Re: [asterisk-users] OT - Which Android handset with Wifi-only ?
On Sun, May 15, 2011 at 10:16 AM, sean darcy seandar...@gmail.com wrote: anyone actually used this on Android to connect to an asterisk server? Yes. I purchased it a while ago from the Marketplace, and had some issues with sound quality as my specific phone (Motorola Atrix) isn't officially supported yet. However, the support people at CounterPath have been extremely responsive, and the latest version works much better. I have not tested the G.729 codec. It's a good app, but I would buy it from CouterPath directly next time as their refund policy is longer than 15 minutes and they list the supported devices. Hopefully they will add video support soon. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA Asterisk
On Fri, Apr 29, 2011 at 7:29 PM, Kaushal Shriyan kaushalshri...@gmail.com wrote: I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf, but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA - High Availability.) The current production ready versions of Asterisk (1.4, 1.6, 1.8) do not have any native HA support. You have to engineer that on your own, or purchase a commercial product that handles it for you. How this is engineered would be based on your specific requirements. Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? AsteriskNOW is a simple to install complete Asterisk setup, just add hardware. While that is great, it would probably be more of a pain to make AsteriskNOW into an HA install than build one yourself based on your specific requirements. I haven't personally tried though, so YMMV. It appears that AsteriskNOW 1.7.1 64-bit contains Asterisk version 1.4.35 and 1.6.2.11. Both versions are now at Security Update Only status (but that's a conversation for another thread) We have been evaluating Asterisk for our Voice Application and it seems it would fit the requirement. Is Asterisk a CPU Intensive or a Memory Intensive application. In my specific experience, I would say Asterisk is neither CPU or Memory intensive. Memory has never been an issue, and we are not transcoding between different codecs. If you plan to do a lot of transcoding in software, then your CPU usage will increase. You would have to test using your specific requirements to know how it will impact your systems. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA Asterisk
On Mon, May 2, 2011 at 1:10 AM, A E [Gmail] all.efor...@gmail.com wrote: Now, I wonder what're the alternatives that people have been using for Asterisk HA other than commercially available solutions like HAAST and Astribanks assuming that kaushal is right and SCF isn't production ready yet. Anyone wants to chime in here with a solution built with readily available linux software like heartbeat , linux-ha, shared filesystems, filesystem replication and of course asterisk realtime? My requirement might be more along the lines of having several asterisk servers in a farm/pool without actually caring about the failover, so it might not even matter for me to worry about all of this, but I'm still curious as to what people are doing out there. For our specific needs we have build an active/passive Asterisk cluster based on CentOS 5 and cman/drbd/gfs2. Two nodes replicate data (configs, voicemail, provisioning data) on a Master/Master DRBD volume, using GFS2 as the shared file system. We use Asterisk Realtime via ODBC (MySQL Backend) for SIP/Extensions/CDR. All services bind to a floating IP Address. CMAN controls what server is running the services at any time, and handles migrating of the IP as well. Lights Out cards (via IPMI) are used for fencing. For access to the PSTN, I prefer to use an external device. We run a mix of Cisco 2800's and AudioCodes Mediant 1000's. I prefer to use PSTN to SIP gateways over cards built-in to the servers, or Astribanks as I feel they are more flexible. You could allow direct media, or allow multiple servers to communicate with the gateways at that same time. So that is the setup that we have chosen, and it might not be right for anyone else. The best advice I can give is to implement something at your comfort level, and test test test! I am aware of the potential issues with our setup, and am prepared to deal with them because of extensive testing. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA Asterisk
On Sun, May 1, 2011 at 3:03 AM, Terry Brummell te...@brummell.net wrote: 8 PRI’s? I’d be using something like an AudioCodes Mediant 1000. No messing around with switches and cables an crap. I agree, use a SIP Gateway. The AudioCodes Mediant 1000 supports up to 4 T1/E1/J1, so use two of them. That also keeps you going in case one of the gateways dies. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime mysql for 1.8
On 11-04-06 03:53 PM, Hans Witvliet wrote: I'm going to have a go with realtime mysql. Just wondering, most examples i came across while googling, was with 1.6 systems. So any drastic changes with 1.8.3, table-layout? other pitfalls? The tables migrate just fine, but you can update them to the newer terms (like directmedia instead of canreinvite). Not all of the configuration options are found in the contributed SQL table definitions, and columns that aren't recognized are ignored (nice for a comment, etc). As far as pitfalls, there have been a few deadlock issues for SIP with Realtime. Most that I have run into have been resolved, but check out issue 18818 if you use any local channels (there is a patch, but the issue hasn't been assigned yet). I don't think realtime has as wide as an audience, so the issues are a little slower to get resolved in SVN / releases. Those that use it are pretty active in helping confirm issues and testing patches though. On Wed, Apr 6, 2011 at 1:32 PM, Paul Belanger pabelan...@digium.com wrote: I suggest using res_odbc, it has better support. Aside from that, I've been testing it with 1.8 for a while; under light load (less then 5 channels). I have not had a problem. +1 for res_odbc -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime mysql for 1.8
On Wed, Apr 6, 2011 at 2:59 PM, Hans Witvliet h...@a-domani.nl wrote: [snip] I think i have to stick with mysql, as info is coming from other applications, but perhaps some of the other code can be tweaked. mysql is nice (lots of tiny programs writen for it), but i'm not religious attached to it ;-) I should have been more clear. res_odbc has better support in Asterisk. We use MySQL as our back-end database. One benefit of using ODBC connectors is that you don't have to be religiously attached to the back-end. The SQL code in Asterisk is generic, and you could swap out for PostgreSQL or anything else that there is a ODBC connector for. The amount of concurent calls will be small, but the amount of pre-defined users is fair (75K) So perhaps i should consider a mix of ldap and a DB. While we don't have that many users defined on any one system, I would test both independently. Both together sounds complex to troubleshoot when something breaks... -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why shouldn't I use 1.8?
On Fri, Mar 25, 2011 at 7:36 AM, Douglas Mortensen d...@impalanetworks.com wrote: But I would like specific reasons why I shouldn't use 1.8 in a production environment if anyone has some? That is a loaded question, in that no two environments are likely to be the same. Some bugs are major issues for 1% of the install base and take time to get merged into the code base. You should read through the open issues for the 1.8 branch and see if there are any show stoppers for your environment. If not, try it in the lab and validate that it works for you. Check out https://issues.asterisk.org For my environment specifically, this issue is currently preventing me from migrating from 1.6.2: - 18818 [patch] Crashing when using local channels and realtime on asterisk There are a lot of benefits to the 1.8 branch (Long term support, Called party id, Multicast RTP, etc) but only you can say if it will work with your configuration in your environment. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 asterisk 1.8.22 ringlist.dat error
On Mon, Feb 14, 2011 at 10:31 AM, James Miller paramedi...@gmail.com wrote: I did that and this is what I got when I tried to play the 24 ringtone: 13:29:16.573318 IP 192.168.1.103.50849 192.168.1.60.69: 39 RRQ Emergency ring_emergency.pcm octet That line should read something like: blah.. RRQ ring_emergency.pcm octet According to the line you send, the phone is requesting the file: Emergency ring_emergency.pcm In the ringlist.dat file in the first column I typed the display name then hit the tab key. Now on some it only moved a couple of spaces over, on others, it tabbed way over. Not sure whats going on there with that. Not sure what editor you are using, but are you certain that it is inserting Tabs, and not spaces when you hit the tab key? If you want, you can send me the file off-list and I'll take a look at it. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime and Local Channel Crash Problem 1.8.3-rc2
On Tue, Feb 15, 2011 at 10:15 AM, Nic Colledge n...@njcolledge.net wrote: I have been having a problem with asterisk crashing when using local channels and realtime on asterisk 1.8.3-rc2. Nic, I can reproduce this using the latest SVN for the 1.8 branch. I don't get the console locking, but SIP definitely deadlocks every time. If you want to open a ticket, I'll upload the bt/threads/locks info that I have. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 asterisk 1.8.22 ringlist.dat error
On Mon, Feb 14, 2011 at 5:40 AM, James Miller paramedi...@gmail.com wrote: Good Day everyone, Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by Cisco, however now the phone does not and will not read the RINGLIST.dat file. I’ve tried rebooting the phone, tried resetting the phone back to factory, have deleted the RINGLIST.dat file and reloaded the phone then reinstalled the RINGLIST.dat, and still the bloody phone will not read the file. I have not been able to locate anything in google about this kind of issue and am at a loss as to what in the world is the issue. Have you run a tcpdump on the tftp server to make sure it is requesting the correct file? It might be asking for RingList.dat, ringlist.dat, RINGLIST.DAT, etc. as capitalization seems to not be one of Cisco's concerns. (FYI, mine was RINGLIST.DAT, but I have no more 79x0's around to test with) Try running this as root on the tftp server and look for a request for the file: # tcpdump -nn port 69 -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 asterisk 1.8.22 ringlist.dat error
On Mon, Feb 14, 2011 at 9:04 AM, James Miller paramedi...@gmail.com wrote: I did the command listed, and its actually requesting RINGLIST.DAT, so I changed the filename to match its request but now its showing in the ring type setting: Chirp 1 Chirp 2 24 24-ring-tone-1.raw Att1 ring_att1.pcm . You should only see the description of the file on the display. Make sure that the description and filename are tab-separated, since spaces are allowed in the description. However, when you attempt to play one it says Loading Ringer File but it doesn’t do anything. So now it’s at least seeing the file, now it just won’t play them. You can run the same command ( tcpdump -nn port 69 ) to view what file the phone is attempting to download from the tftp server. My guess is that it isn't pulling anything down or something like 24 24-ring-tone-1.raw if the file is not tab separated. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.3
On Fri, Feb 11, 2011 at 7:59 AM, satish patel satish...@hotmail.com wrote: I thought it has been resolved in 1.8.2 version Issue 18403 was not resolved in 1.8.2, but in 1.8.3-rc1. Release 1.8.3-rc2 was cut on 1/20/2011, so hopefully the full release will be out soon. You can see where the issue was merged here: http://svn.asterisk.org/svn/asterisk/tags/1.8.3-rc1/ChangeLog -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail-over server
On Wed, Feb 9, 2011 at 6:55 AM, Vieri rentor...@yahoo.com wrote: [snip] Since all of the SIP devices in my LAN have static IP addresses, I can keep track of everyone on my own. For instance, could I do fake SIP registrations from localhost (the * server) and specify a LAN IP address? Have you looked at the 'defaultip' sip configuration option? Or setting host=IP for those devices? I would write a custom script that would execute whenever an Asterisk server takes over. As said earlier, this server would not have any SIP extensions registered at first and they would be registering slowly within 60 seconds or more. However, since I KNOW FOR SURE that some SIP devices are always online and have static IP addresses, can't I fool Asterisk by somehow registering via locahost but spoofing the source IP address? Maybe setting the source port to what it was exactly can be tougher but I *could* try to keep track of it. That sounds more complicated and likely to break than using Realtime. This way, whenever the Asterisk server that took over tries to bridge a call, it will try to connect to the fakely-registered IP address. I'm not using realtime for 2 reasons: 1- I'm using the FreePBX framework and there's no realtime backend unfortunately. Moving to Realtime and losing all the FreePBX goodies is time-consuming. Does anyone know how to use FreePBX + Realtime? This is unfortunate for most of the Asterisk GUI's available. 2- I don't have enough hardware resources to setup a server for the realtime DB that both Asterisk servers would connect to. Also, I wouldn't feel comfortable having just one DB server. For easier maintenance I would use a clustered database for realtime. However, I'm using Mysql 5.0 ndbcluster tables for other non-voip purposes and my experience hasn't been so great. I once had a power outage and all ndb table data was lost. Also, 5.0 ndb crashes in several occasions. As far as I can tell, it isn't reliable. I haven't tried 5.1 though. I have no experience with clustered postgresql. So run the DB on the same server as Asterisk, if your call volume allows it, and either replicate the data using the built-in DB replication or use DRBD between the two existing servers. We use DRBD between two Asterisk nodes on smaller installations for configurations and voicemail. It works very well for us. For MySQL Cluster to work well, the application has to be designed for it, and it is a RAM based storage. But that is a conversation for another list. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail-over server
On Tue, Feb 8, 2011 at 8:07 AM, Vieri rentor...@yahoo.com wrote: Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. This is a typical setup for two node HA. Just be careful when clustering only two servers. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to server2 and that server2 knows that they are actually on-line so calls can be routed to them. It depends on your configuration. If you use Asterisk Realtime to store SIP registrations, then the database will contain information on how to contact the device (fullcontact, ipaddr, and port fields). Then on a failover, Asterisk will do a lookup for the peer in the database, find the needed information and dial the device. Of course any registrations that happen before being written right before the server fails may not work. Also make sure to use the latest version of Asterisk as there was a bug where fullcontact wasn't saved correctly. How can I minimize this time lapse? Can Asterisk notify all SIP clients in its sip.conf that they need to acknowledge being on-line or not (thus forcing re-registration in my scenario)? In the above scenario, I can kill Asterisk, start it again, and place a call from two devices that have not registered again. So, the best timeout is your dead time detection and failover startup time. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP 420
On Mon, Dec 20, 2010 at 9:46 AM, Dovey Forman dovey.for...@idt.net wrote: I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it’s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? Yes. The softphone is requiring x-call-detail, which Asterisk does not support. The softphone either needs to drop that requirement completely, or change it to a Supported header so it can be processed by other SIP servers. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Specifying DID for outbound calls
On Sun, Dec 19, 2010 at 2:57 PM, Stephen Reese rsre...@gmail.com wrote:
I believe I have made a little headway. I have two outgoing DID
contexts and have changed the GotoIf statement to the extension name.
