Re: [asterisk-users] How to register SIP phone on Asterisk 1.6.2.14 on Centos 5.5 64bit
Check X-lite sending register request or not to asterisk buy checking the asterisk console,if not there would some problem in X-lite configuration settings,if sending check the console and see what error logs you are getting.. Thanks Nikhil On 11/18/2010 04:06 PM, Phuong Hoang wrote: Hi all, I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but not successful, Can anyone help me to do it? Thanks and best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] URGENT Help needed
Hello, We tried to upgrade our Asterisk from 1.6.2.13 to 1.6.2.14, after trying to install iksemel (jabber support) and spandsp, but now Asterisk doesn't work anymore and we can't get it to run, althorugh we tried to remove it completely and reinstall 1.6.2.13. when trying to start it via /etc/init.d/asterisk start we get the following error: Asterisk died with code 1. Automatically restarting Asterisk. Asterisk ended with exit status 1 When just trying to run it as asterisk from the command line, we don't see the process being active and we get this message when running asterisk -r, although the file is present: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Any help would be highly appreciated. Thank you in advance, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_musiconhold.c Bug - Patch to solve?
Hello Asterisk community, We are having some problems with crashes in Asterisk, my asterisk versions are 1.4.24.1 and 1.4.23.2. I have found this: ~/work/asterisk-branch-1.4$ svn log -c 260345 r260345 | mmichelson | 2010-04-30 22:08:15 +0200 (Fri, 30 Apr 2010) | 18 lines Fix potential crash from race condition due to accessing channel data without the channel locked. In res_musiconhold.c, there are several places where a channel's stream's existence is checked prior to calling ast_closestream on it. The issue here is that in several cases, the channel was not locked while checking the stream. The result was that if two threads checked the state of the channel's stream at approximately the same time, then there could be a situation where both threads attempt to call ast_closestream on the channel's stream. The result here is that the refcount for the stream would go below 0, resulting in a crash. I have added proper channel locking to res_musiconhold.c to ensure that we do not try to check chan-stream without the channel locked. A Digium customer has been using this patch for several weeks and has not had any crashes since applying the patch. ABE-2147 How can i apply this patch on my asterisk versions: 1.4.24.1 and 1.4.23.2? do i have to apply this patch manually? Thanks in advance for your help -- Ing. Danny Dias www.DannTEL.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT Help needed
Do your asterisk logs say anything -- /var/log/asterisk/messages or full? Also, what happens if you do asterisk -c this may help you figure things out. Michael voip.quest...@gmail.com wrote: Hello, We tried to upgrade our Asterisk from 1.6.2.13 to 1.6.2.14, after trying to install iksemel (jabber support) and spandsp, but now Asterisk doesn't work anymore and we can't get it to run, althorugh we tried to remove it completely and reinstall 1.6.2.13. when trying to start it via /etc/init.d/asterisk start we get the following error: Asterisk died with code 1. Automatically restarting Asterisk. Asterisk ended with exit status 1 When just trying to run it as asterisk from the command line, we don't see the process being active and we get this message when running asterisk -r, although the file is present: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Any help would be highly appreciated. Thank you in advance, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT Help needed
- Original Message - Hello, We tried to upgrade our Asterisk from 1.6.2.13 to 1.6.2.14, after trying to install iksemel (jabber support) and spandsp, but now Asterisk doesn't work anymore and we can't get it to run, althorugh we tried to remove it completely and reinstall 1.6.2.13. when trying to start it via /etc/init.d/asterisk start we get the following error: Asterisk died with code 1. Automatically restarting Asterisk. Asterisk ended with exit status 1 When just trying to run it as asterisk from the command line, we don't see the process being active and we get this message when running asterisk -r, although the file is present: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Any help would be highly appreciated. Thank you in advance, Michael What is being reported in /var/log/asterisk/messages ? Do you see any errors when you run asterisk from the command line in foreground ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.8 SIP register uri: peer field ?
It's a peer name defined below in sip.conf. You may skip secret if it is specified in peer section. I don't know of any other meanings. For example, register = mypeer?u...@host [mypeer] type=peer defaultuser=user secret=blah ... This syntax exists since 1.6.2. 21.10.2010 17:31, Guillaume Bour пишет: Hello, Looking the asterisk 1.8 API documentation (http://www.asterisk.org/astdocs/api/index.html), I see a lot of new fields for sip register uris: register = [peer?][transport://]us...@domain][:secret[:authuse...@host[:port][/extension][~expiry] But the *peer* is not explained anywhere. What it is for ? Regards, Guillaume Bour. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT Help needed
Also, what happens if you do asterisk -c this may help you figure things out. Hi, These are the WARNINGSI found in /var/log/asterisk/messages after running the above command: [Nov 22 12:10:19] WARNING[2316] udptl.c: T38FaxUdpEC in udptl.conf is no longer supported; use the t38pt_udptl configuration option in sip.conf instead. [Nov 22 12:10:19] WARNING[2316] udptl.c: T38FaxMaxDatagram in udptl.conf is no longer supported; value is now supplied by T.38 applications. [Nov 22 12:10:19] WARNING[2316] loader.c: Error loading module 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_odbc_clear_cache [Nov 22 12:10:19] WARNING[2316] res_config_ldap.c: No directory user found, anonymous binding as default. [Nov 22 12:10:19] ERROR[2316] res_config_ldap.c: No directory URL or host found. [Nov 22 12:10:19] WARNING[2316] utils.c: trying to reset empty pool [Nov 22 12:10:19] WARNING[2316] utils.c: trying to reset empty pool [Nov 22 12:10:19] WARNING[2316] utils.c: trying to reset empty pool [Nov 22 12:10:19] WARNING[2316] pbx.c: Already have an application 'SendFAX' [Nov 22 12:10:19] WARNING[2316] pbx.c: Already have an application 'ReceiveFAX' Thanks, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Extensions and loss of Internet connection
- Original Message - Hi all! A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. I'm using Asterisk 1.4.24.1. I was researching on the Internet and I found that it can be related to a bug of chan_sip, can it be? In this case, is there a possible workaround? Thanks in advance for your reply. Regards, Daniel Does you Asterisk server point to an internal DNS or to your router ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Extensions and loss of Internet connection
Hi, Phil. A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. I'm using Asterisk 1.4.24.1. I was researching on the Internet and I found that it can be related to a bug of chan_sip, can it be? In this case, is there a possible workaround? Does you Asterisk server point to an internal DNS or to your router ? The /etc/resolv.conf of the host on which I installed Asterisk points to an internal DNS. Is there a parameter in the Asterisk configuration where also I have to force the use of an internal DNS server? Thanks for your reply. Regards, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avoiding deadlock
