RE: [asterisk-users] Server redundancy
Asterisk realtime hardly provides redundancy. 1. There's no support for realtime SIP where multiple Asterisk systems can reference the same MySQL database for SIP peers. Ask Kevin Fleming about this. It's known not to work. 2. The IP address of the MySQL server is hard coded into the Asterisk config files. In the event of a database failure, Asterisk fails as well. You need to build redundancy into MySQL with a primary and seconday server, and something that can monitor MySQL system, network, and application and then transparently (to Asterisk, because it can't do it itself) switch IP's in the event of failure. 3. Other stuff I can't recollect right now because I am tired. -Original Message- From: RR [mailto:[EMAIL PROTECTED] Sent: Mon 7/10/2006 9:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] Server redundancy Alejandro, doesn't sound like you've read up or done research on ARA (Asterisk Realtime Architecture)? That's what allows you to build asterisk server clusters which draw upon configs either for individual config files OR entire family of processes froma common database (which can then be made redundant/clustered). All examples on the net are based on using MySQL But it's possible to use other drivers in conjunction with the ODBC engine that Asterisk uses to integrate oracle and/or MSSQL to store CDRs and Voicemail. These two can also be stored on a common clustered file system such as GFS or PVFS etc. So all in all, you can deploy ARA along with RedHat CSGFS (built from source, of course) and come up with a fully redundant realtime asterisk cluster. Hope this helps, Cheers \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Server redundancy
DUNDi doesn't provide good redundancy for phone registrations. Each phone is only registered on a single, primary Asterisk system. In the event that the primary system for a given phone becomes available, the phone will not re-register until it's registraiton expirey period, on it's secondary Asterisk system. During this time, the phone cannot be reached. You can cut the phone registration period right down, to some small period of time, say 5min, but is that acceptible? Also, keep in mind that the lower the registration period, the greater the number of registrations, and therefore the greater the network traffic, and hence, the load on each Asterisk system. Doug. -Original Message- From: RR [mailto:[EMAIL PROTECTED] Sent: Mon 7/10/2006 10:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] Server redundancy unplug, thanks for pointing that out as well as opposed to or in complement with ARA where you can either implement DUNDi between clusters of Asterisk servers or have a redundant pair of DUNDi lookup servers (just like DNS) somewhere remote to the local asterisk servers. DUNDi is a p2p IAX based protocol to allow for looking up contact information for a particular registered extension. So in ARA you can really store all extension based information in the common database for servers that are local to your network i.e. perhaps on the same private network or even the same VLAN. Then implement DUNDi between asterisk servers that are remote to your location and in their own private network using their own database. Not sure the level of reliability one can expect using the public internet for DUNDi look-ups from a server on the other end of the world but in theory it might be do-able. So if you can't find an extension within your local database, you perform the DUNDi lookup and find it in your remote servers. I'm sure the gurus on the list might have plenty to say on this :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 2 NICs; Asterisk receives on eth1 and replieson eth0
Yes, we tried to do the same thing. We wanted our Asterisk system to be multi-homed. Turned out to be a dissapointing limitation of Asterisk. It would have been nice to have, because then you could have multiple NIC's, have Asterisk listen on both, and if one failed, you had some degree of redundancy. But nope, no go. -Original Message- From: Daniel Lawson [mailto:[EMAIL PROTECTED] Sent: Mon 7/10/2006 11:21 PM To: kjcsb; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] 2 NICs; Asterisk receives on eth1 and replieson eth0 kjcsb wrote: I have an Asterisk server with 2 network cards. One provides the LAN connection and the other provides the Internet connection. Currently this is set up in the following way: eth0 192.168.1.5. This provides LAN connectivity eth1 192.168.1.251, gw 192.168.1.252 (Note that other nodes on the network use a different gateway, not 192.168.1.252). This provides the internet connection. The router is set up with DMZ enabled and pointing to 192.168.1.251. I am going to assume you are using a netmask of 255.255.255.0 above, as you haven't specified it. You can't have both interfaces being on the same network. This is why you are having this problem. I'd suggest making the Asterisk - upstream link inside a different network, such as 192.168.2.xxx. EG: eth0 192.168.1.25 LAN eth1 192.168.2.1 gw 192.168.2.254 set the router up with DMZ enabled and pointing to 192.168.2.1 I'd also suggest you connect eth1 on the asterisk box directly to the gateway, and don't plug them into your LAN switch. You may need to use a crossover cable to do this. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 NICs; Asterisk receives on eth1 and replies on eth0
On Tue, 2006-07-11 at 16:31 +1200, kjcsb wrote: I have an Asterisk server with 2 network cards. One provides the LAN connection and the other provides the Internet connection. Currently this is set up in the following way: eth0 192.168.1.5. This provides LAN connectivity eth1 192.168.1.251, gw 192.168.1.252 (Note that other nodes on the network use a different gateway, not 192.168.1.252). This provides the internet connection. The router is set up with DMZ enabled and pointing to 192.168.1.251. Two NICs in the same address space? Are you sure the rest of the system works as you think it does? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 NICs; Asterisk receives on eth1 and replieson eth0
Douglas Garstang wrote: Yes, we tried to do the same thing. We wanted our Asterisk system to be multi-homed. My head office Asterisk box is multi-homed: I have three networks across two NICs. One dedicated to hardphones, another to the local LAN (and PC-based softphones). The third network is bound to the same NIC as the LAN, but has different IP addressing. This links to our national VPN to connect to Asterisk boxes in other cities. All of the regional Asterisk boxes are also multi-homed. They have two IP addresses (sometimes on one NIC, sometimes on two). One connected to the local LAN, the other to the national VPN. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE: RE: [Asterisk-Users] Very bad quality withAVMFritz!cardPCIandchan_capi
So I've just had the time to swap and disable usb in my bios and it changed nothing the quality is still the same (which means horrible). How could I check where the problem comes from? Ben Hmmm... that's a shame. Apologies if you have already specified this, but what are the versions of chan_capi and the Linuux kernel? Have you tried mISDN? (again, apologies if you've already mentioned this). James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server redundancy
Interesting points on both messages 1) as far as multiple asterisk servers talking to the same database is concerned, I will have to test this out. I know nothing about the database side of things, and a newbie on asterisk and linux so I have no idea what and where the development of either of these are. From your message it sounds like it's just how ARA is designed because I doubt it's to do with the ODBC driver itself. This will cause me a lot of grief if you're right about this for multiple * servers to not be able to access the same database for peer lookup. 2) Clustering of DB isn't an issue, not for me at least. Haven't tested this either but my DBs are clustered A/P providing a single entity to the internal systems. Might further look into a local DNS lookup to add to this. I believe it's possible to do this in the MySQL world with MySQL grid etc? 3) I don't believe frequent registration is that big of an issue for the network load it generates. Most providers out there set devices for a 30-40secs Reg. Refresh to support NAT'ed endpoints and the a reg refresh is hardly about 300-400Byte pkts (I think). The math doesn't add up for a major load esp. if you've got a load balancing mechanism in front of your * boxes. 4) I don't know enough about DUNDi to get into this discussion but DUNDi just lookup extensions? or it also have any part to play in registrations? If they just do extension lookup, then If DUNDi is implemented on an A/P pair of dedicated DUNDi lookup servers which access a clustered database, then barring #1 being true, each * server accesses the same database and pool of registrations. If registrations are refreshed frequently enough, the contact info in the database will always be current and one server dying won't affect anything. At the same time, they just consult the DUNDi lookup server for extension lookups instead of asking the database directly. 5) If you really want to improve on this, supplement your network with SER as proxies and have them deal with Registrations and load-balance feature requests to * servers etc. Once * has done whatever it needs to do (e.g. provide PBX features, voicemail, conference, IVR etc.) it passes the call back to the Proxy to deal with the endpoints. All depends on your scope and budget. If you want to have a SP grade service then you need to breakout your functions. I just hope #1 isn't true though. The only alternative then would be to have /etc/asterisk reside on an NFS share or a CFS for all servers to read massively huge conf files if you're catering for large number of endpoints. Dunno if it helps anyone or I'm just shooting sh*t ;) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Servers problem?
Hello Iam trying to communicate between two asterisk servers using IAX protocol. Details are SERVER 1 IP address-192.168.0.54 Clients 949 and 950 SERVER 2 IP address-192.168.0.11 Clients 449 and 450 When i tried to dial 949 from server 2 its working but when i tried to dial 449 from server 1 it is gving error like AT SERVER 1:*CLI Jul 11 12:12:24 WARNING[4832]: chan_iax2.c:6995 socket_read: Call rejected by 192.168.0.11: No authority found *CLI Jul 11 12:12:24 WARNING[4832]: chan_iax2.c:6995 socket_read: Call rejected by 192.168.0.11: No authority found *CLI Jul 11 12:12:24 WARNING[4832]: chan_iax2.c:6995 socket_read: Call rejected by 192.168.0.11: No authority found *CLI Jul 11 12:12:24 WARNING[4832]: chan_iax2.c:6995 socket_read: Call rejected by 192.168.0.11: No authority found AT SERVER 2: CLI Jul 11 12:17:18 NOTICE[4914]: chan_iax2.c:6802 socket_read: Rejected connect attempt from 192.168.0.54, who was trying to reach '[EMAIL PROTECTED]' Can you please suggest me what is the problem Thanks in advance Find out what India is talking about on Yahoo! Answers India. So, whats NEW about the NEW Yahoo! Messenger? Find out.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialing timeouts
Doug Lytle wrote: Dan Elder wrote: Hey All, probably missing something really obvious here, but when our users are trying to dial the phone, asterisk timesout really quickly if they don't press the digits fast enough. Is there a global timeout value for dialing See: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DigitTimeout And http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ResponseTimeout These two apps are only for IVR stuff. The timeouts for dialing a call are normally handled by the device. i.e. the SIP phone or ATA, or the zaptel code. For Zaptel see this: /path/to/src/asterisk/channels/chan_zap.c: /*! \brief Wait up to 16 seconds for first digit (FXO logic) */ /* static int firstdigittimeout = 16000; */ static int firstdigittimeout = 2; /*! \brief How long to wait for following digits (FXO logic) */ /* static int gendigittimeout = 8000; */ static int gendigittimeout = 2; -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text priority labels not working for me
Wes Santee wrote:
Greetings all,
I'm on 1.2.9.1, and I'm trying to get a dialplan working that uses text
labels, but it's not working. For instance, take the following macro
snippet:
[macro-dosomething]
exten = s,1,GotoIf($[${MACRO_EXTEN:1:1} != 1] ? scid)
exten = s,n,Set(MACRO_EXTEN=1${MACRO_EXTEN})
exten = s,n(scid),SetCallerId(${MY_CID})
exten = s,n,Dial(...)
When I call this macro, I get the following:
-- Executing Macro(SIP/1000-66b0, dosomething) in new stack
-- Executing GotoIf(SIP/1000-66b0, 1 ? scid) in new stack
Jul 10 20:05:52 NOTICE[99803]: pbx.c:1753 pbx_extension_helper: No such
label ' scid' in extension 's' in context 'macro-dosomething'
Jul 10 20:05:52 WARNING[99803]: pbx.c:6514 ast_parseable_goto: Priority
' scid' must be a number 0, or valid label
The last log line suggests I can't use labels, but according to
http://www.voip-info.org/wiki/index.php?page=Asterisk+priorities it
shouldn't be a problem.
Am I doing something wrong?
Don't put spaces around the ?
--
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Chattanooga, and Montgomery.
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Re: [asterisk-users] 2 NICs; Asterisk receives on eth1 and replies on eth0
Two different things I think. For redundancy, if you're using RHEL4U2 or later, you can bond your ethernet channels and configure any of the 6 modes AA, AP, ALB etc. you'll get NIC redundancy. Anyone of the NICs die, the other one takes over the MAC and the IP of the failed NIC and the system is always accessible on that same IP. That should solve your problem. Now with the original question, The original question wasn't redundancy, it was physical separation of traffic flow between the two NICs, which has a side-benefit of redundancy but not really since if eth0 dies, no SIP Service! This is not necessarily a limitation of asterisk but a routing thing. If you do a netstat -r, it'll show you eth0 before eth1, which is what the network stack will then use to send your packets over no matter where it's coming from if both NICs are on the same subnet. You may have to either create a manual entry in the routing table. Will have think about it more in terms of what the exact table would look like but if you have a different gateway for your internal traffic, having a static route for that along with some netmask manipulation might sort your problems. But like other people have suggested, you're MUCH better off by having a different subnet for your 2nd NIC. IF you can't, then just bond your NICs and set them to mode 6 where the OS will load balance your tx/rx traffic between the two NICs and you'll still only have one address that you can assign to the DMZ. Hope this helps. \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Servers problem?
