William L. Thomson Jr.
2004-Oct-06 15:11 UTC
[Asterisk-Users] * to Cisco router with FXO's via SIP
Ok, very frustrated after spending most of the day onthe * irc channel with little to no help. Mostly just a bunch of crap about being a newbie, going and reading voip-info.org. etc. Despite me doing all that already. My situation is not good but here it is. Hurricane came through, power spikes killed PBX. Just trying to replace it affordable and possibly with a few more features. I am using * v 0.9.0 on Gentoo. Tried going to 1.0.0. Several times. It just crashes. Bought the wrong phones Cisco 7910 so I am stuck using SCCP/Skinny. Got SCCP working with * after loosing a couple teeth. All internal phones work and except for festival causing some issues. I am ready to go for the most part. Final step which I had hoped to complete today was connecting * to the outside world. I have a Cisco 827-4v with 4 FXO's on it. I have it configured to use SIP. I used this example http://www.loligo.com/asterisk/cisco/827-4v/cisco-827-4v-with-asterisk-version1 Despite repeated attempts at asking for help on the * irc channel I made no progress. I have a sip softphone installed on my laptop and that's about it. People on the irc channel are just plain rude to newbies. Hope this is not a reflection of the entire * community? Basic ?'s Is * the SIP server or client in my scenario? It seems it's the client and my router is the server. I was trying to test out the router directly via a soft phone but that's not really working. Not sure on syntax. etc. It was recommended to test out the router via a soft phone on the irc. However once I got the soft phone and it would not connect I was left out on my own again. Do I need to have register lines in my sip.conf? register => user:secret:authuser@host:port/extension I tried to create an extension for the inside to dial out. Basically I would like to press 9 get an outside line and dial. Then again at this point anyway to dial out would be great. I just want to dial out and get calls in at this point. exten => 9,1,Dial(SIP/5114) I am really confused Please help. Any help is greatly appreciated. -- Sincerely, William L. Thomson Jr. Support Group Obsidian-Studios, Inc. http://www.obsidian-studios.com
Hi, Hopefully you can get a little more help here. Can you post your cisco config and your * extensions.conf. The example you were looking at is for FXS station ports on the cisco not FXO CO ports. I am trying to get this working in my lab to help you out. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of William L. Thomson Jr. Sent: Wednesday, October 06, 2004 5:11 PM To: asterisk-users Subject: [Asterisk-Users] * to Cisco router with FXO's via SIP Ok, very frustrated after spending most of the day onthe * irc channel with little to no help. Mostly just a bunch of crap about being a newbie, going and reading voip-info.org. etc. Despite me doing all that already. My situation is not good but here it is. Hurricane came through, power spikes killed PBX. Just trying to replace it affordable and possibly with a few more features. I am using * v 0.9.0 on Gentoo. Tried going to 1.0.0. Several times. It just crashes. Bought the wrong phones Cisco 7910 so I am stuck using SCCP/Skinny. Got SCCP working with * after loosing a couple teeth. All internal phones work and except for festival causing some issues. I am ready to go for the most part. Final step which I had hoped to complete today was connecting * to the outside world. I have a Cisco 827-4v with 4 FXO's on it. I have it configured to use SIP. I used this example http://www.loligo.com/asterisk/cisco/827-4v/cisco-827-4v-with-asterisk-versi on1 Despite repeated attempts at asking for help on the * irc channel I made no progress. I have a sip softphone installed on my laptop and that's about it. People on the irc channel are just plain rude to newbies. Hope this is not a reflection of the entire * community? Basic ?'s Is * the SIP server or client in my scenario? It seems it's the client and my router is the server. I was trying to test out the router directly via a soft phone but that's not really working. Not sure on syntax. etc. It was recommended to test out the router via a soft phone on the irc. However once I got the soft phone and it would not connect I was left out on my own again. Do I need to have register lines in my sip.conf? register => user:secret:authuser@host:port/extension I tried to create an extension for the inside to dial out. Basically I would like to press 9 get an outside line and dial. Then again at this point anyway to dial out would be great. I just want to dial out and get calls in at this point. exten => 9,1,Dial(SIP/5114) I am really confused Please help. Any help is greatly appreciated. -- Sincerely, William L. Thomson Jr. Support Group Obsidian-Studios, Inc. http://www.obsidian-studios.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
William L. Thomson Jr.
