Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones
Peter, Did you look at http://www.cudatel.com? Probably just what you are looking for. GUI goodness based on FS. SDR Peter J. Zandvoort wrote: Matthew, I'm about in the same boat as you are, just on a smaller scale. We have a ton of Nortel telephony gear, but it's time to move out of the 90's and enter this millennium. My Cisco quote was in the same ballpark as yours. The Cisco stuff is mature, rock solid, meshes very well with their network gear and is actually relatively easy to set up and maintain if you know your way around IOS. I just refuse to pay that kind of money for yet another semi-proprietary solution. After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall. The developers and the community are great and available, but just starting out with SIP and voip in general, this may not be the best platform. So let the blasphemy begin :) SipX was a breeze to install (insert CD, boot, next next next...) and looks pretty solid. I believe they actually use FreeSWITCH for their voicemail and conferencing, internally. I just couldn't get my head around their GUI, ACD was too basic and had all kinds of issues getting stuff to just work. 3CX (Windows Only) was completely painless. It just worked. But I'm still not convinced that I want to run all my voice on a single windows box. Plus it's not free/open/etc and I don't want to lock myself in again. Although it's an asterisk based solution, I found trixbox to be very easy. Setup is automatic and everything just worked. The GUI is simple and logical enough that I can let somebody else handle the day-to-day phone setup and basic admin. I have my doubts about it scaling to 250 users, though. This may be a completely flawed strategy and I may very well be shooting myself in the foot by doing this, but I plan on piloting a trixbox install with a dozen or so users and see how stable it is. I'll keep a FreeSWITCH box next to it for the more advanced stuff. Once I get more comfortable with the intricacies of SIP and get some time to code a basic GUI for FreeSWITCH, I have a feeling that that trixbox is going to get phased out... Peter -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of mkitchin.pub...@gmail.com Sent: Tuesday, November 03, 2009 11:10 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones Michael Collins wrote: On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.pub...@gmail.com mailto:mkitchin.pub...@gmail.com mkitchin.pub...@gmail.com mailto:mkitchin.pub...@gmail.com wrote: I'm working on an alternative to a $120,000 Cisco phone system that my company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53 http://10.85.0.53. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3. Thanks, Matthew Kitchin Matthew, Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We think you'll find FS is as powerful as any software out there right now. Here's a handy wiki page that will help you get the diagnosing skills you need: http://wiki.freeswitch.org/wiki/Reporting_Bugs I'd say first thing to do is capture the SIP traffic to see if there are any clues. A normal temporary failure doesn't give you a lot of detail. :) If you're new to SIP debugging then the best thing to do is to capture the SIP trace and put it in the pastebin. (http://pastebin.freeswitch.org) You can also join the IRC channel #freeswitch on irc.freenode.net
Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones
Absolutely agreed. To use Matthew's original car metaphor: When you just got your learner's permit, the old Chevy may be a better choice than the Ferrari. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jason White Sent: Wednesday, November 04, 2009 1:42 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones Peter J. Zandvoort pe...@cindyandpeter.com wrote: After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall. The most flexible and sophisticated tools tend to have this characteristic, the best solution to which is a supportive community and good documentation. FreeSWITCH has the community; the documentation is improving thanks to ongoing efforts to extend, clarify and enhance the wiki. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones
I had the exact same problem with the Cisco phones not being able to receive calls. I fixed it by messing around with the NAT settings in the internal sofia profile. From what I remember, I just removed the param name=apply-nat-acl value=nat.auto/ line and everything worked fine. -- mm_202. On Wed, Nov 4, 2009 at 11:33 AM, Peter J. Zandvoort pe...@cindyandpeter.com wrote: Absolutely agreed. To use Matthew's original car metaphor: When you just got your learner's permit, the old Chevy may be a better choice than the Ferrari. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jason White Sent: Wednesday, November 04, 2009 1:42 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones Peter J. Zandvoort pe...@cindyandpeter.com wrote: After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall. The most flexible and sophisticated tools tend to have this characteristic, the best solution to which is a supportive community and good documentation. FreeSWITCH has the community; the documentation is improving thanks to ongoing efforts to extend, clarify and enhance the wiki. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Newbie trying to setup Cisco 7940 phones
