[asterisk-users] Asterisk 1.4 and DAHDI 2.7
Hello, Can someone confirm to me if Asterisk 1.4 can be used with DAHDI 2.7 ? Thanks in advance. -- _ -- 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 1.4 and DAHDI 2.7
On Tue, Nov 05, 2013 at 05:02:13PM +, Rodrigo Borges Pereira wrote: Hello, Can someone confirm to me if Asterisk 1.4 can be used with DAHDI 2.7 ? Thanks in advance. 2.7 is not tested against the head of the Asterisk 1.4 branch, but it *should* work. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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
Re: [asterisk-users] Asterisk 1.4 and DAHDI 2.7
Thanks Shaun. Will give it a go. Regards, On Tue, Nov 5, 2013 at 5:31 PM, Shaun Ruffell sruff...@digium.com wrote: On Tue, Nov 05, 2013 at 05:02:13PM +, Rodrigo Borges Pereira wrote: Hello, Can someone confirm to me if Asterisk 1.4 can be used with DAHDI 2.7 ? Thanks in advance. 2.7 is not tested against the head of the Asterisk 1.4 branch, but it *should* work. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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
[asterisk-users] Asterisk 1.4 CDR vs VoIP Innovations CDR
When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more origination minutes than I am. -- _ -- 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 1.4 CDR vs VoIP Innovations CDR
May be as simple as this: When you terminate a call you start the call before they even get it. When they originate a call, they start the call before you get it. Just a guess without really thinking about this too much. On Thu, Aug 1, 2013 at 10:28 AM, Adam Moffett adamli...@plexicomm.netwrote: When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more origination minutes than I am. -- __**__**_ -- 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-**usershttp://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 1.4 CDR vs VoIP Innovations CDR
A fair guess May be as simple as this: When you terminate a call you start the call before they even get it. When they originate a call, they start the call before you get it. Just a guess without really thinking about this too much. On Thu, Aug 1, 2013 at 10:28 AM, Adam Moffett adamli...@plexicomm.net mailto:adamli...@plexicomm.net wrote: When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more origination minutes than I am. -- _ -- 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 1.4 CDR vs VoIP Innovations CDR
On Thu, Aug 1, 2013 at 10:28 AM, Adam Moffett adamli...@plexicomm.netwrote: When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more origination minutes than I am. If I remember correctly, they bill in sub-minute increments, something like 60 second minimum, then every 6 seconds after that. In other words, if you have a 20 second call, it's billed as 60 seconds, however, if you have a 62 second call, it's billed as 66. I don't remember what they're specific increments are, but I don't believe it was a straight bill. Are you finding that you're off by just a few seconds per call, or by minutes? When you say you're off by 3-4%, are you saying your CDR reports 100 minutes on a call and they are showing 104 minutes, or vice versa? -- _ -- 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 1.4 freezes with queues and iax after virtualization
Thanks a lot Gareth, I just took the astdb file from an older version of the same system and it worked after a few adaptation (registers, and some DB puts/del). Regards, Samuel. On 30 July 2013 11:30, Gareth Blades mailinglist+aster...@dns99.co.ukwrote: On 29/07/13 18:12, samuel wrote: there's no dahdi installed. Following debugging the issue, it looks like the astdb file is broken. Whenever database show command is executed it loops over the same results. The file itself is around 225K but dumping its content via asterisk -rx 'database show' creates and endless file. Is there any easy way to restore the database content? Thanks a lot for the replies, Samuel. There is some information listed here :- http://www.voip-info.org/wiki/**view/Asterisk+databasehttp://www.voip-info.org/wiki/view/Asterisk+database Are you actually storing any data in there yourself? If not it would probably be a lot easier to just rename the file and restart asterisk and it should create a new clean file. -- __**__**_ -- 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-**usershttp://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 1.4 freezes with queues and iax after virtualization
On 29/07/13 18:12, samuel wrote: there's no dahdi installed. Following debugging the issue, it looks like the astdb file is broken. Whenever database show command is executed it loops over the same results. The file itself is around 225K but dumping its content via asterisk -rx 'database show' creates and endless file. Is there any easy way to restore the database content? Thanks a lot for the replies, Samuel. There is some information listed here :- http://www.voip-info.org/wiki/view/Asterisk+database Are you actually storing any data in there yourself? If not it would probably be a lot easier to just rename the file and restart asterisk and it should create a new clean file. -- _ -- 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 1.4 freezes with queues and iax after virtualization
Hi folks, Recently a customer of us moved his old asterisk installation, an 1.4.44 to a VMWARE infraestructure and has started having some weird issues. Asterisk started going slow and even refused to start up. After few tests, it only loaded when deactivating queues and iax2 (with noload in modules file). The thing is that it had been working with these modules loaded and lately it just freezes when trying to use these modules. We've made some checks to the server and there seems to be no issues with load, with swap, with wait (disk access), or other server parameters. Could it be some timing issues? How could we debug further the issue? Thanks a lot in advance, Samuel. -- _ -- 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 1.4 freezes with queues and iax after virtualization
On 29/07/13 12:15, samuel wrote: Hi folks, Recently a customer of us moved his old asterisk installation, an 1.4.44 to a VMWARE infraestructure and has started having some weird issues. Asterisk started going slow and even refused to start up. After few tests, it only loaded when deactivating queues and iax2 (with noload in modules file). The thing is that it had been working with these modules loaded and lately it just freezes when trying to use these modules. We've made some checks to the server and there seems to be no issues with load, with swap, with wait (disk access), or other server parameters. Could it be some timing issues? How could we debug further the issue? Thanks a lot in advance, Samuel. Timing could be an issue. Is dahdi installed? Asterisk 1.4 is old and no longer supported. I would suggest upgrading which would also make the timerfd kernel timing source available if you are running on a recent operating system. See https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces -- _ -- 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 1.4 freezes with queues and iax after virtualization