Hi Moises, Thanks for your opinion. However I still wouldn't want to agree that reducing debug logging is a solution. Let me explain why, we are driving Asterisk using AMI and verbose logging is simply not enough to investigate issues that arises with our software or Asterisk itself. Also we are getting valuable information from the debug logs in order to verify activities in our own logs. Printing Avoiding deadlock message 12000 times in the logs makes system less efficient and causes performance degradation due to massive I/O activity. Would you say this should be ignored too? I'm not implying that Avoiding deadlock is the problem here, maybe its Asterisk debug logging? Regards, Vilius. On 18 November 2010 03:35, Moises Silva moises.si...@gmail.com wrote: On Wed, Nov 17, 2010 at 9:56 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi Chad, Thanks for your suggestions. However I believe decreasing logging, its just like closing your eyes and ignoring what happening behind you, the problem is still there. Also decreased logging will prevent from troubleshooting any other problems in the future. Would you happen to know any potential causes for this message? The problem is you were just told by a Digium engineer who knows the code from many years back that is a debug message and there is nothing to worry about and you insist in believing this is a problem. If you want to know what the message means and why you should not worry you must understand what a lock is, what lock contention is and what a deadlock is. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Extensions and loss of Internet connection
- Original Message - Hi, Phil. A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. I'm using Asterisk 1.4.24.1. I was researching on the Internet and I found that it can be related to a bug of chan_sip, can it be? In this case, is there a possible workaround? Does you Asterisk server point to an internal DNS or to your router ? The /etc/resolv.conf of the host on which I installed Asterisk points to an internal DNS. Is there a parameter in the Asterisk configuration where also I have to force the use of an internal DNS server? Thanks for your reply. Regards, Daniel Do your SIP extensions use your internal DNS server ? are they able to resolve the IP of your Asterisk server ? If you enable SIP debugging do you see them even try and connect ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_musiconhold.c Bug - Patch to solve?
Hasn't this been fixed in later versions? 1.4.37 is current, or at least it was in the last few days. Upgrading with no reason isn't suggested, but in this case you have a good reason, and if you dig deep enough you may find the fix is already in place. John Novack Danny Dias wrote: Hello Asterisk community, We are having some problems with crashes in Asterisk, my asterisk versions are 1.4.24.1 and 1.4.23.2. I have found this: ~/work/asterisk-branch-1.4$ svn log -c 260345 r260345 | mmichelson | 2010-04-30 22:08:15 +0200 (Fri, 30 Apr 2010) | 18 lines Fix potential crash from race condition due to accessing channel data without the channel locked. In res_musiconhold.c, there are several places where a channel's stream's existence is checked prior to calling ast_closestream on it. The issue here is that in several cases, the channel was not locked while checking the stream. The result was that if two threads checked the state of the channel's stream at approximately the same time, then there could be a situation where both threads attempt to call ast_closestream on the channel's stream. The result here is that the refcount for the stream would go below 0, resulting in a crash. I have added proper channel locking to res_musiconhold.c to ensure that we do not try to check chan-stream without the channel locked. A Digium customer has been using this patch for several weeks and has not had any crashes since applying the patch. ABE-2147 How can i apply this patch on my asterisk versions: 1.4.24.1 and 1.4.23.2? do i have to apply this patch manually? Thanks in advance for your help -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call recording format
Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. What could be the reason for this? Is Asterisk updating wav headers every time it writes? What would be recommended hardware setup for over 60 simultaneous call records? Regards, Vilius. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_musiconhold.c Bug - Patch to solve?
Hello John, What i am asking is if i can apply this patch manually or something like this without making any upgrade of Asterisk, has anyone done this before? Or i have to upgrade my Asterisk versions...i don't really want to do this... Thanks in Advance! 2010/11/22 John Novack jnov...@stromberg-carlson.org Hasn't this been fixed in later versions? 1.4.37 is current, or at least it was in the last few days. Upgrading with no reason isn't suggested, but in this case you have a good reason, and if you dig deep enough you may find the fix is already in place. John Novack Danny Dias wrote: Hello Asterisk community, We are having some problems with crashes in Asterisk, my asterisk versions are 1.4.24.1 and 1.4.23.2. I have found this: ~/work/asterisk-branch-1.4$ svn log -c 260345 r260345 | mmichelson | 2010-04-30 22:08:15 +0200 (Fri, 30 Apr 2010) | 18 lines Fix potential crash from race condition due to accessing channel data without the channel locked. In res_musiconhold.c, there are several places where a channel's stream's existence is checked prior to calling ast_closestream on it. The issue here is that in several cases, the channel was not locked while checking the stream. The result was that if two threads checked the state of the channel's stream at approximately the same time, then there could be a situation where both threads attempt to call ast_closestream on the channel's stream. The result here is that the refcount for the stream would go below 0, resulting in a crash. I have added proper channel locking to res_musiconhold.c to ensure that we do not try to check chan-stream without the channel locked. A Digium customer has been using this patch for several weeks and has not had any crashes since applying the patch. ABE-2147 How can i apply this patch on my asterisk versions: 1.4.24.1 and 1.4.23.2? do i have to apply this patch manually? Thanks in advance for your help -- Dog is my Co-pilot -- Ing. Danny Dias www.DannTEL.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Extensions and loss of Internet connection
Does you Asterisk server point to an internal DNS or to your router ? The /etc/resolv.conf of the host on which I installed Asterisk points to an internal DNS. Is there a parameter in the Asterisk configuration where also I have to force the use of an internal DNS server? Do your SIP extensions use your internal DNS server ? are they able to resolve the IP of your Asterisk server ? If you enable SIP debugging do you see them even try and connect ? The extensions have configured the Asterisk server by its IP, so I do not think there is a problem on that side. To enable debug I should use 'sip set debug'? from the Asterisk CLI? I do not see any record in the CLI after running this command. However, from Twinkle, for example, I see the following: - lun 10:49:59 Daniel, registration failed: 503 Service Unavailable - Thanks for your reply. Regards, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording format
What format are the actual calls in? Are they in G.711u/a format or are they in something else (perhaps gsm?) format? I'm asking to find out if Asterisk would need to transcode them. On Mon, Nov 22, 2010 at 6:47 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. What could be the reason for this? Is Asterisk updating wav headers every time it writes? What would be recommended hardware setup for over 60 simultaneous call records? Regards, Vilius. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_musiconhold.c Bug - Patch to solve?