You sure the iax.conf are almost identifcal in both the servers so the two servers are maybe listed as friend and not peer or user? I don't know if that'll fix it, but it's a stab in the dark. Don't know why one would work and the other wouldn't. I also haven't looked at your error messages but will leave it to the experts, if this isn't the issue :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip_poke_noanswer: Peer xxx is now unreachable
I was getting this message throughout yesterday in repitition, anyone experienced this before and what is the best solution? Jul 10 12:19:03 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Last qualify: 124Jul 10 12:19:03 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4002' is now UNREACHABLE! Last qualify: 120Jul 10 12:19:05 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4003' is now UNREACHABLE! Last qualify: 117Jul 10 12:19:07 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4004' is now UNREACHABLE! Last qualify: 116Jul 10 12:19:08 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4005' is now UNREACHABLE! Last qualify: 124Jul 10 12:19:08 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4006' is now UNREACHABLE! Last qualify: 131Jul 10 12:19:08 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4006' is now UNREACHABLE! Last qualify: 130Jul 10 12:19:10 NOTICE[13020]: chan_sip.c:9700 handle_response_peerpoke: Peer '4007' is now TOO LAGGED! (2142 ms / 2000ms)Jul 10 12:19:13 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4001' is now REACHABLE! (112ms / 2000ms)Jul 10 12:19:13 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4002' is now REACHABLE! (120ms / 2000ms)Jul 10 12:19:15 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4003' is now REACHABLE! (118ms / 2000ms)Jul 10 12:19:18 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4004' is now REACHABLE! (118ms / 2000ms)Jul 10 12:19:18 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4005' is now REACHABLE! (114ms / 2000ms)Jul 10 12:19:18 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4006' is now REACHABLE! (139ms / 2000ms)Jul 10 12:19:18 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4007' is now REACHABLE! (136ms / 2000ms)Jul 10 12:19:20 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4008' is now REACHABLE! (142ms / 2 000ms) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE: RE: [Asterisk-Users] Very bad quality withAVMFritz!cardPCIandchan_capi
On Tue, 2006-07-11 at 16:38 +1000, James Harper wrote: So I've just had the time to swap and disable usb in my bios and it changed nothing the quality is still the same (which means horrible). How could I check where the problem comes from? I had absolutely awful sound on my AVM Fritz! with chan_capi until someone pointed me in the direction of the codec setting in capi.conf ;ulaw=yes;set this, if you live in u-law world instead of a-law I thought I am in the u-law world but evidently I am not. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail
Dear I am using trixbox,I want ot disable and enable voicemail from command line At [EMAIL PROTECTED] v 2.8 I was using this command and was working successfully Database put AMPUSER/9990999 voicemail default And Database put AMPUSER.9990999 voicemail disables But at trixbox its not working Any ideas pleas Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SRTP or zrtp
Is SRTP or zrtp available in asterisk? Or how to implement it ? am using trixbox Please if you know send me full configuration I will be thanks full Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server redundancy
I have asked about it here. As Douglas said, it doesn't support mult-asterisk in current version. However, I have questions about why multi-asterisk so difficult to implement. 1. As we can use ARA to store all information, sip user register info, dial plan ... to DB. All asterisks can use ARA to refer to the DB for necessary information even register information. 2. What is mean by multiple Asterisk systems can't reference the same MySQL database for SIP peers.? Does SIP peer information also store in DB? 3. Any difficulty to implement multiple asterisk? 4. If I want to implement multiple asterisk in some extent, how do I begin? Any reference? On 7/11/06, RR [EMAIL PROTECTED] wrote: Interesting points on both messages 1) as far as multiple asterisk servers talking to the same database is concerned, I will have to test this out. I know nothing about the database side of things, and a newbie on asterisk and linux so I have no idea what and where the development of either of these are. From your message it sounds like it's just how ARA is designed because I doubt it's to do with the ODBC driver itself. This will cause me a lot of grief if you're right about this for multiple * servers to not be able to access the same database for peer lookup. 2) Clustering of DB isn't an issue, not for me at least. Haven't tested this either but my DBs are clustered A/P providing a single entity to the internal systems. Might further look into a local DNS lookup to add to this. I believe it's possible to do this in the MySQL world with MySQL grid etc? 3) I don't believe frequent registration is that big of an issue for the network load it generates. Most providers out there set devices for a 30-40secs Reg. Refresh to support NAT'ed endpoints and the a reg refresh is hardly about 300-400Byte pkts (I think). The math doesn't add up for a major load esp. if you've got a load balancing mechanism in front of your * boxes. 4) I don't know enough about DUNDi to get into this discussion but DUNDi just lookup extensions? or it also have any part to play in registrations? If they just do extension lookup, then If DUNDi is implemented on an A/P pair of dedicated DUNDi lookup servers which access a clustered database, then barring #1 being true, each * server accesses the same database and pool of registrations. If registrations are refreshed frequently enough, the contact info in the database will always be current and one server dying won't affect anything. At the same time, they just consult the DUNDi lookup server for extension lookups instead of asking the database directly. 5) If you really want to improve on this, supplement your network with SER as proxies and have them deal with Registrations and load-balance feature requests to * servers etc. Once * has done whatever it needs to do (e.g. provide PBX features, voicemail, conference, IVR etc.) it passes the call back to the Proxy to deal with the endpoints. All depends on your scope and budget. If you want to have a SP grade service then you need to breakout your functions. I just hope #1 isn't true though. The only alternative then would be to have /etc/asterisk reside on an NFS share or a CFS for all servers to read massively huge conf files if you're catering for large number of endpoints. Dunno if it helps anyone or I'm just shooting sh*t ;) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem of Quality
Hello, Sometimes, when i call an outside people, he said me that the communication is bad: The voice is low, far, bad poor quality. How can i know where is the problem, which tests can i make? Best regards, -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IVR DTMF
Dear I want to make billing recharge through receiving digits from IVR through dtmf and store it on a text file , How can I do that ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with JIAXClient
I have already get a register, but I can't make a call.I had to setup a listener in order to get the register, but once the register is set I can't make a call in any way.Any hint with that??Thx in advance.Richard OSS [EMAIL PROTECTED] escribió: I think you have to set where to get the libraries (jiaxc*.jar files).Setup a webserver somewhere and put the jar files there. Then in your code before initialize client.setCodeBase("your URL to the jar files");HTH,richard Enrique Sanchez [EMAIL PROTECTED] wrote:I'm trying to make a little example programfor register to an Asterisk PBX and dial a softphone, but i just can't register to the PBX.package iax; import net.sourceforge.iaxclient.Call;import net.sourceforge.iaxclient.JIAXClient;import net.sourceforge.iaxclient.Registration; public class TestIAX { public static void main(String[] args) { Registration registration; JIAXClient client = JIAXClient.getInstance(); client.initialize (1, 10); registration = client.register("kike", "elkike", "10.32.81.31:4569"); client.setCallerID("Kike", "1001"); client.call("1002"); System.out.println(registration); }} I'm frustrated because JIAX doesn't throw any exception, but the code is not working properly. Greetings, -- Enrique Sanchez ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options
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LLama Gratis a cualquier PC del Mundo.Llamadas a fijos y móviles desde 1 céntimo por minuto.http://es.voice.yahoo.com___
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Re: RE: [asterisk-users] Asterisk with ISDN Fritz PCI card
Hi Ben, Thank you for the help: it works. It seems that the dirty thinks works better that the clean one ;-). bests regards, Guy. At 11:15 10.07.2006 +0200, you wrote: Hi, Maybe this is dirty but this is how I did it (with capi but you can probably do it with anything you want): ***Suppress the Hisax drivers in conflict with capi: [EMAIL PROTECTED]:~# mv /lib/modules/2.6.12-9-386/kernel/drivers/isdn/hisax/hisax.ko /lib/modules/2.6.12-9-386/kernel/drivers/isdn/hisax/hisax.ko.old [EMAIL PROTECTED]:~# mv /lib/modules/2.6.12-9-386/kernel/drivers/isdn/hisax/hisax.fcpcipnp.ko /lib/modules/2.6.12-9-386/kernel/drivers/isdn/hisax/hisax.fcpcipnp.ko.old [EMAIL PROTECTED]:~# mv /lib/modules/2.6.12-9-386/kernel/drivers/isdn/i4l/isdn.ko /lib/modules/2.6.12-9-386/kernel/drivers/isdn/i4l/isdn.ko.old [EMAIL PROTECTED]:~# mv /lib/modules/2.6.12-9-386/kernel/drivers/isdn/i4l/isdn_bsdcomp.ko /lib/modules/2.6.12-9-386/kernel/drivers/isdn/i4l/isdn_bsdcomp.ko.old ***Download http://www.avm.de/ftp/cardware/fritzcrd.pci/linux and make, make install ***move the newly created modules to the good place from /lib/modules/2.6.12-10-386/extra/ to /lib/modules/2.6.12-10-386/kernel/drivers *** add capi and fcpci to /etc/modules (now when you reboot your machine the modules are loaded) then ***apt-get the libraries for capi # apt-get install libcapi20-dev ***download chan_capi on ftp://ftp.chan-capi.org/chan-capi and make,make install, make install_config but this probably works with misdn or anything else. Tell me if this works or if it doesn't (I'm on ubuntu not debian but this should be almost the same) Good luck, Ben - Original Message - From: Guy Corbaz [EMAIL PROTECTED] Date: Sunday, July 9, 2006 2:03 pm Subject: RE: [asterisk-users] Asterisk with ISDN Fritz PCI card Hi, Thank you for the suggestion. I tried to use mISDN first, then CAPI and now I'm trying I4L. As I'm using Debian, I can not load the FRITZ drivers. I got the source from the official site and recompiled it, but there is a strange message in the log and the capi drivers are not loaded. The problem is more linked to drivers that Asterisk. If you have any tips to get this up and running, I would be very happy as my search on the Internet didn't allowed me to solve that issue. Bests regards, Guy. At 11:25 09.07.2006 +1000, you wrote: What are you using (misdn, capi, something else?) and what problems are you having? I submitted a patch recently to mISDN which should have fixed a problem on hangup, if that's the problem you are having then try the latest cvs mqueue branch of mISDN. James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk- users- [EMAIL PROTECTED] On Behalf Of Guy Corbaz Sent: Saturday, 8 July 2006 23:59 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk with ISDN Fritz PCI card Dear all, I'm desperately trying to get Asterisk working with a FRITZ PCI card on Debian with kernel 2.6.15. I'm wondering if anybody has such a working installation. Thank you for your help, Guy. Guy Corbaz ch. du Châtaignier 2 1052 Le Mont Switzerland phone:+41 21 652 26 05 mobile: +41 79 420 26 06 e-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Guy Corbaz ch. du Châtaignier 2 1052 Le Mont Switzerland phone:+41 21 652 26 05 mobile: +41 79 420 26 06 e-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Guy Corbaz ch. du Châtaignier 2 1052 Le Mont Switzerland phone:+41 21 652 26 05 mobile: +41 79 420 26 06 e-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Certain fax types cause problems
Feedback on my testing here. On 7/10/06, Doug Lytle [EMAIL PROTECTED] wrote: Steve Davies wrote: Hi, I was wondering whether anyone has any input into the reliability of faxing (over a PRI) using spandsp and rxfax. 99% of times this is a reliable combination - we use it almost exclusively, but there seem to be certain fax devices which have problems talking to us. Most notably fax modems, and a couple of HP multi-function devices. SpanDSP does very little or no error checking, when an error is encountered, it will fail. Hmmm... I just switched to iaxmodem on a loopback interface as documented, and tried both Hylafax and efax. Both seemed to drop out on any fax larger than 3 pages. When I switched back to app_rxfax, I could receive any length of fax once again. It is also easier to integrate Asterisk and rxfax as asterisk can control where the fax is saved, and what is done to it based on the call that occurred (eg. CLID and dialled number) Also, Hylafax is BIG, especially by the time ghostscript and its prerequisites are installed... Any pointers on how to diagnose or improve this would be appreciated. Install HylaFAX and iaxmodem on your Asterisk box. iaxmodem does use an updated version of spandsp - perhaps that is the source of my problems in this particular case. What I would LOVE is a version of app_rxfax that works with spandsp-0.0.3 and asterisk 1.0.x so I could pin this down further - Perhaps I'll have to backport the test version of rxfax myself. Thanks for the pointers. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Certain fax types cause problems
Steve Davies wrote: Feedback on my testing here. documented, and tried both Hylafax and efax. Both seemed to drop out on any fax larger than 3 pages. I frequently receive 50+ pages. asterisk can control where the fax is saved, and what is done to it based on the call that occurred (eg. CLID and dialled number) This is all controlled via the FaxDispatch script. Works very well. Also, Hylafax is BIG, especially by the time ghostscript and its prerequisites are installed... True. Any pointers on how to diagnose or improve this would be appreciated. Install HylaFAX and iaxmodem on your Asterisk box. I would suggest that you post your questions to the HylaFAX mailing list, Lee is very responsive. iaxmodem does use an updated version of spandsp - perhaps that is the source of my problems in this particular case. What I would LOVE is a version of app_rxfax that works with spandsp-0.0.3 and asterisk 1.0.x Steve Underwood has stated that version 0.0.3 is for developers only (At this time). -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone out there using Junghanns ISDNguard?
