Re: [asterisk-users] SIP/IAX peers UNREACHABLE and audio loss
Well, we have add similar issues - do you use a media gateway /.IP Phones / softphones as your extensions? We were running Audiocodes and for some reason (I suspect a poor ethernet switch), when there are more than 15 people using the line, Audiocodes will not respond to a qualify and asterisk will drop the call. Turned off qualify (removed qualify=yes) and still keeping fingers crossed things seem fine. Rajeev On 3/23/07, Edoardo Serra [EMAIL PROTECTED] wrote: Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently randomly, everything goes ok for days and bad for another few days. I strongly believe the 2 problems are strictly related because in the logs I see REACHABLE / UNREACHABLE messages only for certains days without regularity. The days in wich i see a lot of messages are exactly the days with most of complaint about audio loss I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE) are quite always during business hours, this makes me think at somewhat related to load (cpu load, badwidth load, calls load, etc...) But, looking at hardware specs of our lan, servers and average load I don't think they are over-stressed. Our servers are all: 2 x Intel(R) Xeon(TM) CPU 3.20GHz 1 GB RAM 2 x IDE HDDs Software RAID 1 Asterisk 1.2.13 with res_perl Gentoo Linux Some of them has a Sangoma card connected with an E1 Most ot these are on the same LAN, interconnected with a 1 GB switch (I don't think it should be a bandwidth problem). Load averages of these server is varying from 0.5 to 1.0 (I guess it should be ok) On each server we don't have more than 50 concurrent calls (bridged SIP - IAX2 or IAX2 - ZAP) Used codec is mostly G729 Sometimes on asterisk cli i see some messages like Avoided initial deadlock for '0x9fd130', 10 retries! I don't know if it could be somehow related. Someone of you can point me in the right direction ? Tnx in advance Regards Ing. Edoardo Serra WeBRainstorm S.r.l. ___ --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] SIP/IAX peers UNREACHABLE and audio loss
Hi men, I have already encountered some issue like this with few switches (very known great brand) which doesn't like VoIP traffic ! Check by drectly connected the VoIP equipment - if you can - with temporary long Ethernet cables bypassing the tested switch to see what happens in this case. You can also tell to qualify with a longer delay, but this could not help in case of regulary frames losses. Good luck ! Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Rajeev Natarajan Envoyé : samedi 24 mars 2007 08:14 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] SIP/IAX peers UNREACHABLE and audio loss Well, we have add similar issues - do you use a media gateway /.IP Phones / softphones as your extensions? We were running Audiocodes and for some reason (I suspect a poor ethernet switch), when there are more than 15 people using the line, Audiocodes will not respond to a qualify and asterisk will drop the call. Turned off qualify (removed qualify=yes) and still keeping fingers crossed things seem fine. Rajeev On 3/23/07, Edoardo Serra [EMAIL PROTECTED] wrote: Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently randomly, everything goes ok for days and bad for another few days. I strongly believe the 2 problems are strictly related because in the logs I see REACHABLE / UNREACHABLE messages only for certains days without regularity. The days in wich i see a lot of messages are exactly the days with most of complaint about audio loss I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE) are quite always during business hours, this makes me think at somewhat related to load (cpu load, badwidth load, calls load, etc...) But, looking at hardware specs of our lan, servers and average load I don't think they are over-stressed. Our servers are all: 2 x Intel(R) Xeon(TM) CPU 3.20GHz 1 GB RAM 2 x IDE HDDs Software RAID 1 Asterisk 1.2.13 with res_perl Gentoo Linux Some of them has a Sangoma card connected with an E1 Most ot these are on the same LAN, interconnected with a 1 GB switch (I don't think it should be a bandwidth problem). Load averages of these server is varying from 0.5 to 1.0 (I guess it should be ok) On each server we don't have more than 50 concurrent calls (bridged SIP - IAX2 or IAX2 - ZAP) Used codec is mostly G729 Sometimes on asterisk cli i see some messages like Avoided initial deadlock for '0x9fd130', 10 retries! I don't know if it could be somehow related. Someone of you can point me in the right direction ? Tnx in advance Regards Ing. Edoardo Serra WeBRainstorm S.r.l. ___ --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 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] GTalk/Jabber passing audio in 1.4.1!
On Tue, Mar 06, 2007 at 11:03:33PM -0500, Ronald Lewis wrote: I've just compiled Asterisk 1.4.1 and I'm happy to report that I've got two-way audio between Google Talk and Asterisk! This IS an exciting moment today in VoIP! This is just GREAT! No such luck here. I tried 1.4.1 and 1.4.2 and I have no indication to the fact that Asterisk has successfully connected. Relevant configurations: ;; ;;/etc/asterisk/gtalk.conf: ;; [general] ; over-permissive. But I'm more worried about outgoing calls now. context=from-pstn allowguest=yes ; Based on a sample in voip-info. [asterisk] type=client serverhost=talk.google.com ; I never used gtalk, and hence /GTalk does not exist [EMAIL PROTECTED]/Gaim secret=xx port=5222 usetls=yes usesasl=yes [EMAIL PROTECTED] statusmessage=Asterisk on Tzafrir's laptop timeout=100 ;; ;;In /etc/asterisk/extensions.conf: ;; exten = nickname,1,Dial(gtalk/asterisk/[EMAIL PROTECTED]) I dial from kiax, and this is what I see in the logs: [Mar 24 10:42:27] DEBUG[27181] pbx.c: Launching 'Dial' [Mar 24 10:42:27] VERBOSE[27181] logger.c: -- Executing [EMAIL PROTECTED]: 1] Dial(IAX2/601-1, gtalk/asterisk/[EMAIL PROTECTED]) in new stack [Mar 24 10:42:27] WARNING[27181] chan_gtalk.c: Could not find recipient. [Mar 24 10:42:27] WARNING[27181] app_dial.c: Unable to create channel of type 'gtalk' (cause 0 - Unknown) I see in a different Jabber client that [EMAIL PROTECTED] is on-line. [EMAIL PROTECTED] never appears to be on-line. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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: SIP/IAX peers UNREACHABLE and audio loss