Danny Dias wrote: Hello John, What i am asking is if i can apply this patch manually or something like this without making any upgrade of Asterisk, has anyone done this before? I can't answer that question. Or i have to upgrade my Asterisk versions...i don't really want to do this... Why not? MANY fixes have been included in the upgrades. Improved security at the least. There are 10-15 versions between where you are operating and what is current John Novack Thanks in Advance! 2010/11/22 John Novack jnov...@stromberg-carlson.org mailto:jnov...@stromberg-carlson.org Hasn't this been fixed in later versions? 1.4.37 is current, or at least it was in the last few days. Upgrading with no reason isn't suggested, but in this case you have a good reason, and if you dig deep enough you may find the fix is already in place. John Novack Danny Dias wrote: Hello Asterisk community, We are having some problems with crashes in Asterisk, my asterisk versions are 1.4.24.1 and 1.4.23.2. I have found this: ~/work/asterisk-branch-1.4$ svn log -c 260345 r260345 | mmichelson | 2010-04-30 22:08:15 +0200 (Fri, 30 Apr 2010) | 18 lines Fix potential crash from race condition due to accessing channel data without the channel locked. In res_musiconhold.c, there are several places where a channel's stream's existence is checked prior to calling ast_closestream on it. The issue here is that in several cases, the channel was not locked while checking the stream. The result was that if two threads checked the state of the channel's stream at approximately the same time, then there could be a situation where both threads attempt to call ast_closestream on the channel's stream. The result here is that the refcount for the stream would go below 0, resulting in a crash. I have added proper channel locking to res_musiconhold.c to ensure that we do not try to check chan-stream without the channel locked. A Digium customer has been using this patch for several weeks and has not had any crashes since applying the patch. ABE-2147 How can i apply this patch on my asterisk versions: 1.4.24.1 and 1.4.23.2? do i have to apply this patch manually? Thanks in advance for your help -- Dog is my Co-pilot -- Ing. Danny Dias www.DannTEL.net http://www.DannTEL.net -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_musiconhold.c Bug - Patch to solve?
2010/11/22 John Novack jnov...@stromberg-carlson.org Danny Dias wrote: Hello John, What i am asking is if i can apply this patch manually or something like this without making any upgrade of Asterisk, has anyone done this before? I can't answer that question. ummm why not? is something wrong? Or i have to upgrade my Asterisk versions...i don't really want to do this... Why not? MANY fixes have been included in the upgrades. Improved security at the least. There are 10-15 versions between where you are operating and what is current I'm sure that the upgrade will fix this, but if applying the patch without making any upgrade will be better for me, my asterisk servers are working with many calls, realtime, fop etc...and an upgrade could make something happen... John Novack Thanks in Advance! 2010/11/22 John Novack jnov...@stromberg-carlson.org Hasn't this been fixed in later versions? 1.4.37 is current, or at least it was in the last few days. Upgrading with no reason isn't suggested, but in this case you have a good reason, and if you dig deep enough you may find the fix is already in place. John Novack Danny Dias wrote: Hello Asterisk community, We are having some problems with crashes in Asterisk, my asterisk versions are 1.4.24.1 and 1.4.23.2. I have found this: ~/work/asterisk-branch-1.4$ svn log -c 260345 r260345 | mmichelson | 2010-04-30 22:08:15 +0200 (Fri, 30 Apr 2010) | 18 lines Fix potential crash from race condition due to accessing channel data without the channel locked. In res_musiconhold.c, there are several places where a channel's stream's existence is checked prior to calling ast_closestream on it. The issue here is that in several cases, the channel was not locked while checking the stream. The result was that if two threads checked the state of the channel's stream at approximately the same time, then there could be a situation where both threads attempt to call ast_closestream on the channel's stream. The result here is that the refcount for the stream would go below 0, resulting in a crash. I have added proper channel locking to res_musiconhold.c to ensure that we do not try to check chan-stream without the channel locked. A Digium customer has been using this patch for several weeks and has not had any crashes since applying the patch. ABE-2147 How can i apply this patch on my asterisk versions: 1.4.24.1 and 1.4.23.2? do i have to apply this patch manually? Thanks in advance for your help -- Dog is my Co-pilot -- Ing. Danny Dias www.DannTEL.net -- Dog is my Co-pilot -- Ing. Danny Dias www.DannTEL.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT Help needed
Hi Michael, With regards the following error: 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_odbc_clear_cache You can fix that one by modifying /etc/asterisk/modules.conf and uncommenting the following 2 lines: preload = res_odbc.so preload = res_config_odbc.so That will ensure the odbc resource is available for any other applications that may require it. Thanks Bruce -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Sent: 22 November 2010 10:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] URGENT Help needed Also, what happens if you do asterisk -c this may help you figure things out. Hi, These are the WARNINGSI found in /var/log/asterisk/messages after running the above command: [Nov 22 12:10:19] WARNING[2316] udptl.c: T38FaxUdpEC in udptl.conf is no longer supported; use the t38pt_udptl configuration option in sip.conf instead. [Nov 22 12:10:19] WARNING[2316] udptl.c: T38FaxMaxDatagram in udptl.conf is no longer supported; value is now supplied by T.38 applications. [Nov 22 12:10:19] WARNING[2316] loader.c: Error loading module 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_odbc_clear_cache [Nov 22 12:10:19] WARNING[2316] res_config_ldap.c: No directory user found, anonymous binding as default. [Nov 22 12:10:19] ERROR[2316] res_config_ldap.c: No directory URL or host found. [Nov 22 12:10:19] WARNING[2316] utils.c: trying to reset empty pool [Nov 22 12:10:19] WARNING[2316] utils.c: trying to reset empty pool [Nov 22 12:10:19] WARNING[2316] utils.c: trying to reset empty pool [Nov 22 12:10:19] WARNING[2316] pbx.c: Already have an application 'SendFAX' [Nov 22 12:10:19] WARNING[2316] pbx.c: Already have an application 'ReceiveFAX' Thanks, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording format
Hi Joel, We have a meetme on which we are landing two G.711 alaw calls, one coming from TDM another from SIP. Once we those parties are in the conference we are adding one more leg using Local channel and starting to record it. Surely it would be logical if it would be less overhead recording alaw wav since we are using alaw on both parties, but its not. Thanks, Vilius. On 22 November 2010 14:19, Joel Maslak jmas...@antelope.net wrote: What format are the actual calls in? Are they in G.711u/a format or are they in something else (perhaps gsm?) format? I'm asking to find out if Asterisk would need to transcode them. On Mon, Nov 22, 2010 at 6:47 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. What could be the reason for this? Is Asterisk updating wav headers every time it writes? What would be recommended hardware setup for over 60 simultaneous call records? Regards, Vilius. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Vilius Adamkavicius InVADE Technical Support 3 Berkeley Crescent, Bristol United Kingdom BS8 1HA Company Registration Number: 3660482 Registered in England and Wales this email, and any attachment, is intended only for the attention of the addressee. Its unauthorised use, disclosure, storage or copying is not permitted. If you are not the intended recipient, please destroy all copies and inform the sender by return email. If you have received this email in error, please return it to the sender and highlight the error. We accept no legal liability for the content of the message. Any opinions or views presented are solely the responsibility of the author and do not necessarily represent those of InVADE. We cannot guarantee that this message has not been modified in transit, and this message should not be viewed as contractually binding. Although we have taken reasonable steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. international phone number +44(0) 117 33 555 00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_musiconhold.c Bug - Patch to solve?
