search for: gw1

...deviation. If I do equal trimming at both tail of distribution that I chose, I will use the first ssd formulae which is "a". But if I am doing unequal trimming, then I will be using "b". My problem is I dont know how to come out with a value of ssdw. For example, if the value of gw1=gw2 (which mean equal trimming), the result will come out as "a" not the value of ssdw when equal trimming is done. I f anyone could with this, I really appreciate it. I need the value of ssdw so that I can proceed with computing the F test. Thanks. Regards, Hyo Min UPM Malaysia n1<...

I'm trying to link up Cisco MGCP-enabled router (residential gateway) with Asterisk, and it looks like some sort of protocol mismatch, could it be MGCP 0.1 vs 1.0? Look at this (x.x.x.99 is the router, x.x.x.98 is Asterisk): MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427Verb: 'NTFY', Identifier: '2', Endpoint: 'aaln/0@voip-gw1', Version: 'MGCP 0.1' 3 headers, 0 lines Handling request 'NTFY' on aaln/0@vo...

...pretty close on giving up on Asterisk :-/ I am (still) trying to make a call from a H323 phone to an Asterisk provider using AIX. But H323 does not route the number to AIX. All it is transmitting is an "s". *CLI> -- Executing Dial("OH323/R27865", "IAX2/demo:demo@gw1.musimi.dk/s") in new stack -- Called demo:demo@gw1.musimi.dk/s Jul 2 23:43:55 WARNING[-1137550416]: chan_iax2.c:5231 socket_read: Call rejected by 212.130.58.212: No such context/extension -- Hungup 'IAX2[demo]/3' == No one is available to answer at this time The dialed sho...

...se there is no interaction (yet) from the AMI with our asterisk. Version 1.6.2.7 human_now: 2010-05-20 01:42:03.567385 Event: Newexten Privilege: dialplan,all Timestamp: 1274308923.567385 Channel: SIP/Prov6-000001be Context: NPDB2 Extension: 37062646666 Priority: 75 Application: Dial AppData: SIP/GW1/00737062646666,60,M(connect-jack,737219) Uniqueid: 1274308923.446 human_now: 2010-05-20 01:42:03.568501 Event: Dial Privilege: call,all Timestamp: 1274308923.568501 SubEvent: Begin Channel: SIP/Prov6-000001be Destination: SIP/GW1-000001bf CallerIDNum: <unknown> CallerIDName: <unknown>...

...l Asterisk server. No intelligent routing, PIN, anything else.... I configured it like this: PSTN-To-VoIP Gateway Setup PSTN-To-VoIP Gateway Enable: yes PSTN Caller Auth Method: none PSTN Ring Thru Line 1: no PSTN Caller Default DP: 1 Then I configured the dialplan #1 as: Dial Plan 1: (S0<:@gw1>) And I configured gateway 1 as: Gateway Accounts Gateway 1: my.asterisk.server GW1 NAT Mapping Enable: no GW1 Auth ID: --my-sip-login-- GW1 Password: --my-sip-password-- But it seems to simply ignore incoming calls at all.... Anybody's got a pointer to get me started? Thanks in...

Hi Who concern, I setup TINC VPN follow these. 192.168.1.x / 24 (Client groups) | 192.168.1.1 (eth1) (GW1) 202.44.34.206 (eth0) || Internet || 202.44.45.14 (eth0) (GW2) 192.168.2.1 (eth1) | 192.168.2.x/24 ( Client groups) I use Red Hat 6.2 Kernel 2.2.14 and Tin...

...r to my problem. i have two gateways running with asterisk , my question is : is there any possibility to do failover with gateways with asterisk ? i mean that if one gateway is down , asterisk switch automatically to other gateway . i have succefully used failover with limit number off calls (if gw1 have max calls ,asterisk swith to gw2) , but now i want if it is possible if gw1 is down ,asterisk go out over gw1 ? regards.

...ork). The issue is that I am not able to make outbound calls, because the call fails with the error: res_pjsip_outbound_authenticator_digest.c:125 digest_create_request_with_auth: Unable to create request with auth.No auth credentials for any realms in challenge. CLI> pjsip show endpoint sonnyGW1 ... ========================================================================================= Endpoint: sonnyGW1 Not in use 0 of inf OutAuth: sonnyGW1_auth/sonny Aor: sonnyGW1 0 Contact: sonnyGW1...

..., because the call >> fails with the error: >> >> res_pjsip_outbound_authenticator_digest.c:125 >> digest_create_request_with_auth: Unable to create request with auth.No auth >> credentials for any realms in challenge. >> >> CLI> pjsip show endpoint sonnyGW1 >> >> ... >> ========================================================================================= >> >> Endpoint: sonnyGW1 Not in use >> 0 of inf >> OutAuth: sonnyGW1_auth/sonny >> Aor: sonn...

I have the following in extentions.conf: exten => 2111,1,Dial(SIP/2111 at gw1.langley) exten => 2111,2,Voicemail(u2111) exten => 2111,3,Hangup exten => 2111,100,Voicemail(b2111) exten => 2111,101,Hangup I have the following in sip.conf: ; Cisco 1750 [gw1.langley] type=friend host=172.16.17.1 context=default canreinvite=no Like the ATA, lots of stuff doesn'...