Hi Rajeev, Rajeev Natarajan ha scritto: Well, we have add similar issues - do you use a media gateway /.IP Phones / softphones as your extensions? The problem happens mainly between server with Asterisks ! We were running Audiocodes and for some reason (I suspect a poor ethernet switch), when there are more than 15 people using the line, Audiocodes will not respond to a qualify and asterisk will drop the call. Does Asterisk drop the line if the peer becomes UNREACHABLE ? Even if RTP is still flowing ?? Turned off qualify (removed qualify=yes) and still keeping fingers crossed things seem fine. I'll give it a try Tnx for help Edoardo Rajeev On 3/23/07, *Edoardo Serra* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently randomly, everything goes ok for days and bad for another few days. I strongly believe the 2 problems are strictly related because in the logs I see REACHABLE / UNREACHABLE messages only for certains days without regularity. The days in wich i see a lot of messages are exactly the days with most of complaint about audio loss I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE) are quite always during business hours, this makes me think at somewhat related to load (cpu load, badwidth load, calls load, etc...) But, looking at hardware specs of our lan, servers and average load I don't think they are over-stressed. Our servers are all: 2 x Intel(R) Xeon(TM) CPU 3.20GHz 1 GB RAM 2 x IDE HDDs Software RAID 1 Asterisk 1.2.13 with res_perl Gentoo Linux Some of them has a Sangoma card connected with an E1 Most ot these are on the same LAN, interconnected with a 1 GB switch (I don't think it should be a bandwidth problem). Load averages of these server is varying from 0.5 to 1.0 (I guess it should be ok) On each server we don't have more than 50 concurrent calls (bridged SIP - IAX2 or IAX2 - ZAP) Used codec is mostly G729 Sometimes on asterisk cli i see some messages like Avoided initial deadlock for '0x9fd130', 10 retries! I don't know if it could be somehow related. Someone of you can point me in the right direction ? Tnx in advance Regards Ing. Edoardo Serra WeBRainstorm S.r.l. ___ --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 ___ --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] Re: RE : SIP/IAX peers UNREACHABLE and audio loss
Hi Francois, [EMAIL PROTECTED] ha scritto: Hi men, I have already encountered some issue like this with few switches (very known great brand) which doesn't like VoIP traffic ! I also have switches of a very known great brand !! It was so strange to me that I didn't consider a network problem... Check by drectly connected the VoIP equipment - if you can - with temporary long Ethernet cables bypassing the tested switch to see what happens in this case. I'd try to bypass the switch someway but every server neeeds to have its own public ip address.. I'll put an RTP proxy somewhere... You can also tell to qualify with a longer delay, but this could not help in case of regulary frames losses. What about turning qualify off ? Do you think taht Asterisk is stopping RTP when it loose a qualify packet ? Or is the RTP traffic itself that is lost by the switches ? Good luck ! It couldn't be more appropriate... Tnx for help ;) Edoardo Francois BERGERET, France. -Message d'origine- *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Rajeev Natarajan *Envoyé :* samedi 24 mars 2007 08:14 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* Re: [asterisk-users] SIP/IAX peers UNREACHABLE and audio loss Well, we have add similar issues - do you use a media gateway /.IP Phones / softphones as your extensions? We were running Audiocodes and for some reason (I suspect a poor ethernet switch), when there are more than 15 people using the line, Audiocodes will not respond to a qualify and asterisk will drop the call. Turned off qualify (removed qualify=yes) and still keeping fingers crossed things seem fine. Rajeev On 3/23/07, *Edoardo Serra* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently randomly, everything goes ok for days and bad for another few days. I strongly believe the 2 problems are strictly related because in the logs I see REACHABLE / UNREACHABLE messages only for certains days without regularity. The days in wich i see a lot of messages are exactly the days with most of complaint about audio loss I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE) are quite always during business hours, this makes me think at somewhat related to load (cpu load, badwidth load, calls load, etc...) But, looking at hardware specs of our lan, servers and average load I don't think they are over-stressed. Our servers are all: 2 x Intel(R) Xeon(TM) CPU 3.20GHz 1 GB RAM 2 x IDE HDDs Software RAID 1 Asterisk 1.2.13 with res_perl Gentoo Linux Some of them has a Sangoma card connected with an E1 Most ot these are on the same LAN, interconnected with a 1 GB switch (I don't think it should be a bandwidth problem). Load averages of these server is varying from 0.5 to 1.0 (I guess it should be ok) On each server we don't have more than 50 concurrent calls (bridged SIP - IAX2 or IAX2 - ZAP) Used codec is mostly G729 Sometimes on asterisk cli i see some messages like Avoided initial deadlock for '0x9fd130', 10 retries! I don't know if it could be somehow related. Someone of you can point me in the right direction ? Tnx in advance Regards Ing. Edoardo Serra WeBRainstorm S.r.l. ___ --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 ___ --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] freepbx - DB Error messages...
Hi all, I am probably missing something ultimately obvious, but I have a problem configuring freepbx... Using Edgy Eft with the cvs freePBX 2.2.1 and followed the Ubuntu installation guide on freepbx.org. System pxe-boots from a server with NFS root on same Using * 1.2 current (from source, not .deb's) Using mISDN-streams (from source, not .deb's) Using freePBX-2.2.1 (from source, not .deb's) Installed everything, and mISDN and * load just fine amportal start works fine as well However I keep getting DB Error's in the GUI... The syslog gives two separate errors: 1) Error 127 when reading table ./asterisk/whatever 2) Table is crashed and needs to be repaired I created a special mysql user for * and did an PERMIT ALL PRIVILEGES on the mysql databases When I log in to mysql as root and do 'SELECT username FROM ampusers ORDER BY username' I get the record list. When I do the same as the * user, I get the 'Table is crashed, blablabla' line. I tried changing the login user for freepbx (ampdbuser) to root, but that doesn't help either, as I keep getting the 127 error... Googling wasn't very helpful, and the freepbx forum admins still haven't approved my account, so I thought I'd try here... Any help appreciated! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] Issue with Hamlet ISDN PCI card(Cologne Chipset)
Hi everybody I have installed a TrixBox with Asterisk 1.2.14 and relative upgreaded software. I Bristuffed it with last version of bristuff to use a Hemlet PCI ISDN CARD in a normal Italian EUROISDN installation. The * works fine except for the ISDN CARD. It is always Channel D down, but if a Call comes in, it works perfectly for some time, both inbound and outbound. It prompts Channel D UP! If I disconnect the NT+ termination the Channel D goes down at once. Did I make something wrong? Best regards Mauro Zanin _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] Issue with Hamlet ISDN PCI card(Cologne Chipset)