Application of a patch to any one-or-more-off version of asterisk can be a Russian roulette proposition; If you're applying 1-off you're pretty safe. The more versions between the patch and where you are, the more bullets you are loading into the gun. The best (IMO) procedure for this or any other 'more-than-1-off' patch you want to apply is #1. create a backup copy of the module you're patching #2 apply the patch #3 do a native gcc compile of the module for any obvious gotcha's #4 if nothing happened in step 3, do your make and make install on asterisk to install the patch and check it out. #5 if it works, you're done, if not, put the file back from the backup in step 1 and repeat step 4. FWIW Danny Nicholas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_musiconhold.c Bug - Patch to solve?
On Mon, Nov 22, 2010 at 9:50 AM, Danny Dias ing.diasda...@gmail.com wrote: 2010/11/22 John Novack jnov...@stromberg-carlson.org Danny Dias wrote: Hello John, What i am asking is if i can apply this patch manually or something like this without making any upgrade of Asterisk, has anyone done this before? I can't answer that question. ummm why not? is something wrong? Or i have to upgrade my Asterisk versions...i don't really want to do this... Why not? MANY fixes have been included in the upgrades. Improved security at the least. There are 10-15 versions between where you are operating and what is current I'm sure that the upgrade will fix this, but if applying the patch without making any upgrade will be better for me, my asterisk servers are working with many calls, realtime, fop etc...and an upgrade could make something happen... I would look at a svn diff between the two revisions and see how different they are. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording format
On Mon, Nov 22, 2010 at 8:47 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. Ignoring your real questions, and asking an alternate question: Why not just record in gsm? If your answer is that you have to play these back on Windows, you can build an on-the-fly gsm-to-wav converter using sox. My understanding is that recording in wav doesn't exactly make you have higher audio quality in your recordings, although the experts at codecs could better answer that. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Extensions and loss of Internet connection
On Sun, Nov 21, 2010 at 8:14 PM, Daniel Bareiro daniel-lis...@gmx.net wrote: Hi all! A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. You have to be a bit more specific. For example is your Asterisk box behind a router/nat? Or does your asterisk box have two NICs one for the public and/or natted IP and one for the LAN? You need to specify your exact setup. I'm using Asterisk 1.4.24.1. I was researching on the Internet and I found that it can be related to a bug of chan_sip, can it be? In this case, is there a possible workaround? It's probably not a bug. Maybe you are registering by name and the name resolves to the public IP, and if you are in a DSL cable connection you public IP will change and perhaps you don't even have a public IP. Another possibility is that your ISP does not in fact give you public IPs (like most in the USA) and you have your LAN in the same network definition as theirs. I mean there are so many possibilities but you need to specifiy the exact network setup (IPs, masks, routing, etc.) Thanks in advance for your reply. Regards, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording format
Hi David, Looking at MOS G.711alaw wav most definitely has the higher score than gsm. Moreover recording in gsm is more CPU intense than wav. Therefore your suggestion to do more CPU intense recording and afterwards use system resources to convert it back to wav is not a solution. Also some of our customers require call recordings to be done in wav. Thanks, Vilius. On 22 November 2010 15:03, David Backeberg dbackeb...@gmail.com wrote: On Mon, Nov 22, 2010 at 8:47 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. Ignoring your real questions, and asking an alternate question: Why not just record in gsm? If your answer is that you have to play these back on Windows, you can build an on-the-fly gsm-to-wav converter using sox. My understanding is that recording in wav doesn't exactly make you have higher audio quality in your recordings, although the experts at codecs could better answer that. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT Help needed
Hi Michael, With regards the following error: 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_odbc_clear_cache You can fix that one by modifying /etc/asterisk/modules.conf and uncommenting the following 2 lines: preload = res_odbc.so preload = res_config_odbc.so That will ensure the odbc resource is available for any other applications that may require it. Thank you Bruce. IT works. I also found in FFA manual that SPANDSP and FFA can't exist simultaneously, so since I didn't succeed to run the RxFAX and TxFAX, I went back to FFA and removed the app_fax from the /modules directory and now Asterisk works again. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Someone has hacked into our system
Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom dial w/o Dial, while on-hook?