If so can you comment on how well it has (or hasn't) worked for you? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WG: CDR ist getting wrong status
Hi, It seems the cdr modul always put ANSWERED Status into accounting table, even if it is not answered: Jul 11 12:29:47 DEBUG[18722] app_dial.c: Exiting with DIALSTATUS=CANCEL. Jul 11 12:29:47 VERBOSE[18722] logger.c: == Spawn extension (macro-call-cisco, s, 5) exited non-zero on 'SIP/1000131-093bd318' in macro 'call-cisco' Jul 11 12:29:47 VERBOSE[18722] logger.c: == Spawn extension (macro-call-cisco, s, 5) exited non-zero on 'SIP/1000131-093bd318' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '4989xxx 31' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '31' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '089...' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '10001' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is 'SIP/1000131-093bd318' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is 'SIP/x.x.x.x-093cf108' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is 'Dial' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is 'SIP/[EMAIL PROTECTED]|60' mailto:'SIP/[EMAIL PROTECTED]|60' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '2006-07-11 12:29:41' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '2006-07-11 12:29:41' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '2006-07-11 12:29:47' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '6' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '6' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is 'ANSWERED' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is 'DOCUMENTATION' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '146' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '1152613781.34' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is 'EXTERN_OUTGOING' ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functionserror message
I have recompiled with mpg123 and music on hold is working fine. But the asterisk will not compile the meetme app, using the release 30432. Is there any way to compile the app manually? -Original Message- From: Dean @ INKnBITs [mailto:[EMAIL PROTECTED] Sent: 10 July 2006 13:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functionserror message Is this correct: zaptel: make clean; make; make install asterisk: make clean; make; make install Will this recompile everything needed? I tried, but the meetme app still does not get compiled (and no music) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of BJ Weschke Sent: 10 July 2006 13:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functionserror message You may need to recompile now that you've got zaptel/ztdummy installed so that your install sees that the proper zaptel exists now. On 7/10/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: After using the trunk versions as below, it all compiled ok, and the polycom acd is working great, but the music on hold and meetme will now work. I do not have any digium cards, is the ztdummy installed with the truck version? Or is there any thing I need to change? Thanks, Dean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of BJ Weschke Sent: 04 July 2006 15:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functionserror message On 7/4/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: I have installed libpri 1.2.3 and zaptel 1.2.6 (with make clean, make, make install), there was no errors. I used svn to get the polycom_acd_functions asterisk branch release 30432, I have to run make 3 times as it as it comes up with making opts re-run make. It then completes and I run make install, and get the following error message. chan_zap.c:73:2: #error You need newer libpri chan_zap.c:113:2: #error Your zaptel is too old. please update Does anybody know why I'm getting these error message, as I have the newest versions of both? You need the /trunk versions of libpri and zaptel instead of the branches/1.2 releases. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Provider UNREACHABLE
Hi All I am repeatedly getting a UNREACHABLE and then REACHABLE about 10 sec apart most of the time and then sometimes for about 45 - 74 minutes I have tried a reload and sip reload but neither bring the provider back ? What else could I try and how do I prevent this Thanks in advance Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Provider UNREACHABLE
Teliax ? I'm seeing the same. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop Sent: Tuesday, July 11, 2006 7:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Provider UNREACHABLE Hi All I am repeatedly getting a UNREACHABLE and then REACHABLE about 10 sec apart most of the time and then sometimes for about 45 - 74 minutes I have tried a reload and sip reload but neither bring the provider back ? What else could I try and how do I prevent this Thanks in advance Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Having trouble to receive fax from samsung sf3200
I have following issue with receiving fax from Samsung sf2000 fax machine. Faxes with other machine works OK; I am running asterisk 1.0.10 with spandsp-0.0.2pre26 on zaptel 1.0.10. The following is the log Jun 26 09:20:03 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Jun 26 09:20:03 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Changed from phase 1 to 4 Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW DIS: Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: 80 Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: 00 Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: ce Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: f4 Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: 80 Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: 80 Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: 81 Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: 80 Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: 80 Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: 80 Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: 18 Jun 26 09:20:06 DEBUG[14882]: app_rxfax.c:70 span_message: Jun 26 09:20:06 DEBUG[14882]: chan_zap.c:4059 zt_read: DTMF digit: f on Zap/1-1 Jun 26 09:20:06 DEBUG[14882]: chan_zap.c:4101 zt_read: Fax already handled Jun 26 09:20:07 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW HDLC underflow in state 9 Jun 26 09:20:07 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Changed from phase 4 to 3 Jun 26 09:20:07 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Jun 26 09:20:07 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW HDLC framing OK Jun 26 09:20:07 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Jun 26 09:20:08 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Jun 26 09:20:08 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW HDLC framing OK Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW TSI: Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 43 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 20 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 20 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 20 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 20 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 20 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 20 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 20 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 20 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 20 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 38 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 30 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 30 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 30 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 35 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 32 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 34 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 31 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 37 Jun 26 09:20:09 DEBUG[14882]:app_rxfax.c:70 span_message: 38 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: 30 Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW TSI without final frame tag Jun 26 09:20:09 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Remote fax gave TSI as: Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW DCS: Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: 83 Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: 00 Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: 86 Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: a0 Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: 80 Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: 80 Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: 00 Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW DCS with final frame tag Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW In state 9 Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Get at 9600bps, modem 1 Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Changed from phase 3 to 5 Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM carrier up Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM carrier
Re: [asterisk-users] Having trouble to receive fax from samsung sf3200
On 7/11/06, Muhammad Zaka [EMAIL PROTECTED] wrote: I have following issue with receiving fax from Samsung sf2000 fax machine. Faxes with other machine works OK; I am running asterisk 1.0.10 with spandsp-0.0.2pre26 on zaptel 1.0.10. The following is the log [snip FLOW trace] Please can you help me what is wrong. I have to add a me too here I am afraid. In my case it is a HP OfficeJet 7310, but the trace is almost identical. I am using very similar versions of the asterisk codebase too. Cheers, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] stuck/phantom zap channels
Hello, Using 1.2.9.1 with bristuff and a QuadBRI card, phantom/zombie channels accumulate throughout the day and end up blocking all incoming calls. It's the first time we have this problem and several similar installations work fine. We suspect bad cabling between the telco and the QuadBRI card. Has anyone dealt with this before? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New Asterisk server crashes daily
Hi -This is the first Linux server I have ever built with an installation of [EMAIL PROTECTED] 2.7. For development I have been running on VMWare on an XP box and sustained no crashes or reboots. After moving Asterisk to it's own server I am experiencing daily crashses (around 4am) and I'm not quite sure what the problem is, nor am I sure where exactly to look for logs of any errors prior and during the crash. During the crash there should be nothing running so I'm not sure why it crashes at this time (perhaps some system job that is running at this time?).My hardware is: AMD Athlon 64bit 3200 CPU, 1 gig memory, 100gb hd and a gigabit NIC card. The BIOS is set with defaults.Many thanks, Al. Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tired of fax calls... :-/
Zeeshan Zakaria wrote: Is NVFaxDetect for PSTN calls or works for SIP/IAX as well? Works for SIP/IAX, for PSTN you only need to switch on faxdetect in zapata.conf and have Answer() in that part of your dialplan. On 7/9/06, *Thomas Kenyon* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Olivier wrote: 2006/7/6, Maxim Vexler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: NVFaxDetect does just that ;) Do you think NVFaxDetect is reliable ? Could you use it along a voicemail (I mean : someone having a single extension for voice and fax call, forward all incoming calls to its voicemail when leaving the office) Cheers If you use NVBackgroundDetect(not-here-greeting) then VoiceMail(sBOX) (s means no announcement), It appears to work. You will need a [fax] context to handle the fax. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk server crashes daily
On Tue, Jul 11, 2006 at 06:23:05AM -0700, Al Lougher wrote: Hi - This is the first Linux server I have ever built with an installation of [EMAIL PROTECTED] 2.7. For development I have been running on VMWare on an XP box and sustained no crashes or reboots. After moving Asterisk to it's own server I am experiencing daily crashses (around 4am) Daily crons? log rotation not done well? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk server crashes daily
Al, try /var/log/asterisk/full that's where asterisk typically stores its logs. Might be a good place to start to read into what's going on. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk server crashes daily
On Tue, Jul 11, 2006 at 06:23:05AM -0700, Al Lougher scribbled: Hi - This is the first Linux server I have ever built with an installation of [EMAIL PROTECTED] 2.7. For development I have been running on VMWare on an XP box and sustained no crashes or reboots. After moving Asterisk to it's own server I am experiencing daily crashses (around 4am) and I'm not quite sure what the problem is, nor am I sure where exactly to look for logs of any errors prior and during the crash. During the crash there should be nothing running so I'm not sure why it crashes at this time (perhaps some system job that is running at this time?). My hardware is: AMD Athlon 64bit 3200 CPU, 1 gig memory, 100gb hd and a gigabit NIC card. The BIOS is set with defaults. Many thanks, Al. Your comment about it happening around 4am leads me to think it might be the default daily-scheduled cron jobs somehow affecting you. Are you sure you have enough swap space configured? Roshan -- http://roshan.info Being normal is driving me crazy. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rate or rank ITSP
Hi There I know of wiki there is a list of VOIP providers, but is there a list or can we create / suggest one that will list VoIP providers, their location and quality of service ? Too me this will be very valuable, plus looking at some of the requests of late I'm sure others would like that too? Thanks Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk server crashes daily
Al Lougher wrote: Hi - This is the first Linux server I have ever built with an installation of [EMAIL PROTECTED] 2.7 mailto:[EMAIL PROTECTED]. For development I have been running on VMWare on an XP box and sustained no crashes or reboots. After moving Asterisk to it's own server I am experiencing daily crashses (around 4am) and I'm not quite sure what the problem is, nor am I sure where exactly to look for logs of any errors prior and during the crash. During the crash there should be nothing running so I'm not sure why it crashes at this time (perhaps some system job that is running at this time?). My hardware is: AMD Athlon 64bit 3200 CPU, 1 gig memory, 100gb hd and a gigabit NIC card. The BIOS is set with defaults. Many thanks, Al. Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta. http://us.rd.yahoo.com/evt=42297/*http://advision.webevents.yahoo.com/handraisers Al, Sounds like a possible memory issue. Memtest might help diagnose. You'll probably need a bootable CD such as the Fedora rescue CD. Otherwise, take a look in /var/log for the system logs. the main log is messages. The asterisk logs are in /var/log/asterisk. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy MOH (Cisco gateway)
VAD option on Cisco Gateways maube are causing this. Check This http://www.cisco.com/warp/public/788/voice-qos/hissing.html#topic3 Its a feature , to have Zeeshan Zakaria escribió: My service provider had issue with his Cisco hardware when it came to MoH. They were new with Asterisk at that time. I told them many times that they had problem in their system, but they never agreed, until one day when one of their engineers figured out that the Cisco hardware was compressing the MoH data to conserve bandwidth, causing choppy MoH. That was some simple feature which he switched off and I didn't have MoH problem after that. I am not a Cisco expert, but those who are, may know what I am talking about. Zeeshan A Zakaria On 7/10/06, Bill Gibbs [EMAIL PROTECTED] wrote: Yes that is correct. Bill -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Monday, July 10, 2006 12:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway) On Jul 10, 2006, at 4:49 AM, Bill Gibbs wrote: And of course I just found this article http://www.cisco.com/warp/public/788/voice-qos/hissing.html#topic3 Hope this helps some other people out as well! So was the fix to reconfigure your gateway to notuse VAD? Just want to be clear... Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Yet another problem with incoming SIP calls and 407
Hi all, when I receive incoming SIP calls on my Asterisk (1.2.9.1) where the caller has a username in it's From-Address which also exists in my sip.conf then my system answers with 407 Proxy Authentication Required. If it's nonexistent username then callin works fine! It seems that this is a problem in the SIP implementation of Asterisk and found a few hints on how to resolve this (allowguest=yes, insecure=invite,port etc.). But none of them does help! Can anyone suggest what I else could try? Thanks, Wolfgang ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Certain fax types cause problems
Steve Davies wrote: Hmmm... I just switched to iaxmodem on a loopback interface as documented, and tried both Hylafax and efax. Both seemed to drop out on any fax larger than 3 pages. Then it's time to consider that your problems are not with spandsp, iaxmodem, or HylaFAX. My suggestion would be to continue using iaxmodem and begin working with us at the iaxmodem users list with some HylaFAX session logs. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Server Optimization and Load Balancing
I'm hoping to get some guidance on some of our asterisk growing pains. Any help is greatly appreciated.Over the last few months, our call center has grown considerably and we're now experiencing choppy calls and dropped calls under full capacity. We have around a 60/40 outgoing to incoming ratioAt the moment, we've got the following configuration:Asterisk SVN-trunk-r7230All calls recorded to diskExternal mysql server for CDR + IVR operations Dual Xeon 2.8 4GB Ram (Dell)Digium TE210P2 x PRI lines72 Ploycom 301P SIP phones using ulaw codecWe have a second identical server ready to offset some of the load, but we're not sure how to balance the sip phones and configuration files between the two servers. If we balance the sip registrations between the two servers, then there's the issues of both servers having to handle one call via IAX in some situations. What kind of experiences, problems and solutions have y'all had when adding servers to your center?Should we try to have incoming on one server and outgoing on the other?Should we have both servers capable of handling all the IVR operations, so the other server doesn't have to? Should we try to have an identical configuration between both servers and load balance?What kind of general optimizations should we look at to improve network / server performance?Is there a way to easily register each phone with all asterisk servers, and have the phone choose a random server to dial, and then be available as a SIP to each server if it needs to contact it? Is it a bad idea to register all phones with each server instead of distributing registration?-Here's some of the things we're got planned in the next few days:-Make sure we have all audio files in all codec formats to reduce the need for transcoding in IVR -Convert all music on hold from mp3 to native codec formats-Reduce database operations from within extensions.conf-Upgrade switches on each set of desks to midrange enterprise 100MB switches with gigabit uplinks, from SOHO netgear 100MB switches Thanks,/mitch/fidelity reserves ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING[30954]: chan_sip.c:2734 sip_indicate: Don't know how to indicate condition 9
Hi, is there anybody who knows what the following warnings mean? WARNING[30954]: chan_sip.c:2734 sip_indicate: Don't know how to indicate condition 9 WARNING[30954]: channel.c:2051 ast_indicate: Unable to handle indication 9 for 'SIP/8-3c6e' TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Server Optimization and Load Balancing
