search for: outoing

I am trying to get Asterisk 1.6.2.5 working with a Grandstream HT-503 ATA. The FXO part is giving me fits. Every call I try to make to the FXO port outbound I get 403 Forbidden coming back. I've been through every configuration setting I can see, and Uncle Google is not helping me much. I updated the firmware to the current version, and that didn't help. If anyone has this working, I

Outoing calls I make using Motif Google Voice Calls continue ringing even after the other end picks up. I have to restart Asterisk to resolve the issue. I don't see any errors. It's not recognizing that the other party picked up the phone and restarting Asterisk fixes it only for a day. -- Co...

>Asterisk and it works fine untill the following >situation: > >- one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that >line) - both via old and new PBX. >- zap show channel <n> would show that line as 'Offhook', though no telephone is off hook. > >If physical line would be

...= yes ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc [1001] type=...

Hello. I've seen several posts talking about line quality using Digium cards that are sharing IRQs or on machines where X is running but after trying all of those fixes I am still having a problem with line static on outoing calls. BTW, calls that are from one extension to another extension have no static, however, they have occasional clicks and pops. At any rate, I was wondering if someone might be able to help figure out how to fix this problem. Here is some information about my setup: Celeron 2.4Ghz w/ 256MB...

...; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = frode.dyndns.org localnet = 192.168.0.0/16 context=stighess tos=lowdelay maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=60 ; Default length of incoming/outoing registratio bandwidth=high disallow=all allow=ulaw ....... [stanaphone] type=friend username=91438xxxx fromuser=91438xxxxx secret=********* host=sip.stanaphone.com context=stighess fromdomain=216.128.82.18 insecure=very nat=yes canreinvite=no disallow=all allow=ulaw __ Stig Hess -------------- n...

...= 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw [1004] type=friend username=1004 secret= reinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1004 nat=1 disallow=all allow=ulaw allow=al...

...; IP QoS parameter, either keyword or value ; like tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow realm=asterisk ; Our global authentication realm ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=all ; Disallow all codecs allow=all ; Allow codecs in order of preference ;allow=ulaw...

...kay so far? I think the basic extension stuff can get me the first part (answer & record the incoming call): - answer the call - play the canned greeting - wait for the caller to talk and then hang up - record the conversation to a file - invoke my script Okay, now my script... It creates an outoing call in /var/spool/asterisk/outgoing, pulling information from a database (assuming I learn some perl and mysql, or something!) THAT file (outgoing call queue) would have to... - call the given number - if it gets an answer, play the recorded message - then if it gets a # key just quit - otherwise...

...to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls tos=lowdelay ; lowdelay,throughput,reliability,mincost,none maxexpirey=7200 ; Max length of incoming registration we allow defaultexpirey=3600 ; Default length of incoming/outoing registration videosupport=yes ; Turn on support for SIP video disallow=all ; First disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw allow=g729 allow=gsm rtcachefriends=yes rtnoupdate=yes rtautoclear=yes externip = 59.14.2.1 localnet=192.1...

...Stefan > > > -- > (o_ Stefan Gofferje | Linux Systems >Specialist > //\ Reg'd Linux User #247167 | Network Security >Specialist > V_/_ Linux is like a Wigwam - No gates, no windows, >Apache inside defaultexpirey=120 :Default length of incoming/outoing registration I believe that is the correct option. This site is your friend. Try searching... http://www.voip-info.org/wiki-Asterisk+config+sip.conf

Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

...yes ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registrati ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=all ; Disallow all codecs ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc ; ;register =...

...; Enable slow, pedantic checking for tos=lowdelay ; IP QoS parameter, either keyword or ; like tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registratio n ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video externip = xxxxxxxxxxxxxxxxx ; Address that we're going to put in ; if we're behind a NAT localnet = 1...

snat not working my local ip is aaa.aaa.aaa.aaa asterisk sitting on the internet at ip bbb.bbb.bbb.bbb my firewall''s internal ip is 192.168.0.254 i did snat: iptables -t nat -A POSTROUTING -o ppp0 -j SNAT --to aaa.aaa.aaa iptables -t nat -L -v gives: Chain POSTROUTING (policy ACCEPT 23663 packets, 2182K bytes) pkts bytes target prot opt in out source destination 33056

...#################### [general] context=from-sip port=5060 bindaddr=0.0.0.0 srvlookup=yes register=userid:password@rs.swifttel.com/5552225059 register=userid:password@rs.swifttel.com/5552225046 register=userid:password@rs.swifttel.com/5552225049 defaultexpirey=1200 ; Default length of incoming/outoing registration disallow=all allow=ulaw allow=alaw allow=g723.1 allow=g729 nat=yes externip=216.246.192.144 localnet=192.168.0.1/255.255.255.0 [5046] type=friend context=from-sip defaultip = 192.168.0.5 host=dynamic canreinvite=yes dtmfmode=info mailbox=5046@local disallow=all allow=ulaw allow=alaw...

...alls srvlookup = yes ; Enable SRV lookups on outbound calls pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw...

Hi, I''m trying to configure IMQ to work on the same machine with frottle (http://frottle.sourceforge.net). The problem is both feed themselves packets through netfilter queueing mechanism, but currently there can only be one netfilter queue per protocol family. To explain why I need IMQ in the first place I have to explain what frottle does. It is a deamon that tweaks the behaviour of a

Hello, Is there a asterisk developer guide/source code doc or something like that? I want to see if I can implement the admin menu function for meetme. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030922/3ff8a388/attachment.htm

...my config: ; ; SIP Configuration for Asterisk [general] port = 5060 ; Port to bind to tos=reliability ; Packet Retagging maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY bandwidth=high ; Bandwidth Limits disallow=all allow=gsm allow=g726 srvlookup=yes context=incoming register => 847*******:XXXXXXX@sip.broadvoice.com/9999 ;; ; Providers [bvoice] type=friend...