[asterisk-users] Extensions Configuration
Hi all, I am building out a new platform and I need help with a couple of items. I need to have an extension 101 that is public (on business cards, in the directory, etc...) however I want this extension to exist as a hunt group with a ring all strategy so two phones (107 which is the private extension for the 101 user is run, and the 102 extension). The 107 extension should not have a separate voicemail and when the user at 107 presses the messages button they need to log into the 101 mailbox. When 107 dials other users internally it should show the callerid as 101. What is the best way to configure asterisk to to this? Second question, for the hunt groups I want to change the callerid display for incoming calls so the phone displays Boss's Line:123456789, but I want to make sure that when the user redials via the phone directory the number 123456789 is dialed directly. How do I change the caller id display for inbound calls and still have the directory work properly? Thanks in advance, Max ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 Schedule and Features/Changes
Hi all, Asterisk 1.4 was originally scheduled to be released early July 2006. Is there an update on the expected release of this version? Also is there a changelog or feature list available that lists the differences over 1.2? TIA, Max -- Max Clark http://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Follow Me
Thank You. On 2/21/06, C F [EMAIL PROTECTED] wrote: http://bugs.digium.com/view.php?id=5574 That is a patch that will do just that. On 2/21/06, Max Clark [EMAIL PROTECTED] wrote: Hi all, I am interested in a follow me script for Asterisk - specifically I am looking for one that will prompt the calling party to record their name and then call through a list of numbers playing the recording. If a digit is pressed by the recipient then the call is put through. Is there anything like this available as an example for Asterisk? TIA, Max -- Max Clark http://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Max Clark http://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fromuser required but overrides SetCallerID
Hi all, I have an asterisk box connecting to a SER instance for outbound (termination) calling. In order to authenticate with SER it seems that I have to use fromuser in the sip.conf in the peer section for the SER connection - with fromuser set I can make calls, without it I get a Forbidden - wrong password on authentication for INVITE error. The problem is that setting fromuser in the sip.conf overrides anything that I have set in the dialplan with SetCallerID. How do I work around this? TIA, Max -- Max Clark http://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Follow Me
Hi all, I am interested in a follow me script for Asterisk - specifically I am looking for one that will prompt the calling party to record their name and then call through a list of numbers playing the recording. If a digit is pressed by the recipient then the call is put through. Is there anything like this available as an example for Asterisk? TIA, Max -- Max Clark http://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Teliax Down?
I hate to burst your bubble but DOS attacks are a fact of life for IP based services. The bigger you get the more of a target you are. There are a ton of DOS prevention/mitigation appliances/services available in today's world. But they all rely on the same thing: having more bandwidth/capacity than your attacker. I've seen DOS attacks against ISP customers of mine that were pushing over a million packets per second across 50+ peering points. Not many networks can absorb that kind of thing. If your phones are that critical to your business you need to get dedicated service (aka T1), or switch to a service with static registration that can be protected with a good firewall. Max On 1/23/06, JCC [EMAIL PROTECTED] wrote: I've had problems for the last couple of weeks regarding incoming calls. Cant hear the party calling me (their voice sounds garbled/scrambled). If you haven't done so yet, I would recommend you post your complaint on their online forum as well under 'bugs'. You usually get some good responses from other Teliax users regarding the problem. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Ross CSent: Friday, January 20, 2006 8:40 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Teliax Down? I was having trouble too. I had trouble yesterday as well. I called and David said it was a "massive DDOS". Seems to get fixed pretty quickly when it does happen (5 minutes or so); however, for a business, 5 minutes without phones (people can't get a hold of your company) isn't really acceptable IMO. Also on co3. I couldn't even access their website during that time… From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Rusty DekemaSent: Friday, January 20, 2006 5:42 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Teliax Down? Is anyone else experiencing trouble with Teliax? I can only intermittently register to, and am not able to place any outgoing calls through my assigned gateway; voip-co3.teliax.com. -Rusty___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Max Clarkhttp://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OFF TOPIC: Core router upgrade for a voip colocation center
Josh,You are in a position that many of our customers have found themselves in. For ISPs/Colo operations starting out Cisco 720x, Foundry BigIron, RiverStone, and Extreme switches all present an aggressive price point for the performance. However once you pass this point Foundry, RiverStone and Extreme all start to become exponentially expensive due to the lack of parts on the resale market. I would advise you against the 7206/12008 upgrade for a couple of different reasons. 1) Like you said they arenearing their EOL date,2) Processor performance is limited based on today's standards (unless you want to shell out for the NPE-G1 or the PRP), 3) They have a limited number of fixed interface ports, and additional line cards are expensive. With the exception of sites running Sonet the 6500 platform is the only way to go. We have sites running the SUP 720 3BXL cards with over 20 full BGP sessions pushing Gigs worth of traffic through them. When you look at processor/memory utilization you wouldn't even know the switch was being used. For your configuration I would recommend a 6503/6504 with a Sup 32 (WS-SUP32-GE-3B)supervisor module... 1) The Sup 32 comes with enough processor/memory to handle BGP in real world situations (256MB standard, upgradable to 1GB),2) It has 8 SFP ports on the supervisor module which is enough for most mid tier applications, 3) 32GB shared bus,4) 15 million packets per second, and my favorite reason, 5) it runs IOS My company is based in Los Angeles, give me a call and I will be more than happy to go over all of this with you. Best, MaxMax ClarkCreative Thought, Inc.