On Sat, March 24, 2007 11:54, Mauro Zanin wrote: Hi everybody I have installed a TrixBox with Asterisk 1.2.14 and relative upgreaded software. I Bristuffed it with last version of bristuff to use a Hemlet PCI ISDN CARD in a normal Italian EUROISDN installation. The * works fine except for the ISDN CARD. It is always Channel D down, but if a Call comes in, it works perfectly for some time, both inbound and outbound. It prompts Channel D UP! If I disconnect the NT+ termination the Channel D goes down at once. Did I make something wrong? Not really... It's a bristuff quirk... It doesn't gracefully handle the forced D-channel down that most European ISDN operators implement. That is why I switched to testing vISDN, but that has been stagnant for over half a year without any fixes for a few very annoying bugs, because the programmer dedicated all his time to rewriting the vGSM part... I am now testing mISDN as someone on the vISDN list mentioned that it's chan_misdn voice support had greatly improved... The only way I can *somewhat* keep bristuff working without contacting the ISDN carrier to turn on the D channel permanently is by initiation a 100ms outbound call every minute using the manager interface... (Yes, a very ugly kludge indeed, but I do not want permanent channel up, as I want to be able to test everything in a normal environment, as I am planning to install this in other location too once I have a stable, reliable environment) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] Issue with Hamlet ISDN PCI card(Cologne Chipset)
Hi David and the list, It's normal ;-) Near all European BRI operators cut off the line between calls. So, you must trieve the correct parameter avoiding to survey the line as for mISDN : pmp_l1_check=no I use mISDN without any issue with B410P. I hope this help. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Francesco Peeters (Asterisk) Envoyé : samedi 24 mars 2007 12:40 À : Asterisk Users Mailing List - Non-Commercial Discussion Cc : asterisk-users@lists.digium.com Objet : Re: [asterisk-users] Issue with Hamlet ISDN PCI card(Cologne Chipset) On Sat, March 24, 2007 11:54, Mauro Zanin wrote: Hi everybody I have installed a TrixBox with Asterisk 1.2.14 and relative upgreaded software. I Bristuffed it with last version of bristuff to use a Hemlet PCI ISDN CARD in a normal Italian EUROISDN installation. The * works fine except for the ISDN CARD. It is always Channel D down, but if a Call comes in, it works perfectly for some time, both inbound and outbound. It prompts Channel D UP! If I disconnect the NT+ termination the Channel D goes down at once. Did I make something wrong? Not really... It's a bristuff quirk... It doesn't gracefully handle the forced D-channel down that most European ISDN operators implement. That is why I switched to testing vISDN, but that has been stagnant for over half a year without any fixes for a few very annoying bugs, because the programmer dedicated all his time to rewriting the vGSM part... I am now testing mISDN as someone on the vISDN list mentioned that it's chan_misdn voice support had greatly improved... The only way I can *somewhat* keep bristuff working without contacting the ISDN carrier to turn on the D channel permanently is by initiation a 100ms outbound call every minute using the manager interface... (Yes, a very ugly kludge indeed, but I do not want permanent channel up, as I want to be able to test everything in a normal environment, as I am planning to install this in other location too once I have a stable, reliable environment) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] Asterisk Viruses?
The Skype network is circulating a virus that has appeared there before: http://www.informationweek.com/news/showArticle.jhtml?articleID=198500135 . The virus sends a URL to other Skype users in the infected user's contacts, which the target Skype displays as clickable. Clicking downloads the virus. Asterisk supports features like these, in combination with certain clients (which aren't themselves Asterisk), including IM and URL redirection. Any reports of this kind of attack on Asterisk itself, or using Asterisk to support those potentially vulnerable clients? Any analysis of Asterisk's vulnerability to these? Any mitigations? -- (C) Matthew Rubenstein ___ --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] Shoretel integration into Salesforce.com
http://www.theregister.co.uk/2007/03/24/shortel_sforce Shoretel has integrated its IP phone system with Salesforce.com's call centre software. Using the two together will mean call centre agents get a reduced admin workload, with automatic call logging and screen pop-ups with the customer's record. Is anyone in the asterisk community providing integration into Salesforce.com? Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation www.Mexuar.com http://www.mexuar.com/ Want to voice enable your website? Use Corraleta to reach your customers in 10 seconds or less. image001.gif Description: image001.gif ___ --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] Can be called on FreeWorldDialup/IAX channel, but can't make calls
Hi, I have an FWD account and it's configured in asterisk. I can be called by people using FWD, but I cannot make FWD calls myself. Every number dialed with a 8 prefix goes to FWD, if for example I call the echo servie I get this: Connected to Asterisk 1.2.13 currently running on asterisk (pid = 2865) Verbosity is at least 35 -- Executing SetCallerID(SIP/timothy-08224f08, Het Bos) in new stack -- Executing Dial(SIP/timothy-08224f08, IAX2/814179:[EMAIL PROTECTED]/613|60|r) in new stack -- Called 814179:[EMAIL PROTECTED]/613 Mar 24 15:26:28 NOTICE[2875]: chan_iax2.c:2869 auto_congest: Auto-congesting call due to slow response -- IAX2/192.246.69.186:4569-5 is circuit-busy -- Hungup 'IAX2/192.246.69.186:4569-5' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Congestion(SIP/timothy-08224f08, ) in new stack == Spawn extension (internal, 8613, 3) exited non-zero on 'SIP/timothy-08224f08' Mar 24 15:28:55 ERROR[2873]: chan_sip.c:11076 handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 172.17.249.253, but there is no hint for that extension Who can help me with this? Thank 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] Doorphone vs. Grandstream BT101
Dean Collins wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, 23 March 2007 5:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Doorphone vs. Grandstream BT101 And contrary to what someone asked me in private, wiring isn't an issue -- I do have cat5 at the door bell :) Thanks, JM Like all good geeks should - correct Jay J So did you run two lengths so that you have access to a IP Door camera as well? Don’t forget a few pairs for the electric strike to open the door remotely from a web interface as well. 17,000 ft of Cat5, cat6 and rg6 in the house, somewhere around 120 drops, along with multiple 2 PVC from basement to attic. The front and back door do have dual cat5s, but I'm not planning on a remote door strike for either. CCTV is separate and runs on utp baluns into two 4-channel BT cards, so there's cat5 in all the places where I need (or may later need) a camera. ___ --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 Viruses?