User One acts as expected and User two now displays unknown when
calling so I believe it is trying to to goto 20 but it's not quite
making it. Any tips? Thanks
[outgoing]
exten = _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten = _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten = _1NXXNXX,n,GotoIf($[${Outgoing} = 201]?20:10)
exten = _1NXXNXX,10,Set(CALLERID(all)=User One 3012323434)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten = _1NXXNXX,n,Goto(h,1)
exten = _1NXXNXX,20,Set(CALLERID(num)=User Two 3013232322)
This should either be CALLERID(all) or just set the number on the line
above. As a side note, I prefer to use labels an not line numbers.
Less to change later...
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
exten = _1NXXNXX,n,Goto(h,1)
I'll also give a +1 to using setvar. It allows you to abstract the
dial plan much more. I use this feature a lot in both static and
Realtime configurations. For example (not tested, but based on live
production code):
sip.conf:
[101]
...
setvar=EXTERNAL_CALLERID=User One 3012323434
[201]
...
setvar=EXTERNAL_CALLERID=User Two 3013232322
extensions.conf:
[outgoing]
exten = _1NXXNXX,1,Verbose(1, Someone is making a call out)
exten =
_1NXXNXX,n,ExecIf($[${EXISTS(${EXTERNAL_CALLERID})}]?Set(CALLERID(all)=${EXTERNAL_CALLERID}))
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
But then I am sure there are 100 other ways to do this same thing.
-Jonathan
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Re: [asterisk-users] Specifying DID for outbound calls
On Sat, Dec 18, 2010 at 4:03 PM, Stephen Reese rsre...@gmail.com wrote: The host I am working with has two accounts from the same DID provider. Incoming calls work correctly and dial the appropriate extensions. This also allows incoming calls to be billed appropriately to the individual DID accounts. Outgoing calls from either extension default to the first DID, i.e. calls from either extension have the same callerID. How can an extension specify separate outgoing contexts so the correct number is associated with it, also allowing the SIP provider to recognize the difference for billing purposes, or is there a better way? The outgoing caller-id is probably just the extension number, so the provider is setting it to a default (usually the main billing number). You can set what Asterisk sends as the outbound Caller-ID in the outbound context before the Dial statement. Make sure your provider will honor what you set, as many filter what you can send to only the DIDs they provide for you. Take a look here for more information on setting the caller-id in the dialplan: http://www.voip-info.org/wiki/view/Asterisk+func+callerid -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with 1.8 and BlindTransfer
On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com wrote: I am having issues with Blind Transfer on asterisk 1.8 What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS? Verison 1.8.0, Suse 11.1 Try the latest SVN branch for 1.8 and see if that resolves your issue: $ svn checkout http://svn.asterisk.org/svn/asterisk/branches/1.8 (that will create a 1.8 directory in your current working directory) On Thu, Dec 2, 2010 at 8:44 AM, Karsten Wemheuer k...@gmx.de wrote: There was an issue with blind transfer in 1.8.0, fixed in SVN (and maybe in 1.8.1 ?) See https://issues.asterisk.org/view.php?id=18185 According to the ChangeLog, the fix for issue 18185 was committed after 1.8.1-rc1 was released. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push central phone book to phones
On Thu, Dec 2, 2010 at 6:56 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Thu, 2 Dec 2010, Jonas Kellens wrote: I have Snom, Cisco, Grandstream YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Grandstreams support an XML format phone book download - it would susprice me if the others didn't, but I've no 1st hand experience of them. Cisco (at least the 79x1 series) phones also have a special XML format for the directory. I have implemented it before as an interactive web app the phones query. No information is stored on the phone itself. Well, that's what I do anyway. It's better than mucking about downloading phone books to all the different types of phones. Real-time query (Live XML/LDAP) back-ended on a database are really the best way to go for Corporate style directory. Unfortunately, you have to get a license from Polycom for LDAP, and static XML files get out of sync way to fast... -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files corrupted
On Thu, Dec 2, 2010 at 4:56 PM, Mike l...@net-wall.com wrote: Hi, I know I am using SVN, but I was wondering if anybody ever came across this error. There is nothing wrong with using SVN. Well, there isn’t a msg.txt file, I can see that. There is a msg0003.txt and msg0005.txt (along with the appropriate wav files). Looking into the directory, all files seem there. Except the sequence doesn’t start at . 1) How do I fix this? I don’t mind manually fixing it when it happens, but what’s wrong exactly? I have seen this once on a 1.6.2 system a while back. I just renamed the TXT and audio files to be sequencial numbers starting at and everything worked again. Asterisk assumes the voicemail message files are named that way, and it errors out if that is not the case. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with 1.8 and BlindTransfer
On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com wrote: I am having issues with Blind Transfer on asterisk 1.8 What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS? If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. Does the remote party (being transferred) initially hear hold music, then the line go silent after completing the transfer? Does the Grandstream show the call still on hold, but you are unable to pick it up? Are you using Realtime and/or Direct media? It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use the # to initiate the transfer everything works. But our customers are use to using the transfer key on the phone. I found several bugs out there on the bug tracker but do not see if there is any work around. Any ideas or help would be appreciated. I have been chasing a deadlock (issue #18403) on blind transfers with 1.8SVN and have not found a work-around yet. While I can deadlock every time (Polycom and Cisco handsets), at least one other has reported different results with the Bria Softphone and Grandstream handsets. You could try a softphone and see if you get the same results as the physical phones. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Preserve CallerID on transfers
On Sat, Nov 27, 2010 at 11:40 AM, Fabiano Carlos Heringer b...@grupoheringer.com.br wrote: Hi, it´s possible to mantain the original CallerId when making transfers? (atx or blind) Example: A calls to B, A transfer to C, C see the CallerID of B, and not A... It´s possible? Asterisk 1.8 added Connected Party Identification Support. Try 1.8 in a test environment and see if it meets your needs. For more info, see: http://lists.digium.com/pipermail/asterisk-announce/2010-October/000277.html -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and cisco 7970 - multiple lines
On Mon, Nov 22, 2010 at 11:24 AM, Peter Kowalski kowalla...@gmail.com wrote: Below is my xml button 1 and button 2 portion. Any help will be appreciated. line button=1 name130/name authName130/authName authPasswordpass/authPassword contact7b452e87-4496-4762-e11f-b26751a1884b/contact /line line button=2 name160/name authName160/authName authPasswordpass/authPassword contact7b452e87-4496-4762-e11f-b26751a1884b/contact /line I don't use 7970s, but on the 7941/61s I set the name, authName, and contact all to the SIP username. The first thing that I see is that the Contact is set to the same thing on both lines, which might cause your problem. Try changing the contact to the SIP account name for each line. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending calls to a particular T1 port.
On Fri, Nov 12, 2010 at 10:17 AM, Ernie Dunbar maill...@lightspeed.ca wrote: that goes from port 4 on the live server to port 1 on the backup server. In /etc/asterisk/chan_dahdi.conf: group=4 context=local switchtype = national signalling = pri_cpe channel = 73-95 context = default group = 63 What is the configuration on the backup server? One side needs to act as the network side with signalling=pri_net In /etc/asterisk/extensions.conf: exten = _*88,1,Dial(DAHDI/g4/123456789) However, in the Asterisk console, I get this error on the live server: -- Executing [...@lightspeedout:1] Dial(SCCP/lightspeed7-0062, DAHDI/g4/123456789) in new stack [Nov 12 09:24:41] WARNING[1970]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SCCP/lightspeed7-0062' status is 'CHANUNAVAIL' What is the output of 'pri show spans' or 'dahdi show channels'? Specifically does Asterisk recognize the channels as up/active an In Service. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?
On Mon, Sep 20, 2010 at 8:58 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Mon, Sep 20, 2010 at 11:48 AM, Olivier CALVANO o.calv...@gmail.com wrote: Anyone have a AudioCodes with Asterisk ??? I use many AudioCode devices with Asterisk. Mostly Mediant 1000s and MP-114s, no Mediant 2000s. I would suggest you contact AudioCodes or your reseller, as AudioCodes has configuration guides that may help you. Here is a quick summary if I remember correctly: - Create a peer in Asterisk for the gateway - Configure the E1 on the Mediant (Provider specific) - Configure the SIP proxy (Asterisk) on the Mediant - Create a Trunk Group on the Mediant for the E1 - Configure IP to Trunk Group Routing to send calls out the Trunk Group If you have problems beyond that, contact whoever sold you the device. For the price they better offer some basic configuration support! You can also purchase support directly from AudioCodes. Yes, but why? Both do the same thing. It would be like me asking 'I have a bike and need to get to work. Can I use the bike with a car?' I would have to disagree with that statement. It is quite common to separate termination, call routing, and media for larger installations or to add some HA. Since termination is only part of the system, a better analogy might be different type of tires on the car. Sure you don't need snow tires, but you might want them when things get slick out! -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Include and Realtime
On Tue, Aug 24, 2010 at 5:48 AM, Dan Journo d...@keshercommunications.com wrote: Hi, I think I already know the answer to this question, but is there any way to do the following using realtime? Or do I have to create a full dialplan for each client without using includes? One way that I know works, as I used it on 1.6.2 is to create the contexts like you have listed below with all of the includes, then create a dummy context for each one using Realtime. For example, expanding on your existing Client1_phones context you could add: [client1_phones] include = client1_internal include = client1_outgoing_calls include = test_calls include = parkedcalls [client1_internal] switch = Realtime [client1_outgoing_calls] switch = Realtime You would have to create the base contexts for each client. I put each client/site/logical group in a different file and #include that to keep the extensions.conf file short, and easy to remove a specific section without impacting others. Now this I have not tried, and have no idea if it would work. Maybe someone more familiar with the code can comment. You can specify the context in the switch statement, but can you have multiple switch statements under a context? It would be worth at least trying in a test environment. So to change client2_phones with an untested idea: [client2_phones] include = client2_internal include = client2_outgoing_calls include = test_calls include = parkedcalls [client2_phones] switch = Realtime/client2_internal switch = Realtime/client2_outgoing_calls include = test_calls include = parkedcalls If that doesn't work, maybe just having one switch and an include? Again, I haven't tested any of that, but it seems like an interesting way to do what I think you want. Good luck! -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for MIB description
On Tue, Aug 24, 2010 at 7:03 PM, Bruce Ferrell bferr...@baywinds.org wrote: Hi, I've gone through the source tree and I don't see a MIB description file for the SNMP agent in asterisk. can someone point me to it. There is an asterisk-mib.txt and a diguim-mib.txt in the doc directory, and here are some links to the SVN: http://svnview.digium.com/svn/asterisk/trunk/doc/asterisk-mib.txt?revision=124392view=markup http://svnview.digium.com/svn/asterisk/trunk/doc/digium-mib.txt?revision=124392view=markup -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk realtime SIP configuration
On Wed, Jul 21, 2010 at 3:09 AM, Murali Vasu vimurli@gmail.com wrote:
Hi All,
I am trying to configure asterisk realtime. But i am unable to get the
extensions listed successfully when i type sip show peers in the asterisk
CLI . i am unable to see any failure logs when i do a reload
If you want to see the peers on the CLI, then you have to enable
caching of the peers. Add this to your sip.conf file:
[general]
rtcachefriends=yes
-Jonathan
i can able to connect to the data source through odbc show in the
CLI, Any hep in this regard is highly appreciated. Following is the
configuration and specification.
Server Specification:
1) asterisk-1.6.2.6
2) CentOS- 5.2 (64-bit)
3) Postgresql- 8.1
Configuration:
odbc.ini
[PostgreSQL]
Description = Test to Postgres
Driver = PostgreSQL
Trace = Yes
TraceFile = /tmp/sql.log
Database = bedrock
Servername = localhost
UserName =
Password =
Port = 5432
Protocol = 6.4
ReadOnly = No
RowVersioning = No
ShowSystemTables = No
ShowOidColumn = No
FakeOidIndex = No
ConnSettings =
odbcinst.ini
[PostgreSQL]
Description = ODBC for PostgreSQL
Driver = /usr/lib64/libodbcpsql.so
Setup = /usr/lib64/libodbcpsqlS.so
FileUsage = 1
res_odbc.conf
[postgres]
enabled = yes
dsn = PostgreSQL
username =postgres
password =postgres
pre-connect = yes
Database table in postgres sip :
Column | Type | Modifiers
++--
id | integer | not null default
nextval('sip_id_seq'::regclass)
name | character varying(80) | not null
accountcode | character varying(20) |
amaflags | character varying(7) |
callgroup | character varying(10) |
callerid | character varying(80) |
directmedia | character varying(3) | default 'yes'::character varying
context | character varying(80) | default 'default'::character
varying
defaultip | character varying(15) |
dtmfmode | character varying(7) |
fromuser | character varying(80) |
fromdomain | character varying(80) |
host | character varying(31) | not null default
'dynamic'::character varying
insecure | character varying(4) |
language | character varying(2) |
mailbox | character varying(50) |
md5secret | character varying(80) |
nat | character varying(5) | not null default 'no'::character
varying
permit | character varying(95) |
deny | character varying(95) |
mask | character varying(95) |
pickupgroup | character varying(10) |
port | character varying(5) |
qualify | character varying(3) |
restrictcid | character varying(1) |
rtptimeout | character varying(3) |
rtpholdtimeout | character varying(3) |
secret | character varying(80) |
type | character varying | not null default
'friend'::character varying
username | character varying(80) |
disallow | character varying(100) | default 'all'::character varying
allow | character varying(100) | default 'alaw,ulaw'::character
varying
musiconhold | character varying(100) |
regseconds | integer | not null default 0
ipaddr | character varying(15) |
regexten | character varying(80) |
cancallforward | character varying(3) | default 'yes'::character varying
lastms | character varying(80) |
useragent | character varying(100) |
defaultuser | character varying(100) |
fullcontact | character varying(100) |
regserver | character varying(100) |
Indexes:
sip_conf_pkey PRIMARY KEY, btree (id)
name UNIQUE, btree (name)
extconfig.conf
sipusers = odbc,postgres,sip
sippeers = odbc,postgres,sip
Thanks Regards
Murali Vasu
--
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--
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To
Re: [asterisk-users] not sure what to change to point the timing to the att circuits?