there's no dahdi installed. Following debugging the issue, it looks like the astdb file is broken. Whenever database show command is executed it loops over the same results. The file itself is around 225K but dumping its content via asterisk -rx 'database show' creates and endless file. Is there any easy way to restore the database content? Thanks a lot for the replies, Samuel. On 29 July 2013 14:07, Gareth Blades mailinglist+aster...@dns99.co.ukwrote: On 29/07/13 12:15, samuel wrote: Hi folks, Recently a customer of us moved his old asterisk installation, an 1.4.44 to a VMWARE infraestructure and has started having some weird issues. Asterisk started going slow and even refused to start up. After few tests, it only loaded when deactivating queues and iax2 (with noload in modules file). The thing is that it had been working with these modules loaded and lately it just freezes when trying to use these modules. We've made some checks to the server and there seems to be no issues with load, with swap, with wait (disk access), or other server parameters. Could it be some timing issues? How could we debug further the issue? Thanks a lot in advance, Samuel. Timing could be an issue. Is dahdi installed? Asterisk 1.4 is old and no longer supported. I would suggest upgrading which would also make the timerfd kernel timing source available if you are running on a recent operating system. See https://wiki.asterisk.org/** wiki/display/AST/Timing+**Interfaceshttps://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces -- __**__**_ -- 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-**usershttp://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
[asterisk-users] asterisk 1.4 and SMS module
Hello; As I am using vicidial and still vicidial is using asterisk 1.4, so how is the SMS module with asterisk 1.4? Is it stable? Also, I am looking to integrate with social medial like whatsapp and facebook, so how is asterisk 1.4 with jaber channel? Regards Bilal -- _ -- 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] DAHDI 1.4 on Kernel 3.0
Thanks a lot for the link and the tip. Have been trying it these days and think it wil work on my system. Thanks again Shaun. 2012/11/8 Shaun Ruffell sruff...@digium.com On Tue, Nov 06, 2012 at 06:49:09PM -0600, Alyed wrote: Hello listers, I'm trying to run DAHDI 1.4 on a 3.0 Debian Kernel in an embedded system, but have faced lots of problems mainly because it has lots of functions looking for the PCI. Have seen so many problems, I'm in fact thinking it cannot be possibly done (at least not in a couple of weeks, by one only man). Has anyone out there had any experience on something like this? or can someone shed some light on how to overcome this issues? Any ideas are very welcome There isn't a 1.4 version of DAHDI. However version v2.6.0 will not build any PCI drivers if the Kernel does not have the PCI bus configured. [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10397 -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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
Re: [asterisk-users] DAHDI 1.4 on Kernel 3.0
On Tue, Nov 06, 2012 at 06:49:09PM -0600, Alyed wrote: Hello listers, I'm trying to run DAHDI 1.4 on a 3.0 Debian Kernel in an embedded system, but have faced lots of problems mainly because it has lots of functions looking for the PCI. Have seen so many problems, I'm in fact thinking it cannot be possibly done (at least not in a couple of weeks, by one only man). Has anyone out there had any experience on something like this? or can someone shed some light on how to overcome this issues? Any ideas are very welcome There isn't a 1.4 version of DAHDI. However version v2.6.0 will not build any PCI drivers if the Kernel does not have the PCI bus configured. [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10397 -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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
[asterisk-users] DAHDI 1.4 on Kernel 3.0
Hello listers, I'm trying to run DAHDI 1.4 on a 3.0 Debian Kernel in an embedded system, but have faced lots of problems mainly because it has lots of functions looking for the PCI. Have seen so many problems, I'm in fact thinking it cannot be possibly done (at least not in a couple of weeks, by one only man). Has anyone out there had any experience on something like this? or can someone shed some light on how to overcome this issues? Any ideas are very welcome -- _ -- 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 1.4 and configuration to be via Database instead of conf files
Dear Binni; My asterisk version is: Connected to Asterisk 1.4.39.1-vici RPM by dem...@goautodial.com So it is only by 1.4.19? By the way, the version I am using has been installed using goautodial. Regards Bilal Hi, I've played around with using a database configuration for Asterisk and it certainly works in 1.4.19. If you want any help setting up the configuration you can contact me directly. Regards Binni -- _ -- 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 1.4 and configuration to be via Database instead of conf files
Hi All; Because vicidial is working with asterisk 1.4, so I would like to know in case of using asterisk 1.4, can I have the configuration to be in the database? As I know that version 1.8 is supporting configuration to be via Database instead of conf files, but what about 1.4? From the other side, if I am going to use Asterisk Now to be a GUI for asterisk, in that case the configuration will be in database or in conf files? By the way: Asterisk Now can work with Asterisk 1.4 and Asterisk 1.8? Regards Bilal -- _ -- 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 1.4.x segfaulting daily
Hello list,
An Asterisk installation that was doing fine suddenly stared segfaulting
a couple of times per day. I enabled all the logging and debugging to
try to find a pattern but there was too much information to see exactly
where it broke. So I enabled core dump and did backtraces and all of
them seem to break on ast_setstate, setting the state to AST_STATE_DOWN.
That's pretty much the only thing I can make of it, don't even know if
that's correct.
Does anyone have any ideas on why this is happening? The backtrace is
attached.
P.S.: I've switched the whole hardware already, including the BRI card
(B400P, OpenVox). Also tried different versions of Asterisk, Dahdi and
mISDN. I'm stuck with 1.4 Asterisk branch and mISDN v1.
Best regards,
Paulo Santos
Core was generated by `/usr/sbin/asterisk'.
Program terminated with signal 11, Segmentation fault.