Hi On Sat, Mar 24, 2007 at 09:21:01AM -0400, Matthew Rubenstein wrote: The Skype network is circulating a virus that has appeared there before: http://www.informationweek.com/news/showArticle.jhtml?articleID=198500135 . The virus sends a URL to other Skype users in the infected user's contacts, which the target Skype displays as clickable. Clicking downloads the virus. This is not a skype virus per-se. Skype's instant messanging is used to transfer the URL of the file. According to the description, the user even has to confirm the execusion of the program. If there is an issue here it is with the user interface of the client program or with other parts of the client system. No inherent feature of Skype's protocol is used here. Otherwise it is yet another variation of the stupid programmer virus (I'm a programmer from ___. In my coutry we're still primitive and don't know how to write viruses. So when you get this mesage, please delete some important files and send this message to all the people n your contacts list). Variations on this theme have been available for just about any instant messaging service. Asterisk supports features like these, Sadly, not enough, in combination with certain clients (which aren't themselves Asterisk), including IM and URL redirection. Any reports of this kind of attack on Asterisk itself, or using Asterisk to support those potentially vulnerable clients? This is a purely client issue. Asterisk cannot be expected to filter URLs passing through it (and even if someone would be foolish enough to try to do that, there are enough ways around this. Not the least of them is some trivial javascript redirection). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Issue with Hamlet ISDN PCI card(Cologne Chipset)
On Sat, Mar 24, 2007 at 12:40:24PM +0100, Francesco Peeters (Asterisk) wrote: On Sat, March 24, 2007 11:54, Mauro Zanin wrote: Hi everybody I have installed a TrixBox with Asterisk 1.2.14 and relative upgreaded software. I Bristuffed it with last version of bristuff to use a Hemlet PCI ISDN CARD in a normal Italian EUROISDN installation. The * works fine except for the ISDN CARD. It is always Channel D down, but if a Call comes in, it works perfectly for some time, both inbound and outbound. It prompts Channel D UP! If I disconnect the NT+ termination the Channel D goes down at once. Did I make something wrong? Not really... It's a bristuff quirk... It doesn't gracefully handle the forced D-channel down that most European ISDN operators implement. It's not a bristuff quirck. From my anecdotial knowledge, bristuff/vzaphfc seems to just work there. Maybe the zaphfc driver lacks the support for setting the span up. That is why I switched to testing vISDN, but that has been stagnant for over half a year without any fixes for a few very annoying bugs, because the programmer dedicated all his time to rewriting the vGSM part... I figure vzaphfc may share a bit of code with vISDN ... zaphfc seems to be the least worked-on driver in bristuff. You should really try vzaphfc instead. (/me wonders if bristuff will ever get to z) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Doorphone vs. Grandstream BT101
Jay Milk wrote: Doug Lytle wrote: Jay Milk wrote: I've done all the googling I can on this, and have come to the conclusion that a Grandstream BT101 can be abused to be a door phone. Could someone with access to one, confirm that the following is possible? Researched: 1. When set to auto-answer, dialing the phone will result in a short beep and instant speaker-phone connection. I have this setup now, but don't recall the short beep. It may be configurable. 2. When pressing the message button while on-hook, the phone will activate speaker-phone and dial the number configured for voice mail retrieval. Correct. Assumptions: 3. Pressing the message button additional times will simply be ignored by the phone. I have several, I can check this weekend. 4. Hanging up the other end of the call will deactivate the speaker phone and cause the phone to go on-hook. (This is the behavior I see on a Polycom 430). I would have to say correct as well, since I'm using it as a paging unit and it does hang up after playing back the audio file. Something to consider, the BT101's speak phone has no Echo cancellation whatsoever and sounds just awful in a two way conversation. Doug Thanks to Dave and Doug for the quick responses. I'm looking forward to hearing the response on #3, but I think I'll get get one of these devices to play with this weekend. At worst, it'll be a usable garage or basement phone. Doug, I didn't even consider audio-quality on this, as even with the most rudimentary speaker phone circuits, phones seem pretty usable these days. I was planning to put this in a custom door-box anyway, along with a water-resistant speaker (plastic membrane). Considering our wide-open porch and some physical separation of the mic/speaker, the echo may not be as much of an issue as protection from the elements. And contrary to what someone asked me in private, wiring isn't an issue -- I do have cat5 at the door bell :) Thanks, JM A few notes about your idea. Yes it will work, you can set it auto answer, you can also set it to dial a number automatically when taken off-hook in addition to pressing the message button. You will probably want some sort or script to reboot the phone regularly (everyday) or it will just stop working (lose registration with *). The speaker phones really do stink on these but for a simple doorphone application, it should be fine and may even function better with the water-resistant mods you are doing. You actually do not even need cat5 (even though you have it) you can run 10mbit over cat3. Thanks, 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] Voicemail for an ATT System 75
Does anyone know if Asterisk can be used to provide voicemail (500-800 mailboxes) for a ATT System 75, Definity G-3? I was approached about this lately, and really know little about the ATT hardware. Any opinions would be appreciated! ___ --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] Need feedback on vitelity
Hello Anyone here uses Vitelity as voip provider ? Their pplans looks good but i need some feedback from existing customers if any here . ___ --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-biz] Need feedback on vitelity
Mail list wrote: Hello Anyone here uses Vitelity as voip provider ? Their pplans looks good but i need some feedback from existing customers if any here . A couple of our Enswitch customers use them heavily, and seem to be happy with them. Alistair Cunningham +1 888 468 3111 +44 20 799 39 799 sip:[EMAIL PROTECTED] http://integrics.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] Doorphone vs. Grandstream BT101
Steve Totaro wrote: You will probably want some sort or script to reboot the phone regularly (everyday) or it will just stop working (lose registration with *). The speaker phones really Really? I have several of them in use and have yet to reboot any of them. Doug -- 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] TDM analog cards, volume, echo, fxotune, ztmonitor and HPEC
Hi, everyone: I am developing a system using Asterisk, TDM-400 analog cards, analog lines, and Polycom SIP phones for internal extensions. Initially there was bad echo but after a series of efforts, I've managed to reduce it to a negligible level (it only happens when both parties speak simultaneously, and even there, only for a few hundred milliseconds). From an echo standpoint, things are very satisfactory. The HPEC has been very effective. The remaining problem is extremely variable volume, particularly as perceived by our internal callers. The remote party is often, but not always, too quiet. I've determined that it depends heavily on where that remote party is. - If the remote party is on a circuit on the same local exchange, the volume is perfect or nearly perfect. - If the party is on a remote exchange, the volume can vary, from barely audible, to audible but still too quiet to be really comfortable. In these cases, the users will crank the handset or headset volume on their Polycoms to make the remote party audible, but that ends up causing distortion which is bad enough to be irritating. The volume level does seem to depend on the exchange. For instance, all calls to a specific exchange are at the same general volume level. Now, standard analog sets have a varistor circuit to compensate for these variations in signal level, but it would appear that the TDM cards don't incorporate this kind of dynamic gain control. Zaptel allows you to control the transmit and receive gain at the interface card, but isn't that what fxotune is supposed to do? Tune those gains so that the echo is minimized? I don't want to play with gain only to undo the echo tuning done by fxotune. Does tinkering with the gain undo the tunings done by fxotune? Similarly with ztmonitor: Is tinkering with the gain using ztmonitor going to undo the tuning done by fxotune, or can I do both? Have you encountered gain issues like this on analog lines, and if so, how did you address them? Is cranking the rx gain on the Polycom phones a viable solution, or is it likely to make echo worse? I've tried a few adjustments, and they seemed to aggravate echo; once even so badly that the remote party could hear echo. -Stephen- ___ --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] Need feedback on vitelity
I did a trial as a wholesale provider and it seemed to work pretty good, but I could never get them to activate our account. I emailed the sales guy probably five times over a month to go ahead and fire it up and he never responded. Also, their tech support is horrible So basically they are like every other pay per minute VoIP provider Mail list wrote: Hello Anyone here uses Vitelity as voip provider ? Their pplans looks good but i need some feedback from existing customers if any here . ___ --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 -- Network stuff you didn't know http://www.networkoblivion.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] SRTP testers needed
any recommendation for usable softphone for windows with srtp support? I'm using CounterPath eyeBeam, but seems, that doesn't support secure rtp at all. ... and minisip seems to be quite death project, without activity. PJ marek cervenka wrote: On Fri, 2007-03-23 at 16:12 +0100, marek cervenka wrote: please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compilerun clients with srtp (linksys,gxp-2000, minisip, twikle, ...) ___ --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] freepbx - DB Error messages...