On Thu, Jul 8, 2010 at 8:30 AM, Jared Terrell jared.terr...@mcc.edu wrote: # Span 1 span=1,1,0,esf,b8zs bchan=1-23 dchan=24 echocanceller=mg2,1-23 # Span 2 span=2,2,0,esf,b8zs bchan=25-47 dchan=48 echocanceller=mg2,25-47 # Span 3 span=3,3,0,esf,b8zs bchan=49-71 dchan=72 echocanceller=mg2,49-71 # Span 4 span=4,4,0,esf,b8zs bchan=73-95 dchan=96 echocanceller=mg2,73-95 # Global loadzone = us defaultzone = us You have it configured correctly. Here is an quote from http://www.voip-info.org/wiki/view/Asterisk+PRI # span=span num,timing source,line build out (LBO),framing,coding[,yellow] The timing sources are 1-4 (in your example), which is the priority of the remote clock. A value of zero means that the Asterisk server is the master (i.e. to a channel bank you are acting as the CO). Only one timing source is used for all spans. Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) B8ZS/ESF ClockSource This is your primary timing source (the far end is used as a reference clock for ALL spans) Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 B8ZS/ESF Timing slips: 38526 This is your secondary timing source. Looks like maybe something wrong with the T1? Try replacing the cable, that has bit me before. If that doesn't work call ATT and have them test that circuit. Span 3: TE2/1/1 T2XXP (PCI) Card 1 Span 1 B8ZS/ESF Third timing choice, looks clean. Span 4: TE2/1/2 T2XXP (PCI) Card 1 Span 2 B8ZS/ESF Forth timing choice, looks clean. Hope that helps. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom firmware: split vs. combined
On Mon, Jun 21, 2010 at 10:19 AM, Warren Selby wcse...@selbytech.com wrote: On Mon, Jun 21, 2010 at 12:10 PM, Ken D'Ambrosio k...@jots.org wrote: Howdy, all. What's the difference between split and combined firmware, which can be seen at the above link? I've googled to no avail, I'm afraid. The release notes talk about which one to choose and why in Section 1.4 Distribution Files http://downloads.polycom.com/voice/voip/relnotes/spip_ssip_vvx_release_notes_v3_2_3.pdf The split contains all the firmwares for the different model phones as separate files, the combined combines all of the firmwares into one big firmware file. The combined will cover any supported polycom phone model, but it takes longer to load. You also need to use the combined if you have a BootRom release older than 4.0.0. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best way to limit outgoing calls per trunk
On Sun, May 30, 2010 at 9:37 AM, bruce bruce bruceb...@gmail.com wrote: Thanks for the tip. I have been checking those two options. Would you be able to provide an example of how GROUP or GROUP_COUNT may check for a trunk usuage? Here is how I do it. It is based on Asterisk 1.6.1.x, and I created a generic sub-routine to call for limiting trunks to a specific number of calls. The code is documented, so it should give you a good idea of how to use it. http://thurmantech.com/node/7 -Jonathan From what I see is that you have to assing certain routes a group and then count the group, but how I do include a trunk in the group? Thanks On Sat, May 29, 2010 at 7:07 PM, Steve Edwards asterisk@sedwards.com wrote: On Sat, 29 May 2010, bruce bruce wrote: I am looking to use System() function along with some bash scripting to determine if a Trunk is being used during certain time of the day or not. Here is what I have in mind. Please guide me if you know a better way: Using the GROUP/GROUP_COUNT functions in the dialplan is a better way. Using system() will mean creating a bunch of processes (each sed/awk/cut/etc command). -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best way to limit outgoing calls per trunk
On Sat, May 29, 2010 at 2:02 PM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, I am looking to use System() function along with some bash scripting to determine if a Trunk is being used during certain time of the day or not. Here is what I have in mind. Please guide me if you know a better way: I don't know what version you are running, but check out GotoIfTime. I use it frequently for office hours. GROUP, and GROUP_COUNT can help with limiting on a trunk too exten = s,1,answer exten = s,n,System(/tmp/check.sh) exten = _X.,n,GotoIfTime(7:30-16:30,mon-fri,*,*?multicall) ; Within this time, go to the label 'multicall' code to limit to one call exten = _X.,n(multicall),Verbose(3,We can call more than once) code to call multiple times) http://www.voip-info.org/wiki/view/Asterisk+cmd+GotoIfTime http://www.voip-info.org/wiki/view/Asterisk+func+group check.sh: check EPOCH time = do an IF for certain times = Allow mutiple calls in certain times and only single call at certain times return back to Asterisk context and report if Trunk would allow more channels or not... Something along those lines. Should this be a solid thing to do? I am looking to use GotoIF and `asterisk -rx sip show channels` to grab results or `asterisk -rx core show channels` Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Cluster
On Tue, May 18, 2010 at 9:38 PM, Adolphe Cher-Aime achera...@gmail.com wrote: Hello Everyone, I must deploy an asterisk system that can support at least 500 pstn outbound calls. It's a challenge as it's the first time i'm gonna build such a large system. I want to have your advice on hardware, software and so on . What i have in my plan is a cluster of servers with quad PRI cards. I will appreciate your advice. Your up front cost is going to be a little higher with TDM - SIP devices, but your management will be a lot easier. AudioCodes also has equipment that can support a DS3 connection, or multiple T1s directly to SIP. For example, If you are getting 22 T1s then get two AudioCodes Mediant 2000s with 16 spans (maxed out) or two Mediant 3000s with 21 spans. If you are getting a DS3, get a Mediant 3000. The M3000 supports up to 84 T1s or three DS3 or one OC3. Then leave call management up to Asterisk. Of course, have redundancy everywhere you can. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Cluster
On Wed, May 19, 2010 at 6:13 PM, Adolphe Cher-aime achera...@gmail.com wrote: Jonathan for redundancy which software do you recomand? Without knowing exactly what you are trying to do beside having at least 500 outbound calls, that would be impossible to say. I mostly use a home grown HA Linux configuration (CentOS, cman, MySQL, GFS2) with Asterisk Realtime. I would use what you know, as long as it scales to what you need. If it doesn't, then I would get someone to help that has a solution. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GXW4024
On Fri, Apr 30, 2010 at 5:26 AM, Peter peterp...@aboutsupport.com wrote: I consider buying three GrandStream GXW4024 and connect 72 analogue phones to asterisk I recommend against that product. I have two that now sit on a shelf due to bad call quality, echo issues, and random one way audio... Do you have any feedback how well it works with Asterisk ? I am on a budget, do you have other recommendation for similar setup that get into same budget - connect around 70 analogue phones to asterisk. They are easy to setup and connect to Asterisk. That is about the only thing that they do well. I purchased two of these to try and fit within my budget, and ended up replacing them after about a month. The call quality was sub-par, and I had all kinds of echo issues. Firmware updates didn't seem to make anything better. I ended up replacing them with AudioCodes MP-124 which have been rock solid. Of course they cost about twice as much, but you get what you pay for. In the long run I went way over budget, but learned a good lesson! -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway E1 = Asterisk ?
On Wed, Apr 28, 2010 at 7:58 AM, Tim Nelson tnel...@rockbochs.com wrote: - Olivier CALVANO o.calv...@gmail.com wrote: Hi i want change my asterisk server. Actually, Asterisk work's on a IBM Server with a internal digium E1 card. For High availability, i don't want now use internal E1 card. In my new asterisk systems, i have two server and two E1 not in the same site. I am search a hardware gateway, if possible in 1U Rack with 2/4 or 8 E1 capacity with echo cancellation. I want that this gateway connect in trunk sip to my asterisk. Anyone have idea of good products for this ? Redfone ? but no SIP i thnk's, only in MAC/Ethernet Patton ? Not in rack other ? Audiocodes Mediant gateways are top notch. Their Mediant-1000 supports up to 4 interfaces (T1/E1/J1). +1 for AudioCodes Median 1000. The AudioCodes Median 2000 supports up to 16 T1/E1s if you need more than 4. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High Availability - Shared Database - Ideas?
On Wed, Apr 21, 2010 at 1:09 PM, Robert Grignon rgrig...@fleetone.com wrote: I am investigating High Availability solutions for my front end servers. Always good to hear. I got into a discussion regarding replicated local databases versus a single fiber connected shared database on an EMC. I will guess that you mean MySQL Master/Slave replication. Is anyone running a shared database on a SAN? Care to comment on your findings... I am running MySQL on shared SAN LUN, but not for Asterisk. Since SANs are expensive, I have been using DRBD/GFS2/MySQL for most of my low budget HA Asterisk installations. Some things to think about: 1. If you are using MySQL, then only one server can have the database open at a time. You will have some lag/downtime when the active server fails and the secondary has to take over. You are going to have this anyway even with a Master/Master replication as the IP has to shift. Same with Master/Slave plus you add time for a script to promote the Slave. 2. Don't even think about using MyISAM... InnoDB *only*. MyISAM doesn't check improperly closed tables until they are accessed which can cause some major lag. Not to mention no transaction support. You won't have another copy if things get corrupted (besides all of your backups of course) 3. While nice SANs are redundant, you are still adding another dependency to the system (a few if you are using FC switches). Make sure everything has multiple paths, and don't forget to configure fencing for the nodes. 4. If you have PRI/Analog lines to the server, then it becomes more of a headache. Use dependable redundant SIP gateways, or have some action plan in place. 5. Test, test, test then test some more. Break it in the lab and know how to fix it. Setup is easy, repair can be a pain. (You also want to know it will actually work =) That's my quick $0.02, and there is a lot more to think about too. Overall, if designed right I think it is a good option. Just depends on your level of comfort with the technologies, and the risk/benefit that goes along with it. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime changes not reflected realtime
On Sun, Apr 18, 2010 at 12:30 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Jonathan, 'sip show peers' works just fine... Sorry, I wasn't clear. It has been my experience in 1.6.1.x that 'sip show peers' does not work without rtcachefriends=yes for realtime implementations. asterisk*CLI sip show peers Name/username Host Dyn Nat ACL Port Status Realtime testcorp4 (Unspecified) D N 0 UNREACHABLE Cached RT testcorp3/testcorp3 192.168.1.100 D N 5061 OK (25 ms) Cached RT Only you see the 'Realtime'-column, and the 'Cached RT'. With rtcachefriends enabled it does show if the peer is Cached RT or a static peer in sip.conf. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime changes not reflected realtime
On Sat, Apr 17, 2010 at 4:42 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Do I need to 'sip prune realtime all' after every change ?? If you change a sip peer and you have caching enabled, then you need to prune that peer for the change to take effect. On 1.6.1 I issue the following: sip prune realtime peername sip show peername load That will only clear the caching for peername and not all of the peers. The load statement re-caches the peer immediately. I haven't tried this on 1.4, so I don't know if those options exist or not. Is rtcachefriends=yes a wrong setting ?? No, not if you want caching enabled. I enable sip realtime caching on all of my systems. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime changes not reflected realtime
On Sat, Apr 17, 2010 at 11:14 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Is rtcachefriends=yes a wrong setting ?? No, not if you want caching enabled. I enable sip realtime caching on all of my systems. What if I do not enable caching ? What would be the effect on my realtime configuration with sip_buddies in my mysql-DB ? At the bottom of the page it talks a little about caching: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip I know that sip show peers doesn't work, and I believe that qualify does not work without caching (but I haven't tested that). I enable caching because I don't change the names of sip_accounts that frequently, and why have Asterisk hit the database constantly if you aren't changing the information? Asterisk will then save all of the results in RAM, and only do a look-up for an unknown account. If you have a web interface for updating information you could always use AMI to issue the prune/reload after committing the changes. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iptables miss up phone calls if not used properly
On Tue, Apr 13, 2010 at 11:17 AM, khalid touati khalidtou...@gmail.com wrote: Hi Guys, i wanted to share this with u and ask for little help at the same time: i used iptables to secure my server, so i wnet ahead and blocked avery thing except a couple of domain protocols and UDP ports of SIP, IAX2 and that range 15000 to 2, tested it and OK. when in production, the calls were taking a huge time 7s to be established and somtimes after call setup people cannot hear ech other (but not all the time which weird), so iptables can miss up performance if not set correctly (even if it's working, stuff like this can happen). so if any body have some lines of iptables that secure server and don't cause performence trouble to phone calls please share with me (i am using Centos 5.3 asterisk 1.4.24). You don't need to open up all of the UDP ports like that if you enable connection tracking for sip. Of course you don't say how many ongoing sessions you are using, but I haven't had any issues with connection tracking for SIP. All of this is based on INBOUND connections to the server, but make sure you are allowing OUTBOUND connections too. Here are some changes for an example that is NOT complete and you can use AT YOUR OWN RISK. Make sure you have something like this in the following files. Notice that this does not restrict who can talk to your server either, and only covers IAX/SIP. This is based on CentOS 5.4. /etc/sysconfig/iptables: # Anything we already know about -A Fwall-IN -m state --state ESTABLISHED,RELATED -j ACCEPT # IAX -A Fwall-IN -m state --state NEW -m udp -p udp --dport 4569 -j ACCEPT # SIP -A Fwall-IN -m state --state NEW -m udp -p udp --dport 5060 -j ACCEPT -A Fwall-IN -m state --state NEW -m tcp -p tcp --dport 5060 -j ACCEPT /etc/sysconfig/iptables-config: IPTABLES_MODULES=ip_conntrack_sip If you need more specifics, you will have to post your iptables configuration for some more advise. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + DRBD Performance