Keep us posted! You have a good real world load with some decent horsepower behind it so it will be interesting to see how your temporary changes you have planned in the next few days pan outI suspect the SOHO switches could be part of the problem. What is the load on the server? Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mitch Jackson Sent: Tuesday, July 11, 2006 10:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Server Optimization and Load Balancing I'm hoping to get some guidance on some of our asterisk growing pains. Any help is greatly appreciated. Over the last few months, our call center has grown considerably and we're now experiencing choppy calls and dropped calls under full capacity. We have around a 60/40 outgoing to incoming ratio At the moment, we've got the following configuration: Asterisk SVN-trunk-r7230 All calls recorded to disk External mysql server for CDR + IVR operations Dual Xeon 2.8 4GB Ram (Dell) Digium TE210P 2 x PRI lines 72 Ploycom 301P SIP phones using ulaw codec We have a second identical server ready to offset some of the load, but we're not sure how to balance the sip phones and configuration files between the two servers. If we balance the sip registrations between the two servers, then there's the issues of both servers having to handle one call via IAX in some situations. What kind of experiences, problems and solutions have y'all had when adding servers to your center? Should we try to have incoming on one server and outgoing on the other? Should we have both servers capable of handling all the IVR operations, so the other server doesn't have to? Should we try to have an identical configuration between both servers and load balance? What kind of general optimizations should we look at to improve network / server performance? Is there a way to easily register each phone with all asterisk servers, and have the phone choose a random server to dial, and then be available as a SIP to each server if it needs to contact it? Is it a bad idea to register all phones with each server instead of distributing registration? -Here's some of the things we're got planned in the next few days: -Make sure we have all audio files in all codec formats to reduce the need for transcoding in IVR -Convert all music on hold from mp3 to native codec formats -Reduce database operations from within extensions.conf -Upgrade switches on each set of desks to midrange enterprise 100MB switches with gigabit uplinks, from SOHO netgear 100MB switches Thanks, /mitch /fidelity reserves ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Optimization and Load Balancing
I'm 100% sure that your problem is the call recording. Disable it for some minutes to see for yourself. Zoa Bill Gibbs wrote: Keep us posted! You have a good real world load with some decent horsepower behind it so it will be interesting to see how your temporary changes you have planned in the next few days pan out…I suspect the SOHO switches could be part of the problem. What is the load on the server? Bill * From: * [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mitch Jackson *Sent:* Tuesday, July 11, 2006 10:44 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Server Optimization and Load Balancing I'm hoping to get some guidance on some of our asterisk growing pains. Any help is greatly appreciated. Over the last few months, our call center has grown considerably and we're now experiencing choppy calls and dropped calls under full capacity. We have around a 60/40 outgoing to incoming ratio At the moment, we've got the following configuration: Asterisk SVN-trunk-r7230 All calls recorded to disk External mysql server for CDR + IVR operations Dual Xeon 2.8 4GB Ram (Dell) Digium TE210P 2 x PRI lines 72 Ploycom 301P SIP phones using ulaw codec We have a second identical server ready to offset some of the load, but we're not sure how to balance the sip phones and configuration files between the two servers. If we balance the sip registrations between the two servers, then there's the issues of both servers having to handle one call via IAX in some situations. What kind of experiences, problems and solutions have y'all had when adding servers to your center? Should we try to have incoming on one server and outgoing on the other? Should we have both servers capable of handling all the IVR operations, so the other server doesn't have to? Should we try to have an identical configuration between both servers and load balance? What kind of general optimizations should we look at to improve network / server performance? Is there a way to easily register each phone with all asterisk servers, and have the phone choose a random server to dial, and then be available as a SIP to each server if it needs to contact it? Is it a bad idea to register all phones with each server instead of distributing registration? -Here's some of the things we're got planned in the next few days: -Make sure we have all audio files in all codec formats to reduce the need for transcoding in IVR -Convert all music on hold from mp3 to native codec formats -Reduce database operations from within extensions.conf -Upgrade switches on each set of desks to midrange enterprise 100MB switches with gigabit uplinks, from SOHO netgear 100MB switches Thanks, /mitch /fidelity reserves ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] WARNING[30954]: chan_sip.c:2734 sip_indicate: Don't know how to indicate condition 9
Condition 9 looks like a flash hook (from frame.h): /*! Flash hook */ #define AST_CONTROL_FLASH 9 so I guess the incoming channel is indicating flash hook to your SIP channel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Giorgio Incantalupo Sent: 11 July 2006 15:48 To: asterisk-users@lists.digium.com Subject: [asterisk-users] WARNING[30954]: chan_sip.c:2734 sip_indicate: Don't know how to indicate condition 9 Hi, is there anybody who knows what the following warnings mean? WARNING[30954]: chan_sip.c:2734 sip_indicate: Don't know how to indicate condition 9 WARNING[30954]: channel.c:2051 ast_indicate: Unable to handle indication 9 for 'SIP/8-3c6e' TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to do load balancing (1:1) with IAX and two different ISPs
Hello folks, Does anyone have an idea how I could setup a load balancing (1:1) solution with IAX and two different Internet service providers. The idea is to increase the bandwidth between offices with cheap Internet access (DSL/Cable). I understand that with SIP, I could do that with a SIP Proxy (SER) but how could I do that with IAX (round robin in DNS? / IAX Proxy? ) Any help is appreciated! Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI tutorials
Rizwan Hisham schrieb: Anybody who knows a good source of AGI tutorials on the net? plz share try one of the mirrors and then the pages on AGI, http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 have Phun Kai ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: [Asterisk-video] Asterisk as an MCU
Hi Curt, At the moment Asterisk does not perform the functionality you are looking for (there is no single server solution for what you are looking for at the moment). We were looking to sponsor video conferencing development on Asterisk a year ago but put it into the too hard basket. We were then looking to build an application using Adobe Flash media Server but have ceased work on this because of licensing changes which made it uneconomical for less than 100 seats. www.cognation.net/unisona At the moment we use Breeze ASP service to do presentations and Asterisk for Voip (and would use LCS or Jabber for internal messaging but just use MSN messenger). We are doing this with the view that things will change in the next 12 months and will re-look at an all in one service based solution at this time. If I had to buy a video/web presentation server solution at the moment it would be www.wiredred.com Best advice I can offer after spending a lot of time looking at this in the past. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-video- [EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Tuesday, 11 July 2006 10:47 AM To: 'Development discussion of video media support in Asterisk' Subject: RE: [Asterisk-video] Asterisk as an MCU Thanks for the clarification. So if I want some functionality of an MCU I could use Asterisk as long as the clients were talking the same (supported) codec? I have never had to build an MCU so I don't know much about them. What we are looking for is video conferencing from workstations through a central system with the ability to dial in from the PSTN and to do IP calls and possibly include some sort of presence features. As far as I can see then Asterisk can fit this bill or am I missing key functionality or performance from not having full MCU capabilities? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeffrey C. Ollie Sent: Tuesday, July 11, 2006 10:41 AM To: Development discussion of video media support in Asterisk Subject: RE: [Asterisk-video] Asterisk as an MCU On Tue, 2006-07-11 at 09:57 -0400, Curt Shaffer wrote: Odd... http://www.voip-info.org/wiki/view/Asterisk+video looks like it does there unless I am missing something. Yes, that page is extremely misleading. Asterisk does not include video codecs. The video support that is mentioned on that page is pass through only. That means that it cannot convert between video formats (which would be required for MCU functionality). Jeff ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Yet another problem with incoming SIP calls and 407
Wolfgang Zweimueller wrote: Hi all, when I receive incoming SIP calls on my Asterisk (1.2.9.1) where the caller has a username in it's From-Address which also exists in my sip.conf then my system answers with 407 Proxy Authentication Required. If it's nonexistent username then callin works fine! It seems that this is a problem in the SIP implementation of Asterisk and found a few hints on how to resolve this (allowguest=yes, insecure=invite,port etc.). But none of them does help! Can anyone suggest what I else could try? in sip.conf [general] context=INVALID Then put the correct context= line for each sip user/friend/peer. Unauthenticated calls use the options in [general] -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IPKALL direct to asterisk bypassing FWD
I found FWD to be unreliable so I was trying to get IPKALL direct to asterisk. I finally got this working direct with asterisk after several attempts. Below is my working configuration if anyone is interested. My asterisk server is directly on the internet. It is also running shorewall, NAT and a couple other services for me. http://phone.ipkall.com/ipphone/login.asp IPKall Phone Number:425XXX Password: Settings at www.ipkall.com SIP Phone Number: 123 SIP Proxy: myserver.myhome.com:5060 --- this could a dyndns address as well Email Address: [EMAIL PROTECTED] ---Must be a valid email address Password: Voice Mail on/off Seconds to Voice Mail: Settings for asterisk [123] type=peer qualify=no port=5060 nat=no insecure=very this is very important host=voiper.ipkall.com dtmfmode=rfc2833 context=from-pstn canreinvite=no ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk stops abruptly
Hello, I'm recently having the problem where Asterisk just stops working. The console gets disconnected and the process appears to die. I am using Asterisk version 1.2.9.1. Anyone have any ideas on where I should be looking for the cause of my problem? Also, I notice there is a /var/log/asterisk/messages log file but it doesn't contain any information that I can use to help troubleshoot the application crashing. Is there a way to put more debugging in the log file? Thank you for your help, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Optimization and Load Balancing
Mitch Jackson [EMAIL PROTECTED] wrote: Over the last few months, our call center has grown considerably and we're now experiencing choppy calls and dropped calls under full capacity. ... All calls recorded to disk ... 72 Ploycom 301P SIP phones using ulaw codec If you run this command vmstat 5 Does it show lots of processes that are blocked, waiting for the disk? If I had to do this I would have a battery backed writeback RAID controller. -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Server redundancy
-Original Message- From: RR [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 11, 2006 12:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Server redundancy Interesting points on both messages 1) as far as multiple asterisk servers talking to the same database is concerned, I will have to test this out. I know nothing about the database side of things, and a newbie on asterisk and linux so I have no idea what and where the development of either of these are. From your message it sounds like it's just how ARA is designed because I doubt it's to do with the ODBC driver itself. This will cause me a lot of grief if you're right about this for multiple * servers to not be able to access the same database for peer lookup. Be prepared for some grief then! :) 2) Clustering of DB isn't an issue, not for me at least. Haven't tested this either but my DBs are clustered A/P providing a single entity to the internal systems. Might further look into a local DNS lookup to add to this. I believe it's possible to do this in the MySQL world with MySQL grid etc? 3) I don't believe frequent registration is that big of an issue for the network load it generates. Most providers out there set devices for a 30-40secs Reg. Refresh to support NAT'ed endpoints and the a reg refresh is hardly about 300-400Byte pkts (I think). The math doesn't add up for a major load esp. if you've got a load balancing mechanism in front of your * boxes. OMG. 30-40seconds? That's insane. We're planning on provisioning 16,000 users on our system. With a registration period of 40s, that's an average of over 400 registrations per second. Actually, it could be over 800 per second, as I think phones re-reregister at half their expirey period. 4) I don't know enough about DUNDi to get into this discussion but DUNDi just lookup extensions? or it also have any part to play in registrations? If they just do extension lookup, then If DUNDi is implemented on an A/P pair of dedicated DUNDi lookup servers which access a clustered database, then barring #1 being true, each * server accesses the same database and pool of registrations. If registrations are refreshed frequently enough, the contact info in the database will always be current and one server dying won't affect anything. At the same time, they just consult the DUNDi lookup server for extension lookups instead of asking the database directly. DUNDi can only lookup the extensions (ie phones) if they are registered. If they aren't registered on any system in the DUNDi peering arrangement, then DUNDi wouldn't return a path to their location... until the phones re-reregister. 5) If you really want to improve on this, supplement your network with SER as proxies and have them deal with Registrations and load-balance feature requests to * servers etc. Once * has done whatever it needs to do (e.g. provide PBX features, voicemail, conference, IVR etc.) it passes the call back to the Proxy to deal with the endpoints. Not as simple as it sounds. So, if the SER boxes handle registrations, how do they propogate this information back to Asterisk? If you don't propogate the information back to Asterisk, then you have to route all dialling from Asterisk back to SER. This causes problems withn certain applications like the Queue command, that as far as I know, can't work with this. There's probably more too. You also have to keep call transfer and call forward in mind. When transferring a call, if you pass the call from Asterisk over to SER for some reason, it has to come back to the same Asterisk box to handle the transfer, Asterisk will puke. All depends on your scope and budget. If you want to have a SP grade service then you need to breakout your functions. I just hope #1 isn't true though. The only alternative then would be to have /etc/asterisk reside on an NFS share or a CFS for all servers to read massively huge conf files if you're catering for large number of endpoints. Dunno if it helps anyone or I'm just shooting sh*t ;) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text priority labels not working for me
Aaron Daniel wrote: The problem is in the space. You've got it as ? scid)... In order for the label to work, you need to get rid of the space. Make it ?scid) and it should work fine. The error's in the details: pbx_extension_helper: No such label ' scid' in extension 's' in context 'macro-dosomething' Removing the spaces worked! Is that just a parser oddity? I've used spaces with numeric labels in the past (e.g.: ? 3:2) and it's worked just fine. Cheers, -Wes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issues with making Transfers
Hello, I am having a problem with transferring calls that come in from the outside. Users have been calling in to the PRI that's on the Cisco GW, then they are passed into Asterisk via SIP and to the end phone (Polycom 501/601) using SIP. When that user tries to transfer that call to another extension, the call disconnects and hangs in the air and doesn't do anything. The call shows active in the Cisco GW but no where to be found in asterisk. Here is some log output of a transfer attempt: -- Stopped music on hold on SIP/10.25.118.2-b7b4e520 == Spawn extension (ANC, 4023, 2) exited non-zero on 'SIP/4023-ebbfZOMBIE' -- SIP/2198-3780 answered SIP/10.25.118.2-b7b4e520 -- Attempting native bridge of SIP/10.25.118.2-b7b4e520 and SIP/2198-3780 -- Incoming call: Got SIP response 500 "Internal Server Error" back from 10.45.25.12 I'm not sure if the SIP 500 error is relative to my issue. Any ideas on what could be causing SIP transfers to hang or drop? Thank you, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RE: [Asterisk-video] Asterisk as an MCU
Thanks for the information. I guess just as a follow up, is it not possible then to utilize something like MSN messenger or Video capable chat clients that support SIP, like MSN, some sort of jabber or iChat that will allow Asterisk to just pass through the video but handle the voice? I think that would suit our needs for now. Thanks again Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, July 11, 2006 11:05 AM To: Development discussion of video media support in Asterisk Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RE: [Asterisk-video] Asterisk as an MCU Hi Curt, At the moment Asterisk does not perform the functionality you are looking for (there is no single server solution for what you are looking for at the moment). We were looking to sponsor video conferencing development on Asterisk a year ago but put it into the too hard basket. We were then looking to build an application using Adobe Flash media Server but have ceased work on this because of licensing changes which made it uneconomical for less than 100 seats. www.cognation.net/unisona At the moment we use Breeze ASP service to do presentations and Asterisk for Voip (and would use LCS or Jabber for internal messaging but just use MSN messenger). We are doing this with the view that things will change in the next 12 months and will re-look at an all in one service based solution at this time. If I had to buy a video/web presentation server solution at the moment it would be www.wiredred.com Best advice I can offer after spending a lot of time looking at this in the past. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-video- [EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Tuesday, 11 July 2006 10:47 AM To: 'Development discussion of video media support in Asterisk' Subject: RE: [Asterisk-video] Asterisk as an MCU Thanks for the clarification. So if I want some functionality of an MCU I could use Asterisk as long as the clients were talking the same (supported) codec? I have never had to build an MCU so I don't know much about them. What we are looking for is video conferencing from workstations through a central system with the ability to dial in from the PSTN and to do IP calls and possibly include some sort of presence features. As far as I can see then Asterisk can fit this bill or am I missing key functionality or performance from not having full MCU capabilities? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeffrey C. Ollie Sent: Tuesday, July 11, 2006 10:41 AM To: Development discussion of video media support in Asterisk Subject: RE: [Asterisk-video] Asterisk as an MCU On Tue, 2006-07-11 at 09:57 -0400, Curt Shaffer wrote: Odd... http://www.voip-info.org/wiki/view/Asterisk+video looks like it does there unless I am missing something. Yes, that page is extremely misleading. Asterisk does not include video codecs. The video support that is mentioned on that page is pass through only. That means that it cannot convert between video formats (which would be required for MCU functionality). Jeff ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk stops abruptly
On Tue, 2006-07-11 at 08:20 -0700, Dan Brummer wrote: Hello, I'm recently having the problem where Asterisk just stops working. The console gets disconnected and the process appears to die. I am using Asterisk version 1.2.9.1. Anyone have any ideas on where I should be looking for the cause of my problem? Also, I notice there is a /var/log/asterisk/messages log file but it doesn't contain any information that I can use to help troubleshoot the application crashing. Is there a way to put more debugging in the log file? Yes take a look at logger.conf. There is a default of 'full' which will create /var/log/asterisk/full for example, and will have more info, but you can add the individual elements to the messages one if you would rather. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] flash button on asterisk + legacy pbx system
Hi C F,
I managed to send the flash code...thanks for your help. Now I'm trying
to send digits after the flash code so that the user can send another
extension. Is it possible to have something like
key = _X.,caller,SendDTMF,X.
inside [applicationmap] in order to send digits to the legacy pbx?