(866)231-7371 x 3874(213)784-3874 Direct(866)369-0953 24/7 SupportIT should facilitate business, we can help.On 1/23/06, josh harrington [EMAIL PROTECTED] wrote: Hello, hope this isn't too far offtopic here but being a troller for a long time here I've realized there is a great knowledge base so I wanted to at least see if i could get some tips.I help run a small colocation company in California and I am in the middle of recommending a new 'core router' platform for our network.We offer mainly colo and dedicated servers, and several of our clients use our space for VOIP services so quality even under high peak usage is a must.We are not huge, but as we have had near 200% growth in the past 12 months and need to expand our network asap to keep up. Simply put, I'd love to hear feedback and/or suggestions from any of you guys who have gone through this already. Our network map is real simple: [Carrier 7609] -- 100 mbit -- Our cisco 7206 -- 100 mbit -- racks [the racks on our end are a series of switches, mainly 2948gl3's] We push about 60 mbit to/from our (1) carrier at peak right now, and the router keeps up fine [its a cisco 7206 npe 150 btw, very low end on the 7206 line], and at peak we have under 50,000 packets per second, and our 7206 File router-choices.txt not changed so no update needed. [EMAIL PROTECTED] Hardware]$ cat router-choices.txt #1 - http://list.linux-vserver.org/archive/vserver/ #2 - webhostingtalk.com: jharington68/adam123 http://www.webhostingtalk.com/forumdisplay.php?f=44 #3 - asterisk mail list http://lists.digium.com/mailman/listinfo/asterisk-users #4 - cisco mail list? HOTMAIL [EMAIL PROTECTED]pw/adam123 Hello, hope this isn't too far offtopic here but being a troller for a long time here I've realized there is a great knowledge base so I wanted to at least see if i could get some tips.I help run a small colocation company in California and I am in the middle of recommending a new 'core router' platform for our network.We offer mainly colo and dedicated servers, and several of our clients use our space for VOIP services so quality even under high peak usage is a must.We are not huge, but as we have had near 200% growth in the past 12 months and need to expand our network asap to keep up. Simply put, I'd love to hear feedback and/or suggestions from any of you guys who have gone through this already. Our network map is real simple: [Carrier 7609] -- 100 mbit -- Our cisco 7206 -- 100 mbit -- racks [the racks on our end are a series of switches, mainly 2948gl3's] We push about 60 mbit to/from our (1) carrier at peak right now, and the router keeps up fine [its a cisco 7206 npe 150 btw, very low end on the 7206 line], and at peak we have under 50,000 packets per second, and our 7206 has little/no features enabled [just static routes and passing all traffic between 2 Ethernet 100 mbit interfaces]. To date we have had 2 problems, both were DOS attacks launched FROM one of our customer's servers flooding a full 100 mbit wire with more packets per second than the router could handle (the 2948gl3's spiked to about 50% cpu load during the attack but the 7200 literally just died for 3 minutes as the interface(s) all rebooted].So our main goal to grow is something that can handle a lot more in this arena against a DOS, and handle our future growth. In then next 12 months we plan
Re: [Asterisk-Users] Need a good extensions.conf sm bus config w/polycom phones
I'd love to see this as well. TIA, Max On 1/21/06, Thomas Johnson [EMAIL PROTECTED] wrote: Thanks!I'd love to see your extensions.conf file.I appreciate it.TomOn Jan 20, 2006, at 8:31 PM, Alexander Lopez wrote: Contact me off list, I have a sample extensions.conf file that I can share. It has Paging (one to one and One to Many) Ivr includes, time of da routing and it is geared towards Polycoms. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Thomas Johnson Sent: Friday, January 20, 2006 8:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Need a good extensions.conf sm bus config w/polycom phones Hello- We've got a patched-together extensions.conf that's barely working for us, and we need to get real about using Asterisk. We've got a couple of remote workers with Polycom IP-601 phones, and a single asterisk server, using a couple of incoming DIDs from teliax and sixtel. Does anyone have a good extensions.conf that they'd be willing to share, that provides a real-world tested dialplan?We'd love to see what other people are doing - (preferably those using all these cool features that polycom phones are capable of). Thanks- Tom ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Max Clarkhttp://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Failover Registration
Hi all, I would like to have two asterisk servers in a cluster. From what I understand using a mysql database I can store all of my peer/user information in the db and share this between servers. I can then take my polycom phone and register it to both of the asterisk servers at the same time - so if one were to go offline traffic would be redirected to the second. This works in theory for the end user - but how do I provide redundancy with my upstream providers? I.e. how do I fail over my registration to an upstream sip provider? Thanks in advance, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: DS3 - VoIP Hardware Recommendations
http://www.lucent.com/products/solution/0,,CTID+2013-STID+10443-SOID+589-LOCL+1,00.html Consolidates a variety of access lines (including channelized T1/E1, channelized DS3, ISDN PRI, leased T1/E1 or Fractional T1/E1 Frame Relay, and channelized T1/E1 frame relay) over high-speed digital trunks for up to 960 simultaneous data connections The Lucent TNT has been the standard for years. It works like a charm. Brian C. Fertig wrote: Trust me dude.. You don't want a lucent TNT. If your going all out for an DS3 and you don't want to multiplex it then you will need something that will take a DS3 which I don't believe TNT's do. Purchase an AS5400HPX they will and work very well. Set yourself up with some dialpeers etc and your good to go. Trust me. I have done it. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Sent: Wednesday, July 13, 2005 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: DS3 - VoIP Hardware Recommendations At 10:06 AM 7/13/2005, you wrote: Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM - VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features are needed, I think it can handle the routing. If not, I can whip up a SER box. We currently have a Cisco 7206VXR (1 voice resource) and a Cisco AS5300 (120 voice resources). The DS3 will also have SS7 signaling on it. Recommendations/comments/concerns/rants are graciously welcomed. Lucent TNT Thanks, Matthew ___ 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 This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ 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 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] Multiple Timezones with Asterisk
Hi all, I am curious if it is possible to have multiple timezones registered on an Asterisk server for Voicemail (i.e. so that PST users get PST time, and EST users get EST time)? Ideally I would like to set my Asterisk box to GMT and have a switch depending on where the user was registered from. Is this possible? Thanks, Max ___ 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] Multiple Timezones with Asterisk
In [zonemessages] pacific=America/Los_Angeles|'vm-received' Q 'digits/at' IMp in the [context] exten = pass,user,email,,tz=pacific -Max hank wrote: for that matter how do you set it up for pst? mine is set to est and its really anoying - Original Message - From: Max Clark [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 29, 2005 12:53 PM Subject: [Asterisk-Users] Multiple Timezones with Asterisk Hi all, I am curious if it is possible to have multiple timezones registered on an Asterisk server for Voicemail (i.e. so that PST users get PST time, and EST users get EST time)? Ideally I would like to set my Asterisk box to GMT and have a switch depending on where the user was registered from. Is this possible? Thanks, Max ___ 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 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 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] SIP Phone Config Generator
Hi all, Cisco/Polycom phones will pull their configuration via a tftp server to help manage mass deployments of phone systems. Are there any config generators available that will create the file for the tftp server? TIA, Max ___ 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] Level 3 SIP -- asterisk