...finitely throwing me for a loop. We have two IP addresses but our downstream gateway is the same. So I have 100.200.300.400 with gateway 6.7.8.9 and 1.2.3.4 with gateway 6.7.8.9. (We are allocated two IPs from the same subnet.) So far I have: ... # setup our routes to our gateways ip route add $GW1 src $IP1 dev $IF1 ip route add $GW2 src $IP2 dev $IF2 # setup interface specific routing tables ip route add $NET1/$CIDR1 dev $IF1 table 200 ip route add default via $GW1 dev $IF1 table $TABLE1 ip route add $NET2/$CIDR2 dev $IF2 table 100 ip route add default via $GW2 dev $IF2 table $TABLE2 # cre...

...led. Defined the PSTN Line (fxo) to register with asterisk via sip using a second sip.conf entry (extn 2222). PSTN User, defined PSTN Ring Thru Line 1 Ring Settings as "1". In Line 1, defined Gateway Account #1 to point to asterisk, and created a dialplan entry like: (*xx|81xxx.<:@gw1>|3xxx<:@gw1>|0<:@gw0>|[2-9]11<:@gw0>|xxxxxx.<:@gw0>) Note: gw0 defaults to the pstn line per the spa-3000 doc. Result: 1. If asterisk is down for any reason, all incoming pstn home calls still ring through to the analog house phones. 2. Incoming pstn calls ring through...

...one, willing to take a lead on this one, since Tom is taking a rest: " I am hosting all servers by myself. I have five static IP addreses with a DSL line. My DSL router from the ISP provider is configured as bridge, so no traffic is filtered. I checked the logs and getting: Jan 5 23:05:12 gw1 kernel: Shorewall:all2all:REJECT:IN= OUT=eth0 SRC=66.58.99.86 DST=216.35.73.164 LEN=68 TOS=0x00 PREC=0xC0 TTL=255 ID=1508 PROTO=ICMP TYPE=3 CODE=1 [SRC=216.35.73.164 DST=66.58.99.84 LEN=40 TOS=0x00 PREC=0x00 TTL=236 ID=55762 DF PROTO=TCP SPT=51131 DPT=25 WINDOW=8760 RES=0x00 RST URGP=0 ] Jan 5 23:...

...n them which confused the heck out of me): [transport-udp] type=transport protocol=udp bind=0.0.0.0 local_net=172.31.32.0/20 local_net=192.168.1.0/24 external_media_address=aa.bb.cc.dd ; replaced real public IP address external_signaling_address=aa.bb.cc.dd ; replaced real public IP address [sonnyGW1] type=registration transport=transport-udp outbound_auth=sonnyGW1_auth server_uri=sip:registrar at gw1.sip.us ; no registrar@ in URI client_uri=sip:sonny at gw1.sip.us contact_user=16175551212 ; replaced real DID retry_interval=60 forbidden_retry_interval=600 expiration=3600 [sonnyGW1_auth] type=a...

Hello, I have a strange situation with an SPA3102 FXO/FXS device. I'm in situation that when i receive a call from PBX line I must forward the calls to 2 VoIP numbers. Right now i have the following settings: (S0<:1010 at GW1>). I want to forward at 1020 too. I tested (S0<:1010|1020 at GW1>) and doesn't work. Did you have any other ideea? Thank you.

I have a working Asterisk 13.1.0 running, and I am trying to configure a SIP trunk for outbound and inbound calling, and a DID for the Asterisk server, which is used for incoming calls from PSTN. I configured my SIP.US trunks (showing one gateway, gw1, here for brevity, have two: gw1 & gw2, which are both configured on my end): [sonnyGW1] type=registration transport=transport-udp outbound_auth=sonnyGW1_auth server_uri=sip:gw1.sip.us client_uri=sip:sonny at gw1.sip.us contact_user=sonny retry_interval=60 forbidden_retry_interval=600 expirati...

Hi, I am trying to configure the SMTP service on DMZ host. Added the rule: ACCEPT wan dmz:66.58.99.84 tcp pop3 - ACCEPT wan dmz:66.58.99.84 tcp 25 - ACCEPT dmz:66.58.99.84 wan tcp 25 - ACCEPT dmz:66.58.99.84 wan tcp pop3 - issued shorewall clear, shorewall restart, but still couldn''t telnet to the mail server

...se this or what I would need to do to look at solving the issue ? I am now offline :( and for some reason rolling back to the older version now does not want to run :( My IAX conf [general] tos=lowdelay jitterbuffer=no disallow=all allow=speex allow=ilbc allow=gsm register => user:password@gw1.austechpartnerships.com [guest] type=user context=default auth=none [2347] type=friend username=user secret=password auth=md5 host=gw1.austechpartnerships.com context=default trunk=yes qualify=3000 disallow=all allow=ilbc Thanks David

Hello all! I applied http://www.ssi.bg/~ja/routes-2.6.8-10.diff patch to kernel 2.6.8.1 and it works fine, or almost fine. It does the load balancing well, but when one link is dropped it continues to try it. At the end of http://www.ssi.bg/~ja/nano.txt it is said to ping gateway 1 and gateway 2, for the kernel to know if that route is working, but since my linux is connected to the links

...to achieve this. ; The NuFone Network's ; Open H.323 driver configuration ; [general] port = 1720 bindaddr = 0.0.0.0 amaflags = default disallow=all allow=all ; turns on all installed codecs allow=g711 dtmfmode=rfc2833 gatekeeper = 200.84.48.85 AllowGKRouted = yes context=gateway [gw1] exten =>9999XXXXXXXXXXX,1,Dial(h323/${Exten:1}@217.199.177.152) [gw1] type=h323 prefix=9999 context=gateway So when I start asterisk with -vvvvvvgc the last part of the information looks like this [chan_h323.so] => (The NuFone Network's Open H.323 Channel Driver) == Parsing '/e...