[New process 21726]
[New process 24376]
[New process 24375]
[New process 24374]
[New process 24371]
[New process 24344]
[New process 23560]
[New process 22868]
[New process 22329]
[New process 22327]
[New process 22325]
[New process 22324]
[New process 22323]
[New process 22322]
[New process 22321]
[New process 22320]
[New process 22319]
[New process 22318]
[New process 22317]
[New process 22316]
[New process 22315]
[New process 22259]
[New process 22208]
[New process 22203]
[New process 22185]
[New process 22184]
[New process 22160]
[New process 21515]
[New process 21725]
[New process 21687]
[New process 21686]
[New process 21685]
[New process 21681]
[New process 21659]
[New process 21658]
[New process 21648]
[New process 21647]
[New process 21609]
[New process 21594]
[New process 21542]
[New process 21540]
[New process 21516]
#0 0x080851ee in ast_setstate (chan=0xb3401c00, state=AST_STATE_DOWN) at
/usr/src/asterisk-1.4.42/include/asterisk/strings.h:37
37 return (!s || (*s == '\0'));
#0 0x080851ee in ast_setstate (chan=0xb3401c00, state=AST_STATE_DOWN) at
/usr/src/asterisk-1.4.42/include/asterisk/strings.h:37
name =
mISDN/4\000u\000ݴ��\177�\020\000@�\b(@�H0ݴf\211q�\020\000@�\b(@�\000\000@�Хe�\b(@�X\b@�h0ݴ\203\225a�P[
3] \000\000\000\000\000
#1 0xb561975d in release_chan (ch=0xb3400858, bc=0x88e8f5c) at
chan_misdn.c:3750
ast = (struct ast_channel *) 0xb3401c00
#2 0xb5622275 in cb_events (event=EVENT_CLEANUP, bc=0x88e8f5c, user_data=0x0)
at chan_misdn.c:4845
msn_valid = -1287644160
held_ch = value optimized out
ch = (struct chan_list *) 0xb3400858
__PRETTY_FUNCTION__ = cb_events
#3 0xb5632d9f in handle_cr (stack=0x88e82d8, frm=value optimized out) at
misdn/isdn_lib.c:1684
channel = 255
bc = (struct misdn_bchannel *) 0x88e8f5c
dummybc = {send_lock = 0xb67feff4, dummy = -1260570753, nt =
-1260572104, pri = -1234083825, port = -1260572068, b_stid = -1260571776,
layer_id = -1260570753, layer = -1234274741, need_disconnect = -1233129484,
need_release = -1260572068, need_release_complete = -1260571776,
dec = -1260571832, l3_id = -1234111388, pid = -1260572068, ces = -1251638304,
restart_channel = -1260570700, channel = -1260571776,
channel_preselected = 0, in_use = -1260571908, last_used = {tv_sec = 1023,
tv_usec = -72515583}, cw = -1260571776, addr = -1260571776,
bframe = 0xb4dd3380 handle_frm: frm-addr:42000303 frm-prim:3f182\n,
bframe_len = -1260571776, time_usec = -1260571729,
astbuf = 0xb4dd377f, misdnbuf = 0xb4dd3380, te_choose_channel = -1260570753,
early_bconnect = 0, dtmf = 0, send_dtmf = 0,
need_more_infos = 0, sending_complete = 0, nodsp = 1635021600, nojitter = 0,
dnumplan = NUMPLAN_UNKNOWN, rnumplan = 1308622848,
onumplan = NUMPLAN_UNKNOWN, cpnnumplan = NUMPLAN_UNINITIALIZED,
progress_coding = 824193585, progress_location = 942881334,
progress_indicator = 3617594, fac_in = {Function = Fac_GetSupportedServices,
u = {Listen = {NotificationMask = 21}, Suspend = {
CallIdentity = \025\000\000\000\000\000\000\000\000\000\000},
Resume = {
CallIdentity = \025\000\000\000\000\000\000\000\000\000\000},
CFActivate = {Handle = 21, Procedure = 0, BasicService = 0,
ServedUserNumber = \000\000\000\000Хe�\001\000\000,
ForwardedToNumber = @�\177�\000\000\000\000�wa�\0203ݴ,
ForwardedToSubaddress = \000\004\000\000�ze�7ݴ@�\177�}, CFDeactivate
= {Handle = 21, Procedure = 0, BasicService = 0,
ServedUserNumber = \000\000\000\000Хe�\001\000\000},
CFInterrogateParameters = {Handle = 21, Procedure = 0, BasicService = 0,
ServedUserNumber = \000\000\000\000Хe�\001\000\000},
CFInterrogateNumbers = {Handle = 21}, CDeflection = {
PresentationAllowed = 21, DeflectedToNumber =
\000\000\000\000\000\000\000\000\000\000Х,
DeflectedToSubaddress = e�\001\000\000\000@�\177�\000\000\000\000�w},
AOCDchu = {chargeNotAvailable = 21, freeOfCharge = 0,
recordedUnits = 0, typeOfChargingInfo = -1, billingId = 0}, AOCDcur =
{chargeNotAvailable
Re: [asterisk-users] Asterisk 1.4.x segfaulting daily
On 14 December 2011 12:56, Paulo Santos paulo.r.san...@sapo.pt wrote: Hello list, An Asterisk installation that was doing fine suddenly stared segfaulting a couple of times per day. I enabled all the logging and debugging to try to find a pattern but there was too much information to see exactly where it broke. So I enabled core dump and did backtraces and all of them seem to break on ast_setstate, setting the state to AST_STATE_DOWN. That's pretty much the only thing I can make of it, don't even know if that's correct. Does anyone have any ideas on why this is happening? The backtrace is attached. P.S.: I've switched the whole hardware already, including the BRI card (B400P, OpenVox). Also tried different versions of Asterisk, Dahdi and mISDN. I'm stuck with 1.4 Asterisk branch and mISDN v1. If I was guessing, I'd say that the channel structure that is being modified by the ast_setstate() call is incomplete, and contains some garbage pointers. If I was guessing further, I'd say that the callerID pointers are the most likely candidate - Does the issue happen when a caller-id withheld call is hung-up? hung-up before being answered perhaps? You'd need to add some debug reporting into ast_setstate() to know for sure. Just my 2p - 1.4.42 is an old version, so the chance of a solid answer is fairly low. Steve -- _ -- 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 1.4.x segfaulting daily
On 14-12-11 13:56, Paulo Santos wrote: Hello list, An Asterisk installation that was doing fine suddenly stared segfaulting a couple of times per day. I enabled all the logging and debugging to try to find a pattern but there was too much information to see exactly where it broke. So I enabled core dump and did backtraces and all of them seem to break on ast_setstate, setting the state to AST_STATE_DOWN. That's pretty much the only thing I can make of it, don't even know if that's correct. Does anyone have any ideas on why this is happening? The backtrace is attached. P.S.: I've switched the whole hardware already, including the BRI card (B400P, OpenVox). Also tried different versions of Asterisk, Dahdi and mISDN. I'm stuck with 1.4 Asterisk branch and mISDN v1. If the suggestion from Steve Davies doesn't work out for you then my suggestion would be to try out the latest DAHDI libpri with the latest Asterisk 1.8 because those versions have built-in support for the 4x BRI HFC chipset which can be found on the Digium b410p and the Openvox B400P. This way you no longer need mISDN V1 and have recent versions with tons of bugs fixed. Here are instructions from Openvox: http://wiki.openvox.cn/index.php/OpenVox_B400P_User_Manual_for_dahdi Please note that in the instructions they use older versions. I would use the latest DAHDI, libpri (don't forget this one) and asterisk 1.8 available here: https://www.asterisk.org/downloads Regards, Patrick -- _ -- 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 1.4.x segfaulting daily