TIA
Giorgio Incantalupo
C F wrote:
Yes I have seen this before and it creates confuseion, but the
solution is that you create 2 application maps, one that works for
inbound calls, and the other that works for outbound calls.
The following is what works for me:
/etc/asterisk/features.conf:
[applicationmap]
inflash = *4,caller,Flash,()
outflash = *3,callee,Flash,()
/etc/asterisk/extensions.conf:
exten = s,1,Set(DYNAMIC_FEATURES=inflash);this is an incoming call on
the FXO port and g2 are the FXS ports
exten = s,2,Dial(Zap/g2,,t)
exten = _1XX,1,Set(DYNAMIC_FEATURES=outflash);this is outbound
exten = _1XX,2,Dial(Zap/g2/${EXTEN},,T)
With the above they dial *4 on incoming calls, and *3 on outgoing
calls to get this working.
I know it's confusing, but the users get used to it.
On 7/4/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi C F,
ok, I also thought to make the user to press some keys for example * and
3 so I setup a little test made using an Asterisk box with a TDM400P (2
FXS + 2 FXO) connected to an analog phone (fxs port) and an analog line
(fxo port).
I searched on internet and found some interesting stuff so I made my
extensions.conf:
My extension.conf is (in brief):
[zap]
exten = s,1,Set(DYNAMIC_FEATURES=zapflash)
exten = s,2,Dial(Zap/3,15,tw) --- Zap/3 is my analog phone
exten = s,3,HangUp
My zapata (Zap/1 is the line and Zap/3 is the phone):
context = zap
language = it
signalling = fxs_ks
threewaycalling=yes
transfer = yes
channel = 1
language = it
signalling = fxo_ks
callerid = tel1 100
threewaycalling=yes
transfer = yes
channel = 3
and my features.conf:
[applicationmap]
...
zapflash = *3,caller,flash,()
When I call the number xxx, Asterisk answers on zap line passing the
call to zap/3. I pick up zap/3 phone and then I press *3 but all I get
is (on asterisk console):
WARNING[3082]: app_flash.c:101 flash_exec: Zap/3-1 is not an FXO Channel
Why? It seems Asterisk sends Flash command to the phone but it is not
what I want.
Is this the right way to follow? Press *3 (or other code) to send
command to host pbx while the callee is on the phone? Is this what you
meant? If yes, why Asterisk does not send the flash command to the line?
Thanks for patience
Giorgio Incantalupo
C F wrote:
Sorry I didn't realize this is how you wanted it to work - that the
user is on a FXS and you want when the user flashes that it flashes
the host pbx.
I disagree with you on this setup the user should be requried to press
some DTMF and not just flash the phone. The main reason being that
otherwise you will lose 3way and callwaiting features on asterisk. I'm
assuming your answer to this is that you don't care since you just
want to make the phone an extended extension on the host PBX, and want
it to be as much an extension of the old PBX as posible. I still
disagree because as much as you are going to try, your users will
still not see this as a direct extension, and sooner or later you/they
will have to learn how to deal with it anyhow.
On 7/4/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi C F,
I read the comments but the problem remains...after some tests, I
changed some parameters inside zapata.h and recompiled to make flash
button work so now my asterisk knows when the user presses the flash
button /during a call./
My problem now is how to transfer the flash signal to the old PBX,
infact seems like asterisk accept it (even if I cannot use it inside
extensions.conf for example with a _FLASH,1,...) but then doesn't
re-send it to the line.
TIA
Giorgio Incantalupo
C F wrote:
Use features.conf,
look here at the comments:
http://www.voip-info.org/wiki-Asterisk+cmd+flash
On 7/3/06, Giorgio Incantalupo [EMAIL PROTECTED]
wrote:
Hi C F,
you say Flash asterisk command send a flash signal to old pbx so
that it
sees that command as coming from an analog phone. But since Flash
is not
a digit, how can I catch it from within asterisk? How can I tell
asterisk (es inside extensions.conf) to do something whene
receive it
from a phone?
TIA
Giorgio Incantalupo
C F wrote:
The flash command will do just that. However flash only works on
FXO
ports and not on SIP FXO ATAs, if you use the later then you
will have
to find out how your ATA supports it.
The easiest way to set this up is to use the features.conf
On 7/3/06, Giorgio Incantalupo [EMAIL PROTECTED]
wrote:
Hi,
I have to connect an old PBX to a new Asterisk box. but I must
keep the
same flash button functionality of the old system. Is it
possible to
tell asterisk to send a Flash signal to old pbx when
receiving it
from
[asterisk-users] [announcement] kansas city asterisk user group
For those that are in the Kansas City area I would like to announce the formation of the Kansas City Asterisk User Group. You can find more information about the group at http://www.kcaug.net/ . Currently there is a mailling list for the group, but if enough interest arises we will probably look into doing meet-ups. If you are in the KC area and at all interested in Asterisk please join up.Thanks, Kyle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Issues with making Transfers
Asterisk 1.2.9.1 is the version I'm on. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan BrummerSent: Tuesday, July 11, 2006 8:30 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Issues with making Transfers Hello, I am having a problem with transferring calls that come in from the outside. Users have been calling in to the PRI that's on the Cisco GW, then they are passed into Asterisk via SIP and to the end phone (Polycom 501/601) using SIP. When that user tries to transfer that call to another extension, the call disconnects and hangs in the air and doesn't do anything. The call shows active in the Cisco GW but no where to be found in asterisk. Here is some log output of a transfer attempt: -- Stopped music on hold on SIP/10.25.118.2-b7b4e520 == Spawn extension (ANC, 4023, 2) exited non-zero on 'SIP/4023-ebbfZOMBIE' -- SIP/2198-3780 answered SIP/10.25.118.2-b7b4e520 -- Attempting native bridge of SIP/10.25.118.2-b7b4e520 and SIP/2198-3780 -- Incoming call: Got SIP response 500 "Internal Server Error" back from 10.45.25.12 I'm not sure if the SIP 500 error is relative to my issue. Any ideas on what could be causing SIP transfers to hang or drop? Thank you, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk stops abruptly
enable debug logging in /etc/asterisk/logger.conf then do a logger reload then if asterisk dies, search the log for relevant events and post it here. I'm also using 1.2.9.1 so im interested. On 7/11/06, Dan Brummer [EMAIL PROTECTED] wrote: Hello, I'm recently having the problem where Asterisk just stops working. The console gets disconnected and the process appears to die. I am using Asterisk version 1.2.9.1. Anyone have any ideas on where I should be looking for the cause of my problem? Also, I notice there is a /var/log/asterisk/messages log file but it doesn't contain any information that I can use to help troubleshoot the application crashing. Is there a way to put more debugging in the log file? Thank you for your help, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk stops abruptly
Thank you for the quick response. I assume this change will require an Asterisk reload? Thanks! -Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Tuesday, July 11, 2006 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk stops abruptly On Tue, 2006-07-11 at 08:20 -0700, Dan Brummer wrote: Hello, I'm recently having the problem where Asterisk just stops working. The console gets disconnected and the process appears to die. I am using Asterisk version 1.2.9.1. Anyone have any ideas on where I should be looking for the cause of my problem? Also, I notice there is a /var/log/asterisk/messages log file but it doesn't contain any information that I can use to help troubleshoot the application crashing. Is there a way to put more debugging in the log file? Yes take a look at logger.conf. There is a default of 'full' which will create /var/log/asterisk/full for example, and will have more info, but you can add the individual elements to the messages one if you would rather. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Test E1 channel
On 7/7/06, Moises Silva [EMAIL PROTECTED] wrote: One of the ends must be configured as pri_net and the other aspri_cpe. By the error I think the problem is with your configuration,does zttool says no alarms in spans?Post your configuration files zapata.conf and zaptel.confRegardsOn 7/7/06, Marco Mouta [EMAIL PROTECTED] wrote: by Ports i mean Spans :) On 7/7/06, Marco Mouta [EMAIL PROTECTED] wrote: Newbie guess, Don't you need to set one of the ports NT mode and the other one as TE mode? hope it helps Best regards, PS. give me some feed back if it solved.Hi folks,that was my first try.I had set all the first E1 channel as pri_net and all the second E1 channel as pri_cpe but I got this error. chan_zap.c: PRI Error: We think we-re the network, but they think they're the network, too.When I set everybody as pri_net this message stops.Today, I put the E1 channel to work, it was only set the channel to pri_cpe and dial ! I still without know why the previous tests didn´t work.Thanks everybody.-- Ralph Liebessohn ICQ: 74835911Skype: liebessohn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RE: [Asterisk-video] Asterisk as an MCU
:) nope not that either. Best thing you can do is go to the voip-wiki bounty page and kick in and wait for development. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Tuesday, 11 July 2006 11:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] RE: [Asterisk-video] Asterisk as an MCU Thanks for the information. I guess just as a follow up, is it not possible then to utilize something like MSN messenger or Video capable chat clients that support SIP, like MSN, some sort of jabber or iChat that will allow Asterisk to just pass through the video but handle the voice? I think that would suit our needs for now. Thanks again Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, July 11, 2006 11:05 AM To: Development discussion of video media support in Asterisk Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RE: [Asterisk-video] Asterisk as an MCU Hi Curt, At the moment Asterisk does not perform the functionality you are looking for (there is no single server solution for what you are looking for at the moment). We were looking to sponsor video conferencing development on Asterisk a year ago but put it into the too hard basket. We were then looking to build an application using Adobe Flash media Server but have ceased work on this because of licensing changes which made it uneconomical for less than 100 seats. www.cognation.net/unisona At the moment we use Breeze ASP service to do presentations and Asterisk for Voip (and would use LCS or Jabber for internal messaging but just use MSN messenger). We are doing this with the view that things will change in the next 12 months and will re-look at an all in one service based solution at this time. If I had to buy a video/web presentation server solution at the moment it would be www.wiredred.com Best advice I can offer after spending a lot of time looking at this in the past. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-video- [EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Tuesday, 11 July 2006 10:47 AM To: 'Development discussion of video media support in Asterisk' Subject: RE: [Asterisk-video] Asterisk as an MCU Thanks for the clarification. So if I want some functionality of an MCU I could use Asterisk as long as the clients were talking the same (supported) codec? I have never had to build an MCU so I don't know much about them. What we are looking for is video conferencing from workstations through a central system with the ability to dial in from the PSTN and to do IP calls and possibly include some sort of presence features. As far as I can see then Asterisk can fit this bill or am I missing key functionality or performance from not having full MCU capabilities? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeffrey C. Ollie Sent: Tuesday, July 11, 2006 10:41 AM To: Development discussion of video media support in Asterisk Subject: RE: [Asterisk-video] Asterisk as an MCU On Tue, 2006-07-11 at 09:57 -0400, Curt Shaffer wrote: Odd... http://www.voip-info.org/wiki/view/Asterisk+video looks like it does there unless I am missing something. Yes, that page is extremely misleading. Asterisk does not include video codecs. The video support that is mentioned on that page is pass through only. That means that it cannot convert between video formats (which would be required for MCU functionality). Jeff ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server redundancy
What about having the softphone/hardphones configured with a Main Asterisk server and an Alternative Asterisk Server?. I just did a couple of tests with a Cisco ATA 186 and worked quit well after changing the RegInterval and Alternative Proxy Timeout. thks, Alejandro, On Tuesday 11 July 2006 05:04 am, unplug wrote: I have asked about it here. As Douglas said, it doesn't support mult-asterisk in current version. However, I have questions about why multi-asterisk so difficult to implement. 1. As we can use ARA to store all information, sip user register info, dial plan ... to DB. All asterisks can use ARA to refer to the DB for necessary information even register information. 2. What is mean by multiple Asterisk systems can't reference the same MySQL database for SIP peers.? Does SIP peer information also store in DB? 3. Any difficulty to implement multiple asterisk? 4. If I want to implement multiple asterisk in some extent, how do I begin? Any reference? On 7/11/06, RR [EMAIL PROTECTED] wrote: Interesting points on both messages 1) as far as multiple asterisk servers talking to the same database is concerned, I will have to test this out. I know nothing about the database side of things, and a newbie on asterisk and linux so I have no idea what and where the development of either of these are. From your message it sounds like it's just how ARA is designed because I doubt it's to do with the ODBC driver itself. This will cause me a lot of grief if you're right about this for multiple * servers to not be able to access the same database for peer lookup. 2) Clustering of DB isn't an issue, not for me at least. Haven't tested this either but my DBs are clustered A/P providing a single entity to the internal systems. Might further look into a local DNS lookup to add to this. I believe it's possible to do this in the MySQL world with MySQL grid etc? 3) I don't believe frequent registration is that big of an issue for the network load it generates. Most providers out there set devices for a 30-40secs Reg. Refresh to support NAT'ed endpoints and the a reg refresh is hardly about 300-400Byte pkts (I think). The math doesn't add up for a major load esp. if you've got a load balancing mechanism in front of your * boxes. 4) I don't know enough about DUNDi to get into this discussion but DUNDi just lookup extensions? or it also have any part to play in registrations? If they just do extension lookup, then If DUNDi is implemented on an A/P pair of dedicated DUNDi lookup servers which access a clustered database, then barring #1 being true, each * server accesses the same database and pool of registrations. If registrations are refreshed frequently enough, the contact info in the database will always be current and one server dying won't affect anything. At the same time, they just consult the DUNDi lookup server for extension lookups instead of asking the database directly. 5) If you really want to improve on this, supplement your network with SER as proxies and have them deal with Registrations and load-balance feature requests to * servers etc. Once * has done whatever it needs to do (e.g. provide PBX features, voicemail, conference, IVR etc.) it passes the call back to the Proxy to deal with the endpoints. All depends on your scope and budget. If you want to have a SP grade service then you need to breakout your functions. I just hope #1 isn't true though. The only alternative then would be to have /etc/asterisk reside on an NFS share or a CFS for all servers to read massively huge conf files if you're catering for large number of endpoints. Dunno if it helps anyone or I'm just shooting sh*t ;) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] what single PRI interface, from which manufacturer
List, I was wondering what would be the best PRI (E1) interface card from which manufacturer. Digium, Sangoma, Eicon, Junghanns, ??? AFAIK there are some IRQ sharing issues with a Digium TE110P; or can someone confirm stable operation of this card? Any suggestions? Thanks, FO -- LocaNet oHG - http://www.loca.net Lindemannstrasse 81, D-44137 Dortmund tel +49 231 91596-23, mobil +49 172 2120354 sip [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] recompiling/updateing zaptel
When I want to recompile zaptel with different echo cancelers, do I need to unload the modules from the kernel before I hit "make install"? Stop asterisk?What would the commands be and in the proper order, ie(Fedora Core 4):make clean make service asterisk stop rmmod -r zaptel make install modprobe asterisk startIs there any way to tell which echo canceler the current module is using?thanks Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail Beta.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MFC/R2 country and carrier specific protocol variants
Hi all,Those of you whoare using or who have used the Unicall channel for MFC/R2 may be familiar with 'protocolvariant' field, in the unicall.conf file. It changes from country to country and even in the same country it may change from carrier to carrier.I googled around looking for a list of those protocol variants, but I only found scattered information, given by users who had been using Unicall in a specific country, sometimes without providing the carrier name.I have added to the bottom of http://www.voip-info.org/wiki/view/Asterisk+MFC+R2the list of protocol variants that I have gathered so far. I would kindly ask those of you who may have successfullyused other variants to add them to the list, so that everyone can benefit from that information.Many thanks, Paulo___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what single PRI interface, from which manufacturer
On Tue, 11 Jul 2006, Frank Ochmann wrote: List, I was wondering what would be the best PRI (E1) interface card from which manufacturer. Digium, Sangoma, Eicon, Junghanns, ??? AFAIK there are some IRQ sharing issues with a Digium TE110P; or can someone confirm stable operation of this card? Any suggestions? I cannot say which is the best, because I didn't try/use all of them. But I can recommend the Eicon DIVA Server card. These cards are very good, reliable and have powerful features. We do have best results on projects using these cards. Some people here might say that the cards are very expensive, but I say it's worth. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] flash button on asterisk + legacy pbx system
Yes you can do that just change the application map to a Goto command
that goes to an exten in the dialplan that does it all for you.