I've had phones before where, with the phone on-hook, it still implements the local dialplan. E.g., if I dialed 0 (on-hook), after three seconds, it would dial the operator, and have the call on speakerphone. Does Polycom allow this functionality? Clearly, not a necessary feature... but it would be a nice one. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording format
On Mon, Nov 22, 2010 at 03:28:27PM +, Vilius Adamkavicius wrote: Hi David, Looking at MOS G.711alaw wav most definitely has the higher score than gsm. Moreover recording in gsm is more CPU intense than wav. Therefore your suggestion to do more CPU intense recording and afterwards use system resources to convert it back to wav is not a solution. Also some of our customers require call recordings to be done in wav. wav with signed linear payload? I wonder what would happen if you record it as .sl (raw signed linear) and convert it to wav at the end of the call (while mixing). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording format
WAV or wav? One of these has GSM-encoding inside a WAV formatted envelope. That said, I wouldn't expect that to have any noticeable CPU utilization above that of GSM. If you are using the non-GSM version of WAV, then I am as baffled as you - hopefully someone who knows more about this can help. On Mon, Nov 22, 2010 at 7:58 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi Joel, We have a meetme on which we are landing two G.711 alaw calls, one coming from TDM another from SIP. Once we those parties are in the conference we are adding one more leg using Local channel and starting to record it. Surely it would be logical if it would be less overhead recording alaw wav since we are using alaw on both parties, but its not. Thanks, Vilius. On 22 November 2010 14:19, Joel Maslak jmas...@antelope.net wrote: What format are the actual calls in? Are they in G.711u/a format or are they in something else (perhaps gsm?) format? I'm asking to find out if Asterisk would need to transcode them. On Mon, Nov 22, 2010 at 6:47 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. What could be the reason for this? Is Asterisk updating wav headers every time it writes? What would be recommended hardware setup for over 60 simultaneous call records? Regards, Vilius. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Vilius Adamkavicius InVADE Technical Support 3 Berkeley Crescent, Bristol United Kingdom BS8 1HA Company Registration Number: 3660482 Registered in England and Wales this email, and any attachment, is intended only for the attention of the addressee. Its unauthorised use, disclosure, storage or copying is not permitted. If you are not the intended recipient, please destroy all copies and inform the sender by return email. If you have received this email in error, please return it to the sender and highlight the error. We accept no legal liability for the content of the message. Any opinions or views presented are solely the responsibility of the author and do not necessarily represent those of InVADE. We cannot guarantee that this message has not been modified in transit, and this message should not be viewed as contractually binding. Although we have taken reasonable steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. international phone number +44(0) 117 33 555 00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Someone has hacked into our system
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz Sent: Monday, November 22, 2010 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Someone has hacked into our system Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz For #1, start with the CDR. You know that X is calling an overseas number. Determine who X is (or is supposed to be) For #2 (and the rest of #1) restrict your dialing access to a known set of IP's. If you have 5 phones (softphones or actual handsets), block everything that doesn't start with those 5 IP addresses. The first thing I would do is to change all of your passwords in sip.conf and do a sip reload. That will slow down or temporarily stop the hacker. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Extensions and loss of Internet connection
Hi, Alejandro. A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. You have to be a bit more specific. For example is your Asterisk box behind a router/nat? Or does your asterisk box have two NICs one for the public and/or natted IP and one for the LAN? You need to specify your exact setup. Asterisk is not behind the router. The problem I'm having is in the LAN. As I told Phil, I am experiencing the same problem both from a softphone on a workstation with fixed IP as a Grandstream phone (which gets network configuration via DHCP). In both extensions, the Asterisk server is configured with IP, so in that sense, I don't think the server is inaccessible to customers. On the other hand, I made sure to have commented in the sip.conf file any reference to providers such as Ekiga or iptel, so the server should not be trying to get to the Internet. It would appear that the server for some reason was 'locked'. For example, when I try to register from Twinkle softphone, I get the following: - lun 13:41:56 Daniel, registration failed: 503 Service Unavailable - Thanks for your reply. Regards, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Someone has hacked into our system
Blocking udp 5060 in the packet filter in unwanted directions should keep asterisk from setting up SIP connections. The real remedy is to figure out how the hacker got in and close the backdoor. I think a lot of us would be interested in what was the vulnerability. And if it turns out that it was a configuration mistake, don't be shy: for every mistake you did in your config, there are at least a thousand people who did the same mistake. You help them (us) by disclosing the error, and if you have already changed the configuration you should not have the error at that time. On 2010-11-22 17:37, Danny Nicholas wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gary Kuznitz *Sent:* Monday, November 22, 2010 10:23 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Someone has hacked into our system Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz For #1, start with the CDR. You know that X is calling an overseas number. Determine who X is (or is supposed to be) For #2 (and the rest of #1) restrict your dialing access to a known set of IP's. If you have 5 phones (softphones or actual handsets), block everything that doesn't start with those 5 IP addresses. The first thing I would do is to change all of your passwords in sip.conf and do a sip reload. That will slow down or temporarily stop the hacker. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Extensions and loss of Internet connection
On Mon, Nov 22, 2010 at 11:44 AM, Daniel Bareiro daniel-lis...@gmx.net wrote: Hi, Alejandro. A few days I have problems connecting to the Internet on my house [...] It would appear that the server for some reason was 'locked'. For example, when I try to register from Twinkle softphone, I get the following: - lun 13:41:56 Daniel, registration failed: 503 Service Unavailable - I have had a similar problem when we have some sort of network disruption, but it _never_ affects clients on the LAN, it only affects my SIP registrations on the public network. I have 2 NICs one on the public network with a public IP (but dynamic), and one on the LAN. I also have a cron to a dyndns service that updates the name of this server so other PBX can register to it. Anyway, sometimes, but very rare, something happens and the is no way that it re-registers to external SIP sources, and no other external SIP can register with it either. Nothing works except to reboot the server a-la Windoze. I have Asterisk 1.6 on FreeBSD 8. I have always attributed this problem to my set-up or a quirky NIC but maybe it's related to your problem (although it has _never_ happened to us in the LAN extensions). Unable to find a solution, and since it's really very rare, we have test calls every day to make sure everything is working ;-) Best, Alejandro Imass -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording format
We are using wav, not WAV. I believe WAV is the one with GSM. Its a very good idea to compare WAV against wav, will run some tests and come back with outcome, will try Tzafrir's suggestion as well. Thanks guys Vilius. On 22 November 2010 16:31, Joel Maslak jmas...@antelope.net wrote: WAV or wav? One of these has GSM-encoding inside a WAV formatted envelope. That said, I wouldn't expect that to have any noticeable CPU utilization above that of GSM. If you are using the non-GSM version of WAV, then I am as baffled as you - hopefully someone who knows more about this can help. On Mon, Nov 22, 2010 at 7:58 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi Joel, We have a meetme on which we are landing two G.711 alaw calls, one coming from TDM another from SIP. Once we those parties are in the conference we are adding one more leg using Local channel and starting to record it. Surely it would be logical if it would be less overhead recording alaw wav since we are using alaw on both parties, but its not. Thanks, Vilius. On 22 November 2010 14:19, Joel Maslak jmas...@antelope.net wrote: What format are the actual calls in? Are they in G.711u/a format or are they in something else (perhaps gsm?) format? I'm asking to find out if Asterisk would need to transcode them. On Mon, Nov 22, 2010 at 6:47 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. What could be the reason for this? Is Asterisk updating wav headers every time it writes? What would be recommended hardware setup for over 60 simultaneous call records? Regards, Vilius. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Vilius Adamkavicius InVADE Technical Support 3 Berkeley Crescent, Bristol United Kingdom BS8 1HA Company Registration Number: 3660482 Registered in England and Wales this email, and any attachment, is intended only for the attention of the addressee. Its unauthorised use, disclosure, storage or copying is not permitted. If you are not the intended recipient, please destroy all copies and inform the sender by return email. If you have received this email in error, please return it to the sender and highlight the error. We accept no legal liability for the content of the message. Any opinions or views presented are solely the responsibility of the author and do not necessarily represent those of InVADE. We cannot guarantee that this message has not been modified in transit, and this message should not be viewed as contractually binding. Although we have taken reasonable steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. international phone number +44(0) 117 33 555 00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk?