On Sat, Apr 10, 2010 at 9:50 AM, James Lamanna jlama...@gmail.com wrote: Hi, Has anyone had any experience using DRBD to mirror an entire asterisk machine? Entire, no. Specific/Important mounts yes. If so, is there a performance issue at all when people are recording voicemails and the like? I haven't seen any performance issues, but most installs that I have done aren't recording a ton of messages at a time. I don't have any statistics, but if you had more information on the installation size it would be easier to say. I also have MySQL running on a DRBD mirror, and don't have any problems with updates. I also do all of the replication for DRBD on a cluster interface, which does not pass any VoIP traffic. It seems like that could generate quite a bit of traffic. Also, do you bother to mirror the log files as well? I don't mirror the log files. just: /etc/asterisk/ /var/lib/asterisk /var/spool/asterisk and tftp/http/mysql directories if used. If you really want to test it out, create the mirror and do some disk throughput testing. That way you can validate your specific hardware and network infrastructure to what you expect. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1 Voicemail users.conf
On Wed, Feb 17, 2010 at 8:50 AM, Dave Poirier dpoir...@mesd.k12.or.us wrote: Hello, We recently upgraded our Asterisk box from 1.4 to 1.6.1. In both versions of voicemail you can press 3 for advanced options, 5 to leave a message and enter an extension to leave a voicemail. This feature worked fine under 1.4. Now under 1.6.1 all the prompts are the same but when you enter the extension it reads back the extension (or says the recorded name if present) then goes straight back to the main menu with the following error on the console app_voicemail.c:5019 leave_voicemail: No entry in voicemail config file for '1562' I should note here that we use users.conf file. We don't have individual entries in voicemail.conf. Does anyone have a fix for this or suggestions to try? I don't use users.conf, but here are some questions / things to try: What exact version of 1.6.1 are you using? Are you using the 'default' voicemail context? If not, do you have 'searchcontexts' enabled in voicemail.conf? Does it work if you add a dummy mailbox to voicemail.conf in the 'default' context? Can you use the directory to forward a message to another user using the directory? -Jonathan -- _ -- 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
Re: [asterisk-users] Popular Gigabit Phones
On Thu, Jan 21, 2010 at 4:56 PM, Matt Darnell mattdarn...@gmail.com wrote: Most manufacturers charge in excess of $80 to upgrade from a 10/100 switch to a 10/100/1000 switch built into the phone. The cost might have been in the chipset 5 years ago but I can get a 5 port gigabit switch for $30. What are most folks using for people that need gigabit to the desktop and don't want to run another cable? For our engineering staff we use Polycom SoundPoint IP 560's. Cubes with two drops for heavy users who have to be dual homed were build without VoIP in mind (or an tech department at all for that matter)... I haven't run iperf through them, so I don't have any performance statistics. No one has complained except for our fiscal department, the phones do come at a premium above the standard phones =). -Jonathan -- _ -- 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
Re: [asterisk-users] How to enable a range of IP addresses in realtime sip_buddies
You need to set: host=dynamic Otherwise only .112 is allowed. -Jonathan On Tue, Jan 19, 2010 at 1:17 PM, Bruce Ferrell bferr...@baywinds.org wrote: I'm using realtime sip peers and I need to enable a range of IP addresses for a peer. I have: deny = 0.0.0.0/0.0.0.0 permit = xxx.yyy.zzz.0/255.255.255.0 mask = 255.255.255.0 defaultIP = xxx.yyy.zzz.112 host = xxx.yyy.zzz.112 Addresses other than .112 are being denied. Can someone offer assistance? Am I doing something wrong? Bruce -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Grandstream GXW-4024
On Sun, Jan 10, 2010 at 1:17 PM, C F shma...@gmail.com wrote: Anyone using the above mentioned SIP Gateway made by grandstream? I would like to hear some feedback on real life experience using this gateway. I have a few that I used for about 2 days before I replaced them with AudioCodes MP-124s. They worked fine in the lab, but could not hold up to production use for faxing. When using them for a fax gateway, I had about a 5-10% success rate with multiple page faxes going through... I also have tried using them for an IVR system, but have been unhappy with the results. Most of the issues have to do with echo on the lines. I would not recommend using them, but YMMV. I do know that they make good paperweights or dust collectors =) -Jonathan -- _ -- 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
Re: [asterisk-users] Grandstream GXW-4004
On Sat, Jan 2, 2010 at 4:27 PM, hin lee hi...@yahoo.com wrote: yes, fxs for my fax machines. I don't have any experience with the 4004, but I do with the GXW-4024. I purchased one for a Fax gateway, tested fine, had it in production for two days and ordered an AudioCodes MP-124 to replace it before the secretaries found my cubical... Fax didn't work the majority of the time, but if you don't need to send multiple pages all day long it *might* work for you. An AudioCodes MP-114 costs more, but saves in frustration and lost of time for those who use the faxes. -Jonathan From: Lyle Giese l...@lcrcomputer.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sat, January 2, 2010 3:29:13 PM Subject: Re: [asterisk-users] Grandstream GXW-4004 hin lee wrote: I am consider replacing my TDM card for a FXS gateway. Anyone has any issues with the Grandstream GXW-4004 on Asterisk? I would like some feedback before I spend the $$ this device. http://www.voip-info.org/wiki/view/Grandstream+GXW-4004 Thanks! Just to be clear, the Grandstream gateway is used to interface analog telephones to Asterisk, not for bringing in outside dialtone from your local telco to Asterisk. Why not buy SIP phones instead? I have not used it, so I have no opinion on it, but whose TDM card are you using now. Lyle Giese LCR Computer Services, Inc. ___ -- 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 ___ -- 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
Re: [asterisk-users] Audiocodes MP-114 2FXO/2FXS help registering with Asterisk
The web interface is a bit confusing at first. Here are some of the steps that I remember off hand. Change as little as possible, makes it easier to troubleshoot later. Get the latest code from your vendor (5.6 is what I run) Configure the proxy to register with Configuration - Protocol Config - Protocol Def - Proxy and Registration - Enable registration - Set the registration per endpoint Configure your call routing Configuration - Protocol Config - Routing Tables - IP to Trunk Group If you send a prefix for outgoing calls, you will need to configure that in the manipulation table too Configuration - Protocol Config - Manipulation tables - Dest number IP to Tel Configure authentication Configuration - Protocol Config - Endpoint settings - Authentication Now the part that took me a while to find... Configure the Channel to phone number mapping: Configuration - Protocol Config - Endpoint Number - EndPoint Phone Number Configure the Hunt group settings Configuration - Protocol Config - Hunt/IP Group - Hunt group settings Hope that helps. These are great devices, once you figure out how to get them configured... -Jonathan On Sat, Dec 26, 2009 at 11:39 PM, Joseph syscon...@gmail.com wrote: I have AudioCodes MP-2FXO/2FXS but have a problem registering it with Asterisk. Any links or pointers to configuration how it is done? -- Joseph ___ -- 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 ___ -- 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
Re: [asterisk-users] Audiocodes MP-114 2FXO/2FXS help registering with Asterisk
The best document is the two page quick start guide that came in the box. You want 5.6, and 5.8 should be out soon if you are an early adopter. -Jonathan Sent from a mobile device. On Dec 27, 2009, at 9:02 AM, Joseph syscon...@gmail.com wrote: What what everybody says, it is a good hardware but configuration samples are not easy to find and going through 500page manual is not easy. What they are missing is short configuration guide with samples for specific software like asterisk. My software version is 5.40A I see early next week what is the latest available. On 12/27/09 07:56, Jonathan Thurman wrote: The web interface is a bit confusing at first. Here are some of the steps that I remember off hand. Change as little as possible, makes it easier to troubleshoot later. I did not change much and trying to register just one line first, but is not easy all I'm getting is: chan_sip.c:15593 handle_request_register: Registration from 'sip: 3...@10.0.0.109' failed for '10.0.0.157' - Wrong password 369 is my extension, 10.0.0.109 is my Asterisk server, 10.0.0.157 is AudioCodes IP Get the latest code from your vendor (5.6 is what I run) Configure the proxy to register with Configuration - Protocol Config - Protocol Def - Proxy and Registration - Enable registration - Set the registration per endpoint So I have Use Default Proxy: Yes Proxy Set Table: == What did you enter here (I enter: 10.0.0.109 UDP; do I need to set: Enable Proxy Keep Alive?) Proxy Name: 10.0.0.109 The below two settings (what to put in there, setting from sip.conf: eg.: but which one? Registrar Name Registrar IP Address Under: Gateway Name (I entered asterisk IP) 10.0.0.109 Again below is: User Name Password Not sure what to put in above. Configure your call routing Configuration - Protocol Config - Routing Tables - IP to Trunk Group Is above sections for routing calls to asterisk? If you send a prefix for outgoing calls, you will need to configure that in the manipulation table too Configuration - Protocol Config - Manipulation tables - Dest number IP to Tel No, I don't use prefixes they are dropped by asterisk; so I configured single stage dialing under: Advanced Applications - FXO Settings - Dialing Mode Configure authentication Configuration - Protocol Config - Endpoint settings - Authentication Here I entered authentication from one of my sip.conf entry: [369] [369] ; outgoing/incoming call on fxs port type=friend host=dynamic context=internal secret=523 username=369 mailbox=369 ;dtmfmode=rfc2833 ;dtmfmode=inband disallow=all allow=ulaw allow=alaw canreinvite=yes nat=no callgroup=1 pickupgroup=1 Now the part that took me a while to find... Configure the Channel to phone number mapping: Configuration - Protocol Config - Endpoint Number - EndPoint Phone Number Configure the Hunt group settings Configuration - Protocol Config - Hunt/IP Group - Hunt group settings Hope that helps. These are great devices, once you figure out how to get them configured... -Jonathan I need to find out from the manual what these setting do. I was hoping to find some setting reference on Wiki but there are none :-/ it seems to me the device is not very popular among asterisk users, if it was somebody would create detailed configuration for asterisk. -- Joseph ___ -- 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 ___ -- 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
Re: [asterisk-users] ATA FXO
On Fri, Dec 11, 2009 at 12:44 PM, Connor Spiess cspi...@idea-ma.com wrote: Joseph You could also check out the Audio Codes gateways if the Grandstream doesn't work out for you. They make FXO/FXS gateways. They were reliable boxes for us but this was to a non-asterisk PBX over MGCP. I mention them cause I know they make a SIP based one. We use AudioCodes MP-114 2FXS/2FXO and they have been rock solid. I have a bunch used for faxing connected back to Asterisk over SIP. I will say that I have had a LOT of issues with faxing on the larger GrandStream GXW-4024s and had to replace them. I put a AudioCodes MP-124 in and have had no complaints since. -Jonathan ___ -- 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
Re: [asterisk-users] ATA FXO
On Fri, Dec 11, 2009 at 7:52 PM, Joseph syscon...@gmail.com wrote: [snip] Thank for suggestion. Well, it is not that cheap but the problem with their equipment is luck support and decent manual. I actually find the Quick-start guide that comes in the box the most useful, if you aren't doing anything strange. Whatever I google about AudioCodecs everybody seems to be straggling with the setup; I don't think this should be that hard to write a decent instructions if they want to sell their product. Maybe they have a good product but without support it will not mean much. While I agree about the manual being a little difficult, the actual support from AudioCodes is great. They want you to get support through the reseller or distributor, but you can purchase direct support too. If you do that then you can call them up and talk to an engineer. They will even Web-X in and show you how to do something if they don't have a quick how-to document to email you on the subject. The interface is also a bit overwhelming at first, and forget the console. However once you get the configuration set, export it out as a text file and make a template. I can't speak specifically to Caller-ID on FXO ports, as I mainly use them for FXS and local 911 gateways. -Jonathan ___ -- 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
Re: [asterisk-users] Free Polycom Provisioning Tool
On Fri, Nov 27, 2009 at 11:17 PM, Michael Munger mich...@highpoweredhelp.com wrote: In 2007, I released a Polycom Provisioning Tool. I retired the package earlier this year, and have had so many requests for it, I have revived the concept, new, improved, and still FREE. Any chance of you releasing the source? -Jonathan ___ -- 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
Re: [asterisk-users] VoiceMail greetings
On Sat, Nov 28, 2009 at 5:22 PM, matthieu Nicaise techni...@thinkrosystem.com wrote: Hello everybody, I'm using Asterisk ( 1.6.1.9 ) Voicemail. For example, if i call extension *11 which is not registered, from extension *12, i have no greetings at all, i only have the please leave a message after the beep. I tried to record the busy, unavailable and temporary greetings for extension *11 using VoiveMailMain and the file are well created on the file system. I cannot understand why those files are not played. If i use VoiceMail(*11) in the extension.conf i have exactly the same behaviour. If i user VoiceMail(*11,b) the busy message is read. Is that a normal behaviour ? I can't understand why Asterisk is not using the Dial status automaticaly. Thank you for your help ! The default option for voicemail is to play only the instructions. Take a look at http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail for more details on the options. You will have to parse the Dial status in the dialplan, and pass 'u' for unavailable message to be played. You can see one way to parse the dial status in the sample extensions.conf file under the stdexten subroutine. There are lots of reasons to let the admin decide which greeting to play. For example, my canned 'receptionist' context plays the busy greeting as the after-hours greeting, otherwise playing the unavailable greeting. -Jonathan ___ -- 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
Re: [asterisk-users] VoiceMail greetings
On Sat, Nov 28, 2009 at 7:34 PM, matthieu Nicaise
techni...@thinkrosystem.com wrote:
I made an error in my first mail, i'm calling voicemail in extensions.conf
this way :
exten = _*.,1,Dial(SIP/${EXTEN:0},60)
exten = _*.,n,VoiceMail(${EXTEN:0},u)
exten = _*.,n,Playback(ss-noservice)
You don't need the :0, but that shouldn't cause any issues.