On 7/11/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi C F,
I managed to send the flash code...thanks for your help. Now I'm trying
to send digits after the flash code so that the user can send another
extension. Is it possible to have something like
key = _X.,caller,SendDTMF,X.
inside [applicationmap] in order to send digits to the legacy pbx?
TIA
Giorgio Incantalupo
C F wrote:
Yes I have seen this before and it creates confuseion, but the
solution is that you create 2 application maps, one that works for
inbound calls, and the other that works for outbound calls.
The following is what works for me:
/etc/asterisk/features.conf:
[applicationmap]
inflash = *4,caller,Flash,()
outflash = *3,callee,Flash,()
/etc/asterisk/extensions.conf:
exten = s,1,Set(DYNAMIC_FEATURES=inflash);this is an incoming call on
the FXO port and g2 are the FXS ports
exten = s,2,Dial(Zap/g2,,t)
exten = _1XX,1,Set(DYNAMIC_FEATURES=outflash);this is outbound
exten = _1XX,2,Dial(Zap/g2/${EXTEN},,T)
With the above they dial *4 on incoming calls, and *3 on outgoing
calls to get this working.
I know it's confusing, but the users get used to it.
On 7/4/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi C F,
ok, I also thought to make the user to press some keys for example * and
3 so I setup a little test made using an Asterisk box with a TDM400P (2
FXS + 2 FXO) connected to an analog phone (fxs port) and an analog line
(fxo port).
I searched on internet and found some interesting stuff so I made my
extensions.conf:
My extension.conf is (in brief):
[zap]
exten = s,1,Set(DYNAMIC_FEATURES=zapflash)
exten = s,2,Dial(Zap/3,15,tw) --- Zap/3 is my analog phone
exten = s,3,HangUp
My zapata (Zap/1 is the line and Zap/3 is the phone):
context = zap
language = it
signalling = fxs_ks
threewaycalling=yes
transfer = yes
channel = 1
language = it
signalling = fxo_ks
callerid = tel1 100
threewaycalling=yes
transfer = yes
channel = 3
and my features.conf:
[applicationmap]
...
zapflash = *3,caller,flash,()
When I call the number xxx, Asterisk answers on zap line passing the
call to zap/3. I pick up zap/3 phone and then I press *3 but all I get
is (on asterisk console):
WARNING[3082]: app_flash.c:101 flash_exec: Zap/3-1 is not an FXO Channel
Why? It seems Asterisk sends Flash command to the phone but it is not
what I want.
Is this the right way to follow? Press *3 (or other code) to send
command to host pbx while the callee is on the phone? Is this what you
meant? If yes, why Asterisk does not send the flash command to the line?
Thanks for patience
Giorgio Incantalupo
C F wrote:
Sorry I didn't realize this is how you wanted it to work - that the
user is on a FXS and you want when the user flashes that it flashes
the host pbx.
I disagree with you on this setup the user should be requried to press
some DTMF and not just flash the phone. The main reason being that
otherwise you will lose 3way and callwaiting features on asterisk. I'm
assuming your answer to this is that you don't care since you just
want to make the phone an extended extension on the host PBX, and want
it to be as much an extension of the old PBX as posible. I still
disagree because as much as you are going to try, your users will
still not see this as a direct extension, and sooner or later you/they
will have to learn how to deal with it anyhow.
On 7/4/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi C F,
I read the comments but the problem remains...after some tests, I
changed some parameters inside zapata.h and recompiled to make flash
button work so now my asterisk knows when the user presses the flash
button /during a call./
My problem now is how to transfer the flash signal to the old PBX,
infact seems like asterisk accept it (even if I cannot use it inside
extensions.conf for example with a _FLASH,1,...) but then doesn't
re-send it to the line.
TIA
Giorgio Incantalupo
C F wrote:
Use features.conf,
look here at the comments:
http://www.voip-info.org/wiki-Asterisk+cmd+flash
On 7/3/06, Giorgio Incantalupo [EMAIL PROTECTED]
wrote:
Hi C F,
you say Flash asterisk command send a flash signal to old pbx so
that it
sees that command as coming from an analog phone. But since Flash
is not
a digit, how can I catch it from within asterisk? How can I tell
asterisk (es inside extensions.conf) to do something whene
receive it
from a phone?
TIA
Giorgio Incantalupo
C F wrote:
The flash command will do just that. However flash only works on
FXO
ports and not on SIP FXO ATAs, if you use the later then you
will have
to find out how your ATA supports it.
The easiest way to set this up is to use the
RE: [asterisk-users] Provider UNREACHABLE
At 04:59 AM 7/11/2006, you wrote: I am repeatedly getting a UNREACHABLE and then REACHABLE about 10 sec apart most of the time and then sometimes for about 45 - 74 minutes I have tried a reload and sip reload but neither bring the provider back ? I see this with sipdiscount and it's brethren occasionally. I'm guessing it has something to do with latency in the outside world, like why is the internet blazing fast one minute and painfully slow 5 minutes later. Here(Los Angeles) it tends to happen late at night, mostly when I'm asleep. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Provider UNREACHABLE
On Tue, 2006-07-11 at 09:08 -0700, Ira wrote: At 04:59 AM 7/11/2006, you wrote: I am repeatedly getting a UNREACHABLE and then REACHABLE about 10 sec apart most of the time and then sometimes for about 45 - 74 minutes I have tried a reload and sip reload but neither bring the provider back ? I see this with sipdiscount and it's brethren occasionally. I'm guessing it has something to do with latency in the outside world, like why is the internet blazing fast one minute and painfully slow 5 minutes later. Here(Los Angeles) it tends to happen late at night, mostly when I'm asleep. with sipdiscount its mostly due to the fact that their sip server doesnt respond in the predefined time set by 'qualify' in the peer definition. If you do like qualify=2000 it will only display them if they cant respond within 2 seconds. Which at night in LA is daytime where they are located (Europe) and they probably see a higher call volume and thus their servers arent as speedy to respond. Note this number isnt the network lag by itself, its also the time it takes for them to respond to sip messages, as well as the network lag, so it can be midleading to performance by itself. These types of errors are generally harmless, if they bother you set the qualify to a higher number or turn it off with qualify=no. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Provider UNREACHABLE
they had a short outage today, it was fixed already, dunno if related to your issue, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Provider UNREACHABLE
teliax had a 2.5 hour outage today. I wouldn't call that short. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Paglayan Sent: Tuesday, July 11, 2006 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Provider UNREACHABLE they had a short outage today, it was fixed already, dunno if related to your issue, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] several asterisk servers questions
We have a 3 stage implementation plan with a customer, we're still documenting the structure, it is of course subject to change so I welcome all your comments on this matter. customer has one main building and 5 more. All inter-connected with fiber optics links for data/voice traffic. Main building holds the datacenter. proper network gear (switches and routers) are being deployed by another company to the customer. On stage one, the customer wants to have a FAX server. They read about using asterisk as a fax server and also read about Hylafax they also read about astfax+trixbox. The setup must compete against MS windows solutions that do fax-to-email and email-to-fax, we must keep deployment of software to the client machines to a minimum. The customer is looking to deploy an E1 so faxes have 30 channels to receive and send faxes, the server must communicate with an MS Exchange 2003 server. On stage two, there will be an asterisk server to handle PSTN calls (in and out)using E1 lines /about 4 E1s. We think that due to the load (500+ SIP users in main building) voicemail should be handled by a different server. Then On stage 3, another server???, serving SIP users in the main building to connect to the other buildings that will also have a little less powerful IP-to-SIP and/or IP-to-FXS asterisk (those server may have PSTN connectivity). Some form of config backups and/or disaster recovery plans must be documented as well as taking images of the RAID systems that will be using asterisk. I'm expecting full server details on this one, because the customer will provide the equipments (servers). So your comments will be appreciated. Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk stops abruptly
no. just logger reload On 7/11/06, Dan Brummer [EMAIL PROTECTED] wrote: Thank you for the quick response. I assume this change will require an Asterisk reload? Thanks! -Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Tuesday, July 11, 2006 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk stops abruptly On Tue, 2006-07-11 at 08:20 -0700, Dan Brummer wrote: Hello, I'm recently having the problem where Asterisk just stops working. The console gets disconnected and the process appears to die. I am using Asterisk version 1.2.9.1. Anyone have any ideas on where I should be looking for the cause of my problem? Also, I notice there is a /var/log/asterisk/messages log file but it doesn't contain any information that I can use to help troubleshoot the application crashing. Is there a way to put more debugging in the log file? Yes take a look at logger.conf. There is a default of 'full' which will create /var/log/asterisk/full for example, and will have more info, but you can add the individual elements to the messages one if you would rather. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Provider UNREACHABLE
hehe yeah.. still when you see that qualify breaks newer xlites' you would wonder why to use it anyhow ?On 7/11/06, Rick Smith [EMAIL PROTECTED] wrote:teliax had a 2.5 hour outage today. I wouldn't call that short. -Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] ] On Behalf Of AndresPaglayanSent: Tuesday, July 11, 2006 1:58 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Provider UNREACHABLEthey had a short outage today, it was fixed already,dunno if related to your issue,___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Managerhttp://www.theclubvoip.comMaking it happen 1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Provider UNREACHABLE
On Tue, 2006-07-11 at 14:10 -0400, Rick Smith wrote: teliax had a 2.5 hour outage today. I wouldn't call that short. its all relative, nufone had a 30 day outage :P -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk stops abruptly
Thanks, got it. -Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez Sent: Tuesday, July 11, 2006 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk stops abruptly no. just logger reload On 7/11/06, Dan Brummer [EMAIL PROTECTED] wrote: Thank you for the quick response. I assume this change will require an Asterisk reload? Thanks! -Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Tuesday, July 11, 2006 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk stops abruptly On Tue, 2006-07-11 at 08:20 -0700, Dan Brummer wrote: Hello, I'm recently having the problem where Asterisk just stops working. The console gets disconnected and the process appears to die. I am using Asterisk version 1.2.9.1. Anyone have any ideas on where I should be looking for the cause of my problem? Also, I notice there is a /var/log/asterisk/messages log file but it doesn't contain any information that I can use to help troubleshoot the application crashing. Is there a way to put more debugging in the log file? Yes take a look at logger.conf. There is a default of 'full' which will create /var/log/asterisk/full for example, and will have more info, but you can add the individual elements to the messages one if you would rather. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server redundancy
#1.. most the failures and network bottle necks on asterisk in a 1k + user sip /iax are registrations polling'syou are right .. get SER ... dont be dumb.#2 the config file with asterisk hardcode ips is a simple matter of running a script that parses it and puts in whatever it needs #3 basic failover will actually steal the ip form other box.. wich in this case would steal ip of down box.#3b.. we need multiple listening addies.. since asterisk can only listen to one ip its sucks for now On 7/11/06, Alejandro Acosta [EMAIL PROTECTED] wrote: What about having the softphone/hardphones configured with a Main Asterisk server and an Alternative Asterisk Server?.I just did a couple of tests with a Cisco ATA 186 and worked quit well after changing the RegInterval and Alternative Proxy Timeout. thks,Alejandro,On Tuesday 11 July 2006 05:04 am, unplug wrote: I have asked about it here. As Douglas said, it doesn't support mult-asterisk in current version. However, I have questions about why multi-asterisk so difficult to implement. 1. As we can use ARA to store all information, sip user register info, dial plan ... to DB.All asterisks can use ARA to refer to the DB for necessary information even register information. 2. What is mean by multiple Asterisk systems can't reference the same MySQL database for SIP peers.?Does SIP peer information also store in DB? 3.Any difficulty to implement multiple asterisk? 4. If I want to implement multiple asterisk in some extent, how do I begin?Any reference? On 7/11/06, RR [EMAIL PROTECTED] wrote: Interesting points on both messages 1) as far as multiple asterisk servers talking to the same database is concerned, I will have to test this out. I know nothing about the database side of things, and a newbie on asterisk and linux so I have no idea what and where the development of either of these are. From your message it sounds like it's just how ARA is designed because I doubt it's to do with the ODBC driver itself. This will cause me a lot of grief if you're right about this for multiple * servers to not be able to access the same database for peer lookup. 2) Clustering of DB isn't an issue, not for me at least. Haven't tested this either but my DBs are clustered A/P providing a single entity to the internal systems. Might further look into a local DNS lookup to add to this. I believe it's possible to do this in the MySQL world with MySQL grid etc? 3) I don't believe frequent registration is that big of an issue for the network load it generates. Most providers out there set devices for a 30-40secs Reg. Refresh to support NAT'ed endpoints and the a reg refresh is hardly about 300-400Byte pkts (I think). The math doesn't add up for a major load esp. if you've got a load balancing mechanism in front of your * boxes. 4) I don't know enough about DUNDi to get into this discussion but DUNDi just lookup extensions? or it also have any part to play in registrations? If they just do extension lookup, then If DUNDi is implemented on an A/P pair of dedicated DUNDi lookup servers which access a clustered database, then barring #1 being true, each * server accesses the same database and pool of registrations. If registrations are refreshed frequently enough, the contact info in the database will always be current and one server dying won't affect anything. At the same time, they just consult the DUNDi lookup server for extension lookups instead of asking the database directly. 5) If you really want to improve on this, supplement your network with SER as proxies and have them deal with Registrations and load-balance feature requests to * servers etc. Once * has done whatever it needs to do (e.g. provide PBX features, voicemail, conference, IVR etc.) it passes the call back to the Proxy to deal with the endpoints. All depends on your scope and budget. If you want to have a SP grade service then you need to breakout your functions. I just hope #1 isn't true though. The only alternative then would be to have /etc/asterisk reside on an NFS share or a CFS for all servers to read massively huge conf files if you're catering for large number of endpoints. Dunno if it helps anyone or I'm just shooting sh*t ;) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MikeSales Managerhttp://www.theclubvoip.comMaking it
[asterisk-users] So many configuration files!