Jim, I had a similar experience recently, a workaround to my problem is published on the web here: http://www.clarksys.com/archives/2005/06/25/handle-request-failed-to-authenticate-user/ http://www.nolata.org/wiki/Handle_request:_Failed_to_authenticate_user HTH, Max Jim Gottlieb wrote: Hi. Can anyone point me to some docs detailing how to set up a connection with Level 3 Communications? A customer of ours wants us to terminate some inbound service via Level 3 to our asterisk server. I've tried all sorts of settings but nothing yet has worked. SIP debug shows a 407 Proxy Authentication Required error. I haven't been able to find anything on the web, and the techs at Level 3 say they've never heard of asterisk and have no idea what I'm doing wrong. Any help would be appreciated. Thanks... ___ 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 Jim Gottlieb wrote: Hi. Can anyone point me to some docs detailing how to set up a connection with Level 3 Communications? A customer of ours wants us to terminate some inbound service via Level 3 to our asterisk server. I've tried all sorts of settings but nothing yet has worked. SIP debug shows a 407 Proxy Authentication Required error. I haven't been able to find anything on the web, and the techs at Level 3 say they've never heard of asterisk and have no idea what I'm doing wrong. Any help would be appreciated. Thanks... ___ 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 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] Asterisk server with remote monitoring capabilities
Maya, Where are you colocated? Remote reboot is something that we offer our customers standard with the rack space. I have found the Baytech products to be fantastic for remote reboot/remote serial access. You might want to look for something like this: http://www.baytech.net/products/showprod.php?prod=DS2-RPC (DS2-RPC) that offers both power switching and serial control in a 1U form factor. It's a must have for remote systems. -Max beonice wrote: Hello, all. I'm tired of having to drive out to the colocation facility each time my dedicated asterisk server craps out, just to press the button to do a hard reboot. (I'm running 1.05 stable at present, no telephony hardware, as this is mainly a system that receives calls, no dial-out ability is needed.) I've been looking at the fancy xeon-based systems listed on ebay for a couple of hundred dollars, in the hope that some of them have remote reboot capabilities, but most of the sellers don't mention this ability, and by the time I send out email, the item is already taken anyway. :) So, to cut the long story short, has anyone used one of these server-class machines with remote reboot capability, and does it really help? Are there any particular configurations to stay away from? The wiki doesn't talk specifically about issues regarding dual-CPU machines, but in following the chat here on asterisk-users, it seems there are definitely issues there ... can anyone elaborate? I don't want to spend money on a fancy system that turns out to be useless for my purposes. Thanks for any insight! Cheers, Maya Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.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 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] Sip Sidecar Options
Hi all, One of the things that I keep being asked for is a sidecar for the receptionist phone. Are there any SIP phones available on the market with a sidecar in addition to the snom? Or is the snom my only option? Any help would be appreciated. Thanks, Max ___ 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] accountcode not present in cdr
But what do people do with large LCR rules... Build special contexts for each peer/user and then include the main LCR context? This seems a little cludgy. Is there any way to have the dialplan context set the account for cdr based on the accountcode defined in the sip.conf? At least this way I could have a single, generic dialplan. -Max Andres wrote: Max Clark wrote: Hi all, I have what I hope will be a simple problem. In my sip.conf I have defined the accountcode field (see below), and they system does not report any errors when I reload the configuration. However when I look at my cdr detail (either the csv on disk, or the mysql info) the accountcode that I have specified is missing. I have scoured the list and have seen a few postings on this with no solutions. What should I be looking at to debug this. Thanks in advance, Max Setting the accountcode in sip.conf is totally unreliable. It does not work in many cases. Your best bet is to set it in a context via the command: SetAccount([account]): Set the channel account code for billing I am running Asterisk 1.0.7 on CentOS 4.0: Linux 2.6.9-11.EL #1 Wed Jun 8 16:59:52 CDT 2005 i686 i686 i386 GNU/Linux [provider1] accountcode=provider1 type=friend host=10.1.1.1 dtmfmode=rfc2833 username=user secret=12345 qualify=no canreinvite=no insecure=very disallow=all allow=ulaw allow=gsm [provider2] type=peer accountcode=provider2 secret=54321 username=user host=10.1.1.10 dtmfmode=rfc2833 ___ 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] accountcode not present in cdr
But what do people do with large LCR rules... Build special contexts for each peer/user and then include the main LCR context? This seems a little cludgy. Is there any way to have the dialplan context set the account for cdr based on the accountcode defined in the sip.conf? At least this way I could have a single, generic dialplan. -Max Andres wrote: Max Clark wrote: Hi all, I have what I hope will be a simple problem. In my sip.conf I have defined the accountcode field (see below), and they system does not report any errors when I reload the configuration. However when I look at my cdr detail (either the csv on disk, or the mysql info) the accountcode that I have specified is missing. I have scoured the list and have seen a few postings on this with no solutions. What should I be looking at to debug this. Thanks in advance, Max Setting the accountcode in sip.conf is totally unreliable. It does not work in many cases. Your best bet is to set it in a context via the command: SetAccount([account]): Set the channel account code for billing I am running Asterisk 1.0.7 on CentOS 4.0: Linux 2.6.9-11.EL #1 Wed Jun 8 16:59:52 CDT 2005 i686 i686 i386 GNU/Linux [provider1] accountcode=provider1 type=friend host=10.1.1.1 dtmfmode=rfc2833 username=user secret=12345 qualify=no canreinvite=no insecure=very disallow=all allow=ulaw allow=gsm [provider2] type=peer accountcode=provider2 secret=54321 username=user host=10.1.1.10 dtmfmode=rfc2833 ___ 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] accountcode not present in cdr
Hi all, I have what I hope will be a simple problem. In my sip.conf I have defined the accountcode field (see below), and they system does not report any errors when I reload the configuration. However when I look at my cdr detail (either the csv on disk, or the mysql info) the accountcode that I have specified is missing. I have scoured the list and have seen a few postings on this with no solutions. What should I be looking at to debug this. Thanks in advance, Max I am running Asterisk 1.0.7 on CentOS 4.0: Linux 2.6.9-11.EL #1 Wed Jun 8 16:59:52 CDT 2005 i686 i686 i386 GNU/Linux [provider1] accountcode=provider1 type=friend host=10.1.1.1 dtmfmode=rfc2833 username=user secret=12345 qualify=no canreinvite=no insecure=very disallow=all allow=ulaw allow=gsm [provider2] type=peer accountcode=provider2 secret=54321 username=user host=10.1.1.10 dtmfmode=rfc2833 ___ 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] Broadvoice Down?