You might get a faster response on freepbx/amp mailing list. On 3/24/07, Francesco Peeters (Asterisk) [EMAIL PROTECTED] wrote: Hi all, I am probably missing something ultimately obvious, but I have a problem configuring freepbx... Using Edgy Eft with the cvs freePBX 2.2.1 and followed the Ubuntu installation guide on freepbx.org. System pxe-boots from a server with NFS root on same Using * 1.2 current (from source, not .deb's) Using mISDN-streams (from source, not .deb's) Using freePBX-2.2.1 (from source, not .deb's) Installed everything, and mISDN and * load just fine amportal start works fine as well However I keep getting DB Error's in the GUI... The syslog gives two separate errors: 1) Error 127 when reading table ./asterisk/whatever 2) Table is crashed and needs to be repaired I created a special mysql user for * and did an PERMIT ALL PRIVILEGES on the mysql databases When I log in to mysql as root and do 'SELECT username FROM ampusers ORDER BY username' I get the record list. When I do the same as the * user, I get the 'Table is crashed, blablabla' line. I tried changing the login user for freepbx (ampdbuser) to root, but that doesn't help either, as I keep getting the 127 error... Googling wasn't very helpful, and the freepbx forum admins still haven't approved my account, so I thought I'd try here... Any help appreciated! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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 -- Bruce Reeves Nortex 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] Need feedback on vitelity
Mail list wrote: Hello Anyone here uses Vitelity as voip provider ? Their pplans looks good but i need some feedback from existing customers if any here . I would like to express an opposite opinion. I have two accounts with them with lots of DIDs. Everything works fine, and they have been very quick to respond to the few issues we had to work through trying to implement, I think, 16 incoming lines in 14 area codes. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 with Dialplan or TrixBox for this case?
Hi all - Been using Asterisk installed on Debian and love it. But it's time to rearrange some lines and looking for a few features I didn't enable or have in the dial plan the first time around and wondering if you would recommend doing it through configs again or if one of the prepackaged solutions would more easily support these needs. One that caught my eye was TrixBox but I'd be open to other suggestions. I have a Wildcat TDM400 (IIRC) with 2 FXS and 2 FXO ports. Currently I'm terminating a POTS line and a VoicePulse VOIP line (via the supplied adapter) into the FXS ports (forgive me if I confused the FXO/FXS it gets me every time.) I have the dialplan set up to ring all extensions when either incoming line rings. Ring available extensions if one is in use. For dial out, it only dials out the VOIP line unless I override by dialing 9 first (because we pay per call on the POTS line so I want to know I'm doing it rather than have asterisk do it for me if the VOIP line is already in use.) - - - What I'm looking to do is keep the functionality above but drop the POTS line and add a SunRocket line also terminated with a VOIP adapter just like the VoicePulse line. Although the net connection will be a single point of failure, at least I'll have two different VOIP providers for some redundancy. I'd like to: - ring all extensions when a call comes in either VOIP line. - distinctive ring for calls coming in the SunRocket line (which Asterisk will know by the port that the line comes in on.) - do not disturb functionality to disable all extensions from ringing by dialing a *XX number from any phone in the house. Ability to toggle ringing back on easily. - dial out any available line (now that both are VOIP) Easy to do with TrixBox or better off installing the latest Asterisk and doing it through the command line and configuration file interface? Thanks! PS - Oddly, the SunRocket VOIP adapter doesn't seem to give a dialtone but a regular old phone works fine when connected to it. Will this cause problems for Asterisk? ___ --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] Shoretel integration into Salesforce.com
Hello Dean, We're currently working on improving integration with SugarCRM, with Salesforce.com integration is next on my list. FWIW, I've enhanced the SugarCRM integration to log and screen pop incoming calls identified by CID, and done a little work to make click to call a bit more robust. I'm currently working on dynamically routing a call to the correct agent based on the SugarCRM data. The goal is to duplicate as much of this functionality as possible with Salesforce.com after the current project is complete. All projects will be released as open-source. Contact me offlist if you'd like more info, Wendell From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Saturday, March 24, 2007 7:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Shoretel integration into Salesforce.com http://www.theregister.co.uk/2007/03/24/shortel_sforce Shoretel has integrated its IP phone system with Salesforce.com's call centre software. Using the two together will mean call centre agents get a reduced admin workload, with automatic call logging and screen pop-ups with the customer's record. Is anyone in the asterisk community providing integration into Salesforce.com? Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation www.Mexuar.com http://www.mexuar.com/ Want to voice enable your website? Use Corraleta to reach your customers in 10 seconds or less. This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. image001.gif Description: image001.gif ___ --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: RE : SIP/IAX peers UNREACHABLE and audio loss
On 2007-03-24 01:53:16 -0700, Edoardo Serra [EMAIL PROTECTED] said: Hi Francois, [EMAIL PROTECTED] ha scritto: Hi men, I have already encountered some issue like this with few switches (very known great brand) which doesn't like VoIP traffic ! I also have switches of a very known great brand !! It was so strange to me that I didn't consider a network problem... Check by drectly connected the VoIP equipment - if you can - with temporary long Ethernet cables bypassing the tested switch to see what happens in this case. I'd try to bypass the switch someway but every server neeeds to have its own public ip address.. I'll put an RTP proxy somewhere... You can also tell to qualify with a longer delay, but this could not help in case of regulary frames losses. What about turning qualify off ? Do you think taht Asterisk is stopping RTP when it loose a qualify packet ? Or is the RTP traffic itself that is lost by the switches ? The fact that qualify fails means you have a network issue. The same reason qualify fails (ie servers can't communicate) is the reason your users are experiencing quality issues in call. turn off Qualify isn't going to fix anything IMO. It's just going to hide it from you. If the asterisk servers are all on your LAN then the network issue should be easily fixable. If the Asterisk servers are at remote locations and are using public internet, you might have problems resolving this completely. 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
[asterisk-users] Re: Refund from SellVoip?