Hello, Thank you all for the replies. Steve Davies wrote: If I was guessing, I'd say that the channel structure that is being modified by the ast_setstate() call is incomplete, and contains some garbage pointers. If I was guessing further, I'd say that the callerID pointers are the most likely candidate - Does the issue happen when a caller-id withheld call is hung-up? hung-up before being answered perhaps? It was an outgoing call that tried to call through the port 2, then 1 and finally 3. The third port has a quite different debug output than the other 2. Maybe it's a problem on that connection, appears to be common on all segfaults. Apparently that third port is something of a strange group of BRI lines between that one and the line on the second port, but behaves differently. I'll try to find out more about it. Patrick Lists wrote: If the suggestion from Steve Davies doesn't work out for you then my suggestion would be to try out the latest DAHDI libpri with the latest Asterisk 1.8 because those versions have built-in support for the 4x BRI HFC chipset which can be found on the Digium b410p and the Openvox B400P. This way you no longer need mISDN V1 and have recent versions with tons of bugs fixed. Unfortunately I can't do that, at least not now. I will, however, migrate it eventually to either mISDN v2 or Dahdi, depending on the state of Dahdi then. P.S.: Attached the log. Best regards, Paulo Santos [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = No match Their Call ID: 333232837-5062-310@192.168.0.8 Their Tag 1036797295 Our tag: as5b7769e2 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = No match Their Call ID: 1693981358-5068-505@192.168.0.7 Their Tag 692402733 Our tag: as170cc25e [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = No match Their Call ID: 1394539361-5064-828@192.168.0.7 Their Tag 1627163612 Our tag: as5f15bf50 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = No match Their Call ID: 1708030692-5060-122@192.168.0.8 Their Tag 52015999 Our tag: as24b80c2d [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on RTP to Off [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on UDPTL to Off [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Allocating new SIP dialog for 1547819775-5062-295@192.168.0.7 - INVITE (With RTP) [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Received INVITE (5) - Command in SIP INVITE [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 0.0.0.0 [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 192.168.0.0 [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 10.0.0.0 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on RTP to On [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on UDPTL to On [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = Found Their Call ID: 1547819775-5062-295@192.168.0.7 Their Tag 2074339809 Our tag: as2515e4b3 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Received ACK (6) - Command in SIP ACK [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Stopping retransmission on '1547819775-5062-295@192.168.0.7' of Response 2940: Match Found [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: = Found Their Call ID: 1547819775-5062-295@192.168.0.7 Their Tag 2074339809 Our tag: as2515e4b3 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Received INVITE (5) - Command in SIP INVITE [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 0.0.0.0 [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 192.168.0.0 [Dec 12 16:38:36] DEBUG[22160] acl.c: # Testing 192.168.0.7 with 10.0.0.0 [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on RTP to On [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Setting NAT on UDPTL to On [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP o=11 8002 8000 IN IP4 192.168.0.7... UNSUPPORTED. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.7... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:4 G723/8000... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Dec 12 16:38:36] DEBUG[22160] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000... OK. [Dec 12 16:38:36] DEBUG[22160]
Re: [asterisk-users] Asterisk 1.4 - Help/Doc for Park() application [SOLVED]
2011/12/5 Olivier oza_4...@yahoo.fr Hi, Porting a dialplan from 1.6.1 to and old 1.4 install, it seems Park() application uses different arguments. The only doc I could get a hand on is (core show application Park) this one : [Synopsis] Park yourself [Description] Park():Used to park yourself (typically in combination with a supervised transfer to know the parking space). This application is always registered internally and does not need to be explicitly added into the dialplan, although you should include the 'parkedcalls' context (or the context specified in features.conf). If you set the PARKINGEXTEN variable to an extension in your parking context, park() will park the call on that extension, unless it already exists. In that case, execution will continue at next priority. More specifically, I'm getting this : -- Executing [9200@autopark:49] Park(SIP/9140-0991dd30, 1000*30|9200|local|s) in new stack == Parked SIP/9140-0991dd30 on 701@parkedcalls. Will timeout back to extension [autopark] s, 1 in 45 seconds Above that, silent option 's' is ignored (parking position is read to incoming channel). So it seems, my timeout, return context and feedback options are not correctly understood. Suggestions ? Cheers Hi, Replying to myself, I worked around this using ParkAndAnnounce app instead (of Park). Too bad I could find by myself what was missing in documentation. 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
Re: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8
On 12/09/11 09:48 PM, Joseph wrote: Was NAT problem fixed in 1.8.7 ? I'm using 1.4 but I've tried 1.6 and 1.8.5 and in both cases connection is not working with my provider with SIP + NAT. The connection is showing up as registered but the call is not coming IN (congestion). Can you define NAT problem? I'm unaware of any issues with Asterisk (or end points) behind NAT. It is mostly likely a configuration issue rather than a bug. -- Leif Madsen http://www.oreilly.com/catalog/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 1.4 Vs 1.6 Vs 1.8
Hello i am not sure if this has been discussed before.. i have an asterisk 1.4 server that i managed to test it with 500+ concurrent calls and hit 800 concurrent calls with no problem CPU USAGE 90% i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed at 100 concurrent calls. my question is .. is there a different in resource consumption between all versions? how come 1.4 could handle over 500 calls while 1.6 crashed at 100? please advise? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- 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 1.4 Vs 1.6 Vs 1.8
On Mon, Sep 12, 2011 at 11:19 AM, Tarek Sawah tareksa...@hotmail.com wrote: i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed at 100 concurrent calls. please advise? Nobody will know why your asterisk crashed unless you follow the instructions here: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace Please try that, and then rerun your call test. -- _ -- 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 1.4 Vs 1.6 Vs 1.8