[asterisk-users] Using AMI to harvest / record HOLD time - Using FreePBX
Hi Everyone, I am looking into AMI (using PHP) to record every instance of HOLD that is generated by putting a caller on HOLD (press hold button on the phone set). There is no HOLD in Asterisk but the event Music on Hold is generated when HOLD is pressed. The complexity is that all of the the calls are handled by FreePBX so I don't have the channel IDs etc... Can someone please point out how I can have an AMI session connected at all times (if that is wise) to harvest these Music on Hold events and to record the duration of the HOLD? I would be able to place it in the asteriskcdrdb then for reporting purposes. Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is existing CDR in Asterisk is enough for complete billing
From our experience it is not enough. We had to rewrite CDR generation to suite our billing needs. That was on 1.4.xx, we are not using 1.6+ Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com Find us on Facebook -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil Sent: Monday, November 22, 2010 7:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Is existing CDR in Asterisk is enough for complete billing Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on this. Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN-FAX with Asterisk
On Thu, Nov 18, 2010 at 10:54:53PM +0100, Thorolf Godawa wrote: since some time I am looking for a current and reliable solution to send and receive faxes (probably fax-2-mail and mail-2-fax) in conjunction with Asterisk. [snip] What are you using? mISDN, CAPI4linux, HylaFAX, IAXmodem, chan_misdn, ... ? Hylafax/IAXmodem hasn't let me down so far, it works independent of technology (it only needs alaw/ulaw). Jitter has the ability to kill the transfers, but that shouldn't be any problem with ISDN. Just create a bunch of iaxmodems and configure them in hylafax. For incoming faxes to email I set the callerID name to the emailadress in the dialplan and in etc/FaxDispatch set SENDTO to $CIDNAME. For outgoing faxes from email read the manpage of sendfax (save the attachment, convert it when necessary, call sendfax with the senders emailadress so notification get send back to the sender). -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and cisco 7970 - multiple lines
I can't believe nobody uses cisco 7970 with asterisk to help with my issue. 2 sip lines registered: Line 1: ext 260 Line 2: ext 160 How to get Line 2 blinking when Line 2 (ext 160) is called? For some reason with my setup when I call Line 2 - Line 1 is blinking. I use firmware 8.0.3 Anyone has the same problem or is it just me? Please give me some hint. Thanks, Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call forward problem
On Fri, Nov 19, 2010 at 12:04:47PM +0530, Aparna Narayan wrote:
I tried to perform call forward in asterisk by writing the following in the
dial plan.The data base is getting updated with the caller ID number how
ever the call is not getting forwarded.
[apps]
exten = _*21*XX,1,Set(DB(CFIM/${CALLERID(number)})=${EXTEN:4})
exten = _*21*XX,2,Hangup
exten = #21#,1,DBDel(CFIM/${CALLERID(num)})=${EXTEN:4}
exten = #21#,2,Hangup
You are not actually forwarding the call, just storing a number to
forward to. You need to implement a dialplan where calls to internal
numbers check whether ${DB(CFIM/${EXTEN})} is empty (and do nothing) or
set (and dial that number instead).
--
Daniel Tryba
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Re: [asterisk-users] asterisk and cisco 7970 - multiple lines
Post the germane portions of your xml. How does your phone register each line button? Cassius From: Peter Kowalski kowalla...@gmail.com Organization: GreatValueMart Reply-To: kowalla...@gmail.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 22 Nov 2010 12:38:22 -0600 To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk and cisco 7970 - multiple lines I can¹t believe nobody uses cisco 7970 with asterisk to help with my issue. 2 sip lines registered: Line 1: ext 260 Line 2: ext 160 How to get Line 2 blinking when Line 2 (ext 160) is called? For some reason with my setup when I call Line 2 Line 1 is blinking. I use firmware 8.0.3 Anyone has the same problem or is it just me? Please give me some hint. Thanks, Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Early audio (long distance codes) not working after upgrading to 1.8?
