similar to: Asterisk to sip client behind Firewall/NAT - cancall but cannot receive calls ?

As far as I can remember I only opened sip and tftp ports for the phone. For some reason (didn't look into it too much) the call stays with sip and doesn't use RTP. The problem you describe (the call doesn't even ring on the other side) is something I had and was solved by upgrading the firmware. Checkpoint's tracker explicitly said what connection attempts were blocked and why.

Hi, I have following setup: BT100 ---- Firewall/nat 1 (www.ipcop.org) ---- Internet ----Firewall/nat2 (Vigor) ---- Asterisk . I'd like to use BT100 as local extension to Asterisk. I've done simple setup and BT100 can call Asterisk and place outgoing calls. However I cannot set him to qualify, cause it is claimed as unreachable. I have port redirection at Firewall 1 (to 5060 and rtp

When I made a call using an older version I saw, using checkpoint's user monitor that the call was indeed using RTP (somewhere between 10000 and 20000, dynamically set for each call). After I upgraded the firmware, the entire conversation stays on the sip port. > -----Original Message----- > From: Jon Lawrence [mailto:jon@lawrence.org.uk] > Sent: Tuesday, December 14, 2004 10:30 PM

Hi, I'd like to organize my Asterisk to properly handle incoming SIP/IAX/H323 callerids so they can be called back if needed. I have three incoming contexts for sip, iax and h323 calls. To each incoming call I'd like to prepend certain number that will be catched with pattern matching on output calls. For instance for iax I have: [from-iax] exten => s,1,NoOp(IAX call from outside

I've been using Asterisk for a few weeks now. I have a (1) BT100 phone and a Sipura-2000 for all my analog phones. All has worked rather flawlessly, until today. I was on the BT100 phone today. During my phone conversation, the BT100 disconnected and went into a "click" mode. 2 "clicks" per second I think. Asterisk was fine, I picked up one of the analog phones,

I'm seeing all sorts of problems and it's probably more of my lack of experience than anything else. I have a BT100 running 1.0.6.7 code. When I go to the status page it says it's not registered (hmm, that's not good). I also can't get dial tone but I can dial! I'm afraid I'm lost any good pointers? I've done a sip debug and all I'm seeing for the BT100 -

Hi All, I'm trying to configure 2 GS BT100 connected to asterisk and Welltech 2 ports FXO gateway. I configure WellTech 2ports FXO and GS BT100, both GS BT100 can call each other without any problem but when I tried to call a local extensions connected to my Welltech FXO gateway, I couldn't hear any voice on both ends. I would like to ask if anyone has ever encountered this kind of

Hi All, I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes monitor-format = gsm|wav|wav49 monitor-join = yes eventwhencalled = yes member => Agent/1000

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, I am getting the following error when I attempt to listen to voice messages by dialing 9999 (I can hear nothing): --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. I read in previous posts that this may be to do with the dtmf

Does anyone know why the bt100 cannot park and pickup a parked call? attendant announces the call is parked at extension 701 but the call cannot be retrieved by any other phone. also, the bt100 constantly rings when the phone is hung up after parking. anyone experienced this? using the basic features.conf [general] parkext => 700 ; What ext. to dial to park parkpos =>

Following on from my below email, things have taken another bizarre twist.. I have continued getting the error when 2092 tries to listen to messages by dialing 9999. --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. Then I

can anybody identify why the CLI is issuing a failure message while debug shows everything is fine???? this makes no sense to me. also, why is the username being updated? this has got to be wrong from CLI -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 Dec 24 12:16:35 NOTICE[15776]:

I see my own number(or remote called num) instead of caller id on incoming calls on my BT-102. but on Xlite everyything is OK. I'm using * latest CVS. - shabanip

Hi All, I have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as well as any configuration in asterisk thats needed, on this phone? Thank you, James -------------- next part -------------- An HTML attachment was scrubbed...

hi all i upgrade a bt100 phone and it can't resgister with asterisk Dec 1 13:25:49 NOTICE[1112980400]: chan_sip.c:7519 handle_request: Registration from '<sip:@172.16.4.249>' failed for '172.16.4.226' is was working with the version 1.0.5.3 some bady now what is hapening? thanks in advance Rodney

I have 2 firewall in HA with keepalived. Can I use active the same tinc configuration on 2 firewalls ? using tun Interface with same ip on all 2 nodes is a problem ? tun device advertise itself on the network having an IP/MAC pairs (ARP) or the IP is only used by the system internally for routing so using the same configuration is right ? so one firewall be active, the other is passive. With this

I think it should work at least for TUN virtual interface as TUn works at IP level. This is a sample configuration. firewall1 lan = 172.16.1.11/19 (ALWAYS ACTIVE) - "Physical Network Interface" - system config as ifcfg-... 172.16.1.10/19 (VIP Keepalived Make active) - Active/Passive configuration with firewall2 firewall1 vpndr1

Hi, I cannot get SIP channel working with folowing codec configuration: [sip] disallow=all allow=g723.1 ;I need this codec between sip phones (BT100) allow=ilbc ;Use this codec to others Calling between BT100 SIP phones is OK - asterisk makes native bridge (with g723.1) between them. When I'm calling from SIP to other channel (iax,zap,...), asterisk is not able to chose right codec