Is anyone else having difficulty with their Broadvoice service? When I dial my number right now it rings either fast busy or tells me it cannot complete the call. I can make outgoing calls from my system through broadvoice however. Seems their inbound trunks hit capacity? Am I alone in this? -Max -- Max Clark max [at] clarksys.com http://www.clarksys.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] QOS Routers
Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. TIA, Max -- Max Clark max [at] clarksys.com http://www.clarksys.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] DID via SIP/IAX
Hi all, Can anyone point me to configuration examples for DID across SIP/IAX trunks? I want to be able to take in a DID from a T1 and forward it to another PBX via SIP/IAX. Thanks in advance, Max -- Max Clark max [at] clarksys.com http://www.clarksys.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-write callerid?
What about Callerid on call forwarding? I.e. an external call comes in and is forwarded to a cell phone, how do I make the callerid that is displayed on the cell phone the same as the inbound call? Thanks, Max Max Clark max [at] clarksys.com http://www.clarksys.com Remco Barende wrote: Is it possible to rewrite caller id's? I would like to have sip phones appear by their local cid (like Henk 208) but when they call out using the PRI I would like their full DID (MSN) to appear (like 0031201234567) I could ofcourse set callerid to the main phonenumber but surely there must be a better solution? Thanks!! Remco ___ 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 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] Advanced Cisco 7960 Config
I can't believe that the 7960 doesn't have a hot keypad. That has to be one of the more annoying things I've heard. Can you point me to a good dialplan.xml example that I can use on my phones? Thanks, Max Max Clark max [at] clarksys.com http://www.clarksys.com Shaun Ewing wrote: On Thu, 24 Mar 2005 21:26:27 -0800, Max Clark [EMAIL PROTECTED] wrote: Hi all, Good evening I have a working (it was a pain) set of Cisco 7960 phones. In order to dial I have to either pick up the handset or select the line and then dial the extension or outside line. How do I configure the dialplan so I can: - Start dialing via the keypad and have the phone automatically go to speaker on the first line? The 7960 doesn't have a hot keypad (the cheaper and less featured in other ways 7905G/7912G phones do though - go figure). You need to press Speaker first. - Give the user dialtone after they dial '9'? In your dialplan, add a , after 9. eg: TEMPLATE MATCH=9,.* Timeout=3 User=Phone/ A while ago I found a cool asterisk/penguin logo to use on the phone, can anyone point me to a place I can download this again? Wouldn't have a clue, but would also like to know :) -Shaun Thanks in advance, Max -- Max Clark max [at] clarksys.com http://www.clarksys.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 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] Advanced Cisco 7960 Config
Hi all, I have a working (it was a pain) set of Cisco 7960 phones. In order to dial I have to either pick up the handset or select the line and then dial the extension or outside line. How do I configure the dialplan so I can: - Start dialing via the keypad and have the phone automatically go to speaker on the first line? - Give the user dialtone after they dial '9'? A while ago I found a cool asterisk/penguin logo to use on the phone, can anyone point me to a place I can download this again? Thanks in advance, Max -- Max Clark max [at] clarksys.com http://www.clarksys.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] Polycom vs. Cisco IP Phones
Hi all, I am working on building a new VoIP PBX. Looking at the current market for phones it seems my best enterprise options are the Cisco and Polycom phones. I have some experiance with the Cisco 7940G, but the process of flashing the phone with the SIP firmware left a bad taste in my mouth (not to mention the added expense for the phone). What is the general consensis about the polycom IP phones? Are they good? Are they better than Cisco? What do I do for the receptionist's station? Thanks in advance, Max -- Max Clark max [at] clarksys.com http://www.clarksys.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] Help With Broadvoice {Scanned}
David, Thanks for the reply. Just to clarify, is the register and first type=friend block all within the [general] section of sip.conf? Thanks, Max Max Clark max [at] clarksys.com http://www.clarksys.com David Shaw wrote: Here is my conf files. sip.conf register = phone#:sip/[EMAIL PROTECTED] type=friend username=phone# fromuser=phone# secret=sip/passwd host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice1 dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=no [bv-in-1] type=friend host=sip.broadvoice.com context=from-broadvoice1 dtmfmode=inband canreinvite=no nat=no allow=ulaw extensions.conf exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _011,1,Dial(SIP/[EMAIL PROTECTED]) On Wed, 2005-02-16 at 00:52 -0500, Randy Johnson wrote: Remember that the password is not your broadvoice website password but the one you need to get from broadvoice support. Randy Greg Hill wrote: On Tue, 15 Feb 2005, Max Clark wrote: I have experimented with several configs based on different pages and threads but nothing is working. How do I properly configure my broadvoice account? [general] register = [EMAIL PROTECTED]:pass:[EMAIL PROTECTED] the register I'm using looks like this: register = 310584:pass@sip.broadvoice.com [broadvoice] type=peer host=sip.broadvoice.com secret=pass fromuser=310584 fromdomain=sip.broadvoice.com context=incoming dtmfmode=inband canreinvite=no nat=yes qualify=yes try: [broadvoice] type=peer username=310584 secret=pass host=sip.broadvoice.com port=5060 context=incoming fromuser=310584 fromdomain=sip.broadvoice.com dtmfmode=inband canreinvite=no insecure=very permit=147.135.8.128/32 qualify=yes and adjust your permit= line to match the IP of the BV proxy you've set in your /etc/hosts (proxy.[chi|lax|dca|one other I forgot].broadvoice.com). Try a blend of this stuff with whatever the most recent recommendation on their support page says. Greg ___ 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 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 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] Help With Broadvoice
