[Asterisk-Users] Re: (no subject)
On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote: ok. didnt work :( i thinks is a pbx problem. because E1 is incomming in the pbx. and all incomming calls go to 100. thats the problem i will try to solve this. It could potentially be both. I would look at your extensions.conf first though. What does the extension entry for that context look like. For instance I have an entry in my extensions.conf for dialing outside lines (outside being from asterisk to my PBX and then onto the outside world from there). The entry looks like this: [to-analog] exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN}) exten = _9XXX.,2,Congestion exten = _9XXX.,103,Hangup To dial a PBX extension the entry would look almost the same: [to-pbx-extension] exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1}) exten = _9XXX.,2,Congestion exten = _9XXX.,103,Hangup Hope this helps, -Matt On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote: hi all, i have a box with a te110p and a pbx siemens... connect both with a e1. with a xten soft i can call extensions numbers in my office example 100 102 etc. but when i truy to go outside with the 9 before the call rings in the first extensions (100). this is a asterisk problem? or a pbx problem? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: (no subject)
This is not a siemens pbx problem you set the pridialplan = to national and that adds a number to the outgoing call or something just use Pridialplan = local prilocaldialplan = local and it should work I tried to open the file kds again and now it showed me your configuration :) don't know why it did not show me before Sander -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: woensdag 14 september 2005 17:31 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Re: (no subject) On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote: ok. didnt work :( i thinks is a pbx problem. because E1 is incomming in the pbx. and all incomming calls go to 100. thats the problem i will try to solve this. It could potentially be both. I would look at your extensions.conf first though. What does the extension entry for that context look like. For instance I have an entry in my extensions.conf for dialing outside lines (outside being from asterisk to my PBX and then onto the outside world from there). The entry looks like this: [to-analog] exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN}) exten = _9XXX.,2,Congestion exten = _9XXX.,103,Hangup To dial a PBX extension the entry would look almost the same: [to-pbx-extension] exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1}) exten = _9XXX.,2,Congestion exten = _9XXX.,103,Hangup Hope this helps, -Matt On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote: hi all, i have a box with a te110p and a pbx siemens... connect both with a e1. with a xten soft i can call extensions numbers in my office example 100 102 etc. but when i truy to go outside with the 9 before the call rings in the first extensions (100). this is a asterisk problem? or a pbx problem? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: (no subject)
On Wed, Sep 14, 2005 at 07:52:26PM +0200, Sander wrote: This is not a siemens pbx problem you set the pridialplan = to national and that adds a number to the outgoing call or something just use Pridialplan = local prilocaldialplan = local and it should work no uuuaaa the same problem.. ring in the extension 100. I tried to open the file kds again and now it showed me your configuration :) don't know why it did not show me before Sander -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: woensdag 14 september 2005 17:31 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Re: (no subject) On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote: ok. didnt work :( i thinks is a pbx problem. because E1 is incomming in the pbx. and all incomming calls go to 100. thats the problem i will try to solve this. It could potentially be both. I would look at your extensions.conf first though. What does the extension entry for that context look like. For instance I have an entry in my extensions.conf for dialing outside lines (outside being from asterisk to my PBX and then onto the outside world from there). The entry looks like this: [to-analog] exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN}) exten = _9XXX.,2,Congestion exten = _9XXX.,103,Hangup To dial a PBX extension the entry would look almost the same: [to-pbx-extension] exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1}) exten = _9XXX.,2,Congestion exten = _9XXX.,103,Hangup Hope this helps, -Matt On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote: hi all, i have a box with a te110p and a pbx siemens... connect both with a e1. with a xten soft i can call extensions numbers in my office example 100 102 etc. but when i truy to go outside with the 9 before the call rings in the first extensions (100). this is a asterisk problem? or a pbx problem? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: (no subject)