On 2007-03-23 14:37:18 -0700, Tom Lynn [EMAIL PROTECTED] said: Now I know where they've been spending my remaining balance... I still use Sellvoip as my primary terminator, and have found the call quality to be superior to any other ITSP from my location (Seattle). I agree completely that there is no support from this company, which is a major issue if you are trying to support other customers. Still, I remain a happy customer of sellvoip, with Teliax and Nufone configured as backups... I wouldn't expect a refund for cancellation of prepaid phone usage, does the original agreement you have with then suggest that they owe you a refund? 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
[asterisk-users] Re: RE : SIP/IAX peers UNREACHABLE and audio loss
Martin Joseph ha scritto: The fact that qualify fails means you have a network issue. The same reason qualify fails (ie servers can't communicate) is the reason your users are experiencing quality issues in call. It was also my first though, but my LAN is very SIMPLE, so I was wondering if something else could cause the problem. turn off Qualify isn't going to fix anything IMO. It's just going to hide it from you. You're probably right, but it depends on Asterisk internals (which I don't know well). If Asterisk would stop to send RTP audio when just a qualify packet get lost it can make the situation worst. If the asterisk servers are all on your LAN then the network issue should be easily fixable. It should, but my LAN is very simple... I have a 10/100 Mbit switch with no more than 15 servers on it. Traffic on the LAN is not heavy even if the time of the day I see in the logs make me think it could be an issue related to network load trafic Anyhow I'll try to generate some heavy traffic on the LAN to see if it could be related to that. I also noticed that this problem began to happen when I upgraded my Asterisk to 1.2, but it can be a concidence. Do you think it could be related to bugs in ethernet drivers, kernel or whatever at the OS level ?? If the Asterisk servers are at remote locations and are using public internet, you might have problems resolving this completely. We have some Asterisk spread all over the public Internet, but firstly we should solve this problem at a LAN level Tnx for attention Regards 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
[asterisk-users] Remote host can't match request NOTIFY to call
Evnin'... Anybody got an idea where those CLI messages come from? [Mar 24 20:30:05] WARNING[4518]: chan_sip.c:12296 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. Interestingly all are caused by local IP used by asterisk-1.4.1 cheers rick ___ --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: RE : SIP/IAX peers UNREACHABLE and audio loss
Have you taken care of any eventual IRQ sharing ? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Edoardo Serra Envoyé : samedi 24 mars 2007 20:27 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Re: RE : SIP/IAX peers UNREACHABLE and audio loss Martin Joseph ha scritto: The fact that qualify fails means you have a network issue. The same reason qualify fails (ie servers can't communicate) is the reason your users are experiencing quality issues in call. It was also my first though, but my LAN is very SIMPLE, so I was wondering if something else could cause the problem. turn off Qualify isn't going to fix anything IMO. It's just going to hide it from you. You're probably right, but it depends on Asterisk internals (which I don't know well). If Asterisk would stop to send RTP audio when just a qualify packet get lost it can make the situation worst. If the asterisk servers are all on your LAN then the network issue should be easily fixable. It should, but my LAN is very simple... I have a 10/100 Mbit switch with no more than 15 servers on it. Traffic on the LAN is not heavy even if the time of the day I see in the logs make me think it could be an issue related to network load trafic Anyhow I'll try to generate some heavy traffic on the LAN to see if it could be related to that. I also noticed that this problem began to happen when I upgraded my Asterisk to 1.2, but it can be a concidence. Do you think it could be related to bugs in ethernet drivers, kernel or whatever at the OS level ?? If the Asterisk servers are at remote locations and are using public internet, you might have problems resolving this completely. We have some Asterisk spread all over the public Internet, but firstly we should solve this problem at a LAN level Tnx for attention Regards 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
[asterisk-users] Re: RE : Re: RE : SIP/IAX peers UNREACHABLE and audio loss
[EMAIL PROTECTED] ha scritto: Have you taken care of any eventual IRQ sharing ? I don't think so. (how cuold I detect it ? ) Servers are not self assembled but brand machines They have no other pci cards (some of them have, but the problem happens also between server with no added pci cards) this is my /proc/interrupts # cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 486589652 44037074 667253957 76390228IO-APIC-edge timer 8: 2 0 0 0IO-APIC-edge rtc 9: 0 0 0 0 IO-APIC-level acpi 10: 0 0 0 0 IO-APIC-level ohci_hcd:usb1 14: 22211 0 26603304 0IO-APIC-edge ide0 15: 22977880 0 03575943IO-APIC-edge ide1 16: 1526340728 1176101099 992391841 1707199189 IO-APIC-level eth0 17: 98016813 642505 961195082349025 IO-APIC-level eth1 NMI: 0 0 0 0 LOC: 1274300159 1274300179 1274300196 1274300195 ERR: 0 MIS: 0 My kernel is a # uname -ar Linux switch1 2.6.18-gentoo-r6 #1 SMP Wed Jan 24 21:08:48 CET 2007 i686 Intel(R) Xeon(TM) CPU 3.20GHz GenuineIntel GNU/Linux Tnx for attention Edoardo -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Edoardo Serra Envoyé : samedi 24 mars 2007 20:27 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Re: RE : SIP/IAX peers UNREACHABLE and audio loss Martin Joseph ha scritto: The fact that qualify fails means you have a network issue. The same reason qualify fails (ie servers can't communicate) is the reason your users are experiencing quality issues in call. It was also my first though, but my LAN is very SIMPLE, so I was wondering if something else could cause the problem. turn off Qualify isn't going to fix anything IMO. It's just going to hide it from you. You're probably right, but it depends on Asterisk internals (which I don't know well). If Asterisk would stop to send RTP audio when just a qualify packet get lost it can make the situation worst. If the asterisk servers are all on your LAN then the network issue should be easily fixable. It should, but my LAN is very simple... I have a 10/100 Mbit switch with no more than 15 servers on it. Traffic on the LAN is not heavy even if the time of the day I see in the logs make me think it could be an issue related to network load trafic Anyhow I'll try to generate some heavy traffic on the LAN to see if it could be related to that. I also noticed that this problem began to happen when I upgraded my Asterisk to 1.2, but it can be a concidence. Do you think it could be related to bugs in ethernet drivers, kernel or whatever at the OS level ?? If the Asterisk servers are at remote locations and are using public internet, you might have problems resolving this completely. We have some Asterisk spread all over the public Internet, but firstly we should solve this problem at a LAN level Tnx for attention Regards 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 ___ --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] Doorphone vs. Grandstream BT101
Am Samstag, den 24.03.2007, 11:43 -0400 schrieb Steve Totaro: You will probably want some sort or script to reboot the phone regularly (everyday) or it will just stop working (lose registration with *). The speaker phones really do stink on these but for a simple doorphone application, it should be fine and may even function better with the water-resistant mods you are doing. I only saw that behaviour with an unreliable network cable - actually, the tab of the only free ethernet cable that ran behind my desk had broken, so the plug would not sit perfectly. This could do all kinds of things to the BT101, losing connection during a call being the obvious phenomenon. Sometimes, that phone crashed hard, only power-cycling would do. Replacing the cable solved this problem. You actually do not even need cat5 (even though you have it) you can run 10mbit over cat3. Right. But who uses cat3 these days, with cable prizing as it is? I run CAT5 everywhere instead of any low-voltage phone line or whatever, just to be future proof. I have an old PC mouse, with a RJ45 plug, that features two LEDs and open the garage door, giving a door status feedback. Just a free CAT5 from the hall to the basement, and it looks nice, somehow techy :-) BR Anselm ___ --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] Can be called on FreeWorldDialup/IAX channel, but can't make calls