I personally would not bother with 1.6 unless you needed some feature in that branch. 1.4 is the stable branch, but it seems that all of the resources are being channeled into 1.8 and 10.0, so 1.6 is a rabbit hole you really shouldn't be headed into. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah Sent: Monday, September 12, 2011 10:19 AM To: Asterisk Users Subject: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8 Hello i am not sure if this has been discussed before.. i have an asterisk 1.4 server that i managed to test it with 500+ concurrent calls and hit 800 concurrent calls with no problem CPU USAGE 90% i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed at 100 concurrent calls. my question is .. is there a different in resource consumption between all versions? how come 1.4 could handle over 500 calls while 1.6 crashed at 100? please advise? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- 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 1.4 Vs 1.6 Vs 1.8
Actually i had to upgrade to 1.6 due to a provider problem with session-timers and RTP data .. then i downgraded again to 1.4. do you suggest that i test 1.8 instead of 1.6? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 12 Sep 2011 10:54:35 -0500 Subject: Re: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8 I personally would not bother with 1.6 unless you needed some feature in that branch. 1.4 is the stable branch, but it seems that all of the resources are being channeled into 1.8 and 10.0, so 1.6 is a rabbit hole you really shouldn't be headed into. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah Sent: Monday, September 12, 2011 10:19 AM To: Asterisk Users Subject: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8 Hello i am not sure if this has been discussed before.. i have an asterisk 1.4 server that i managed to test it with 500+ concurrent calls and hit 800 concurrent calls with no problem CPU USAGE 90% i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed at 100 concurrent calls. my question is .. is there a different in resource consumption between all versions? how come 1.4 could handle over 500 calls while 1.6 crashed at 100? please advise? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- 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 1.4 Vs 1.6 Vs 1.8
I think that is your best bet. 1.8.6 unless somebody has a good reason not to. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah Sent: Monday, September 12, 2011 11:00 AM To: Asterisk Users Subject: Re: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8 Actually i had to upgrade to 1.6 due to a provider problem with session-timers and RTP data .. then i downgraded again to 1.4. do you suggest that i test 1.8 instead of 1.6? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 12 Sep 2011 10:54:35 -0500 Subject: Re: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8 I personally would not bother with 1.6 unless you needed some feature in that branch. 1.4 is the stable branch, but it seems that all of the resources are being channeled into 1.8 and 10.0, so 1.6 is a rabbit hole you really shouldn't be headed into. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah Sent: Monday, September 12, 2011 10:19 AM To: Asterisk Users Subject: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8 Hello i am not sure if this has been discussed before.. i have an asterisk 1.4 server that i managed to test it with 500+ concurrent calls and hit 800 concurrent calls with no problem CPU USAGE 90% i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed at 100 concurrent calls. my question is .. is there a different in resource consumption between all versions? how come 1.4 could handle over 500 calls while 1.6 crashed at 100? please advise? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- 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 -- _ -- 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 1.4 Vs 1.6 Vs 1.8
On 11-09-12 12:07 PM, Danny Nicholas wrote: I think that is your best bet. 1.8.6 unless somebody has a good reason not to. You actually might want to test with 1.8.7.0-rc1, this will fix 2 big issue. A performance regressions and timerfd. -- 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
Re: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8
On 09/12/11 14:08, Paul Belanger wrote: On 11-09-12 12:07 PM, Danny Nicholas wrote: I think that is your best bet. 1.8.6 unless somebody has a good reason not to. You actually might want to test with 1.8.7.0-rc1, this will fix 2 big issue. A performance regressions and timerfd. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org Was NAT problem fixed in 1.8.7 ? I'm using 1.4 but I've tried 1.6 and 1.8.5 and in both cases connection is not working with my provider with SIP + NAT. The connection is showing up as registered but the call is not coming IN (congestion). -- Joseph -- _ -- 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 1.4 func_odbc frustrations
Maybe somebody can help me here.
I've finally got another server together, so I can test and upgrade a
couple of my older 1.4.x installations.
I figured that while I'm at it, I'll give func_odbc a try (have been
using the mysql addon), knowing full well that when I finally move over
to 1.8.x, it's what I'm planning on using.
I've installed all the requisites listed for ODBC, compiled and install
the current 1.4.41.1 (Was current a couple days ago) and set out
Googling how-tos and digging into voip-info.org
After an hour, I had what seemed to be a good test case, so did a copy
of my dialplan and started making changes.
On querying my database, everything is working as expected, but for the
life of me, I cannot get entries in my database to update the master
mysql server
I've seen lots of conflicting data on how it should be written out in
the func_odbc.conf and a lot of the info is for 1.6.
My setup:
1 master mysql server
1 test slave
The test Asterisk system reads from the local mysql database and writes
back to the master.
My /etc/odbc.ini
[MySQL-Conferencing]
Description = Conferencing MySQL ODBC
Driver = MySQL
Socket = /var/lib/mysql/mysql.sock
Server = 127.0.0.1
User= username
Password= password
Database= Conferencing
Option = 3
[MySQL-Corporate]
Description = Conferencing MySQL ODBC
Driver = MySQL
Server = 192.168.104.142
User= username
Password= password
Database= Conferencing
Option = 3
My /etc/asterisk/res_odbc.conf
[MySQL-Conferencing]
enabled = yes
dsn = MySQL-Conferencing
username = username
password = password
preconnect = yes
[MySQL-Corporate]
enabled = yes
dsn = MySQL-Corporate
username = username
password = password
preconnect = yes
My /etc/asterisk/func_odbc.conf
[CONFERENCE]
dsn=MySQL-Conferencing
read=SELECT room, password, admin, scheduled, owner, comments FROM
${ARG1} WHERE ${ARG2}=${SQL_ESC(${ARG3})}
[CONFERENCE_WRITE]
dsn=MySQL-Corporate
write=UPDATE Corporate SET room=${VAL1}, password=${VAL2},
admin=${VAL3}, scheduled=${VAL4}, owner=${VAL5}, comments=${VAL6} WHERE
admin=${VAL3}
The reading from ODBC works fine:
exten =
s-verify,n,Set(ARRAY(conference.room,conference.password,conference.admin,conference.scheduled,conference.owner,conference.comments)=${ODBC_CONFERENCE_WRITE(Corporate,admin,${get-admin-password})})
The Writing does not work.