[Nov 29 03:38:13] WARNING[24635]: app_dial.c:1528 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [...@local:2] VoiceMail(SIP/*15-0849a370, *11) in new
stack
That last line should look like (from my 1.6.1.1 system):
-- Executing [...@local:2] VoiceMail(SIP/*15-0849a370, *11,u)
in new stack
Did you reload the dialplan after the change?
-Jonathan
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Re: [asterisk-users] Experience with LLDP
On Tue, Nov 24, 2009 at 12:49 AM, Olivier oza-4...@myamail.com wrote: Hello, LLDP is more and more available on various network elements (endpoint, switches, ...). It seems to ease network configuration. Makes Voice VLAN assignment much easier for sure. Do you have any experience with it ? I work with customers that have mixed environments for access level switches (Cisco, Linksys, Extreme, Juniper, etc) and prefer to use LLDP when the phones support it. It makes sense if you are in an all Cisco environment to use CDP. How would you rate LLDP ? I would rate LLDP as a very useful vendor-agnostic protocol. -Jonathan ___ -- 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
Re: [asterisk-users] Changing labels on Phones
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian We have several types of phones, Cisco 79xx, Aastra 9133i etc. We have a hotdesk type system where anyone can log on to an extension - however what I would love to do is relabel the phone with the current owner when this logon happens. I know that I can change the sip.conf and phones tftp file, however this is a big problem with the Cisco's as they take *forever* (ok, maybe 2 / 3 minutes) to reboot (VLAN problem) 1) Has anyone actually solved this VLAN issue with the cisco ? 2) Is there any way of changing a label without rebooting the phone ? TIA I have not personally tried this, but I remember someone had posted a way to script the change of the background image on Cisco 79x1 phones. You could create a dynamic image in PHP that had the user info on it, then kick off the script to change the background image. Might be a little tricky, but no reboot required! -Jonathan ___ -- 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
Re: [asterisk-users] Changing labels on Phones
On Mon, Nov 16, 2009 at 7:29 AM, Peder pe...@networkoblivion.com wrote: I'm pretty sure it only pulls the background image during a reboot. On a 79x0, yes. On the 79x1 phones the user can change the background to a list of custom images that you provide. It downloads the image on the fly, and applies it. Here is a link to a reboot script http://www.dave.vc/wordpress/?p=14 that uses your ability to press keys on the phone. You could apply the same idea to press the correct buttons to change the background without rebooting. I can't find the script that I found to do this, but I'll keep looking when I get a chance. -Jonathan ___ -- 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
Re: [asterisk-users] digium fax: can't indicate condition 19?
On Wed, Nov 11, 2009 at 2:04 PM, Kevin P. Fleming kpflem...@digium.com wrote: Scott L. Lykens wrote: Any progress on new Fax for Asterisk modules? Last update here was October 19 as quoted above; Original discussion is now over six weeks old. FAA Download Selector still shows modules for 1.6.1.4 as the latest available. Yes, there has been progress. The new modules are undergoing testing in Digium's Product Quality department and (should they not have any regressions) will be released next week. Any chance that 64 bit Linux will be supported? -Jonathan ___ -- 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
Re: [asterisk-users] Cisco 1751 setup with asterisk
On Mon, Oct 19, 2009 at 3:42 PM, Joseph syscon...@gmail.com wrote: How hard is to setup Cisco 1751 w/2x FXO with asterisk? I was googling but couldn't find much information; how to access unit interface for programming? I haven't personally used a 1751, but I have used the 1760 series and 2800 series. It depends on what you are trying to do, but in general they are not that difficult to configure if you just want to send calls to it and have inbound calls route in. If you want to register each port, you are out of luck (or tell me how!) You also can't qualify these devices. It might be a good replacement for Linksys. Not likely. Cisco works great with CallManager, but seems to be somewhat broken with anything else... wonder why? If you want something that is dependable and easy to configure I have had great success with the AudioCodes MP-114 devices. -Jonathan ___ -- 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
Re: [asterisk-users] OT - Can't upgrade Cisco 7942 to SIP
On Thu, Oct 15, 2009 at 12:24 PM, Olivier oza-4...@myamail.com wrote: Hi, I've downloaded for a demo, a P0S3-08-12.zip file which is suppose to work with 7960. Is it supposed to be the same file that the one needed to 7942 model ? No. The SIP firmware for each model are different except for the 794x/796x models. You can download the correct SIP software from Cisco, but are required to have the correct licensing and SmartNet coverage. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with a Cisco AS5300 gateway
destination-pattern .T What does destination-pattern .T mean? I'm not familiar with what .T would match. I would suggest using a more specific pattern that you expect to be coming down the line. One or more characters (up to 31 characters), waiting timeouts inter-digit before sending. http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/dial_peer/dp_plan.html You could be more specific, if you know what is always going to be coming down the line, like 503... if you only have Oregon numbers, and get 10 digits from the provider. T is useful for outbound calls with a trunk number such as 9T because you never know what number those crazy users will try to call. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with a Cisco AS5300 gateway
I don't have any experience with E1, but here are some comments from the T1 perspective (on a 2800 series Cisco). Here is also a link to my collection of Cisco voice debugging commands: http://thurmantech.com/node/5 On Thu, Oct 15, 2009 at 3:27 AM, Phibee Network Operation Center n...@phibee.net wrote: ! Something like this to define what clock to use (internal usually causes a lot of Slip/Error seconds) ! but don't quote me on the following line, I haven't used E1 or the 5300 ! network-clock-select 1 E1 0 isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec preference 2 g729r8 codec preference 3 g723r63 codec preference 4 g711ulaw ! ! You don't seem to use either voice class, do you need both? voice class codec 500 codec preference 1 g729r8 codec preference 2 g723r63 ! controller E1 0 framing NO-CRC4 ! linecode ? ! cablelength ? pri-group timeslots 1-31 description E1 Beta-Test ! interface Serial0:15 no ip address encapsulation ppp isdn switch-type primary-net5 isdn incoming-voice voice no cdp enable voice-port 0:D ! ! ! dial-peer voice 10 voip destination-pattern . redirect ip2ip session protocol sipv2 session target ipv4:IP_OF_ASTERISK:5060 session transport udp dtmf-relay rtp-nte ! codec g711alaw ! If you define the codec class, might as well use it voice-class codec 400 dtmf-relay rtp-nte no vad ! dial-peer voice 42 pots ! Don't make both patterns the same, maybe add a trunk prefix here ! destination-pattern .T destination-pattern 8T incoming called-number .T direct-inward-dial port 0:D ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 ! Don't need this, since you specified it on the dial-peer ! sip-server ipv4:IP_OF_ASTERISK ! Good luck! -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS to SIP gateway
On Wed, Oct 14, 2009 at 12:27 AM, jonas kellens jonas.kell...@telenet.be wrote: Hello list ! I don't have the money to test out all the products and reading the manuals is not always that enlightening... Does someone here know a good gateway-product that lets analogue telephones communicate with an Asterisk-server. I have found the Grandstream GXW-400x to be able to add SIP-accounts to analogue telephone devices that are connected to the FXS-ports. Moreover this product has a backup-PSTN line for emergency calls and backup. Could you advice other products/manufacturers ? We have used Cisco 2800 series routers with voice cards that work fine for PRI, but don't implement SIP the way they should. Analog was not so great. We also tried to convert some Cisco VG-224s to SIP with limited success. I don't recommend using either of those (plus they are expensive...) Grandstream (GXW-4024) had major issues with Fax, so we only use them for connecting for voice only applications. They seem to work well with Asterisk, and are easy to configure. Don't count on fax working at all though, or even worse working in some cases... AudioCodes is where we finally found a product that does what we need. They are about twice as much as a Grandstream (at least for the MP-124 vs GXW-4024) but have been rock solid for faxing so far. They also come in multiple configurations which is handy. We use the MP-114 2FXS/2FXO device at our remote sites for local PSTN access and to connect a fax machine. They also support survivability (proxy registration) in case of WAN failure. The complaints that I have are that the web interface has A LOT going on, and there is no real CLI to speak of. Neither of these are real issues, just takes you a few more minutes up front to read the manual. I haven't tried any Adtran devices but have thought about purchasing one to test with if I ever get the time. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco router
Depends on what the router is. If you get a 2800 series router (we use 2801s and 2811s for T1s in production with no major issues). You need the T1/E1 module, DSPs, and an IOS that supports voice. For a 2800 series you would need something like: - VWIC2-MFT-T1/E1 ( or VWIC2-2MFT-T1/E1 if you want two ports) - PVDM2-32 (PVDM2-64 if you want two E1 ports, or two PVDM2-32s) - IOS that supports voice (I use spservicesk9) If you are looking at an older router like a 2651XM or something, you will need something like: - NM-HDV2-2T1/E1 (not the VWIC2, as the NM has build in ports) - PVDM2-32 If you have a specific router in mind, I can be more specific. -Jonathan On Wed, Oct 14, 2009 at 11:00 AM, Julian Lyndon-Smith aster...@dotr.com wrote: I was thinking of putting a cisco router on the E1 line for my test system, so I can have multiple test servers accessing the ISDN, rather than a dedicated server and a TE410 card. I *am* confused at all of the modules for the cisco :) What would be the best router to use to connect 30 channels E1 to SIP ? What modules would I need ? I was going to purchase off ebay as this is purely for testing purposes. TIA ;) Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco router
On Wed, Oct 14, 2009 at 12:57 PM, Julian Lyndon-Smith aster...@dotr.com wrote: Thanks for the info. I didn't have any model in mind, just wondering what was required. If you haven't purchased anything yet, or don't have anything, it might serve you better to look at other products. While the Cisco 2800s that we use work with Asterisk, we use them because that's what we had. I would look at an AudioCodes M1000, or an Adtran 908e or the like. I don't have any experience with E1, but I would guess that there is some support for them by those devices. The AudioCodes is about the same cost as a new Cisco solution, but the Adtran would probably be a lot less. I haven't had a chance to play with Adtran and Asterisk, but you can register at their website and play with all of the CLI / GUIs for all the devices which is really cool. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Door Phones
On Wed, Oct 14, 2009 at 1:52 PM, Dan Journo d...@keshercommunications.com wrote: Hi, Can anyone recommend a cheap SIP doorphone? Please only respond if you’ve had personal experience of a doorphone. I searched around for a while and couldn't find a hardened SIP external phone. We ended up using an ATA and a regular outside door phone from Ceeco (http://www.ceeco.net/ Model WPP-531-F). For a analog phone in a metal box, they aren't exactly cheap. You could say that an Analog phone would be more secure if someone ripped it off the wall, they wouldn't have network access. Then you just lock down what numbers can be called on your PBX. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls being dropped - Cisco 7940 with SIP 8.12 image
On Sun, Oct 11, 2009 at 8:03 AM, James Stocks stoc...@stocksy.co.uk wrote: OK. For anyone finding this thread, the problem exists in Asterisk 1.4, but upgrading to Asterisk 1.6.1.6 appears to eliminate the problem. Sorry, I lost your last response in my inbox... Your phone configs look fine. The only thing that we do differently is disable VAD on the phones. Never used 1.4, only the 1.6 branch. Glad to see that you got it working. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXW4024 experience
Don't use them for Fax... I spent too much time trying to use one for a faxing ATA. (We went with the AudioCodes MP-124 instead, which rocks). We to have some analog phones and an analog IVR system hooked up to one with no issues. They are easy to configure if you just need to hook up some analog handsets. -Jonathan On Mon, Oct 5, 2009 at 2:14 AM, Olivier oza-4...@myamail.com wrote: Hi, In this http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/204725/focus=221375 dating from 2008, experiences with Grandstream GXW4024 were asked. Has anyone something up-to-date to share about this ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls being dropped - Cisco 7940 with SIP 8.12 image
On Sat, Oct 3, 2009 at 6:17 AM, James Stocks stoc...@stocksy.co.uk wrote: Hi everyone, I hope someone can help me with a problem I'm having with Cisco 7940 phones on the SIP 8.12 image. When I place a call from one of the handsets, the call proceeds as normal for 20 seconds and is then terminated by Asterisk (1.4.26.2): We are runing 08-12-00 on 7940/60s just fine (Asterisk 1.6.1.1), and have been for a while. As far as I can tell, the 'a=silenceSupp:off - - - -' header is not accepted by the 7940, which seems like a bug in the SIP image to me. However, I can't find a way to report this problem to Cisco without a support contract (which I do not have). Reverting to version 7.5 fixes the problem, but it is still present in 8.11. The problem is not present if the PSTN initiates the call, nor is it present if I allow the handsets to reinvite each other. Here's the sip.conf snippet if it helps: That all looks fine to me. What do your SIPDefault.cnf and SIPMAC.cnf files look like? -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk HA Current Thoughts (Centos 5.3 Platform)
I have been working on a HA procedure for Asterisk on CentOS 5.3, but haven't had time to publish it. It is a little complex, but here are the components used: - CentOS 5.3 - Asterisk 1.6 (version doesn't matter) - MySQL - Cluster services - GFS2 - DRBD A basic run-down is: * Two servers configured with DRBD in Master-Master mode. All data is replicated between the two so in case of a failure there should be very limited data loss (voicemail) if any at all. * MySQL and Asterisk run on the same node. If you have an external MySQL server / don't use MySQL, then this is not an issue. The MySQL data directory is also mounted on a GFS2/DRBD partition. The most important thing here is to use INNODB, NOT MYISAM! MyISAM doesn't take kindly fail-over... * Using Cluster services enables you to create GFS2 file systems (on top of DRBD) so that both nodes can see the data at the same time. This is important to reduce the time required for fail-over. Cluster services also handles starting/stopping the services, and migrating the Virtual IP address between nodes. * DHCP (if needed) runs on both nodes, as DHCP has native support for fail-over configuration. It's pretty easy to get installed and running. I also create RPMS for Asterisk, so that the version on each service is the exact same. I can upgrade one node, use the cluster manager to fail-over to the other node (during a maintenance window of course!). The biggest issue now is that the CentOS Repo is somewhat broken for Cluster... but there is a work around on the bug tracker for CentOS. Hopefully that will be resolved soon. Let me know off list if you need any help! -Jonathan On Fri, Oct 2, 2009 at 10:58 AM, James Hankins j...@allpointsmediaworks.com wrote: I'm looking into doing an HA setup for a Asterisk 1.4 install on Centos. I've seen a number of different pointers to packages for this some of which are packages that seem quite dated from an update perspective (Ultra Monkey links I've seen haven't been updated in a while). What is the current best practice on this for this platform? My first foray into any of the Linux HA setups but not afraid of the command line. Jim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk HA Current Thoughts (Centos 5.3 Platform)
On Fri, Oct 2, 2009 at 11:41 AM, Fred Posner f...@teamforrest.com wrote: * Two servers configured with DRBD in Master-Master mode. All data is replicated between the two so in case of a failure there should be very limited data loss (voicemail) if any at all. If you put the asterisk spool, lib, and config files on the DRBD then you shouldn't lose voicemail or any configuration. If someone is in the middle of recording a message, and the server fails, you will probably lose that message. That's all I was getting at. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone having issues with 1.6.1.6 res_snmp?