Ok it was a problem with my provider it could not see the right numbers comming in :) You can start maintenence in the manager e tool from siemens and start a trace or start call monitoring on extension 100 then you can see the number the asterisk has given to the pbx to dial. -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: woensdag 14 september 2005 21:17 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Re: (no subject) On Wed, Sep 14, 2005 at 07:52:26PM +0200, Sander wrote: This is not a siemens pbx problem you set the pridialplan = to national and that adds a number to the outgoing call or something just use Pridialplan = local prilocaldialplan = local and it should work no uuuaaa the same problem.. ring in the extension 100. I tried to open the file kds again and now it showed me your configuration :) don't know why it did not show me before Sander -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: woensdag 14 september 2005 17:31 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Re: (no subject) On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote: ok. didnt work :( i thinks is a pbx problem. because E1 is incomming in the pbx. and all incomming calls go to 100. thats the problem i will try to solve this. It could potentially be both. I would look at your extensions.conf first though. What does the extension entry for that context look like. For instance I have an entry in my extensions.conf for dialing outside lines (outside being from asterisk to my PBX and then onto the outside world from there). The entry looks like this: [to-analog] exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN}) exten = _9XXX.,2,Congestion exten = _9XXX.,103,Hangup To dial a PBX extension the entry would look almost the same: [to-pbx-extension] exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1}) exten = _9XXX.,2,Congestion exten = _9XXX.,103,Hangup Hope this helps, -Matt On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote: hi all, i have a box with a te110p and a pbx siemens... connect both with a e1. with a xten soft i can call extensions numbers in my office example 100 102 etc. but when i truy to go outside with the 9 before the call rings in the first extensions (100). this is a asterisk problem? or a pbx problem? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: No subject by Steve M
Just responding in case this may be of help to somebody with firewalling issues. Not sure if this is off on a tangent to the original question... Here are three different forms of common firewall scripts and ways of getting SIP to work behind them. The third one has some additional stuff beyond just SIP although I can't remember why I wrote it that way. I've been having no fun using sip phones that try to figure things out with third party STUN servers. It seems better to use a good linux firewall like it was intended. *** Redhat 9 scripts - basic firewall rules for SIP forwarding: file: /etc/rc.d/init.d/firewall In the Services section, add this: SIP=your.internal.ip.here # VIOP SIP Add the following code amongst the service scripts toward the bottom: ## #SIP # ## function SIP_WAN { $IPT -A INPUT -p udp -i $WANIFACE --dport 5060 -j ACCEPT $IPT -A INPUT -p udp -i $WANIFACE --dport 5004 -j ACCEPT } function SIP_PORT_FORWARDING { $IPT -A PREROUTING -t nat -i $WANIFACE -p udp --dport 5060 -j DNAT --to $SIP:5060 $IPT -A FORWARD -i $WANIFACE -p udp --dport 5060 -j ACCEPT $IPT -A PREROUTING -t nat -i $WANIFACE -p udp --dport 5004 -j DNAT --to $SIP:5004 $IPT -A FORWARD -i $WANIFACE -p udp --dport 5004 -j ACCEPT } if [ $SIP = ON ]; then SIP_WAN else if [ $SIP != OFF ]; then SIP_PORT_FORWARDING fi fi Note: the two lines above beginning with two dashes (--) wrapped, they should be at the end of the lines above them. *** Basic rule for SuSEfirewall2: file: /etc/sysconfig/SuSEfirewall2 In the section for FW_FORWARD_MASQ= insert the following two lines: 0/0,internal.sip.ip.address,udp,5060,external.ip.address.here 0/0,internal.sip.ip.address,udp,5004,external.ip.address.here Note: quotation marks are used in this section- although only at the beginning and the end... it's a goofy syntax for writing a config file- so if it doesn't work, and these are the only two ports you're forwarding, it should look like this: FW_FORWARD_MASQ=0/0,internal.sip.ip.address,udp,5060,external.ip.address.heree 0/0,internal.sip.ip.address,udp,5004,external.ip.address.here ** Ruleset for an old reliable IPChains firewall: file: /etc/rc.d/init.d/firewall This actually opens up a few more holes for some outbound streams. Can't remember exactly why I did it this way but it works good. # VIOP - asterisk # vars $EXT_IP=your.external.ip.here $ASTERISK_IP=your.asterisk.server.ip # #chains ipmasqadm portfw -a -P udp -L $EXT_IP 5060 -R $ASTERISK_IP 5060 ipchains -A portfw -s 0/0 1024: -d $EXT_IP 5060 -p 17 -j ACCEPT ipmasqadm portfw -a -P udp -L $EXT_IP 4569 -R $ASTERISK_IP 4569 ipchains -A portfw -s 0/0 1024: -d $EXT_IP 4569 -p 17 -j ACCEPT ipmasqadm portfw -a -P udp -L $EXT_IP 5036 -R $ASTERISK_IP 5036 ipchains -A portfw -s 0/0 1024: -d $EXT_IP 5036 -p 17 -j ACCEPT # loop for a bunch of ports for streams port2=10001 while [ $port2 -lt 10699 ] do ipmasqadm portfw -a -P udp -L $EXT_IP $port2 -R $ASTERISK_IP $port2 ipchains -A portfw -s 0/0 1024: -d $EXT_IP $port2 -p 17 -j ACCEPT ipchains -A portfw -s 0/0 $port2 -d $EXT_IP $port2 -p 17 -j ACCEPT port2=$((port2+1)) done # * I've also been able to run sip traffic over a vpn. My ISP here seems like it's doing some weird stuff with delaying packets on certain ports - so I stuff a lot of stuff through a tunnel. Problem is that the encryption slows it down- which I fixed by running a pptpd daemon without encryption. This will be out of thread, excuse me, but my mozilla is broken and won't take mailto registry fixes. Will need to use a different client in future posts. TJH ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users