2004-Oct-06 16:10 UTC
[Asterisk-Users] * to Cisco router with FXO's via SIP
On Wed, 2004-10-06 at 18:33, Henry Devito wrote:> Hi, Hopefully you can get a little more help here.Thank you thank you thank you.> Can you post your cisco > config and your * extensions.conf.Sure I will keep the cisco to what's relevant. Doubt you care about the rest. The rest I will include as an attachment/link. http://www.obsidian-studios.com/extensions.conf http://www.obsidian-studios.com/sip.conf voice-port 1 cptone DK ! voice-port 2 cptone DK ! voice-port 3 cptone DK ! voice-port 4 cptone DK ! dial-peer voice 1 pots destination-pattern 5111 port 1 ! dial-peer voice 2 pots destination-pattern 5112 port 2 ! dial-peer voice 3 pots destination-pattern 5113 port 3 ! dial-peer voice 4 pots destination-pattern 5114 port 4 ! dial-peer voice 10 voip destination-pattern .+T session protocol sipv2 session target sip-server codec g711alaw bytes 80 ! sip-ua retry invite 3 retry cancel 2 sip-server ipv4:192.168.1.10:5060 ! Although I do need to update that last one to reflect * new IP. 192.168.1.1> The example you were looking at is for > FXS station ports on the cisco not FXO CO ports.They are what I plug the outside phone lines into no?> I am trying to get this > working in my lab to help you out.Great. Really appreciate it. I will post a wiki or something once I get a working config for others. So they can keep their teeth. -- Sincerely, William L. Thomson Jr. Support Group Obsidian-Studios, Inc. http://www.obsidian-studios.com
William L. Thomson Jr.
2004-Oct-06 17:04 UTC
[Asterisk-Users] * to Cisco router with FXO's via SIP
On Wed, 2004-10-06 at 19:37, Henry Devito wrote:> Meant to ask you. What type of router is this?Cisco 827-4v> Which FXO module do you have > in the Router?? It's got FXS's on it. I get confused by that. So that means it does VOIP to analog. Not analog to VOIP? Arrgghh So I guess I will need a Digium card FXO or something.> What is the current IOS level?Cisco Internetwork Operating System Software IOS (tm) C820 Software (C820-OV6Y6-M), Version 12.2(15)T, RELEASE SOFTWARE (fc1 ) TAC Support: http://www.cisco.com/tac Copyright (c) 1986-2003 by cisco Systems, Inc. Compiled Tue 11-Mar-03 18:15 by ccai Image text-base: 0x80013148, data-base: 0x80B9B00C ROM: System Bootstrap, Version 12.2(1r)XE2, RELEASE SOFTWARE (fc1) adsl uptime is 39 minutes System returned to ROM by power-on System image file is "flash:c820-ov6y6-mz.122-15.T.bin" CISCO C827-4V (MPC855T) processor (revision 0x502) with 48128K/1024K bytes of memory. Processor board ID JAD04380OUW (346214163), with hardware revision 1987 CPU rev number 5 Bridging software. 4 POTS Ports 1 Ethernet/IEEE 802.3 interface(s) 1 ATM network interface(s) 128K bytes of non-volatile configuration memory. 8192K bytes of processor board System flash (Read/Write) 2048K bytes of processor board Web flash (Read/Write) Configuration register is 0x2102 -- Sincerely, William L. Thomson Jr. Support Group Obsidian-Studios, Inc. http://www.obsidian-studios.com
Tenorio, Leandro
2004-Oct-06 17:17 UTC
[Asterisk-Users] * to Cisco router with FXO's via SIP
You could use the Cisco GW. Try adding in the dialplan Exten => exten => _9.,1,Dial(SIP/${EXTEN:1}@192.168.1.254) Also you wont want the register in the SIP conf file. LTenorio -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of William L. Thomson Jr. Sent: Wednesday, October 06, 2004 9:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] * to Cisco router with FXO's via SIP On Wed, 2004-10-06 at 19:37, Henry Devito wrote:> Meant to ask you. What type of router is this?Cisco 827-4v> Which FXO module do you have > in the Router?? It's got FXS's on it. I get confused by that. So that means it does VOIP to analog. Not analog to VOIP? Arrgghh So I guess I will need a Digium card FXO or something.> What is the current IOS level?Cisco Internetwork Operating System Software IOS (tm) C820 Software (C820-OV6Y6-M), Version 12.2(15)T, RELEASE SOFTWARE (fc1 ) TAC Support: http://www.cisco.com/tac Copyright (c) 1986-2003 by cisco Systems, Inc. Compiled Tue 11-Mar-03 18:15 by ccai Image text-base: 0x80013148, data-base: 0x80B9B00C ROM: System Bootstrap, Version 12.2(1r)XE2, RELEASE SOFTWARE (fc1) adsl uptime is 39 minutes System returned to ROM by power-on System image file is "flash:c820-ov6y6-mz.122-15.T.bin" CISCO C827-4V (MPC855T) processor (revision 0x502) with 48128K/1024K bytes of memory. Processor board ID JAD04380OUW (346214163), with hardware revision 1987 CPU rev number 5 Bridging software. 4 POTS Ports 1 Ethernet/IEEE 802.3 interface(s) 1 ATM network interface(s) 128K bytes of non-volatile configuration memory. 8192K bytes of processor board System flash (Read/Write) 2048K bytes of processor board Web flash (Read/Write) Configuration register is 0x2102 -- Sincerely, William L. Thomson Jr. Support Group Obsidian-Studios, Inc. http://www.obsidian-studios.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