No dice on finding a fix for this. I've been looking through the bug tracker and through the config files and haven't found anything... - Original Message - From: Jonathan C. Bailey jbai...@co.marshall.ia.us To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 21, 2010 9:42:44 PM Subject: Re: [asterisk-users] Early audio (long distance codes) not working after upgrading to 1.8? I know about the Progress command, but isn't that only for *inbound* channels? It's only outbound calls that I have an issue with. My two test chases are: SIP Phone - Asterisk - PRI ...and... Channel Bank - Asterisk - PRI -Jon - Original Message - From: Paul Belanger pabelan...@digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 21, 2010 9:33:21 PM Subject: Re: [asterisk-users] Early audio (long distance codes) not working after upgrading to 1.8? On 10-11-21 09:41 PM, Jonathan C. Bailey wrote: Does anyone know what changed between 1.4 and 1.8 in regards to early audio (both hearing it and interacting with it)? Read UPGRADE.txt and CHANGES *CLI core show application Progress -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and cisco 7970 - multiple lines
Below is my xml button 1 and button 2 portion. Any help will be appreciated. line button=1 featureID9/featureID featureLabelPete(260)/featureLabel proxyproxyip/proxy port5060/port name130/name displayNamePeter/displayName autoAnswer autoAnswerEnabled2/autoAnswerEnabled autoAnswerModeAuto Answer with Speakerphone/autoAnswerMode /autoAnswer callWaiting3/callWaiting authName130/authName authPasswordpass/authPassword sharedLinefalse/sharedLine messageWaitingLampPolicy3/messageWaitingLampPolicy messagesNumber850/messagesNumber ringSettingIdle4/ringSettingIdle ringSettingActive5/ringSettingActive contact7b452e87-4496-4762-e11f-b26751a1884b/contact forwardCallInfoDisplay callerNametrue/callerName callerNumberfalse/callerNumber redirectedNumberfalse/redirectedNumber dialedNumbertrue/dialedNumber /forwardCallInfoDisplay /line line button=2 featureID9/featureID featureLabelIntercom/featureLabel proxyproxyip/proxy port5061/port name160/name displayNamePeter/displayName autoAnswer autoAnswerEnabled3/autoAnswerEnabled autoAnswerModeAuto Answer with Speakerphone/autoAnswerMode /autoAnswer callWaiting3/callWaiting authName160/authName authPasswordpass/authPassword sharedLinefalse/sharedLine messageWaitingLampPolicy3/messageWaitingLampPolicy messagesNumber850/messagesNumber ringSettingIdle4/ringSettingIdle ringSettingActive5/ringSettingActive contact7b452e87-4496-4762-e11f-b26751a1884b/contact forwardCallInfoDisplay callerNametrue/callerName callerNumberfalse/callerNumber redirectedNumberfalse/redirectedNumber dialedNumbertrue/dialedNumber /forwardCallInfoDisplay /line Thanks, Peter From: Cassius Smith [mailto:cass...@cassius.org] Sent: Monday, November 22, 2010 1:12 PM To: kowalla...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and cisco 7970 - multiple lines Post the germane portions of your xml. How does your phone register each line button? Cassius From: Peter Kowalski kowalla...@gmail.com Organization: GreatValueMart Reply-To: kowalla...@gmail.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 22 Nov 2010 12:38:22 -0600 To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk and cisco 7970 - multiple lines I can't believe nobody uses cisco 7970 with asterisk to help with my issue. 2 sip lines registered: Line 1: ext 260 Line 2: ext 160 How to get Line 2 blinking when Line 2 (ext 160) is called? For some reason with my setup when I call Line 2 - Line 1 is blinking. I use firmware 8.0.3 Anyone has the same problem or is it just me? Please give me some hint. Thanks, Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and cisco 7970 - multiple lines
On Mon, Nov 22, 2010 at 11:24 AM, Peter Kowalski kowalla...@gmail.com wrote: Below is my xml button 1 and button 2 portion. Any help will be appreciated. line button=1 name130/name authName130/authName authPasswordpass/authPassword contact7b452e87-4496-4762-e11f-b26751a1884b/contact /line line button=2 name160/name authName160/authName authPasswordpass/authPassword contact7b452e87-4496-4762-e11f-b26751a1884b/contact /line I don't use 7970s, but on the 7941/61s I set the name, authName, and contact all to the SIP username. The first thing that I see is that the Contact is set to the same thing on both lines, which might cause your problem. Try changing the contact to the SIP account name for each line. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and cisco 7970 - multiple lines
I have done something similar; I am using SIP load 8.5.2. I use port 5060 on both line buttons. Cassius From: Peter Kowalski kowalla...@gmail.com Organization: GreatValueMart Reply-To: kowalla...@gmail.com Date: Mon, 22 Nov 2010 13:24:41 -0600 To: Cassius Smith cass...@cassius.org Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: RE: [asterisk-users] asterisk and cisco 7970 - multiple lines Below is my xml button 1 and button 2 portion. Any help will be appreciated. line button=1 featureID9/featureID featureLabelPete(260)/featureLabel proxyproxyip/proxy port5060/port name130/name displayNamePeter/displayName autoAnswer autoAnswerEnabled2/autoAnswerEnabled autoAnswerModeAuto Answer with Speakerphone/autoAnswerMode /autoAnswer callWaiting3/callWaiting authName130/authName authPasswordpass/authPassword sharedLinefalse/sharedLine messageWaitingLampPolicy3/messageWaitingLampPolicy messagesNumber850/messagesNumber ringSettingIdle4/ringSettingIdle ringSettingActive5/ringSettingActive contact7b452e87-4496-4762-e11f-b26751a1884b/contact forwardCallInfoDisplay callerNametrue/callerName callerNumberfalse/callerNumber redirectedNumberfalse/redirectedNumber dialedNumbertrue/dialedNumber /forwardCallInfoDisplay /line line button=2 featureID9/featureID featureLabelIntercom/featureLabel proxyproxyip/proxy port5061/port name160/name displayNamePeter/displayName autoAnswer autoAnswerEnabled3/autoAnswerEnabled autoAnswerModeAuto Answer with Speakerphone/autoAnswerMode /autoAnswer callWaiting3/callWaiting authName160/authName authPasswordpass/authPassword sharedLinefalse/sharedLine messageWaitingLampPolicy3/messageWaitingLampPolicy messagesNumber850/messagesNumber ringSettingIdle4/ringSettingIdle ringSettingActive5/ringSettingActive contact7b452e87-4496-4762-e11f-b26751a1884b/contact forwardCallInfoDisplay callerNametrue/callerName callerNumberfalse/callerNumber redirectedNumberfalse/redirectedNumber dialedNumbertrue/dialedNumber /forwardCallInfoDisplay /line Thanks, Peter From: Cassius Smith [mailto:cass...@cassius.org] Sent: Monday, November 22, 2010 1:12 PM To: kowalla...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and cisco 7970 - multiple lines Post the germane portions of your xml. How does your phone register each line button? Cassius From: Peter Kowalski kowalla...@gmail.com Organization: GreatValueMart Reply-To: kowalla...@gmail.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 22 Nov 2010 12:38:22 -0600 To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk and cisco 7970 - multiple lines I can¹t believe nobody uses cisco 7970 with asterisk to help with my issue. 2 sip lines registered: Line 1: ext 260 Line 2: ext 160 How to get Line 2 blinking when Line 2 (ext 160) is called? For some reason with my setup when I call Line 2 Line 1 is blinking. I use firmware 8.0.3 Anyone has the same problem or is it just me? Please give me some hint. Thanks, Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and cisco 7970 - multiple lines