On 3/24/07, Timothy Parez [EMAIL PROTECTED] wrote: Hi, I have an FWD account and it's configured in asterisk. I can be called by people using FWD, but I cannot make FWD calls myself. I have FWD and IAXTel configured as well. FWD has been having problems with their IAX server for awhile judging by their forums. I have exchanged emails with Juan there and after one reboot of their IAX box I was able to call time, echo test, and the likes, but no 800 numbers. I even could receive calls. Then it quit again. Juan reset my account, but it still does not work. IAXTel registers fine. However, I get the conjestion messges with every 800 number I attempt to dial always. But, these are donated services so we can't complain too much. Chris ___ --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] Can be called on FreeWorldDialup/IAX channel, but can't make calls
On 3/24/07, Timothy Parez [EMAIL PROTECTED] wrote: Mar 24 15:28:55 ERROR[2873]: chan_sip.c:11076 handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 172.17.249.253, but there is no hint for that extension I believe the subscribe error comes from not having a 'hint' in the context of the extension for the sip @ 172.17.249.253 indicating the sip at extension 00032498043823 (what an extension!). I am new myself to * so someone may need to correct me on this one. Chris ___ --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] Doorphone vs. Grandstream BT101
I did something similar one year ago for a friend of mine that was interested to answer to bell door from internal phones. I used an HT286 with a sort of homebuilt analog hybrid with a microcontroller able to automatically answer when the ring was present on the HT286 FXS line (when calling from internal to the external box) and using the auto-call feature of the HT286 when people press the external button. To terminate the call I used a sort of DTMF sequence sent by Asterisk dead-agi script that the micro detects just to hang-up. I've added on the same box an axis camera to have a sort of video on the LAN. To be able to safely open the door, I made a little box ethernet based able to receive some UDP packets sent by Asterisk through agi when the received call was transferred on a predefined internal extension. It's working well! In my spare time, I'm working to have this solution well packed in an easy to build electronic kit (my friend is using a prototype version). If you are interested, I can post my results and the link to my site when they will be ready. Thank you and bye, Marco Signorini. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, 23 March 2007 5:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Doorphone vs. Grandstream BT101 And contrary to what someone asked me in private, wiring isn't an issue -- I do have cat5 at the door bell :) Thanks, JM Like all good geeks should - correct Jay J So did you run two lengths so that you have access to a IP Door camera as well? Dont forget a few pairs for the electric strike to open the door remotely from a web interface as well. 17,000 ft of Cat5, cat6 and rg6 in the house, somewhere around 120 drops, along with multiple 2 PVC from basement to attic. The front and back door do have dual cat5s, but I'm not planning on a remote door strike for either. CCTV is separate and runs on utp baluns into two 4-channel BT cards, so there's cat5 in all the places where I need (or may later need) a camera. -- Passa a Infostrada. ADSL e Telefono senza limiti e senza canone Telecom http://click.libero.it/infostrada ___ --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] Question about DSP in Digium card
Hello. I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to find out if there is any limitation about DSP capabilities, I mean, I am not sure how many phone calls my Digium card supports, simultaneously. The calling flow goes from IAX - ISDN. I am running this card into CPU like this: - Micro PIV 3.0 - 1Gbyte Memory Thanks. Levy.- ___ --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] Timeout for conferences
Hi, The dialin conference via asterisk is over, one person is still in the conference room and accidentally does not hang up properly. Her meter at the phone company keeps running... I'd like to implement something to the effect of checking whether there is only one participant in the conference, and when this is the case, to cancel the call after a predefined time (perhaps 5 or 10 mins. to allow for some waiting for latecomers). Has someone already written some code or a quick idea for this scenario? Regards, --AvH ___ --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] Voicemail for an ATT System 75
Gary Eck wrote: Does anyone know if Asterisk can be used to provide voicemail (500-800 mailboxes) for a ATT System 75, Definity G-3? I was approached about this lately, and really know little about the ATT hardware. Any opinions would be appreciated! Probably... I would expect that system to support T1 trunking and SMDI (probably over RS-232), which would allow Asterisk to provide transparent voicemail services. ___ --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] Timeout for conferences
Way back in the day (1.2.7.1), I did this for a client.
In conf_run() in app_meetme.c, I added this code:
// if an agent abandons a caller, kick the caller after 15 seconds
// check for no agent
if ((conf-isdynamic)
(1 == conf-users)
(0 == (CONFFLAG_ADMIN user-userflags)))
{
if (0 == lonely_timeout)
{
// give the agent 15 seconds to log back in
lonely_timeout = time(0) + 15;
}
if (time(0) lonely_timeout)
{
ret = 0;
return(ret);
}
}
Their goal was if the agent was disconnected, give the agent 15 seconds to
dial back in before bumping the customer.
Reply off-list if you need more :)
On Sat, 24 Mar 2007, Andreas v. Heydwolff wrote:
Hi,
The dialin conference via asterisk is over, one person is still in the
conference room and accidentally does not hang up properly. Her meter at the
phone company keeps running...
I'd like to implement something to the effect of checking whether there is
only one participant in the conference, and when this is the case, to cancel
the call after a predefined time (perhaps 5 or 10 mins. to allow for some
waiting for latecomers).
Has someone already written some code or a quick idea for this scenario?
Regards,
--AvH
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Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] PickUp a call with feature pickup (*8) from a IAX2channel
I had set it On 3/21/07, LKS GMAIL [EMAIL PROTECTED] wrote: Try to set the callgroup and pickupgroup up in the IAX conf. Saludos, Lukassky. -- *De:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *En nombre de *Alvaro Parres *Enviado el:* miércoles, 21 de marzo de 2007 16:55 *Para:* Asterisk Users Mailing List - Non-Commercial Discussion *Asunto:* [asterisk-users] PickUp a call with feature pickup (*8) from a IAX2channel Hi list, i'm trying to do that iax channels can acces the pickup feature(normaly *8 dialing). But always the iax channel when dial *8, search for the extensión *8 on its context. I know i can program the *8 extension with the pickup applicatión. But its doesn't works for me, becouse i need to pickup some calls comming from IVR's o Queues. And there de exten is no the same as the channel, etc. Any idea or help ? Thaks. ___ --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] Can be called on FreeWorldDialup/IAX channel, but can't make calls
That extension is a mobile number, a number I called ealier that day, but does not seem to be related to my problem. On Sat, 2007-03-24 at 17:35 -0400, Chris Nighswonger wrote: On 3/24/07, Timothy Parez [EMAIL PROTECTED] wrote: Mar 24 15:28:55 ERROR[2873]: chan_sip.c:11076 handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 172.17.249.253, but there is no hint for that extension I believe the subscribe error comes from not having a 'hint' in the context of the extension for the sip @ 172.17.249.253 indicating the sip at extension 00032498043823 (what an extension!). I am new myself to * so someone may need to correct me on this one. Chris ___ --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 - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --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: Refund from SellVoip?