Hi all, I am trying to configure my Asterisk PBX to connect to a broadvoice account. I want incoming calls to go to the s extension and to ultimately have outbound calls routed through my broadvoice account(s). I cannot for the life of me get this to register correctly. I am currently receiving this message on the console: Failed to authenticate on Register to sip:[EMAIL PROTECTED] I have experimented with several configs based on different pages and threads but nothing is working. How do I properly configure my broadvoice account? Thanks, Max [general] register = [EMAIL PROTECTED]:pass:[EMAIL PROTECTED] [broadvoice] type=peer host=sip.broadvoice.com secret=pass fromuser=310584 fromdomain=sip.broadvoice.com context=incoming dtmfmode=inband canreinvite=no nat=yes qualify=yes -- Max Clark max [at] clarksys.com http://www.clarksys.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] Help With Broadvoice
Greg, Thanks for the suggestions. I made the changes in your response (and even converted back to the the strange way broadvoice wants you to register. I still get this error message. Feb 15 21:43:55 NOTICE[15983]: chan_sip.c:7547 handle_response: Failed to authenticate on REGISTER to 'sip:[EMAIL PROTECTED];tag=as20df4509' -Max Max Clark max [at] clarksys.com http://www.clarksys.com Greg Hill wrote: On Tue, 15 Feb 2005, Max Clark wrote: I have experimented with several configs based on different pages and threads but nothing is working. How do I properly configure my broadvoice account? [general] register = [EMAIL PROTECTED]:pass:[EMAIL PROTECTED] the register I'm using looks like this: register = 310584:pass@sip.broadvoice.com [broadvoice] type=peer host=sip.broadvoice.com secret=pass fromuser=310584 fromdomain=sip.broadvoice.com context=incoming dtmfmode=inband canreinvite=no nat=yes qualify=yes try: [broadvoice] type=peer username=310584 secret=pass host=sip.broadvoice.com port=5060 context=incoming fromuser=310584 fromdomain=sip.broadvoice.com dtmfmode=inband canreinvite=no insecure=very permit=147.135.8.128/32 qualify=yes and adjust your permit= line to match the IP of the BV proxy you've set in your /etc/hosts (proxy.[chi|lax|dca|one other I forgot].broadvoice.com). Try a blend of this stuff with whatever the most recent recommendation on their support page says. Greg ___ 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 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] Delay answering inbound calls
Hi all, I have a developer kit that I have been playing arround with. I have managed to get the system configured properly and add an analog and a few sip extensions. I have noticied that on inbound calls the system takes 5 rings before it will execute the script (this is the demo script), I have even tried taking out the Wait in the demo but that does not seem to help. How to I configure asterisk to answer calls immediately? Thanks, Max -- Max Clark max [at] clarksys.com http://www.clarksys.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] Cisco 7960 Beating a Dead Horse
Hi all, Gene pointed out that the OS79XX.TXT and the SIP*.cnf files have to be in Unix format without CR/LF. This got me to the point where the phone requests and retrieves the OS79XX.TXT file, and downloads the firmware specified from the tftpd server... only thing is, the phone never finishes upgrading the firmware. My phone now says Upgrading Software and every 10 seconds or so the tftpd server log shows a request for the firmware. The phone never seems to complete this task and is perminately in the Upgrading software phase. I have tried the 3.0, 4.4, 5.3, and 6.3 software, all with the same results. Has anyone experianced this problem before? What should I check? Thanks in advance, Max Max Clark max [at] clarksys.com http://www.clarksys.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] Proper Contexts in extensions.conf
Hi all, I am looking for examples of the extensions.conf that puts all incoming calls into a context where extensions can be dials, and all phones in a context where extensions and outside calls can be dialed. i.e. I have seen: [incoming] include = sip-extensions [sip-extensions] include = longdistance [longdistance] Doesn't this allow any internal callers to make external calls? How do you properly set this up? Thanks, Max -- Max Clark max [at] clarksys.com http://www.clarksys.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] Cisco 7960 Beating a Dead Horse
Hi all, So I have been reading through the docs available online and the different threads on this list, but I cannot seem to get this phone to work. I have configured the OS79XX.TXT and SIP/SEP*.cnf files (see attached), when I configure the phone to point to my tftp server and reboot it I get this message: Connection received from 10.6.0.224 on port 50608 [09/02 12:16:11.750] Read request for file OS79XX.TXT. Mode octet [09/02 12:16:11.750] OS79XX.TXT: sent 1 blk, 12 bytes in 0 s. 0 blk resent [09/02 12:16:11.750] Connection received from 10.6.0.224 on port 50609 [09/02 12:16:11.750] Read request for file P0S3-06-.bin. Mode octet [09/02 12:16:11.765] File P0S3-06-.bin : error 2 in system call CreateFile The system cannot find the file specified. [09/02 12:16:11.765] Connection received from 10.6.0.224 on port 50610 [09/02 12:16:11.765] Read request for file SEP0007EB0630A6.cnf. Mode octet [09/02 12:16:11.765] SEP0007EB0630A6.cnf: sent 7 blks, 3469 bytes in 0 s. 0 blk resent [09/02 12:16:11.781] Which lead me to believe that there was an 8.3 naming issue when using a windows based tftpd server. So I changed the file names of my image to an 8.3 structure, updated the configuration files and rebooted. After I do that I see the request and transfer of the OS79XX.TXT and the SEP0007EB0630A6.cnf but nothing for the firmware, or anything about the SIP configuration files. Just out of curiostity looking at my phone config at this point the call manager is pointing to the tftp server. What gives? How do I get this phone to download the SIP firmware? Best, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com P0S3-06-3-00# SIP Configuration Generic File (start) image_version: P0S3-06-3-00 preferred_codec: g711ulaw #preferred_codec: g729a # Proxy Server proxy1_address: 10.6.0.223 proxy2_address: proxy3_address: proxy4_address: proxy5_address: proxy6_address: # Line 1 Settings line1_name: Cisco7960; Line 1 Extension\User ID line1_displayname: Cisco7960 Line1; Line 1 Display Name line1_shortname: Line1 line1_authname: Cisco7960; Line 1 Registration Authentication line1_password: Cisco7960 ; Line 1 Registration Password # Line 2 Settings line2_name: ; Line 2 Extension\User ID line2_displayname:; Line 2 Display Name line2_shortname: line2_authname: ; Line 2 Registration Authentication line2_password: ; Line 2 Registration Password # Line 3 Settings line3_name: ; Line 3 Extension\User ID line3_displayname:; Line 3 Display Name line3_shortname: line3_authname: ; Line 3 Registration Authentication line3_password: ; Line 3 Registration Password # Line 4 Settings line4_name: ; Line 4 Extension\User ID line4_displayname:; Line 4 Display Name line4_shortname: line4_authname: ; Line 4 Registration Authentication line4_password: ; Line 4 Registration Password # Line 5 Settings line5_name: ; Line 5 Extension\User ID line5_displayname:; Line 5 Display Name line5_shortname: line5_authname: ; Line 5 Registration Authentication line5_password: ; Line 5 Registration Password # Line 6 Settings line6_name: ; Line 6 Extension\User ID line6_displayname:; Line 6 Display Name line6_shortname: line6_authname: ; Line 6 Registration Authentication line6_password: ; Line 6 Registration Password # Emergency Proxy info proxy_emergency: proxy_emergency_port: 5060 # Backup Proxy info proxy_backup: 10.6.0.223 proxy_backup_port: 5060 # Outbound Proxy info outbound_proxy: 10.6.0.223 outbound_proxy_port: 5060 proxy_register: 1 timer_register_expires : 120 # NAT/Firewall Traversal nat_enable: 1 nat_address: voip_control_port: 5060 start_media_port: 16384 end_media_port: 32766 nat_received_processing: 1 # Phone Label (Text desired to be displayed in upper right corner) phone_label: Garrett - ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: PST sntp_server: 136.159.2.254 ; SNTP Server IP Address sntp_mode: directedbroadcast; unicast, multicast, anycast, or directedbroadcast (default) # Telnet Level (enable or disable the ability to telnet into this phone telnet_level: 2 ; 0-Disabled (default), 1-Enabled, 2-Privileged # Phone prompt/password for telnet/console session phone_prompt: 123 ; Telnet/Console Prompt phone_password: 123 ; Telnet/Console Password # Enable_VAD (1-enabled, 0-disabled) enable_vad: 0 # Network Media Type (auto, full100, full10, half100, half10) network_media_type: auto user_info: phone tftp_cfg_dir: / sync: 1 # URL for external Directory location logo_url: http
Re: [Asterisk-Users] Cisco 7960 Beating a Dead Horse
BJ, I tried removing the SEP* files and changing the OS79XX.TXT to the 7.3 version of the firmware and then got the message below in my tftpd server. How do I check the version of the Firmware that the phone is running? Based on what you said I have a feeling that I need to go way back in version numbers to get current. Thanks, Max Connection received from 10.6.0.224 on port 50608 [09/02 13:14:02.218] Read request for file OS79XX.TXT. Mode octet [09/02 13:14:02.234] OS79XX.TXT: sent 1 blk, 12 bytes in 0 s. 0 blk resent [09/02 13:14:02.234] Connection received from 10.6.0.224 on port 50609 [09/02 13:14:02.234] Read request for file P0S3-07-.bin. Mode octet [09/02 13:14:02.234] File P0S3-07-.bin : error 2 in system call CreateFile The system cannot find the file specified. [09/02 13:14:02.234] Connection received from 10.6.0.224 on port 50610 [09/02 13:14:02.234] Read request for file SEP0007EB0630A6.cnf. Mode octet [09/02 13:14:02.250] File SEP0007EB0630A6.cnf : error 2 in system call CreateFile The system cannot find the file specified. [09/02 13:14:02.250] Connection received from 10.6.0.224 on port 50611 [09/02 13:14:02.250] Read request for file SEPDefault.cnf. Mode octet [09/02 13:14:02.250] File SEPDefault.cnf : error 2 in system call CreateFile The system cannot find the file specified. [09/02 13:14:02.250] Max Clark max [at] clarksys.com http://www.clarksys.com BJ Weschke wrote: You're probably going to want to go ahead and remove the SEPmac address.cnf.xml file from the TFTP server and leave the SIPmac address.cnf file in place. I think with version 6 or 7 of the firmware (don't recall which one - probably the one that starts using the Universal App Loader) the phone starts looking for the SEPmac address.cnf.xml file and will default to call manager mode if it finds it. Additionally, once you get into v7.x of the firmware, the OS79XX.TXT file goes from P0S3-XX-X-XX to P0O3-XX-X-XX even though you're going to use SIP. SIPDefault.cnf then contains the version of the software you actually want to use: # eg.. # SIP Default Generic Configuration File # Image Version image_version: P0S3-07-3-00 This threw me for a loop initially as my phone factory shipped with v3.3 of a factory image and while upgrading up through the versions, you had to initially indicate in OS79XX.TXT whether or not you were going to do SIP or SCCP via the string that was in OS79XX.TXT. With the UAL, you no longer do it that way. :-/ On Wed, 09 Feb 2005 12:20:01 -0800, Max Clark [EMAIL PROTECTED] wrote: Hi all, So I have been reading through the docs available online and the different threads on this list, but I cannot seem to get this phone to work. I have configured the OS79XX.TXT and SIP/SEP*.cnf files (see attached), when I configure the phone to point to my tftp server and reboot it I get this message: Connection received from 10.6.0.224 on port 50608 [09/02 12:16:11.750] Read request for file OS79XX.TXT. Mode octet [09/02 12:16:11.750] OS79XX.TXT: sent 1 blk, 12 bytes in 0 s. 0 blk resent [09/02 12:16:11.750] Connection received from 10.6.0.224 on port 50609 [09/02 12:16:11.750] Read request for file P0S3-06-.bin. Mode octet [09/02 12:16:11.765] File P0S3-06-.bin : error 2 in system call CreateFile The system cannot find the file specified. [09/02 12:16:11.765] Connection received from 10.6.0.224 on port 50610 [09/02 12:16:11.765] Read request for file SEP0007EB0630A6.cnf. Mode octet [09/02 12:16:11.765] SEP0007EB0630A6.cnf: sent 7 blks, 3469 bytes in 0 s. 0 blk resent [09/02 12:16:11.781] Which lead me to believe that there was an 8.3 naming issue when using a windows based tftpd server. So I changed the file names of my image to an 8.3 structure, updated the configuration