exten =
s-setup,n,Set(ODBC_CONFERENCE_WRITE(room=${conference.room}\,password=${put-new-password}\,admin=${conference.admin}\,scheduled=${TODAY}\,owner=${conference.owner}\,comments=${conference.comments})
Any suggestions would be appreciated,
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 http://www.api-digital.com --
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Re: [asterisk-users] From 1.4 to 1.8: stdexten issue
On Sun, Apr 03, 2011 at 10:35:52PM +0200, Benny Amorsen wrote: stdexten in the default extensions.conf seems to only handle extensions with at least 2 digits... Good one, I hadn't noticed that. Thanks that fixed it!!! -- Mathieu Chouquet-Stringer math...@csetco.com The sun itself sees not till heaven clears. -- William Shakespeare -- -- _ -- 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] From 1.4 to 1.8: stdexten issue
Hello all,
I'm in the middle of upgrading my asterisk setup to 1.8 (1.8.2.3) and
I'm completely confused by the gosub/stdexten thing.
I used to call the stdexten macro but I haven't been able to figure out
how to use Gosub.
I'm using the sample extensions.conf and added something like this:
=
[home]
include = stdexten
exten = 1,1,Gosub(${EXTEN},stdexten(SIP/phone1))
=
But if I call 1, all I get is:
[Apr 3 18:20:51] NOTICE[9031]: pbx.c:4119 pbx_extension_helper: No such label
'stdexten' in extension '1' in context 'home'
[Apr 3 18:20:51] WARNING[9031]: pbx.c:10174 pbx_parseable_goto: Priority
'stdexten' must be a number 0, or valid label
[Apr 3 18:20:51] ERROR[9031]: app_stack.c:411 gosub_exec: Gosub address is
invalid: '1,stdexten(SIP/phone1)'
I've googled and pretty much tried all forms of the syntax but I've yet
to make it work. For instance I tried not including stdexten and
calling Gosub with (stdexten,${EXTEN},1(SIP/phone1)) but it didn't work
either...
Can anyone sched some light here? I think I got lost trying to figure
this out...
What am I missing here?
Best,
--
Mathieu Chouquet-Stringer math...@csetco.com
The sun itself sees not till heaven clears.
-- William Shakespeare --
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Re: [asterisk-users] From 1.4 to 1.8: stdexten issue
Mathieu Chouquet-Stringer math...@csetco.com writes:
I've googled and pretty much tried all forms of the syntax but I've yet
to make it work. For instance I tried not including stdexten and
calling Gosub with (stdexten,${EXTEN},1(SIP/phone1)) but it didn't work
either...
stdexten in the default extensions.conf seems to only handle extensions
with at least 2 digits...
/Benny
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[asterisk-users] Migrating 1.4 to 1.6.2
much static testing of my realtime configuration and applications I'm almost ready to pull the trigger. The one thing I've been able to determine is what I need to do to migrate my g729 licenses. Has anyone got any advice for me on this? The Digium site is... difficult to navigate TIA Bruce Ferrell -- _ -- 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] Migrating 1.4 to 1.6.2
From: Bruce Ferrell bferr...@baywinds.org much static testing of my realtime configuration and applications I'm almost ready to pull the trigger. The one thing I've been able to determine is what I need to do to migrate my g729 licenses. Has anyone got any advice for me on this? The Digium site is... difficult to navigate TIA Bruce Ferrell--- If you are not changing servers you just download the correct binary for 1.6.2 for your machine. If your are moving machines then you must re-register the license on the new box. If you have moved them before you must call Digium and have them increment the count on the licenses. Here is a link to the general install instructions. http://downloads.digium.com/pub/telephony/codec_g729/README It is not really hard to do you just need to follow the steps. Bryant -- _ -- 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 1.4 and TE420P
I have a site running 1.4.17 with Zaptel. They want to add a TE420P for additional T1 capacity. I'm 99% sure this will work, anyone aware of a reason it wont? Thanks, James -- _ -- 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 1.4 and TE420P
On Fri, Aug 6, 2010 at 10:33 AM, James Texter james.tex...@gmail.comwrote: I have a site running 1.4.17 with Zaptel. They want to add a TE420P for additional T1 capacity. I'm 99% sure this will work, anyone aware of a reason it wont? Thanks, James I've got a client running a TE420P with asterisk 1.4.33.1 and DAHDI 2.2, works fine. They've only got the one card in the box, but it's using all 4 ports. -- Thanks, --Warren Selby http://www.selbytech.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] Asterisk 1.4 and TE420P
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Subject: Re: [asterisk-users] Asterisk 1.4 and TE420P On Fri, Aug 6, 2010 at 10:33 AM, James Texter james.tex...@gmail.com wrote: I have a site running 1.4.17 with Zaptel. They want to add a TE420P for additional T1 capacity. I'm 99% sure this will work, anyone aware of a reason it wont? I've got a client running a TE420P with asterisk 1.4.33.1 and DAHDI 2.2, works fine. They've only got the one card in the box, but it's using all 4 ports. To expand on OP's question, is he going to have to upgrade to 1.4.3X/DAHDI to make the TE420P work? -- _ -- 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 1.4 and TE420P
On Fri, Aug 6, 2010 at 11:24 AM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren Selby *Subject:* Re: [asterisk-users] Asterisk 1.4 and TE420P On Fri, Aug 6, 2010 at 10:33 AM, James Texter james.tex...@gmail.com wrote: I have a site running 1.4.17 with Zaptel. They want to add a TE420P for additional T1 capacity. I'm 99% sure this will work, anyone aware of a reason it wont? I've got a client running a TE420P with asterisk 1.4.33.1 and DAHDI 2.2, works fine. They've only got the one card in the box, but it's using all 4 ports. To expand on OP’s question, is he going to have to upgrade to 1.4.3X/DAHDI to make the TE420P work? I've only ever used it with DAHDI. I've used it with version of asterisk down to 1.4.22 though, I'm pretty sure. -- Thanks, --Warren Selby http://www.selbytech.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] subject: 1.4 vs 1.6