I, on the other hand, have been disappointed repeatedly by their failures to route international calls. I've received e-mails from them promising a refund. I expect them to keep their word. On 3/24/07, Martin Joseph [EMAIL PROTECTED] wrote: On 2007-03-23 14:37:18 -0700, Tom Lynn [EMAIL PROTECTED] said: Now I know where they've been spending my remaining balance... I still use Sellvoip as my primary terminator, and have found the call quality to be superior to any other ITSP from my location (Seattle). I agree completely that there is no support from this company, which is a major issue if you are trying to support other customers. Still, I remain a happy customer of sellvoip, with Teliax and Nufone configured as backups... I wouldn't expect a refund for cancellation of prepaid phone usage, does the original agreement you have with then suggest that they owe you a refund? 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
[asterisk-users] asterisk: error while loading shared libraries: libiksemel.so.3: cannot open shared object file: No such file or directory
How I can disable Gtalk Jabber module?Thanks# asterisk -vcasterisk: error while loading shared libraries: libiksemel.so.3: cannot open shared object file: No such file or directory===Centos4.4 2.6.9-34.0.2.ELzaptel 1.4.1asterisk 1.4.2iksemel 1.2Dmitri Smirnoff msn: [EMAIL PROTECTED]: 613 693 1299 ext 120 ___ --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] Sendmail and exchange for voicemail integration
Can you post the SMTP logs from the exchange server? Or try the mqueue on the Asterisk box. Post them so someone can help you further. On 3/23/07, Jordan Novak [EMAIL PROTECTED] wrote: I am having real trouble getting Asterisk to send to exchange. They are on the same LAN. Does anyone know of a walkthrough for this setup. I have gotten it to work before, but that was to a hotmail account. Jordan Novak ___ --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] PickUp a call with feature pickup (*8) from a IAX2channel
Can you show me your extensions.conf ? I think the problem is right there. Something else... Spanish? El sáb, 24-03-2007 a las 17:12 -0600, Alvaro Parres escribió: I had set it On 3/21/07, LKS GMAIL [EMAIL PROTECTED] wrote: Try to set the callgroup and pickupgroup up in the IAX conf. Saludos, Lukassky. __ De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Alvaro Parres Enviado el: miércoles, 21 de marzo de 2007 16:55 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] PickUp a call with feature pickup (*8) from a IAX2channel Hi list, i'm trying to do that iax channels can acces the pickup feature(normaly *8 dialing). But always the iax channel when dial *8, search for the extensión *8 on its context. I know i can program the *8 extension with the pickup applicatión. But its doesn't works for me, becouse i need to pickup some calls comming from IVR's o Queues. And there de exten is no the same as the channel, etc. Any idea or help ? Thaks. ___ --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] Asterisk with Dialplan or TrixBox for this case?
From: Brian McEntire [EMAIL PROTECTED] Date: Sat, 24 Mar 2007 13:57:38 -0400 Hi all - Been using Asterisk installed on Debian and love it. But it's time to rearrange some lines and looking for a few features I didn't enable or have in the dial plan the first time around and wondering if you would recommend doing it through configs again or if one of the prepackaged solutions would more easily support these needs. One that caught my eye was TrixBox but I'd be open to other suggestions. I have a Wildcat TDM400 (IIRC) with 2 FXS and 2 FXO ports. Currently I'm terminating a POTS line and a VoicePulse VOIP line (via the supplied adapter) into the FXS ports (forgive me if I confused the FXO/FXS it gets me every time.) I have the dialplan set up to ring all extensions when either incoming line rings. Ring available extensions if one is in use. For dial out, it only dials out the VOIP line unless I override by dialing 9 first (because we pay per call on the POTS line so I want to know I'm doing it rather than have asterisk do it for me if the VOIP line is already in use.) - - - What I'm looking to do is keep the functionality above but drop the POTS line and add a SunRocket line also terminated with a VOIP adapter just like the VoicePulse line. Although the net connection will be a single point of failure, at least I'll have two different VOIP providers for some redundancy. I'd like to: - ring all extensions when a call comes in either VOIP line. - distinctive ring for calls coming in the SunRocket line (which Asterisk will know by the port that the line comes in on.) - do not disturb functionality to disable all extensions from ringing by dialing a *XX number from any phone in the house. Ability to toggle ringing back on easily. - dial out any available line (now that both are VOIP) Easy to do with TrixBox or better off installing the latest Asterisk and doing it through the command line and configuration file interface? If your box has the power to run extra stuff that come with TrixBox and you are sure that doing what you need is easier in TrixBox, there's not much difference. (From your description, the requirements are easily implementable with plain config files.) Thanks! PS - Oddly, the SunRocket VOIP adapter doesn't seem to give a dialtone but a regular old phone works fine when connected to it. Will this cause problems for Asterisk? Asterisk does not have to check dial tone. (But it's a really oddball adapter.) However, if you are going all VoIP, why bother providers that require adapters (thus TDM card)? You can get better result by using providers that transmits voice over IP into your Asterisk and get rid of the TDM card. Yuan Liu ___ --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: error while loading shared libraries: libiksemel.
From: Dmitri Smirnoff [EMAIL PROTECTED] Date: Sat, 24 Mar 2007 21:11:17 -0400 How I can disable Gtalk Jabber module?Thanks# asterisk -vcasterisk: error while loading shared libraries: libiksemel.so.3: cannot open shared objectfile: No such file or directory===Centos4.4 2.6.9-34.0.2.ELzaptel 1.4.1asterisk 1.4.2iksemel 1.2Dmitri Smirnoff msn: [EMAIL PROTECTED]: 613 693 1299 ext 120 Rerun make menuselect? Yuan Liu ___ --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 with ztdummy
Although I have a Debian system with prebuilt asterisk package available, I was finding it crashed when I tried to use MeetMe. So I have built asterisk from scratch. However the first thing I try and do is install the ztdummy module with modprobe ztdummy but it always fails to load with FATAL: Error inserting ztdummy (/lib/modules/2.6.18-4-686/misc/ztdummy.ko): Device or resource busy Can someone give me any clues as to what is wrong -- Alan Chandler http://www.chandlerfamily.org.uk ___ --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
