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I upgraded my 7960 to sip v 6.3 and my display time has now disappeared from the top left corner. Loadid: SW: P0S3-06-3-00 ARM: PAS3ARM1 Boot: PC03M030 DSP: PS03AT38 Here is the section dealing with time in my SIPDefault.cnf file. Does anybody see anything wrong with it or have any other ideas? # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)

It looks like this meetup group is becoming the venue for contact as most "Toronto" respondents have signed up here. Can I request that the remainder who responded to the "Toronto" call signup? Shidan is the organizer and has proposed a date. I'd hate to confirm it without the remainder having input - especially those with travel times like Andrew K. out in Listowel.

My cell vm goes to asterisk, not the carrier. Apparently MWI is turned on/off with specially formatted SMS messages. Anyone know how to do this on a Treo 600? Having the phone light from Asterisk would be HUGE ... not to mention extremely cool. dbc.

I am still haveing problems (echo) with my X100P but I'm thinking it has more to do with the server it is in which is not a negotiable item at this time. My question then is to the use of SPA3000's as a replacement from the FXO standpoint. 1. Can you setup the FXO port to recognize distinctinve ring and call a different context like you can do with Zap channels? Being able to call a

> Lito Lampitoc wrote: > > thanks for enlightening. So you mean, if I have 3 lines when the > caller > > dialled the first line and it was busy, the call will be diverted > to the > > next two available lines in random? > > > > I don't think it's random. I think its just sequential. If main > line > is busy, try second. If that is

Hi all, I've tried search this problem on the list... no luck... The case is: without externip/localnet config on sip.conf [general] my SIP trunk works, but with no audio NAT problem (asterisk sends the private 192 address to the outside...) when I configure externip/localnet correctly my SIP trunk simply disappear! Checking the signalling with tcpdump shows me that Im sending the

Will this do the intended thing? This is in sip_nat.conf which is included in sip.conf: externip=192.168.0.200 localnet=192.168.0.200/255.255.255.0 externip=64.168.237.110 localnet=192.168.1.2/255.255.255.0 I have Asterisk running on a box with two Ethernet interfaces and bound to both. One interface, 192.168.1.2 services clients outside the firewall who are led to believe that Asterisk is

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

> From: "James Bean" <james@hdcs.com.au> > Has anyone every setup an external open/close relay, off say a serial > interface, and have an extension trigger the relay? The following will do the trick. Just add a 5vdc solid state relay ('cause you can't sink too much current out of the RS232C port). Substitute "2", "4" or "6" in the

Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the

Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: "It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the

Hi there, I am trying to setup a configuration that requires me to use SIP and asterisk behind a firewall and over a VPN to a remote office and with some local Phones also. I can't use IAX to my provider because they don't offer it and my handsets ( snom 300 ) also don't support IAX so it's all SIP. The configuration is a follows Asterisk PBX 10.202.17.217/24 ------>|

Hi all! Do anybody has a full working environment using externip on an asterisk box behind a nat? I tried with two diferent boxes (Elastix-1.4.24 e Trixbox-1.4.22-3)and the asterisk do not replace neither contact, neither sdp headers info with the externip informed on sip.conf general parameters. I used these two statements: externip=XXX.XXX.XXX.XXX localnet=192.168.200.0/255.255.255.0 Do

Hi All, Is it possible to specify more than 1 localnet? I know this is an odd question. I have a customer that has multiple sites linked by VPN. Main range is 192.168.33.0/24 and a remote site is 10.1.1.0/24 We want to allow some access to the public IP address at the main site. For this to work I need to use the externip and localnet directive. If I do this it rewrites the SDP with the

Hi everybody, I finally want to get rid of 1-way audio problem. Please help me here. I have 3 scenarios. 1. Audio is always one way. Caller who dialed can't listen the called party but called party can listen him. In this scenatio Asterisk is on dynamic IP with dyndns FQDN. sip.conf has externip = abc.dyndns.org and localnet = xxx.xxx.xxx.xxx entry. Trunk and extensions are SIP. Where is

I tried implementing my * and it didn't pass the spouse factor at this time. I wanted to hook it up for outbound only at this point to get a better handle on the dial plans and the echo problem. I thought this might have been done before as a natural part of testing - but maybe not. In wcfxo.c I found this: if (!wc->offhook && !wc->ringdebounce) { if

Hi, I'm using an Asterisk 1.2.18 box with a remote Snom 360. My Snom always rings but sometimes (it happens randomly!) no voice is passing thru (2 ways). Asterisk CLI shows this warning: Jan 10 10:03:26 WARNING[19164] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"unknown" <sip:unknown at xxx.xxx.xxx.xxx> I have already set localnet and

With regards to the IRQ sharing situation on 400P/X100P cards how would I know if I can use IO-APIC? I am running RHEL 3 on a Dell PowerEdge 1400SC. RHEL installs without IO-APIC support. Is this because RH is overly conservative or because it queried my machine and that is the appropriate option? Does RHEL 3 have a kernel for IO-APIC if appropriate or am I expected to do a custom kernel build

I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxxxxxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it "forget" what it has programmed and

Hi all, I'd like to connect a softphone at home (nat, dynamic-ip) to a sip-phone in the office via asterisk 1.4.21 (nat, fixed-ip). SIP works well, the phone is ringing, but when I pickup the call, there's no audio on both sides. I debugged the rtp-traffic at home. As long as the phone is ringing, everything is fine. But after the pickup, asterisk sends a SIP/SDP package with its