I am working on updating to 1.6.1.6 and if I have res_snmp.so auto-loading on CentOS 5.3 Asterisk Seg faults every time. I can load the module manually after the initial startup. I am starting to dig into it further and will open a ticket, just wanted to see if anyone else knew of any issues off hand, or could reproduce it. Thanks -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom auto-install asterisk using ks.cfg
I have with CentOS 5.3 and custom 1.6.1.6 RPMs. If you use RPMs for the installation of Asterisk then it's really easy. As for the Kickstart, if you haven't used it much here I did a quick write-up with example script here: http://thurmantech.com/node/3 Either use RPMs and add them to the packages section, or download the tar.gz file in the post script, and auto-compile. However, auto-compile might have different results on different systems (hence why I use custom RPMs) -Jonathan On Thu, Sep 17, 2009 at 4:27 PM, Neeraj Chand neeraj.ch...@ocis.com.au wrote: Hi guys, Anyone done this with CentOS and asterisk 1.4? thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco call manager version 6.1.3
I have a SIP trunk between CCM 6.1.2 and Asterisk 1.6.1.1 working without any issues. What does your peer section of the sip.conf look like? When do you get the error (call direction)? -Jonathan On Fri, Sep 4, 2009 at 12:00 PM, Jerry Geisge...@pagestation.com wrote: Hi all I have asterisk 1.4.12 that was working on CCM 4.0 they updated to 6.1.3 and it no longer works. I tried updating to 1.4.26.2 but still not working. I get SIP error 503 service unavailable. The guy says he has MTP enabled etc... Anyone connected to CCM 6.1.3 and have it working? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sticky Park
You could put something into the Asterisk Database with DBput/DBget.
I don't have an example off hand, but create a stickypark family and
store which channels go back into which parking slot. Or something to
that effect, and it would exist until you remove it from the database.
-Jonathan
On Thu, Aug 27, 2009 at 10:52 AM, Mat
Murdockmmurd...@kimballequipment.com wrote:
My company for various reasons has asked that I come up with a way to
have previously parked calls be re-parked in the same parking slot. I
have looked at setting up asterisk so that the receptionist chooses
which slot to place a call, but I think there is an easier way. That is
when I came up with the idea of Sticky Park. Here is how it would
work. A call would come in and the receptionist will park the call as
she normally does. Asterisk will the pick the first open parking slot,
let's say 702 because there is already a call on 701. Lets say that the
call parked on 701 is picked up, freeing 701. So, 701 is free and 702
has our call parked on it. Now the call on 702 rings back to the
receptionist because it has timed out. She asks the person if they
would like to continue hold and will again park the call as she normally
does. Asterisk will then re-park the call back onto 702 because that is
where it came from. The normal behavior of Asterisk would of been to
park it on 701 because it is the first free parking slot. That is why I
call it Sticky Park. So what happens if between the time she picks
up the call and re-parks it someone else parks a call on 702? Then I
think Asterisk should then pick the first available parking slot and
that call becomes stuck to that parking slot if additional re-parks are
necessary.
Here is my dialplan on how I thought I could accomplish this with
dial-plan magic.
Here is the relevant features.conf entries.
[general]
parkext = 799 ;We need to use our own 700 extension so lets get this
out of the way.
parkpos = 702-706
comebacktoorigin = no ;This causes calls that have timed out to
come to the parkedcallstimeout context at s,1.
Ok now onto my Dial Plan.
[from_internal]
include = parkedcalls ; Gotta have this or things don't work.
;I do an attended transfer to 700.
exten = 700,1,Answer()
;Just so I can see if anything has been set
exten = 700,n,NoOp(I want to be parked on: ${PARKINGEXTEN})
;Also so I can see what the state of that parking slot is.
exten = 700,n,NoOp(Device State is:
${DEVICE_STATE(park:${parkingext...@parkedcalls)})
;Check to see if PARKINGEXTEN is set. If not then this must be a new
call being park, let's let asterisk find a spot for it.
exten = 700,n,GotoIf($[${LEN(${PARKINGEXTEN})}=0]?parkcall)
;Ok Looks like this call has been parked before. Let's see if we can
repark it in the same spot again. If it not INUSE then let's park the call.
exten =
700,n,GotoIf($[${DEVICE_STATE(park:${parkingext...@parkedcalls)}=INUSE]?:parkcall)
;Previous slot is not occupied lets clear the PARKINGEXTEN variable so
that when we park the call Asterisk will find the first available slot.
exten = 700,n,Set(PARKINGEXTEN=)
;Lets park the call.
exten = 700,n(parkcall),Park()
exten = 700,n,Hangup()
[parkedcallstimeout]
exten = _SIP011XX,1,Answer()
exten = _SIP011XX,n,NoOp(Call Parked on: ${PARKINGSLOT})
exten = _SIP011XX,n,NoOp(This is who parked us: ${EXTEN})
exten = _SIP011XX,n,Set(PARKINGEXTEN=${PARKINGSLOT})
;This sets the PARKINGEXTEN to the parking slot we were parked in.
exten =
_SIP011XX,n,Dial(SIP/${EXTEN:4:4},${RINGTIMER},${INTERNAL_DIAL_OPTIONS})
;This send the call back to the person who parked it. There are a
couple of global variables I use here. Nothing unusual here.
So what is the problem? Well the problem is that the PARKINGEXTEN
variable gets reset after the dial command in parkedcallstimeout. That
makes it so I cannot find out where that call was originally parked If
I can find out how to get that little bit of information when the call
is re-parked then I think this will work. If anyone has any suggestions
on how to accomplish this I would be grateful.
Thanks,
Mat Murdock
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Re: [asterisk-users] mysql sip realtime
When I reload chan_sip.so, it seems that connected terminals are no longer detected by Asterisk because when I tape CLI command sip show peers, there is no results displayed. Any reflexions about that ? They won't be found in the CLI command until Asterisk receives another packet from that peer and a load from the database is forced. Would it be useful to have a way to 'precache' entries in realtime? So you could do a reload then a 'precache', or maybe just some way to have realtime update the cache from the database base on a record modification date. This might make it appear more like a static sip configuration, just using the database for storage. It would also seem to make it more 'realtime' if things were bidirectional. I haven't looked that much into the realtime code, but this could be an interesting project if others think it would be useful. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mysql sip realtime
Ideally, the way realtime works, it shouldn't matter at all whether the record exists in memory or in the database. In reality, there's a few cases where the data needs to exist in memory for a particular event to occur correctly (such as device state notifications). I think a better goal would be to get Realtime integration to a better place where device state notifications (and other events that require a SIP peer to be in-memory) could actually be delivered to a realtime host not in memory and the realtime caching that we currently need to get things to work correctly could go away entirely. Good point. It would be much more effective to improve the realtime integration. Is there a list somewhere of all of these cases? With a little direction I would be willing to work on this. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Gateway - Worth it?