files and rebooted. After I do that I see the request and transfer of the OS79XX.TXT and the SEP0007EB0630A6.cnf but nothing for the firmware, or anything about the SIP configuration files. Just out of curiostity looking at my phone config at this point the call manager is pointing to the tftp server. What gives? How do I get this phone to download the SIP firmware? Best, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com P0S3-06-3-00 # SIP Configuration Generic File (start) image_version: P0S3-06-3-00 preferred_codec: g711ulaw #preferred_codec: g729a # Proxy Server proxy1_address: 10.6.0.223 proxy2_address: proxy3_address: proxy4_address: proxy5_address: proxy6_address: # Line 1 Settings line1_name: Cisco7960; Line 1 Extension\User ID line1_displayname: Cisco7960 Line1; Line 1 Display Name line1_shortname: Line1 line1_authname: Cisco7960; Line 1 Registration Authentication line1_password: Cisco7960 ; Line 1 Registration Password # Line 2 Settings line2_name: ; Line 2 Extension\User ID line2_displayname:; Line 2 Display Name line2_shortname: line2_authname: ; Line 2 Registration Authentication
Re: [Asterisk-Users] Cisco 7960 Beating a Dead Horse
What I don't understand is that the phone will request and download the SIP/SEP*.cnf files which are well over 8.3, and if I change the file name to an 8.3 format the phone doesn't even attempt to download the firmware. -Max Max Clark max [at] clarksys.com http://www.clarksys.com BJ Weschke wrote: Hey there - At second look, this definitely looks like an 8/3 problem because if you look at the second file request, it's trying to get the firmware file but you're only seeing the first 8 chars in the filename it's asking for. On Wed, 09 Feb 2005 13:16:36 -0800, Max Clark [EMAIL PROTECTED] wrote: BJ, I tried removing the SEP* files and changing the OS79XX.TXT to the 7.3 version of the firmware and then got the message below in my tftpd server. How do I check the version of the Firmware that the phone is running? Based on what you said I have a feeling that I need to go way back in version numbers to get current. Thanks, Max Connection received from 10.6.0.224 on port 50608 [09/02 13:14:02.218] Read request for file OS79XX.TXT. Mode octet [09/02 13:14:02.234] OS79XX.TXT: sent 1 blk, 12 bytes in 0 s. 0 blk resent [09/02 13:14:02.234] Connection received from 10.6.0.224 on port 50609 [09/02 13:14:02.234] Read request for file P0S3-07-.bin. Mode octet [09/02 13:14:02.234] File P0S3-07-.bin : error 2 in system call CreateFile The system cannot find the file specified. [09/02 13:14:02.234] Connection received from 10.6.0.224 on port 50610 [09/02 13:14:02.234] Read request for file SEP0007EB0630A6.cnf. Mode octet [09/02 13:14:02.250] File SEP0007EB0630A6.cnf : error 2 in system call CreateFile The system cannot find the file specified. [09/02 13:14:02.250] Connection received from 10.6.0.224 on port 50611 [09/02 13:14:02.250] Read request for file SEPDefault.cnf. Mode octet [09/02 13:14:02.250] File SEPDefault.cnf : error 2 in system call CreateFile The system cannot find the file specified. [09/02 13:14:02.250] Max Clark max [at] clarksys.com http://www.clarksys.com BJ Weschke wrote: You're probably going to want to go ahead and remove the SEPmac address.cnf.xml file from the TFTP server and leave the SIPmac address.cnf file in place. I think with version 6 or 7 of the firmware (don't recall which one - probably the one that starts using the Universal App Loader) the phone starts looking for the SEPmac address.cnf.xml file and will default to call manager mode if it finds it. Additionally, once you get into v7.x of the firmware, the OS79XX.TXT file goes from P0S3-XX-X-XX to P0O3-XX-X-XX even though you're going to use SIP. SIPDefault.cnf then contains the version of the software you actually want to use: # eg.. # SIP Default Generic Configuration File # Image Version image_version: P0S3-07-3-00 This threw me for a loop initially as my phone factory shipped with v3.3 of a factory image and while upgrading up through the versions, you had to initially indicate in OS79XX.TXT whether or not you were going to do SIP or SCCP via the string that was in OS79XX.TXT. With the UAL, you no longer do it that way. :-/ On Wed, 09 Feb 2005 12:20:01 -0800, Max Clark [EMAIL PROTECTED] wrote: Hi all, So I have been reading through the docs available online and the different threads on this list, but I cannot seem to get this phone to work. I have configured the OS79XX.TXT and SIP/SEP*.cnf files (see attached), when I configure the phone to point to my tftp server and reboot it I get this message: Connection received from 10.6.0.224 on port 50608 [09/02 12:16:11.750] Read request for file OS79XX.TXT. Mode octet [09/02 12:16:11.750] OS79XX.TXT: sent 1 blk, 12 bytes in 0 s. 0 blk resent [09/02 12:16:11.750] Connection received from 10.6.0.224 on port 50609 [09/02 12:16:11.750] Read request for file P0S3-06-.bin. Mode octet [09/02 12:16:11.765] File P0S3-06-.bin : error 2 in system call CreateFile The system cannot find the file specified. [09/02 12:16:11.765] Connection received from 10.6.0.224 on port 50610 [09/02 12:16:11.765] Read request for file SEP0007EB0630A6.cnf. Mode octet [09/02 12:16:11.765] SEP0007EB0630A6.cnf: sent 7 blks, 3469 bytes in 0 s. 0 blk resent [09/02 12:16:11.781] Which lead me to believe that there was an 8.3 naming issue when using a windows based tftpd server. So I changed the file names of my image to an 8.3 structure, updated the configuration files and rebooted. After I do that I see the request and transfer of the OS79XX.TXT and the SEP0007EB0630A6.cnf but nothing for the firmware, or anything about the SIP configuration files. Just out of curiostity looking at my phone config at this point the call manager is pointing to the tftp server. What gives? How do I get this phone to download the SIP firmware? Best, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com P0S3-06-3-00 # SIP Configuration Generic File (start) image_version: P0S3-06-3-00 preferred_codec: g711ulaw #preferred_codec: g729a # Proxy Server proxy1_address: 10.6.0.223 proxy2_address: proxy3_address
Re: [Asterisk-Users] Cisco 7960 Beating a Dead Horse