asteriskuptospeed.linuxinnovations.com is also a good resource for spotting many practical differences between the various versions. On Wed, Feb 24, 2010 at 8:36 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Wednesday 24 February 2010 10:16:25 Miguel Molina wrote: Gergo Csibra escribió: Wednesday, February 24, 2010, 3:56:50 PM, David wrote: On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote: Hi Guys We are using asterisk 1.4 on all of our platforms for a while now. Some of our partners recommended to use asterisk 1.6 in order to improve overall stability and performance. Can someone please let me know if you have a such experience? Also, do you have any other negative or positive comments on 1.6 If it isn't broke, don't 'fix' it. There are benefits to 1.6, like dramatically enhanced SIP support, much faster dialplan processing, easier faxing, changes to dialplan syntax, and lots of other features. I would say the improvement of going to 1.6 is only if you are trying to expect more from the same gear, or want the new features. If you're not actually having problems, don't change anything. Yes, and check this page: http://www.asterisk.org/asterisk-versions as you can see, the 1.4 version is LTS, and the 1.6 isn't, but the upcoming 1.8 will be LTS too. So don't change to 1.6 :) That sounds reasonable, but as I have seen through several years following the asterisk project, when 1.8.0 will be released it will be far less stable than the more used and mature 1.6.0.X, for example. I would prefer to do a middle step for upgrading, that would be 1.4.X - 1.6.0.X - 1.8.X when it becomes really stable. Asterisk history has shown us that a newly released branch, no matter if it's LTS on the new release schema, will need time and a large user base that adopts it to report bugs and help stabilize it. I would not underestimate the actual 1.6.X branches. Additionally, it's worth noting that the dates above are meant to be the EARLIEST dates that development, security fixes, etc. will end. It is quite possible that we will elect to extend some of them. The whole idea is to give companies advance notice of at least six months before we stop supporting a release. The end is coming; but it might be delayed. :-) -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 1.4 Followme Question
I have a question related to FollowMe on Asterisk 1.4. Is there a way to force Asterisk to always leave VM on the forwarded extension's cell phone, as opposed to pulling the call back from the forward to cell and depositing in Asterisk voicemail? Thanks in Advance! -- *Cory J Andrews* 725 Powell Lane Lewiston, NY 14092 voice - 716.579.6331 email - ipcbc...@gmail.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] Asterisk 1.4 Followme Question
Isnt that the point of the FMFM – to allow the call to come back into the asterisk server and have your voicemail managed in one location? If not wanted, I guess remove the voicemail step from the FMFM config and just have it end on the forwarded cellphone. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Cory Andrews *Sent:* Friday, March 05, 2010 10:16 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Asterisk 1.4 Followme Question I have a question related to FollowMe on Asterisk 1.4. Is there a way to force Asterisk to always leave VM on the forwarded extension's cell phone, as opposed to pulling the call back from the forward to cell and depositing in Asterisk voicemail? Thanks in Advance! -- *Cory J Andrews* 725 Powell Lane Lewiston, NY 14092 voice - 716.579.6331 email - ipcbc...@gmail.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
[asterisk-users] subject: 1.4 vs 1.6
Hi Guys We are using asterisk 1.4 on all of our platforms for a while now. Some of our partners recommended to use asterisk 1.6 in order to improve overall stability and performance. Can someone please let me know if you have a such experience? Also, do you have any other negative or positive comments on 1.6 Very much thanking you for your help!!! Juan _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] subject: 1.4 vs 1.6
On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote: Hi Guys We are using asterisk 1.4 on all of our platforms for a while now. Some of our partners recommended to use asterisk 1.6 in order to improve overall stability and performance. Can someone please let me know if you have a such experience? Also, do you have any other negative or positive comments on 1.6 If it isn't broke, don't 'fix' it. There are benefits to 1.6, like dramatically enhanced SIP support, much faster dialplan processing, easier faxing, changes to dialplan syntax, and lots of other features. I would say the improvement of going to 1.6 is only if you are trying to expect more from the same gear, or want the new features. If you're not actually having problems, don't change anything. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] subject: 1.4 vs 1.6
Well.. we do from time to time have SIP attacks, Core dumps and lately very weird issues with Cisco phone becoming unreachable. Anyone had issues with Cisco 7940 where by ALL of the phones will for 30-90 seconds become unreachable? All phones are on T1 MPLS network using Cisco 26xx routers.. Juan Date: Wed, 24 Feb 2010 09:56:50 -0500 From: dbackeb...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] subject: 1.4 vs 1.6 On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote: Hi Guys We are using asterisk 1.4 on all of our platforms for a while now. Some of our partners recommended to use asterisk 1.6 in order to improve overall stability and performance. Can someone please let me know if you have a such experience? Also, do you have any other negative or positive comments on 1.6 If it isn't broke, don't 'fix' it. There are benefits to 1.6, like dramatically enhanced SIP support, much faster dialplan processing, easier faxing, changes to dialplan syntax, and lots of other features. I would say the improvement of going to 1.6 is only if you are trying to expect more from the same gear, or want the new features. If you're not actually having problems, don't change anything. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] subject: 1.4 vs 1.6
Wednesday, February 24, 2010, 3:56:50 PM, David wrote: On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote: Hi Guys We are using asterisk 1.4 on all of our platforms for a while now. Some of our partners recommended to use asterisk 1.6 in order to improve overall stability and performance. Can someone please let me know if you have a such experience? Also, do you have any other negative or positive comments on 1.6 If it isn't broke, don't 'fix' it. There are benefits to 1.6, like dramatically enhanced SIP support, much faster dialplan processing, easier faxing, changes to dialplan syntax, and lots of other features. I would say the improvement of going to 1.6 is only if you are trying to expect more from the same gear, or want the new features. If you're not actually having problems, don't change anything. Yes, and check this page: http://www.asterisk.org/asterisk-versions as you can see, the 1.4 version is LTS, and the 1.6 isn't, but the upcoming 1.8 will be LTS too. So don't change to 1.6 :) -- Best regards, Gergomailto:csi...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] subject: 1.4 vs 1.6
Gergo Csibra escribió: Wednesday, February 24, 2010, 3:56:50 PM, David wrote: On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote: Hi Guys We are using asterisk 1.4 on all of our platforms for a while now. Some of our partners recommended to use asterisk 1.6 in order to improve overall stability and performance. Can someone please let me know if you have a such experience? Also, do you have any other negative or positive comments on 1.6 If it isn't broke, don't 'fix' it. There are benefits to 1.6, like dramatically enhanced SIP support, much faster dialplan processing, easier faxing, changes to dialplan syntax, and lots of other features. I would say the improvement of going to 1.6 is only if you are trying to expect more from the same gear, or want the new features. If you're not actually having problems, don't change anything. Yes, and check this page: http://www.asterisk.org/asterisk-versions as you can see, the 1.4 version is LTS, and the 1.6 isn't, but the upcoming 1.8 will be LTS too. So don't change to 1.6 :) That sounds reasonable, but as I have seen through several years following the asterisk project, when 1.8.0 will be released it will be far less stable than the more used and mature 1.6.0.X, for example. I would prefer to do a middle step for upgrading, that would be 1.4.X - 1.6.0.X - 1.8.X when it becomes really stable. Asterisk history has shown us that a newly released branch, no matter if it's LTS on the new release schema, will need time and a large user base that adopts it to report bugs and help stabilize it. I would not underestimate the actual 1.6.X branches. Just IMHO, any opinions welcome. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] subject: 1.4 vs 1.6