I'm working on an alternative to a $120,000 Cisco phone system that my company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3. Thanks, Matthew Kitchin dsi sh conf -- Current *FLASH* Configuration -- Platform : Cisco Systems, Inc. IP Phone CP-7940G Elapsed Time: 01:01:06 dhcp_server : Disabled my_ip_addr : 10.86.11.50 subnet_mask : 255.255.0.0 defaultgw : 10.86.0.1 dyn_dns_addr_1 : 0.0.0.0 dyn_dns_addr_2 : 0.0.0.0 dns_addr : 10.85.0.11 dns_backup_1: 10.85.0.10 primary_tftp_addr : 10.86.10.58 dyn_tftp_addr : 0.0.0.0 my_mac_addr : 0012:7f98:eaa9 domain_name : dsi-corp.net my_name : SIP00127F98EAA9 Status Flags : 1231 image_version : P003-8-12-00 FirmLoadID : PC030301 DSPLoadID : PS03AT38 network_media_type : Auto network_port2_type : Hub/Switch dscpForAudio : 184 phone_label : Matthew Kitchin tftp_cfg_dir : phone_password : ** phone_prompt : dsi language : english sntp_mode : Unicast sntp_server : 10.85.0.10 time_zone : CST dst_offset : 01/00 dst_start_month : March dst_start_day : 0 dst_start_day_of_week : Sunday dst_start_week_of_month : 8 dst_start_time : 02/00 dst_stop_month : October dst_stop_day : 0 dst_stop_day_of_week : Sunday dst_stop_week_of_month : 8 dst_stop_time : 02/00 dst_auto_adjust : 1 time_format_24hr : 1 date_format : M/D/Y nat_enable : 0 nat_address : UNPROVISIONED voip_control_port : 5060 start_media_port : 16384 end_media_port : 32766 sync : 1 xml_card_dir : xml_card_file : CARD.XML telnet_level : 1 services_url : directory_url : logo_url : http_proxy_addr : UNPROVISIONED http_proxy_port : 80 garp_enable : 0 enable_vad : 0 dial_template : dialplan callerid_blocking : 0 anonymous_call_block : 0 autocomplete : 0 messages_uri : dnd_control : 2 preferred_codec : g711ulaw dtmf_outofband : avt_always dtmf_avt_payload : 101 dtmf_db_level : 3 dtmf_inband : 0 call_manager1_addr : UNPROVISIONED call_manager2_addr : UNPROVISIONED call_manager3_addr : UNPROVISIONED call_manager1_sip_port : 5060 call_manager2_sip_port : 5060 call_manager3_sip_port : 5060 call_manager5_addr : UNPROVISIONED call_manager5_sip_port : 5060 call_manager4_addr : UNPROVISIONED call_manager4_sip_port : 0 line1_name : 1008 line2_name : 1001 line1_authname : 1008 line2_authname : 1001 line1_password : ** line2_password : ** line1_shortname : 1008 line2_shortname : 1001 line1_displayname : 1008 line2_displayname : UNPROVISIONED line1_contact : UNPROVISIONED line2_contact : UNPROVISIONED proxy1_address : nshplpbx1.unix proxy2_address : nshplpbx1.unix proxy1_port : 5060 proxy2_port : 5060 sip_retx : 10 sip_invite_retx : 6 timer_t1 : 500 timer_t2 : 4000 timer_invite_expires : 180 timer_register_expires : 3600 proxy_register : 1 proxy_backup : proxy_emergency : proxy_backup_port : 5060 proxy_emergency_port : 5060 outbound_proxy : UNPROVISIONED outbound_proxy_port : 5060 nat_received_processing : 0 mwi_status : 0 call_waiting : 1 user_info : phone cnf_join_enable : 0 remote_party_id : 1 semi_attended_transfer : 1 transfer_onhook_enabled : 0 call_hold_ringback : 3 stutter_msg_waiting : 0 cfwd_url : call_stats : 0 auto_answer : 0 local_cfwd_enable : 1 timer_register_delta : 5 sip_max_forwards : 70 rfc_2543_hold : 0 version_stamp : timer_keepalive_expires : 120 connection_monitor_duration : 120 encrypt_key : ** 2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5067 0 acls to check for proxy 2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5085 network ip is a proxy [0] 2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5113 IP 10.86.10.58 Rejected by acl domains. Falling back to Digest auth. 2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5067 0 acls to check for proxy 2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5085 network ip is a proxy [0] 2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5113 IP 10.86.10.58 Rejected by acl domains. Falling back to Digest auth. 2009-11-03 15:39:50.797901 [NOTICE] switch_channel.c:613 New Channel sofia/internal/1...@nshplpbx1.unix [8c133ad4-67cf-4ffa-8655-56ffa0e3933d] 2009-11-03
Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones
Michael Collins wrote: On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.pub...@gmail.com mailto:mkitchin.pub...@gmail.com mkitchin.pub...@gmail.com mailto:mkitchin.pub...@gmail.com wrote: I'm working on an alternative to a $120,000 Cisco phone system that my company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53 http://10.85.0.53. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3. Thanks, Matthew Kitchin Matthew, Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We think you'll find FS is as powerful as any software out there right now. Here's a handy wiki page that will help you get the diagnosing skills you need: http://wiki.freeswitch.org/wiki/Reporting_Bugs I'd say first thing to do is capture the SIP traffic to see if there are any clues. A normal temporary failure doesn't give you a lot of detail. :) If you're new to SIP debugging then the best thing to do is to capture the SIP trace and put it in the pastebin. (http://pastebin.freeswitch.org) You can also join the IRC channel #freeswitch on irc.freenode.net http://irc.freenode.net and get some real-time help. There are some sharp folks in there, not the least of which are the three main FreeSWITCH developers. -MC Thank you. I think I did what you are looking for. I stopped FS and launched this command. TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch and captured all output to http://pastebin.freeswitch.org/10965 Does this tell you anything? I'm definitely new to SIP and phone system admin in general. I have plenty of network and Linux experience. With that in mind, someone on this mailing list emailed me directly and said SipX would be a better fit for me. Is that blasphemy for me to even mention? I went through the documentation and the provisioning aspect and web interface do look tempting to a novice. I apologize if this is like trying to buy a chevy at a ford dealership. I'm looking to deploy about 150 handsets at a corporate office and then 10 to 12 handsets at 120 remote locations. We are moving from an old key system, so our current features are very limited. We just need a few ACD groups, call history, and the other general basics. I first found Asterisk and read about some of the shortcomings. FS looks like the most robust solution. I have no idea where SipX would fit in. The people here are obviously a very knowledgeable group and I would gladly accept any thoughts, comments, etc. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones
Matthew, I'm about in the same boat as you are, just on a smaller scale. We have a ton of Nortel telephony gear, but it's time to move out of the 90's and enter this millennium. My Cisco quote was in the same ballpark as yours. The Cisco stuff is mature, rock solid, meshes very well with their network gear and is actually relatively easy to set up and maintain if you know your way around IOS. I just refuse to pay that kind of money for yet another semi-proprietary solution. After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall. The developers and the community are great and available, but just starting out with SIP and voip in general, this may not be the best platform. So let the blasphemy begin :) SipX was a breeze to install (insert CD, boot, next next next...) and looks pretty solid. I believe they actually use FreeSWITCH for their voicemail and conferencing, internally. I just couldn't get my head around their GUI, ACD was too basic and had all kinds of issues getting stuff to just work. 3CX (Windows Only) was completely painless. It just worked. But I'm still not convinced that I want to run all my voice on a single windows box. Plus it's not free/open/etc and I don't want to lock myself in again. Although it's an asterisk based solution, I found trixbox to be very easy. Setup is automatic and everything just worked. The GUI is simple and logical enough that I can let somebody else handle the day-to-day phone setup and basic admin. I have my doubts about it scaling to 250 users, though. This may be a completely flawed strategy and I may very well be shooting myself in the foot by doing this, but I plan on piloting a trixbox install with a dozen or so users and see how stable it is. I'll keep a FreeSWITCH box next to it for the more advanced stuff. Once I get more comfortable with the intricacies of SIP and get some time to code a basic GUI for FreeSWITCH, I have a feeling that that trixbox is going to get phased out... Peter -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of mkitchin.pub...@gmail.com Sent: Tuesday, November 03, 2009 11:10 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones Michael Collins wrote: On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.pub...@gmail.com mailto:mkitchin.pub...@gmail.com mkitchin.pub...@gmail.com mailto:mkitchin.pub...@gmail.com wrote: I'm working on an alternative to a $120,000 Cisco phone system that my company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53 http://10.85.0.53. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3. Thanks, Matthew Kitchin Matthew, Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We think you'll find FS is as powerful as any software out there right now. Here's a handy wiki page that will help you get the diagnosing skills you need: http://wiki.freeswitch.org/wiki/Reporting_Bugs I'd say first thing to do is capture the SIP traffic to see if there are any clues. A normal temporary failure doesn't give you a lot of detail. :) If you're new to SIP debugging then the best thing to do is to capture the SIP trace and put it in the pastebin. (http://pastebin.freeswitch.org) You can also join the IRC channel #freeswitch on irc.freenode.net http://irc.freenode.net and get some real-time help. There are some sharp folks in there, not the least of which are the three main FreeSWITCH developers. -MC Thank you. I think I did
Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones
Peter J. Zandvoort pe...@cindyandpeter.com wrote: After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall. The most flexible and sophisticated tools tend to have this characteristic, the best solution to which is a supportive community and good documentation. FreeSWITCH has the community; the documentation is improving thanks to ongoing efforts to extend, clarify and enhance the wiki. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