I'm working with Asterisk 1.2.5 to get a working system. There are 50 Asterisk configuration files in /etc/asterisk. Are they _all_ called by Asterisk or are some only used in a #include? Is there any way to get a list of which ones Asterisk uses by default? There is only a single #include file and it doesn't even exist. I have only messed with 4 files so far. Are there any more I should be editing? Which ones could be safely ignored? So far the system is just SIP with Zaptel to be added next. The 4 files I have changed are: sip.conf extensions.conf extensions_additional.conf voicemail.conf My list of files in /etc/asterisk - sorted most recent last: [EMAIL PROTECTED] asterisk # ls -1tr zapata.conf vpb.conf telcordia-1.adsi skinny.conf sip_notify.conf rtp.conf rpt.conf res_odbc.conf queues.conf privacy.conf phone.conf oss.conf osp.conf musiconhold.conf modules.conf modem.conf misdn.conf mgcp.conf meetme.conf manager.conf logger.conf indications.conf iaxprov.conf iax.conf festival.conf features.conf extensions.ael extconfig.conf enum.conf dundi.conf dnsmgr.conf codecs.conf cdr_tds.conf cdr_pgsql.conf cdr_odbc.conf cdr_manager.conf cdr_custom.conf cdr.conf asterisk.conf asterisk.adsi alsa.conf alarmreceiver.conf agents.conf adtranvofr.conf adsi.conf sip.conf extensions.conf extensions_additional.conf voicemail.conf -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Slackware Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues with making Transfers
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dan Brummer wrote: Asterisk 1.2.9.1 is the version I'm on. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Dan Brummer *Sent:* Tuesday, July 11, 2006 8:30 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Issues with making Transfers Hello, I am having a problem with transferring calls that come in from the outside. Users have been calling in to the PRI that's on the Cisco GW, then they are passed into Asterisk via SIP and to the end phone (Polycom 501/601) using SIP. When that user tries to transfer that call to another extension, the call disconnects and hangs in the air and doesn't do anything. The call shows active in the Cisco GW but no where to be found in asterisk. Here is some log output of a transfer attempt: -- Stopped music on hold on SIP/10.25.118.2-b7b4e520 == Spawn extension (ANC, 4023, 2) exited non-zero on 'SIP/4023-ebbfZOMBIE' -- SIP/2198-3780 answered SIP/10.25.118.2-b7b4e520 -- Attempting native bridge of SIP/10.25.118.2-b7b4e520 and SIP/2198-3780 -- Incoming call: Got SIP response 500 Internal Server Error back from 10.45.25.12 I'm not sure if the SIP 500 error is relative to my issue. Any ideas on what could be causing SIP transfers to hang or drop? Thank you, Dan Interestingly, we saw a very similar issue with 1.2.9.1 and Cisco 7960s (SIP 8.2 fw) and HFC BRI ISDN cards last week, I went back to 1.2.7 and every thing seems fine now. - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRLP1n0tP/KMNOfRbAQI86gf7BW1G8CmMzOo3O3Wu200gFGlYEUwbc+8Q tk39rot1H6Bus0O0qPNoSAgJyxWp5617urprU9th2hreRjh5r3Cb3MOIfDuhCm2W p7b1UyVhFZaehWy8ketykld1mvV5eCBBCu9aKYINRS4aEAx7Snt3txLEB5x1bA7A 7N97O/h821iqR79fTuhBD8GMOF0dwaVmJ8oAeeUZoR+YXngMGt2pXQCL7LyLMmT2 vNgusL28J4Cmw76sHuuXEQ8W/t1ONT7WPWWwj/TNFzIqGvl4dCeMC5yN4XtHnXp0 l7gYY9qoFxjD4Z6sXzqiETqFsuqsZygDhsqdMg5CaaUEbi94uERMFQ== =0k5S -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rate or rank ITSP
i'll be very interested in that it would also be useful that every qos rate comply with some deterministic criteria also, imho, keep in mind that a qos rating should be given on provider +destination country because in my experience, qos varies very much depending on which destination country you are calling that would add a lot of work to the list, which should be heavily community driven of course a 'resuming' score for each provider would be more readable someone have experience on determining an 'mean opinion score' value with asterisk + some software solution ? i've been messing with app_milliwatt but my know-how is scarying empty .mike On Tue, 2006-07-11 at 09:40 -0400, Barry Fawthrop wrote: Hi There I know of wiki there is a list of VOIP providers, but is there a list or can we create / suggest one that will list VoIP providers, their location and quality of service ? Too me this will be very valuable, plus looking at some of the requests of late I'm sure others would like that too? Thanks Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rate or rank ITSP
something that you could drill into.. or even search.. hold on mate i got this..how about a master LCR system that would generate config for users in terms of filters..EX1: filter on qoswould return BEST QOS list of all terminations for providers
like..provider1=sip/[EMAIL PROTECTED]etc514XXX,1,dial(${provider1})etc..but in regards to qos..filter by rate you would get a conf file listed with all providers rates being lowest for each country
so basically you would get a config file dpeending on filters.. thing is you need accounts on all these providers.. so links to the signup on them ? that would be bad.. imagine keeping a 20$ balance on 100 providers .. ;)
the point is we can't uatomate this i think.. unless we keep this down to 5-6 providers..On 7/11/06, mike
[EMAIL PROTECTED] wrote:i'll be very interested in thatit would also be useful that every qos rate comply with some
deterministic criteriaalso, imho, keep in mind that a qos rating should be given on provider+destination countrybecause in my experience, qos varies very much depending on whichdestination country you are calling
that would add a lot of work to the list, which should be heavilycommunity drivenof course a 'resuming' score for each provider would be more readablesomeone have experience on determining an 'mean opinion score' value
with asterisk + some software solution ?i've been messing with app_milliwatt but my know-how is scarying empty.mikeOn Tue, 2006-07-11 at 09:40 -0400, Barry Fawthrop wrote: Hi There I know of wiki there is a list of VOIP providers, but is there a list or
can we create / suggest one that will list VoIP providers, their location and quality of service ? Too me this will be very valuable, plus looking at some of the requests of late I'm sure others would like that too?
Thanks Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list
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re: [asterisk-users] So many configuration files!
first: download the latest version 1.2.5 had some bugs and is already several months old.Depending on how you want your asterisk to behave will be the amount of files you'll need to mess with. Let's say you want a very basic installation with some SIP phones (hard or soft), then you'll have to deal with sip.conf and extensions.conf only so everything else is just vanity :DUsually you would like to have some voicemail, conference rooms, music on hold, pick up your neighbours extension, dial another asterisk or an IAX softphone, and PSTN access, then change some configs in voicemail.conf, meetme.conf, musiconhold.conf, features.conf, iax.conf and zapata.conf respectively.Want more action?Manage asterisk from an external application and mess with manager.conf, change the way logs are being saved and CRMs with logger.conf and the cdr_ *.conf files, try some text to speach (TTS) with festival.confFeel like you are in the right track?try dealing with any ".c" file, recompile asterisk and make it behave just the way you always dream of (btw if it works you might want to share your new feature with all of us :) )Alyed Return-Path: [EMAIL PROTECTED] Tue Jul 11 11:51:59 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Tue, 11 Jul 2006 11:51:59 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) I'm working with Asterisk 1.2.5 to get a working system.There are 50 Asterisk configuration files in /etc/asterisk.Are they _all_ called by Asterisk or are some only used in a #include?Is there any way to get a list of which ones Asterisk uses by default?There is only a single #include file and it doesn't even exist.I have only messed with 4 files so far.Are there any more I should be editing?Which ones could be safely ignored?So far the system is just SIP with Zaptel to be added next.The 4 files I have changed are:sip.confextensions.confextensions_additional.confvoicemail.confMy list of files in /etc/asterisk - sorted most recent last:[EMAIL PROTECTED] asterisk # ls -1trzapata.confvpb.conftelcordia-1.adsiskinny.confsip_notify.confrtp.confrpt.confres_odbc.confqueues.confprivacy.confphone.confoss.confosp.confmusiconhold.confmodules.confmodem.confmisdn.confmgcp.confmeetme.confmanager.conflogger.confindications.confiaxprov.confiax.conffestival.conffeatures.confextensions.aelextconfig.confenum.confdundi.confdnsmgr.confcodecs.confcdr_tds.confcdr_pgsql.confcdr_odbc.confcdr_manager.confcdr_custom.confcdr.confasterisk.confasterisk.adsialsa.confalarmreceiver.confagents.confadtranvofr.confadsi.confsip.confextensions.confextensions_additional.confvoicemail.conf-- Larry Alkoff N2LA - Austin TXUsing Thunderbird on Slackware Linux___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rate or rank ITSP
There is also the issue of origin of the caller. Not just geographically, but which network provider they use, and to some degree when. Most ISPs see higher traffic volumes when school gets out for the day (abnout 3pm) continuing for a few hours, then gradually declining until its later (11pm or so). As a result, a call made at 10am may have totally different characteristics from one made at 6pm (when both school kids and working adults tend to get home). As such the problem may not be the VoIP provider but with the network of the person sending the call, or a tertiary provider in between the two. Therefore I suggest that if such a list is done, people will include who their provider is, and whether they have had problems in the mornings, evenings, dead of night, etc. That way the list can be as fair as possible to all providers. And if you can search for providers from your geographic location off your provider, you can filter out the ones that are known to be bad from your location and network. On Tue, 2006-07-11 at 21:03 -0400, mike wrote: i'll be very interested in that it would also be useful that every qos rate comply with some [snip] On Tue, 2006-07-11 at 09:40 -0400, Barry Fawthrop wrote: Hi There I know of wiki there is a list of VOIP providers, but is there a list or can we create / suggest one that will list VoIP providers, their location and quality of service ? -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] So many configuration files!