I try to off-load specific tasks like PRI-to-SIP to dedicated hardware for the task. It is also easier to have centralized call processing and easy to configure/manage devices in our remote locations. I have colleagues that use Digium PRI cards just fine. Just depends on your budget and philosophy. We currently use the remains of a CallManager system which includes two PRIs into a Cisco 2811 voice bundle. While the Cisco does work, I wouldn't recommend it. The SIP implementation doesn't play well with others, and even some of the configuration commands that you can use don't take effect... While I haven't used AudioCodes for PRI termination (yet) I have been pleased with the analog gateways they make. The cost on a AudioCodes Mediant 1000 with two PRI ports or two Mediant 600's would probably set you back less than a Cisco 2801/2811 voice bundle + VWIC-2MFT-T1 and be easier to configure. I will say that when I need to replace my other 2800s I will probably go with a AudioCodes M1000 (PRI and Analog capabilities), but I don't have any hands on experience with that device at this time. -Jonathan On Thu, Aug 13, 2009 at 4:26 AM, Steve Totaro stot...@asteriskhelpdesk.comwrote: On Thu, Aug 13, 2009 at 6:25 AM, Shashi Dookhee sdook...@fortify.comwrote: Hi all, I was looking to build a SIP-to-PRI gateway using Asterisk (as in my other post), but there is also an alternative of using a Cisco router with something like an NM-HDV module with a T1 VIC module and DSP channel banks. The question is, would it be more reliable to offload all dahdi/zaptel/libpri type stuff to a dedicated gateway device (Asterisk or Cisco) and have the Asterisk PBX only process SIP/IAX? What prompted all this was terrible call quality/dropped calls over our PRI (which currently has a Wildcard TE207P PCI card) and after two weeks of searching I think I've finally found the issue today (Serial ATA vs Auto mode in the BIOS, would you believe which leads us to believe that offloading duties would be very beneficial performance-wise!). I can think of the gateway allowing us to have multiple PBX's to serve our calls more easily, maybe making it easier to failover PRI's too, etc... Although we do have an ADTRAN Atlas that allows us to split call volume by DIDs if we wanted to (right now we only have one Asterisk PBX, one electronic FAX server and a couple of standalone FXO devices - all sharing our PRI by the ADTRAN)... Any comments appreciated. Thanks! S. I try to dedicate a box (sometimes HA heartbeat or just a decent server like the HP DL360 with dual power supplies and RAID) to a certain task. The litmus test is when that task is mission critical and can logically be drawn out on a whiteboard. PRI-SIP is pretty intensive and obviously mission critical and at least to me can sit on it's own little square on the whiteboard. Budget wise, if possible, I would go with two HP DL360s with dual power supplies, RAID and a Redfone device or at the least two Sangoma cards, one for cold spare (unless Digium now has a lifetime warranty that I am unaware of). -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 79XX, SIP and Asterisk
I am also using them quite extensively, but with English menus. I know that the Locale files from Cisco do not come with the firmware, but usually as an update for CallManager. There are a ton of languages that work with the latest firmware, but I have no idea how to actually get the files from Cisco. You may be able to purchase SmartNet on the phones and get it that way, or at least they would listen when you called them... -Jonathan On Wed, Aug 12, 2009 at 5:23 AM, David Gibbons d...@videon-central.comwrote: I am using the phones quite successfully, though I have not tried non-English menus. -Dave *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* Wednesday, August 12, 2009 12:33 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Cisco 79XX, SIP and Asterisk Hi, Is anyone successfully using SIP-enabled Cisco 79XX phones with Asterisk ? Could you then configure this phone to display non-english menus (in french, spanish, german, ...) ? Mine is using a rather old SIP firmware (8.3 ?) with which I could get non-english menus. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 79XX, SIP and Asterisk
On Wed, Aug 12, 2009 at 12:39 PM, Olivier oza-4...@myamail.com wrote: 2009/8/12 Jonathan Thurman jthurma...@gmail.com I am also using them quite extensively, but with English menus. I know that the Locale files from Cisco do not come with the firmware, but usually as an update for CallManager. There are a ton of languages that work with the latest firmware, but I have no idea how to actually get the files from Cisco. You may be able to purchase SmartNet on the phones and get it that way, or at least they would listen when you called them... -Jonathan Using english menus is a show-stopper in non-english-speaking countries ... Very true From memory, situation was the way you described : you need a call manager (or Asterisk with SCCP ?) to get native menus. It seems this is still the case ... You need the locale files. The phones will pull them off of any tftp server. If you have access to the Cisco software site, you can download the locale installer for a lot of languages. You then just have to extract the files and get them onto your TFTP server and configure the phone to pull the correct locale. This will most likely require either SmartNet on the phones or CallManager. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Multiline phone
On Tue, Aug 11, 2009 at 5:12 PM, Jimmy Ezell jez...@hmhca.com wrote: Sorry for not being real clear. What I have is 1 front desk phone only with 6 lines Front Desk Phone line 1 - incoming extension 1 Front Desk Phone line 2 - incoming extension 2 Front Desk Phone line 3 - incoming extension 3 Front Desk Phone line 4 - incoming extension 4 Front Desk Phone line 5 - incoming extension 5 Front Desk Phone line 6 - inside office extension If incoming line 1 is busy I want the next incoming call to come in on line 2. If incoming line 2 and 3 are busy but 1 is free the next call should got to line 1. So lines 1 and 2 might get a lot of calls but only on really busy days will calls make it up to lines 4 and 5. Does that make sense? Anyone have the solution? *Jimmy Ezell* What is the purpose of having the incoming lines show as different line appearances if you are just going to use them as a hunt group? I as because the easiest solution is to route all the incoming calls to the same line appearance. Each line can have multiple calls coming in at the same time, and you can handle easily using the soft buttons. Then you could reuse the extra buttons as speed dials, other specific extensions (i.e. the boss' DID) etc. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CIsco 7960 + asterisk: hepl needed
Are there any other phones registered, or is it just this phone that is having issues? The first thing that I see is the qualify=200 line, and I have not had good experience with Cisco devices and any qualify setting. I would try leaving that out. I also have double quotes around the line1_* parameters. See my comments inline. On Tue, Jul 28, 2009 at 2:14 PM, pepesz76 pepes...@o2.pl wrote: Dear All, I'm trying to configure my new phone Cisco 7960 to work with asterisk. I followed http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html and I got into the point where I can see on the the display line indication showing 55 phone icon with x so it looks like the phone is not registered. The phone and the asterisk are in the same local network. On asterisk side: Cawdor*CLI sip show peers ... 55/55 (Unspecified)D N 5060 UNKNOWN ... sip.conf: [55] type=friend defaultuser=55 secret=12345655 context=home_castle callerid=Lukasz Cisco 7960 55 canreinvite=no host=dynamic dtmfmode=rfc2833 Remove: qualify=200 Add: disallow=all allow=ulaw (Or whatever codecs you are using) buggymwi=yes SIPDefault.cnf: image_version: P0S3-8-12-00 proxy1_address: 192.168.1.109 ; Can be dotted IP or FQDN proxy_register: 1 messages_uri: 80 phone_password: cisco ; Limited to 31 characters (Default - cisco) sntp_mode: directedbroadcast ;unicast sntp_server: 192.168.1.77 time_zone: GMT+01/00 ; assuming you're in GMT time_format_24hr: 1 ; to show the time in 24hour format date_format: D/M/Y ; format you would like the date in dial_template: dialplan SIPMAC.cnf: image_version: P0S3-8-12-00 line1_name: 55 line1_name: 55 line1_authname: 55 line1_password: 12345655 line1_shortname: 55 line1_displayname: Lukasz Cisco7960 line1_authname: 55 line1_shortname: 55 ; displayed on the phones softkey line1_password: 12345655 line1_displayname: Lukasz Cisco7960; the caller id proxy1_port: 5060 proxy1_address: 192.168.1.109 # Phone Label (Text desired to be displayed in upper right corner) phone_label: Castle ; add a space at the end, looks neater phone_password: cisco ; Limited to 31 characters (Default - cisco) user_info: none telnet_level: 2 If that still doesn't work, then telnet into the phone and see what is going on. Commands like show config show register etch are very useful for this kind of troubleshooting. If the phone was attached to a CallManager using SIP before, then there could be some bad configuration still stuck in the phone. If you don't specify a new value, these phones cache the old config. Try factory defaulting the phone if all else fails. I have quite a few of these phones working without issue. Good luck! -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10) - Solved.
Huh? http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0-current.tar.gz is not the same as http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0.10.tar.gz? Their sha1 files are identical. sean I believe he means that: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0-current.tar.gz is not the same as svn checkout http://svn.digium.com/svn/asterisk/branches/1.6.0 Which is true as there are lots of things that have been fixed in the Subversion repo. -Jonathan ___ -- 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
Re: [asterisk-users] Call Parking timeout fails
This was fixed in the 1.6.1 SVN, and I would guess that it was also fixed in the 1.6.0. SVN log: r189951 | russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines Fix call parking callback. Pipes - Commas. You will have to create a patch against the 1.6.0 source, but you could start by looking at the patch in this issue: https://issues.asterisk.org/view.php?id=15162 Please note again that that patch was against 1.6.1.0. -Jonathan On Tue, Jul 14, 2009 at 11:09 AM, Barry L. Kline blkl...@attglobal.netwrote: John A. Sullivan III wrote: Hello, all. I'm having a nasty problem with call parking in Asterisk 1.6.1.1 that smells like a bug. When the call returns, it seems to be returning to a | delimited extension and failing. Here is the output from the console: Hi John. I've just run into the same problem on 1.6.0.10. Have you heard any more about this problem? TIA, Barry ___ -- 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 ___ -- 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
Re: [asterisk-users] Help in oh323 gatekeeper
On Tue, Jul 14, 2009 at 2:19 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear; I would like to ask: when Asterisk was registering on the gnugk, both (asterisk and gnugk) were on the same hardware machine and same IP address? Can they be on the same IP address? If I understand your questions: Can Asterisk and GnuGK both run as an h323 server on the same IP Address, the answer would be no unless they are running on different ports. You can not have two processes on the same machine/ip/port combination. In case they were on the same IP address then: I am afraid the oh323 channel in asterisk will respond for the H323 endpoint (IP Phone) instead of the gnugk (specially if the IP Phone was in routed mode and not register to gnugk)? I mean, if the IP Phone was need to place call via the gnugk in the routed mode, and the call need to be send for Asterisk, so how can u avoid that oh323 channel in asterisk from responding for the IP Phone instead of the gnugk it self? Because if u let the IP Phone send the call for the IP address that asterisk running on it, then the h323 channel in the asterisk will respond as u know, so how to let the gnugk respond and not the asterisk h323 channel? Right. If they both run on the same ip/port then the one started first would win, and listen for connections (the second app should fail to bind and complain). You could change the port, or the IP that the one of the apps is listening on. Hope that helps. -Jonathan ___ -- 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
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On Sat, Jul 11, 2009 at 12:09 PM, Wayne wa...@planetwayne.com wrote: Thanks for all for the feedback with this - I'd like to help where I can - I'm building another 1.6 system for the office to try out the exchange tie in so if the general consensus is SIP is ok - then that's good for me too as I only have access to a SIP phone there. All my phones at home are Skinny so I was trying to kill two birds with one stone so to speak (get me on the latest version and play around with exchange). My own 1.2 system is chugging along ok so far and there's no 'massive' need to move it over (ok other than servers sat in the lounge which the missus has a moan at every so often :-) ). One thing I am unsure of - how do I get the dumps / information you want in a suitable format.I'm still a novice with Linux / Asterisk but I'll gladly get anything to help out (just need some pointers in the right direction). Take a look at file doc/backtrace.txt and doc/valgrind.txt. What is your exact test scenario? I have updated my test box based on the latest SVN of 1.6.1 (I use 1.6.1.1 in production) and I have one Cisco 7940 configured for Skinny. It seems to work just fine, no seg faults. Have you tried the latest SVN for 1.6.0? You should take a look at this issue if you haven't already: https://issues.asterisk.org/view.php?id=13777 -Jonathan ___ -- 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
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On Fri, Jul 10, 2009 at 4:33 PM, Wayne wa...@planetwayne.com wrote: Hi Steve, Thanks for the pointers. I must admit - I was leaning towards 1.6 as this apparently has support for SIP over TCP (?). My end goal with this was to try and get Asterisk talking to Exchange 2007 servers unified messaging. While I haven't used the SIP over TCP in production (yet), I find that the 1.6.1 series is stable for our environment. I don't know about using Exchange, as we are staying as far from unified messaging as possible (for political reasons of course...) I wouldn't install 1.0, so why go back to 1.2 or 1.4. Just more to learn and relearn. The important thing is to have a test environment to get all of the show stopping buts out. As for chan_skinny - I'm currently using this on an existing 1.2 server although from what I've picked up from previous posts (going back a while) the inbuilt version is now quite stable and possibly better than the older 'chan_skinny' (which I think the development has stopped for now?). This is why I opted to use it for the new 1.6 server. What is the main reason for staying with skinny on these phones? I have quite a few 7940/7960 converted to SIP that work great. Next week I will try and duplicate this behavior on my test system with skinny, but you should get a bug report filed with the core and important configurations. -Jonathan ___ -- 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
Re: [asterisk-users] Small site survivability
snip Audiocodes supports SRST on their mediapack analog gateways. This might be a viable option. I haven't used any Audiocodes devices before. Are people pleased with them? snip Deploy a lot of small asterisk based appliances... This way you can completely decentralise your setup and give each office it's own autonomous system, only needing the WAN links for inter-site calls (and maybe your backhaul to the PSTN) We do not want to decentralise our configuration. The whole point of pulling all of these sites together was to centralise management. We also have a lot of users that move to a different site every year and keep their DID as long as they are within the same county. We simply need some way to provide basic call management for local 911 access in the case of WAN failure. Our Cisco devices do this for any phone using SCCP. If you want to buy an additional license you can have SIP too... snip What happens for IT when WAN fails ? Are people still able to work or not ? I work in K-12 education, so while our users will complain that they don't have internet/email/etc, they continue to work with or without the WAN connection. Even if normal phone service is not available, we HAVE to provide 911 access. If they are, then it should be possible to use current routers (if they have such POTS interfaces) as Media gateways and have a local resource to act as a backup Asterisk server. I am trying to avoid adding additional servers at this small sites. Some sites are nothing more than a portable with Metro Ethernet connection and a fan-less router and switch. If they are not, having IT and Telephony to share the same backup WAN is advisable. Backup WAN links... I wish! Thanks for the input. -Jonathan ___ -- 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] Small site survivability
We are currently moving away from a wide-spread Cisco CallManager deployment to Asterisk. For many of our small sites we have the routers configured for what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP registrar. We are converting to SIP, and from what I can tell Cisco wants a license for each router to run SRST over SIP... So my question to the group is: What are you doing for survivability in these small (6-30 phone) sites? I would like to avoid deploying a lot of servers if at all possible. The requirements would be a simple, easy to manage device for the phones to register to in case of WAN failure with 1 or 2 POTS lines attached (also used for 911 calls from that site). Thanks for any suggestions! -Jonathan ___ -- 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
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote: Hello, all. With the assistance of very helpful folks, our brand new multi-tenant setup seems to be working smoothly from start to finish with just a bump or two. The biggest is parking. Now that we got most kinks worked out, I'm a little more comfortable in trying to resolve this. There seem to be two problems: 1. Parking assigns parking spaces from the default group no matter what we do. I haven't tested this. 2. When the parked call timer expires, the callback to the original callee fails because a | delimiter is used in the Dial() function. This has been fixed in the 1.6.1 SVN, and you will have to back port a patch until these changes are rolled into another release. I was disappointed that more bug fixes were not included in 1.6.1.1. -Jonathan ___ -- 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
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
This has been fixed in the 1.6.1 SVN, and you will have to back port a patch until these changes are rolled into another release. I was disappointed that more bug fixes were not included in 1.6.1.1. -Jonathan Asterisk 1.6.1.1 was released for a security issue, AST-2009-001. Why would you think that more bug fixes would be in it? Security release are only supposed to have the fix for the issue that caused the release to take place. - Brad Sorry, I am relatively new to the Asterisk project and probably don't fully understand how the release cycle for this project works. Are you saying that the minor releases are only for security bugs? I haven't seen anything in the on-line documentation that states this. I would think that major usability issues (like parked calls getting dropped if you don't pick them up) would be addressed in a release, not only in SVN. To me the point of a minor release is to fix bugs. It is sometimes quite a headache to download the latest release, have an issue, dig through the issue tracker to find that it was fixed a month ago, then update to SVN or back port a patch. This is especially difficult for those that are new to the project/community. -Jonathan ___ -- 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
Re: [asterisk-users] T38 Fax Gateway for Asterisk 1.6
On Jun 26, 2009, at 10:44 AM, Tim Nelson tnel...@rockbochs.com wrote: - David Backeberg dbackeb...@gmail.com wrote: On Fri, Jun 26, 2009 at 1:31 PM, James Lamannajlama...@gmail.com wrote: The use case is that a customer has a fax machine attached to an ATA. The ATA sends T38 to Asterisk over SIP, then I need to forward that out the PSTN. Got it. I'm saying why not skip the ATA and asterisk, and plug the fax into the PSTN? ... Maybe... just MAYBE... the ATA/Fax will be nowhere near the physical proximity of the PSTN connection? What if the ATA/Fax is going to be at someone's remote cabin that only has internet connectivity and the Asterisk/PSTN is at their office or home? --Tim Or toll bypass, routing to other internal faxes, you want to assign the fax a number out of your DID block... If you are not a small office, there are a lot of reasons to not have a dedicated fax PSTN line. -Jonathan ___ -- 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 ___ -- 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
Re: [asterisk-users] Removing line 2 from CISCO 7940g
David's directions will work on a 7941/7961, not the 7940/7960. You do have
to keep the line configuration for the 79x0 series phones in the
SIP${MAC}.cnf file.. I have not tested setting them to , but I know if
you telnet into the phone they will show UNPROVISIONED as the setting.
