose their registration with asterisk on a regular basis. I see lots of these lines: -- Registered SIP '3030' at 62.74.107.1 port 58825 expires 60 in my console. But I only see them for 2 extensions. Never see them for the other 11. All 13 phones have the exact same

So, assuming your Asterisk box is behind one firewall (Linksys/Tomato Software) and your Wireless SIP phone is behind another firewall (SonicWall 1260 Enhanced). Is there anything special that I have to do to the firewalls. I do have the Asterisk firewall configured to work (ports 5060 & 10001-20000). But I'm not sure about the other end. Do I need STUN at the SIP Phone end? Do I

Hi, I'm trying to get a remote Cisco Call Manager Express (CME) system behind a dynamic IP address routing both inbound and outbound calls via SIP to my local asterisk server. I've got a local CME system working fine on the LAN, where the CME is at a static IP (host=10.5.7.130 in sip.conf), but I can't figure out how to get it working with host=dynamic, even locally on a test

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

Setup is: Asterisk --->NAT--> SIP Proxy I have following entry for SIP Proxy in sip.conf [Proxy] type=peer host=Static IP (NAT Firewalls public IP) username=xxxx secret=xxxxx nat=yes???????????????? canreinvite=no???????? qualify=yes Proxy sends a call and I get this error Found no matching peer or user for <NAT's Public IP:70001 NAT is using 70001 as the source port in the

I have been working with this for a wile and I have been watching the list for about a month on this subject, to no avail. I am wondering if anyone has successfully configured asterisk for clients to connect to it when the clients are behind nat. I mean successfully because I can do everything except for audio, my audio is only one way. I am asking so I can determin if I will be continuing

SetUp: - Asterisk behind a NAT, - Red Hat 9.0 - Asterisk 1.2.14 My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my dial plan set up so that when outside callers dial the DiD, the call is answered by my auto-attendant. The caller can then select who they'd like to speak to and the call is transferred to the external line associated with that person (usually a mobile

SetUp: - Asterisk behind a NAT, - Red Hat 9.0 - Asterisk 1.2.14 My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my dial plan set up so that when outside callers dial the DiD, the call is answered by my auto-attendant. The caller can then select who they'd like to speak to and the call is transferred to the external line associated with that person (usually a mobile

Say, If bandwidth e.g. ADSL goes fuzzy, is there a way to force * to unregister the Peers? I noticed with qualify=200 for example, even if latency goes above and * shows Lagged and then UNREACHABLE The peer's calls are still accepted. Is there a way to automatically prevent this? Thanks Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm just curious if I was to place my * box behind a a FW/NAT box running linux, if my SIP calls will still work. Box right now is a RH9 computer using iptables as the FW. I wouldn't mind placing my * box behind it, but I'm wondering if anyone has actually gotten NAT working with *? Thanks, -- +------------------------------------------+ |Leif Madsen -

Can anybody point or provide working configuration how to register Sipura to Asterisk over the Internet. Both Sipura and Asterisk are behind firewalls. I'll be force to use SIP as that is the only protocol that Sipura is using. Do I need to enter any "STUN Server:" setting in "SIP" tab. On Asterisk I think I only need to make changes in sip.conf isn't it? What

Hi Everyone, Is it possible to have a SIP Phone work remotely if it's behind a Router performing NAT without connecting the Router to a VPN? The Asterisk Box will be in the DMZ. Thanks Dan CYTEXONE ---- Dan Levine dan@cytexone.com CYTEXONE | Your Technology Specialists R 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012