Steve, No, not SIP version 8.3, but the SIP firmware in an 8.3 formatted file name (i.e. DOS). -Max Max Clark max [at] clarksys.com http://www.clarksys.com Steve Blair wrote: I'm not sure of your configuration but last time I looked there wasn't a SIP v8.3. Unless that has changed I'd suggest getting the correct version of SIP if that is what you'd like to test. Max Clark wrote: What I don't understand is that the phone will request and download the SIP/SEP*.cnf files which are well over 8.3, and if I change the file name to an 8.3 format the phone doesn't even attempt to download the firmware. -Max Max Clark max [at] clarksys.com http://www.clarksys.com BJ Weschke wrote: Hey there - At second look, this definitely looks like an 8/3 problem because if you look at the second file request, it's trying to get the firmware file but you're only seeing the first 8 chars in the filename it's asking for. On Wed, 09 Feb 2005 13:16:36 -0800, Max Clark [EMAIL PROTECTED] wrote: BJ, I tried removing the SEP* files and changing the OS79XX.TXT to the 7.3 version of the firmware and then got the message below in my tftpd server. How do I check the version of the Firmware that the phone is running? Based on what you said I have a feeling that I need to go way back in version numbers to get current. Thanks, Max Connection received from 10.6.0.224 on port 50608 [09/02 13:14:02.218] Read request for file OS79XX.TXT. Mode octet [09/02 13:14:02.234] OS79XX.TXT: sent 1 blk, 12 bytes in 0 s. 0 blk resent [09/02 13:14:02.234] Connection received from 10.6.0.224 on port 50609 [09/02 13:14:02.234] Read request for file P0S3-07-.bin. Mode octet [09/02 13:14:02.234] File P0S3-07-.bin : error 2 in system call CreateFile The system cannot find the file specified. [09/02 13:14:02.234] Connection received from 10.6.0.224 on port 50610 [09/02 13:14:02.234] Read request for file SEP0007EB0630A6.cnf. Mode octet [09/02 13:14:02.250] File SEP0007EB0630A6.cnf : error 2 in system call CreateFile The system cannot find the file specified. [09/02 13:14:02.250] Connection received from 10.6.0.224 on port 50611 [09/02 13:14:02.250] Read request for file SEPDefault.cnf. Mode octet [09/02 13:14:02.250] File SEPDefault.cnf : error 2 in system call CreateFile The system cannot find the file specified. [09/02 13:14:02.250] Max Clark max [at] clarksys.com http://www.clarksys.com BJ Weschke wrote: You're probably going to want to go ahead and remove the SEPmac address.cnf.xml file from the TFTP server and leave the SIPmac address.cnf file in place. I think with version 6 or 7 of the firmware (don't recall which one - probably the one that starts using the Universal App Loader) the phone starts looking for the SEPmac address.cnf.xml file and will default to call manager mode if it finds it. Additionally, once you get into v7.x of the firmware, the OS79XX.TXT file goes from P0S3-XX-X-XX to P0O3-XX-X-XX even though you're going to use SIP. SIPDefault.cnf then contains the version of the software you actually want to use: # eg.. # SIP Default Generic Configuration File # Image Version image_version: P0S3-07-3-00 This threw me for a loop initially as my phone factory shipped with v3.3 of a factory image and while upgrading up through the versions, you had to initially indicate in OS79XX.TXT whether or not you were going to do SIP or SCCP via the string that was in OS79XX.TXT. With the UAL, you no longer do it that way. :-/ On Wed, 09 Feb 2005 12:20:01 -0800, Max Clark [EMAIL PROTECTED] wrote: Hi all, So I have been reading through the docs available online and the different threads on this list, but I cannot seem to get this phone to work. I have configured the OS79XX.TXT and SIP/SEP*.cnf files (see attached), when I configure the phone to point to my tftp server and reboot it I get this message: Connection received from 10.6.0.224 on port 50608 [09/02 12:16:11.750] Read request for file OS79XX.TXT. Mode octet [09/02 12:16:11.750] OS79XX.TXT: sent 1 blk, 12 bytes in 0 s. 0 blk resent [09/02 12:16:11.750] Connection received from 10.6.0.224 on port 50609 [09/02 12:16:11.750] Read request for file P0S3-06-.bin. Mode octet [09/02 12:16:11.765] File P0S3-06-.bin : error 2 in system call CreateFile The system cannot find the file specified. [09/02 12:16:11.765] Connection received from 10.6.0.224 on port 50610 [09/02 12:16:11.765] Read request for file SEP0007EB0630A6.cnf. Mode octet [09/02 12:16:11.765] SEP0007EB0630A6.cnf: sent 7 blks, 3469 bytes in 0 s. 0 blk resent [09/02 12:16:11.781] Which lead me to believe that there was an 8.3 naming issue when using a windows based tftpd server. So I changed the file names of my image to an 8.3 structure, updated the configuration files and rebooted. After I do that I see the request and transfer of the OS79XX.TXT and the SEP0007EB0630A6.cnf but nothing for the firmware, or anything about the SIP configuration files. Just out of curiostity looking at my phone config
[Asterisk-Users] Cisco 7940 Configuration
Hello all, I recently purchased a Cisco 7940 IP phone to do some testing with (to validate a migration to asterisk for our internal PBX needs). I understand that I need to update the phone for it to support SIP, so I configured the phone with an IP address and pointed it at my tftp server. When I reboot the phone I am currently getting TFTP File Not Found SEPDefault.cnf in the status messages. How do I get this phone working? Thanks in advance, Max My /tftpboot directory -- -rwxrwxrwx1 root root 562 Jan 18 16:20 dialplan.xml -rwxrwxrwx1 root root 12 Jan 18 16:20 OS79XX.TXT -rwxrwxrwx1 root root 367506 Jan 18 16:20 P0S30200.bin -rwxrwxrwx1 root root 392534 Jan 18 16:20 P0S30203.bin -rwxrwxrwx1 root root 416942 Jan 18 16:20 P0S3-03-0-00.bin -rwxrwxrwx1 root root 463194 Jan 18 16:20 P0S3-04-2-00.bin -rwxrwxrwx1 root root 463194 Jan 18 16:20 P0S30420.bin -rwxrwxrwx1 root root 464690 Jan 18 16:20 P0S3-04-4-00.bin -rwxrwxrwx1 root root 476158 Jan 18 16:20 P0S3-05-3-00.bin -rwxrwxrwx1 root root 485106 Jan 18 16:20 P0S3-06-0-00.bin -rwxr-xr-x1 root root 2348 Jan 18 16:36 SEPDefault.cnf -rwxrwxrwx1 root root 3371 Jan 18 16:20 SIP000D6527BC59.cnf -rwxrwxrwx1 root root 2348 Jan 18 16:20 SIPDefault.cnf [EMAIL PROTECTED] tftpboot]$ cat OS79XX.TXT P0S3-06-0-00 [EMAIL PROTECTED] tftpboot]$ cat SEPDefault.cnf image_version: P0S3-06-0-00 proxy1_address: 192.168.1.200 proxy2_address: proxy3_address: proxy4_address: proxy5_address: proxy6_address: [snip] -- Max Clark max [at] clarksys.com http://www.clarksys.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