larry each of these files do something for a specific needs.. hence the sip.conf is for sip related modules..iax etc etc..voicemail.conf if you need voicemailres_odbc etc for any database usage.. basically read the manual and look into each files to see what they do..asterisk will start and work without modding all of these but you could have a surprise if a demo user is by default in sip .conf and someone uses your system ;) On 7/11/06, Larry Alkoff [EMAIL PROTECTED] wrote: I'm working with Asterisk 1.2.5 to get a working system.There are 50 Asterisk configuration files in /etc/asterisk.Are they _all_ called by Asterisk or are some only used in a #include?Is there any way to get a list of which ones Asterisk uses by default? There is only a single #include file and it doesn't even exist.I have only messed with 4 files so far.Are there any more I should be editing?Which ones could be safely ignored?So far the system is just SIP with Zaptel to be added next. The 4 files I have changed are:sip.confextensions.confextensions_additional.confvoicemail.confMy list of files in /etc/asterisk - sorted most recent last: [EMAIL PROTECTED] asterisk # ls -1trzapata.confvpb.conftelcordia-1.adsiskinny.confsip_notify.confrtp.confrpt.confres_odbc.confqueues.confprivacy.confphone.confoss.confosp.conf musiconhold.confmodules.confmodem.confmisdn.confmgcp.confmeetme.confmanager.conflogger.confindications.confiaxprov.confiax.conffestival.conffeatures.confextensions.ael extconfig.confenum.confdundi.confdnsmgr.confcodecs.confcdr_tds.confcdr_pgsql.confcdr_odbc.confcdr_manager.confcdr_custom.confcdr.confasterisk.confasterisk.adsialsa.conf alarmreceiver.confagents.confadtranvofr.confadsi.confsip.confextensions.confextensions_additional.confvoicemail.conf--Larry Alkoff N2LA - Austin TXUsing Thunderbird on Slackware Linux ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Manager http://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Provider UNREACHABLE
trxtel ping me.On 7/11/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Tue, 2006-07-11 at 14:10 -0400, Rick Smith wrote: teliax had a 2.5 hour outage today. I wouldn't call that short.its all relative, nufone had a 30 day outage :P--Trixter http://www.0xdecafbad.com Bret McDanelBelfast IE +44 28 9099 6461DE +49 801 777 555 3402Utrecht NL +31 306 553058US WA +1 360 207 0479US NY +1 516 687 5200FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you!-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.3 (GNU/Linux)iD8DBQBEs+72+1olxlzQw5cRApVnAKC4ob9F2SZDeU2DidVLwG7YK/xOlwCgrsOF BNqx9bUsHGBWeNCJUumQgdE==1VDH-END PGP SIGNATURE-___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MikeSales Managerhttp://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Issues with making Transfers
Thank you for the response, I will try downgrading. -Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Wellsted Sent: Tuesday, July 11, 2006 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Issues with making Transfers -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dan Brummer wrote: Asterisk 1.2.9.1 is the version I'm on. -- -- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Dan Brummer *Sent:* Tuesday, July 11, 2006 8:30 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Issues with making Transfers Hello, I am having a problem with transferring calls that come in from the outside. Users have been calling in to the PRI that's on the Cisco GW, then they are passed into Asterisk via SIP and to the end phone (Polycom 501/601) using SIP. When that user tries to transfer that call to another extension, the call disconnects and hangs in the air and doesn't do anything. The call shows active in the Cisco GW but no where to be found in asterisk. Here is some log output of a transfer attempt: -- Stopped music on hold on SIP/10.25.118.2-b7b4e520 == Spawn extension (ANC, 4023, 2) exited non-zero on 'SIP/4023-ebbfZOMBIE' -- SIP/2198-3780 answered SIP/10.25.118.2-b7b4e520 -- Attempting native bridge of SIP/10.25.118.2-b7b4e520 and SIP/2198-3780 -- Incoming call: Got SIP response 500 Internal Server Error back from 10.45.25.12 I'm not sure if the SIP 500 error is relative to my issue. Any ideas on what could be causing SIP transfers to hang or drop? Thank you, Dan Interestingly, we saw a very similar issue with 1.2.9.1 and Cisco 7960s (SIP 8.2 fw) and HFC BRI ISDN cards last week, I went back to 1.2.7 and every thing seems fine now. - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRLP1n0tP/KMNOfRbAQI86gf7BW1G8CmMzOo3O3Wu200gFGlYEUwbc+8Q tk39rot1H6Bus0O0qPNoSAgJyxWp5617urprU9th2hreRjh5r3Cb3MOIfDuhCm2W p7b1UyVhFZaehWy8ketykld1mvV5eCBBCu9aKYINRS4aEAx7Snt3txLEB5x1bA7A 7N97O/h821iqR79fTuhBD8GMOF0dwaVmJ8oAeeUZoR+YXngMGt2pXQCL7LyLMmT2 vNgusL28J4Cmw76sHuuXEQ8W/t1ONT7WPWWwj/TNFzIqGvl4dCeMC5yN4XtHnXp0 l7gYY9qoFxjD4Z6sXzqiETqFsuqsZygDhsqdMg5CaaUEbi94uERMFQ== =0k5S -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rate or rank ITSP
I think if someone put together a site like dslreports that allowed people to rate their experince with a ITSP and record the latency, carrier/network etc . . . then it would be helpful in shopping for an ITSP. On 7/11/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: There is also the issue of origin of the caller.Not justgeographically, but which network provider they use, and to some degreewhen.Most ISPs see higher traffic volumes when school gets out for theday (abnout 3pm) continuing for a few hours, then gradually declining until its later (11pm or so).As a result, a call made at 10am may havetotally different characteristics from one made at 6pm (when both schoolkids and working adults tend to get home).As such the problem may not be the VoIP provider but with the network of the person sending the call, or a tertiary provider in between the two.Therefore I suggest that if such a list is done, people will include whotheir provider is, and whether they have had problems in the mornings, evenings, dead of night, etc.That way the list can be as fair aspossible to all providers.And if you can search for providers from your geographic location offyour provider, you can filter out the ones that are known to be bad from your location and network.On Tue, 2006-07-11 at 21:03 -0400, mike wrote: i'll be very interested in that it would also be useful that every qos rate comply with some[snip] On Tue, 2006-07-11 at 09:40 -0400, Barry Fawthrop wrote: Hi There I know of wiki there is a list of VOIP providers, but is there a list or can we create / suggest one that will list VoIP providers, their location and quality of service ?--Trixter http://www.0xdecafbad.com Bret McDanelBelfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058US WA +1 360 207 0479US NY +1 516 687 5200FreeWorldDialup: 635378http://www.trxtel.com the VoIP provider that pays you!-BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux)iD8DBQBEs/v9+1olxlzQw5cRAgOUAKCXio6U31gGpKreljvDuWSGh6y6XQCgi5MVC9nZ981R5GgXxrhgMa1Aee4==bYDy-END PGP SIGNATURE-___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rate or rank ITSP
i think that setting up a qos shaper on the tester gateway could provide a reasonable testing environment not mentioning delay, on worst case you will need 8KB/s (ulaw) you of course are right about origination/network condition but the effort given from such a list could be great not to mention the idea given before from Mike (automation of such list in asterisk) which is something genius! On Tue, 2006-07-11 at 12:29 -0700, trixter aka Bret McDanel wrote: There is also the issue of origin of the caller. Not just geographically, but which network provider they use, and to some degree when. Most ISPs see higher traffic volumes when school gets out for the day (abnout 3pm) continuing for a few hours, then gradually declining until its later (11pm or so). As a result, a call made at 10am may have totally different characteristics from one made at 6pm (when both school kids and working adults tend to get home). As such the problem may not be the VoIP provider but with the network of the person sending the call, or a tertiary provider in between the two. Therefore I suggest that if such a list is done, people will include who their provider is, and whether they have had problems in the mornings, evenings, dead of night, etc. That way the list can be as fair as possible to all providers. And if you can search for providers from your geographic location off your provider, you can filter out the ones that are known to be bad from your location and network. On Tue, 2006-07-11 at 21:03 -0400, mike wrote: i'll be very interested in that it would also be useful that every qos rate comply with some [snip] On Tue, 2006-07-11 at 09:40 -0400, Barry Fawthrop wrote: Hi There I know of wiki there is a list of VOIP providers, but is there a list or can we create / suggest one that will list VoIP providers, their location and quality of service ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inconsistent call detail records
Hi list: I need some help to resolve this situation. I placed a call from one phone to a queue, which the queue has one static member. I got two cdr records for each call, but the duration and billsec have two different pattens, I don't know what is the control for this situation. I'd like to get one format, but don't know how. Please help. For example, I list the information for two different calls: Call # 1's two records (partial fields): calldate src dst dcontext lastapplastdataduration billsec uniqueid 2006-07-10 16:02:4428263666 ext-ququesqueue 3666|t|| 11 11 1152572564.24 2006-07-10 16:02:4428268666 from-internal dial SIP/8666|15|tr 138 127 1152572564.26 Call # 2's two records calldate src dst dcontext lastapplastdataduration billsec uniqueid 2006-07-11 11:09:4128263666 ext-ququesqueue 3666|t|| 72 72 1152641381.84 2006-07-11 11:09:4128268666 from-internal dial SIP/8666|15|tr 80 1152641381.86 Many thanks. Tielin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR Call Status
Hi List, When using cdr_csv, the call status are plain strings, i.e. NO ANSWER, ANSWERED, BUSY, etc. However, when using cdr_odbc, the call status are integer. Is there some docs somewhere that would let me know what the integers map to? Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom ACD, Asterisk, Kernel 2.6
I'm trying to build another asterisk server as I'm having a problem with the current one. Unless anybody can tell me how to compile the meetme app? Everything else works fine, asterisk just will not compile meetme?!? (Under kernel 2.4) I used svn to pull the trunk versions of libpri, zaptel and the polycom_acd_functions (release 30432). I cannot seem to get the zaptel to compile under 2.6, is this correct? Does it only work on 2.4? Is there any release I should be pulling for the zaptel (and libpri)? Or does anybody haveit working or knowa release version that I could pull? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_conference DTMFs?
henry, did you have any luck setting this up? i'm actually working right now to _suppress_ dtmf clicks in app_conference, and would be happy to look at the dtmf pass-through, if you're still in need. j- On 5/29/06, Henry J. Cobb [EMAIL PROTECTED] wrote: We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server redundancy
On Tue, 2006-07-11 at 14:34 -0400, Mike Lynchfield wrote: #3b.. we need multiple listening addies.. since asterisk can only listen to one ip its sucks for now Incorrect. Asterisk most definitely listens on multiple interfaces. We've got several asterisk boxes that are multi-homed... one public and one private interface, so that we can have external phones and internal phones. Works fine. I'm thinking this is a misconception. We even have heartbeat set up to switch ip's around. The server actually listens on the fly to the new ip address that comes up under it. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Issues with making Transfers
This has worked. I downgraded from 1.2.9.1 to 1.2.7.1 and I'm not having the warm transfer issue anymore. Does anyone know if this is a known issue and is going to be fixed in upcoming release? Should I possibly put in a bug request? -Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Wellsted Sent: Tuesday, July 11, 2006 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Issues with making Transfers -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dan Brummer wrote: Asterisk 1.2.9.1 is the version I'm on. -- -- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Dan Brummer *Sent:* Tuesday, July 11, 2006 8:30 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Issues with making Transfers Hello, I am having a problem with transferring calls that come in from the outside. Users have been calling in to the PRI that's on the Cisco GW, then they are passed into Asterisk via SIP and to the end phone (Polycom 501/601) using SIP. When that user tries to transfer that call to another extension, the call disconnects and hangs in the air and doesn't do anything. The call shows active in the Cisco GW but no where to be found in asterisk. Here is some log output of a transfer attempt: -- Stopped music on hold on SIP/10.25.118.2-b7b4e520 == Spawn extension (ANC, 4023, 2) exited non-zero on 'SIP/4023-ebbfZOMBIE' -- SIP/2198-3780 answered SIP/10.25.118.2-b7b4e520 -- Attempting native bridge of SIP/10.25.118.2-b7b4e520 and SIP/2198-3780 -- Incoming call: Got SIP response 500 Internal Server Error back from 10.45.25.12 I'm not sure if the SIP 500 error is relative to my issue. Any ideas on what could be causing SIP transfers to hang or drop? Thank you, Dan Interestingly, we saw a very similar issue with 1.2.9.1 and Cisco 7960s (SIP 8.2 fw) and HFC BRI ISDN cards last week, I went back to 1.2.7 and every thing seems fine now. - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRLP1n0tP/KMNOfRbAQI86gf7BW1G8CmMzOo3O3Wu200gFGlYEUwbc+8Q tk39rot1H6Bus0O0qPNoSAgJyxWp5617urprU9th2hreRjh5r3Cb3MOIfDuhCm2W p7b1UyVhFZaehWy8ketykld1mvV5eCBBCu9aKYINRS4aEAx7Snt3txLEB5x1bA7A 7N97O/h821iqR79fTuhBD8GMOF0dwaVmJ8oAeeUZoR+YXngMGt2pXQCL7LyLMmT2 vNgusL28J4Cmw76sHuuXEQ8W/t1ONT7WPWWwj/TNFzIqGvl4dCeMC5yN4XtHnXp0 l7gYY9qoFxjD4Z6sXzqiETqFsuqsZygDhsqdMg5CaaUEbi94uERMFQ== =0k5S -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Server redundancy
[EMAIL PROTECTED] wrote: On Tue, 2006-07-11 at 14:34 -0400, Mike Lynchfield wrote: #3b.. we need multiple listening addies.. since asterisk can only listen to one ip its sucks for now That was case for asterisk 1.x ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Server redundancy
I think it can listen either on a specific address, or on ALL addresses, not on a subset of available addresses. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 11, 2006 2:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Server redundancy On Tue, 2006-07-11 at 14:34 -0400, Mike Lynchfield wrote: #3b.. we need multiple listening addies.. since asterisk can only listen to one ip its sucks for now Incorrect. Asterisk most definitely listens on multiple interfaces. We've got several asterisk boxes that are multi-homed... one public and one private interface, so that we can have external phones and internal phones. Works fine. I'm thinking this is a misconception. We even have heartbeat set up to switch ip's around. The server actually listens on the fly to the new ip address that comes up under it. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