Solved! Thank you Jonathan. Like you suggested - I've changed port on both lines to 5060 and changed contact so all: name, authName and contact are the same and it is working like charm. Thanks again, Peter -Original Message- From: jthurma...@gmail.com [mailto:jthurma...@gmail.com] On Behalf Of Jonathan Thurman Sent: Monday, November 22, 2010 2:05 PM To: kowalla...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and cisco 7970 - multiple lines On Mon, Nov 22, 2010 at 11:24 AM, Peter Kowalski kowalla...@gmail.com wrote: Below is my xml button 1 and button 2 portion. Any help will be appreciated. line button=1 name130/name authName130/authName authPasswordpass/authPassword contact7b452e87-4496-4762-e11f-b26751a1884b/contact /line line button=2 name160/name authName160/authName authPasswordpass/authPassword contact7b452e87-4496-4762-e11f-b26751a1884b/contact /line I don't use 7970s, but on the 7941/61s I set the name, authName, and contact all to the SIP username. The first thing that I see is that the Contact is set to the same thing on both lines, which might cause your problem. Try changing the contact to the SIP account name for each line. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] libpri 1.4.11.5 Now Available
The Asterisk Development Team has announced the release of libpri 1.4.11.5. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ The release of libpri 1.4.11.5 resolves several issues reported by the community and would not have been possible without your participation. Thank you! The following are some of the issues resolved in this release: * Prevent a CONNECT message from sending a CONNECT ACKNOWLEDGE in the wrong state. (issue #17360. Reported by: shawkris. Patched by rmudgett) * Made Q.921 delay events to Q.931 if the event could immediately generate response frames. (closes issue #17360. Reported by: shawkris. Patched by rmudgett) * BRI PTMP: Active channels not cleared when the interface goes down. (closes issue #17865. Reported by: wimpy. Patched by rmudgett) * Segfault in pri_schedule_del() - ctrl value is invalid. (closes issue #17522) (closes issue #18032. Reported by: schmoozecom. Patched by rmudgett) * Crash when receiving an unknown/unsupported message type. (closes issue #17968. Reported by: gelo. Patched by rmudgett) * B410P gets incoming call packets on ISDN but Asterisk doesn't see the call. (closes issue #18232. Reported by: lelio. Patched by rmudgett) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.11.5 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk pass a call to status answer while still ringing
Hi, I have a problem with dialing status. I'm using Asterisk 1.6 and a patton 4554 gateway for ISDN calls. When I call fixed telephone (not mobile phone) after few ringing the status change to answer but the phone is still ringing, so if I hangup before someone really answer, the call is reported as answered but it isn't. This gives me problem for call charge. Some I idea what can be? Thanks in advance Selva -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Someone has hacked into our system
Thank you very much for help in finding the log. I have the log now. I'd like to know what to look for in trying to figure out how the calls are getting originated. I'd be happy to shere all the information. I just don't want to post information on this public list that might show other people how to get in to our box. Thanks you, Gary Kuznitz On 22 Nov 2010 at 13:11, Danny (Danny Nicholas da...@debsinc.com) commented about RE: [asterisk-users] Someone has hacked into our : From: Gary Kuznitz [mailto:docf...@theoffice.la] Sent: Monday, November 22, 2010 12:20 PM To: Danny Nicholas Subject: Re: [asterisk-users] Someone has hacked into our system Thank you for the quick response. Comments below... I am not familiar with navigating Asterisk. Would you please help me understand how to see the CDR? Thank you, Gary Kuznitz By default, Asterisk keeps the CDR as a flat-file in /var/log/asterisk/cdr-csv/Master.csv which you can open in Excel for easy viewing. If you have a custom cdr (see /etc/asterisk/cdr.conf or /etc/asterisk/cdr_custom.conf for more information), your CDR might be stored in a MYSQL table or some other place.I would start under the assumption that you have the flat file available.Once you have it open, use this link as a guide http://www.voip-info.org/wiki/view/Asterisk+cdr+csv Fields * accountcode: What account number to use: Asterisk billing account, (string, 20 characters) * src: Caller*ID number (string, 80 characters) * dst: Destination extension (string, 80 characters) * dcontext: Destination context (string, 80 characters) * clid: Caller*ID with text (80 characters) * channel: Channel used (80 characters) * dstchannel: Destination channel if appropriate (80 characters) * lastapp: Last application if appropriate (80 characters) * lastdata: Last application data (arguments) (80 characters) * start: Start of call (date/time) * answer: Answer of call (date/time) * end: End of call (date/time) * duration: Total time in system, in seconds (integer) * billsec: Total time call is up, in seconds (integer) * disposition: What happened to the call: ANSWERED, NO ANSWER, BUSY, FAILED * amaflags: What flags to use: see amaflags::DOCUMENTATION, BILL, IGNORE etc, specified on a per channel basis like accountcode. You will want to see if there are any peculiar src fields on your international calls (dst). WPM$68B7.PM$ Description: Mail message body -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Someone has hacked into our system
Use IPTables to lock down your machine to only accept incoming connections from your local network and from the particular IPs that you are expecting connections from (such as your SIP trunk, maybe). That is of course assuming that these calls are made by SIP. Don't forget to also change all the passwords. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz Sent: Monday, November 22, 2010 8:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Someone has hacked into our system Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Someone has hacked into our system
On 11/22/2010 06:44 PM, Kevin Keane wrote: Use IPTables to lock down your machine to only accept incoming connections from your local network and from the particular IPs that you are expecting connections from (such as your SIP trunk, maybe). That is of course assuming that these calls are made by SIP. Don't forget to also change all the passwords. good point - someone can easily just dial in a pots line locally and dial out another one making a long distance call, assuming the dial plan allows this. it doesn't have to be sip involved in any part of the problem. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gary Kuznitz *Sent:* Monday, November 22, 2010 8:23 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Someone has hacked into our system Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk pass a call to status answer while still ringing
UNSUBSCRIBE On Tue, Nov 23, 2010 at 6:46 AM, antselva antse...@tiscali.it wrote: Hi, I have a problem with dialing status. I'm using Asterisk 1.6 and a patton 4554 gateway for ISDN calls. When I call fixed telephone (not mobile phone) after few ringing the status change to answer but the phone is still ringing, so if I hangup before someone really answer, the call is reported as answered but it isn't. This gives me problem for call charge. Some I idea what can be? Thanks in advance Selva -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeffery ___ /\__\ What is the world coming to? \/__/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