You can also clear all of the cached settings by telnetting into the phone,
clearing the config, and resetting it.
-Jonathan
On Thu, Jun 25, 2009 at 8:29 AM, David Gibbons d...@videon-central.comwrote:
Mike,
1. Remove the 'line 2' entries completely from the SEPXX.XML file.
2. Change the 'Version' tag in the SEPXX.XML file. You need only
change one digit; I usually just increment the last digit.
(version1.0.0.0-9/version).
3. Restart the phone (Settings - **#**).
4. This should do it. If it doesn't, proceed to step 5 with caution.
5. If the line still appears, reset the phone to factory defaults
(Hold # while booting, then dial 123456789*0# when the line lights flash
amber back and forth). DO NOT RESET TO FACTORY DEFAULTS IF YOU DON'T HAVE
THE TFTP SERVER SETUP WITH THE FIRMWARE IMAGES. This will force the phone to
re-download the SEP.XML file.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, June 24, 2009 5:12 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Removing line 2 from CISCO 7940g
Folks,
I have CISCO 7940g phone. I have in the past configured the phone with two
lines. Having found the 2nd line wasn't much use, I want to remove it from
the config. I have taken it out of the SIP config file that is TFTPd to the
phone but it is still showing on the phone and it is still trying to log
into Asterisk with that account. I have tried removing the config line and
blanking out the options but it still persists.
Does anyoen know how to get rid of the thing?
Mike.
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Re: [asterisk-users] Removing line 2 from CISCO 7940g
The phone caches the configuration... To remove it update the config like so: line2_name:UNPROVISIONED line2_authname:UNPROVISIONED line2_password:UNPROVISIONED line2_shortname: UNPROVISIONED line2_displayname: UNPROVISIONED For each line that you don't want anymore. So on a 7960 you would have to do this for lines 2-6. The line will then disappear from the phone. -Jonathan On Wed, Jun 24, 2009 at 2:11 PM, Mike asterisk-us...@norgie.net wrote: Folks, I have CISCO 7940g phone. I have in the past configured the phone with two lines. Having found the 2nd line wasn't much use, I want to remove it from the config. I have taken it out of the SIP config file that is TFTPd to the phone but it is still showing on the phone and it is still trying to log into Asterisk with that account. I have tried removing the config line and blanking out the options but it still persists. Does anyoen know how to get rid of the thing? Mike. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkpCloMACgkQmUrfmTU1ohWLzwCg39To92tTSB+6j8TkkJ4QTO+S 1cAAn3a7FvqwKu4Id/LV44JiO8rmR4m/ =Dpe0 -END PGP SIGNATURE- ___ -- 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 ___ -- 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
Re: [asterisk-users] Cisco 7941G Auth
Try resetting the phone to factory defaults. I have had some odd issues when moving phones between CallManager and Asterisk that this was the easiest fix. It might be worth a shot. Here are the directions: http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a00800941bb.shtml -Jonathan On Tue, Jun 23, 2009 at 2:53 AM, Sasa s...@shoponweb.it wrote: Hi, also with your template I have always the same problem ! Thanks. -- Salvatore. - Original Message - From: David Gibbons d...@videon-central.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, June 22, 2009 2:41 PM Subject: Re: [asterisk-users] Cisco 7941G Auth Hey Sasa, I have templates of all the files you need here (SEP file, extension file): http://dave.vc/wordpress/wp-content/uploads/2008/11/phoneadd.zip If you need further assistance, let me know. Thanks Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sasa Sent: Monday, June 22, 2009 4:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G Auth Jonathan Thurman wrote: What does your SEPMacAddress.cnf.xml file look like? In my experience, the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I had to specify the firmware version in each SEP file. I am using 8-4-4S, but for you this would be something like this: device loadInformationSIP41.8-0-2SR1S/loadInformation /device Hi, I have already writed also in SEPMacAddress.cnf.xml file (other at XMLDefault.cnf.xml file) the parameter: loadInformationSIP41.8-0-2SR1S/loadInformation ..but the problem isn't resolved !. Can I try to change some parameters ?..are desperate ! I think I have tried everything ! Thanks. -- Salvatore. - Original Message - From: Jonathan Thurman jthurma...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 19, 2009 6:04 PM Subject: Re: [asterisk-users] Cisco 7941G Auth What does your SEPMacAddress.cnf.xml file look like? In my experience, the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I had to specify the firmware version in each SEP file. I am using 8-4-4S, but for you this would be something like this: device loadInformationSIP41.8-0-2SR1S/loadInformation /device And you shouldn't need the tlv file. -Jonathan On Fri, Jun 19, 2009 at 8:25 AM, Sasa s...@shoponweb.it wrote: David Gibbons wrote: I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. I have tried with tftp on linux machine but the result isn't changed. Thanks. -- Salvatore. - Original Message - From: David Gibbons d...@videon-central.com To: novacks...@gmail.com; 'Asterisk Users MailingList - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, June 19, 2009 4:50 PM Subject: Re: [asterisk-users] Cisco 7941G Auth I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Friday, June 19, 2009 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G Auth Sasa wrote: Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16
Re: [asterisk-users] Cisco 7941G Auth
What does your SEPMacAddress.cnf.xml file look like? In my experience, the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I had to specify the firmware version in each SEP file. I am using 8-4-4S, but for you this would be something like this: device loadInformationSIP41.8-0-2SR1S/loadInformation /device And you shouldn't need the tlv file. -Jonathan On Fri, Jun 19, 2009 at 8:25 AM, Sasa s...@shoponweb.it wrote: David Gibbons wrote: I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. I have tried with tftp on linux machine but the result isn't changed. Thanks. -- Salvatore. - Original Message - From: David Gibbons d...@videon-central.com To: novacks...@gmail.com; 'Asterisk Users MailingList - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, June 19, 2009 4:50 PM Subject: Re: [asterisk-users] Cisco 7941G Auth I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Friday, June 19, 2009 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G Auth Sasa wrote: Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:40.046] Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984] Read request for file Italy/g3-tones.xml. Mode octet [19/06 10:16:40.999] File Italy\g3-tones.xml : error 3 in system call CreateFile Impossibile trovare il percorso specificato. [19/06 10:16:40.999] Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843] Read request for file dialplan.xml. Mode octet [19/06 10:16:42.859] Using local port 3998 [19/06 10:16:42.859] dialplan.xml: sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06 10:16:42.906] In XMLDefault.cnf I have: loadInformation309 SIP41.8-0-2SR1S/loadInformation309 ..and on 7941G I have: App Load IDjar41sip.8-0-1-18.sbn Boot Load ID7941G_64-02070631Amd64megRel.bin VersionSIP41.8-0-2SR1S Thanks. -- Salvatore. I have had sucess with creating a zero length file named CTLSEPmac_address.tlv Or whatever the damn thing wants, and it then seems to be happy. With Cisco 7960's Your results may vary John Novack ___ -- 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 -- Dog is my co-pilot ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Voicemail Password
I believe that 'externpasscheck' was added in the 1.6 branch. Since we use this, I wrote a quick perl script that checks for password length, difficulty, repeated digits, etc. which are required for us. If you get it back-ported to the version you are on you can have the script, just contact me off-list. This is probably one of the best features added to app_voicemail for 1.6. -Jonathan On Thu, Jun 18, 2009 at 11:40 AM, Darrin Henshaw dhens...@ignition.bmwrote: As usual my manager comes up with some obscure reference I didn't find. There seems to be a parameter called minpassword described here: http://www.asterisk.org/doxygen/trunk/Config_vm.html But from further digging it looks like it's a 1.6.1.0 feature. Might see about a backport if possible. Cheers, Darrin Henshaw -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, June 18, 2009 15:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail Password On Thu, 18 Jun 2009, Darrin Henshaw wrote: Does anyone know of a way to force the voicemail password for users to be of a certain length? We've setup operator=yes within our voicemail.conf and want to have the users use a long password to prevent possible guessing by external parties. I'm not seeing any such option in my research. If it doesn't exist it might be a decent feature. Thanks. Sounds like a cool feature. I started looking into it, checking out voicemail.conf (1.2) to get an idea of a good name to call the parameter and I found this: ; If you need to have an external program, i.e. /usr/bin/myapp called when ; a voicemail password is changed, uncomment this: externpass=/usr/bin/myapp Who knew? Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- 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 ___ -- 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
Re: [asterisk-users] howto store local exchange prefixes ?
On Mon, May 25, 2009 at 2:58 PM, John Novack jnov...@stromberg-carlson.orgwrote: sean darcy wrote: The local telco is now going 10 digit dialing even for local (free) calls which used to be 7 digit. For a while no problem, everyone will continue to dial 7 digits, and I'll add the area code. But pretty soon everyone will become used to 10 digits. Lucky you. Other states require 11 digits for all calls, regardless, and yet others require 10 digit for local and 11 digit for toll, they way the NANP was SUPPOSED to evolve, until the inmates took over the asylum and each state ( in the US ) PUC sets the numbering plan and splits vs overlays. John Novack There are about 40 3 digit local exchanges. I'd like to store the exchanges in a database, and use the dialplan to check them. I can figure that out. Very lucky, we have 700 prefixes to check that are 10 digits on one some our trunks and 11 on others, and some that don't care either way! Right now I have a script that parses the prefixes and creates the dial plan in an #include file. Since the prefixes don't change the frequently, it seems to work. Assume that everything is 11 digits, then using a dialing macro, find an open trunk and strip the '1' if needed. Now my users never have to dial a 11, but it works if they do. I would welcome some ideas for a more elegant solution! So if cell phones never require 11 digits... The company line about NANP and consistancy: *We don't care.**We don't have to.**We're the phone company.*-Jonathan ___ -- 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
Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
From the front page ( http://wiki.centos.org/FrontPage ): *What is CentOS?* CentOS is an Enterprise Linux distribution based on the freely available sources from Red Hat Enterprise Linuxftp://ftp.redhat.com/pub/redhat/linux/enterprise/. Each CentOS version is supported for 7 years (by means of security updates). A new CentOS version is released every 2 years and each CentOS version is regularly updated (every 6 months) to support newer hardware. This results in a secure, low-maintenance, reliable, predictable and reproducible Linux environment. CentOS 4 ( http://wiki.centos.org/FAQ/CentOS4 ): We intend to support CentOS-4 updates until Feb 29, 2012 CentOS 5 ( http://wiki.centos.org/FAQ/CentOS5 ): We intend to support CentOS 5 until Mar 31st, 2014 So if you don't want major upgrades for a while you might want to go with the latest version. To put it into Microsoft terms... the minor version is like a service pack. So CentOS 4.7 is really a base lined version 4, service pack 7. You get the new features in major releases (like there are no more smp kernels in 5 to deal with) -Jonathan On Wed, May 20, 2009 at 2:36 PM, Jimmy Ezell jez...@hmhca.com wrote: On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote: multi-processor machine ( I had to remember to specify smp for the kernel) I repeat: why bother with such an old system? Really? Recall the comment from the book. That book had nothing really specific to Centos 4. Why do you shoot yourself in the foot by installing Centos4 now? (not to mention Zaptel) -- Tzafrir Cohen Tzafrir thanks for the comments. I am not done playing with this and in the end I may well use newer software as you suggest. According to wikipedia CentOS 4.7 was released OCT. 2008 (7 months ago) is that really consider that old? I am looking to setup a phone system that I would hope would not require any major software upgrades for many years. Jimmy ___ -- 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 ___ -- 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
Re: [asterisk-users] Cisco phone - can Call manager reflash automatically if we test in Asterisk with SIP?
When the phone is plugged back in to CallManager network, it should get handed a TFTP server via DHCP, and should automatically download the configuration from CallManager which includes what firmware to load. It should then reload the SCCP firmware (if you are not using SIP) and reboot back to how it was. All of this is assuming that you have a standard CallManager environment of course. -Jonathan On Mon, May 4, 2009 at 3:14 PM, David Shauger sollost...@gmail.com wrote: David, Will it happen automatically when you reconnect it to Cisco Call Manager or does it require additional steps? Thanks! On May 4, 2009, at 4:14 PM, David Gibbons wrote: Yes, you can flash them back and forth as you require. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] on Behalf Of David Shauger Sent: Monday, May 04, 2009 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco phone - can Call manager reflash automatically if we test in Asterisk with SIP? Anyone know if we take a Cisco phone off of a Call Manager system and flash it for SIP to demo on Asterisk, can we take it back to Cisco and Call Manager will remember its MAC address and reflash it back to what it is supposed to be? I would anticipate with Cisco Discovery Protocol this would be the case, but would like to be sure. Thanks! David Shauger Vice President Sollos Technology Solutions 678-317-9444 - voice 404-886-7603 - cell 772-679-5830 - fax d...@sollos.com http://www.sollos.com/ This email has been certified by Thawte Email certification helps prevent identity theft Virus scanning provided by Clam Antivirus ___ -- 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 ___ -- 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 ___ -- 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