On Wednesday 24 February 2010 10:16:25 Miguel Molina wrote: Gergo Csibra escribió: Wednesday, February 24, 2010, 3:56:50 PM, David wrote: On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote: Hi Guys We are using asterisk 1.4 on all of our platforms for a while now. Some of our partners recommended to use asterisk 1.6 in order to improve overall stability and performance. Can someone please let me know if you have a such experience? Also, do you have any other negative or positive comments on 1.6 If it isn't broke, don't 'fix' it. There are benefits to 1.6, like dramatically enhanced SIP support, much faster dialplan processing, easier faxing, changes to dialplan syntax, and lots of other features. I would say the improvement of going to 1.6 is only if you are trying to expect more from the same gear, or want the new features. If you're not actually having problems, don't change anything. Yes, and check this page: http://www.asterisk.org/asterisk-versions as you can see, the 1.4 version is LTS, and the 1.6 isn't, but the upcoming 1.8 will be LTS too. So don't change to 1.6 :) That sounds reasonable, but as I have seen through several years following the asterisk project, when 1.8.0 will be released it will be far less stable than the more used and mature 1.6.0.X, for example. I would prefer to do a middle step for upgrading, that would be 1.4.X - 1.6.0.X - 1.8.X when it becomes really stable. Asterisk history has shown us that a newly released branch, no matter if it's LTS on the new release schema, will need time and a large user base that adopts it to report bugs and help stabilize it. I would not underestimate the actual 1.6.X branches. Additionally, it's worth noting that the dates above are meant to be the EARLIEST dates that development, security fixes, etc. will end. It is quite possible that we will elect to extend some of them. The whole idea is to give companies advance notice of at least six months before we stop supporting a release. The end is coming; but it might be delayed. :-) -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
I suggest you install it from source, that way you can learn more about asterisk. 2010/1/16 William Stillwell (Lists) william.stillwell-li...@ablebody.net: Here is the 1.4.x version on centos 5 walk through. http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Nik Sent: Friday, January 15, 2010 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4 Provided there is no comprehensive install guides (or is there?) yes I would like to see an easy install script which can install it all. On Fri, Jan 15, 2010 at 12:27 PM, Bruce Nik brucev...@gmail.com wrote: Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
On 01/16/10 04:27, Bruce Nik wrote: Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Try PBX-in-a-Flash. Undoubtedly it won't do everything you want out of the box, but I suspect it will do /most/ of what you want out of the box. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install
Use kickstart to configure your default packages, and then set up a shell script to install the additional stuff you need. :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
On Sat, Jan 16, 2010 at 08:48:27PM +1100, Rob Hillis wrote: On 01/16/10 04:27, Bruce Nik wrote: Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Try PBX-in-a-Flash. Undoubtedly it won't do everything you want out of the box, but I suspect it will do /most/ of what you want out of the box. But will not let you debug that install script. I tend to distrust running such a hidden script. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
On 01/17/10 01:15, Tzafrir Cohen wrote: Try PBX-in-a-Flash. Undoubtedly it won't do everything you want out of the box, but I suspect it will do /most/ of what you want out of the box. But will not let you debug that install script. I tend to distrust running such a hidden script What hidden script are you referring to? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
On Fri, Jan 15, 2010 at 12:27 PM, Bruce Nik brucev...@gmail.com wrote: Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Thanks, Bruce Do you like 'kitchen sink' installs? I can't think of any way to decide on an asterisk configuration that out-of-the-box would be right for everybody... as in, fax support? g729 licenses? whether or not to build against DAHDI? You get the idea. The only way I can think to do it would to be to build in a lot of stuff that most people would never want in their asterisk, which would then result in having to restart asterisk because you need a software update to a package that is a dependency for a part of asterisk you don't use anyway. Anybody who was using asterisk in a serious production environment would probably prefer the control of having most of what they don't want compiled out. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
Provided there is no comprehensive install guides (or is there?) yes I would like to see an easy install script which can install it all. On Fri, Jan 15, 2010 at 12:27 PM, Bruce Nik brucev...@gmail.com wrote: Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
On Fri, Jan 15, 2010 at 3:15 PM, Bruce Nik brucev...@gmail.com wrote: Provided there is no comprehensive install guides (or is there?) yes I would like to see an easy install script which can install it all. tar xvzf ./configure make (optional, do a 'make menuconfig') make install But the problem is that there are steps before the configure you need if you want support for more than barebones asterisk. Nobody knows what you personally need except you. Maybe I'm the only one who doesn't think it's so bad to build from source. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
Here is the 1.4.x version on centos 5 walk through. http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Nik Sent: Friday, January 15, 2010 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4 Provided there is no comprehensive install guides (or is there?) yes I would like to see an easy install script which can install it all. On Fri, Jan 15, 2010 at 12:27 PM, Bruce Nik brucev...@gmail.com wrote: Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR_MYSQL 1.4 Database Structure
So I'm noticing from the docs/ on Asterisk Addons 1.4.10 that the database structure for cdr_mysql is: CREATE TABLE cdr ( calldate datetime NOT NULL default '-00-00 00:00:00', clid varchar(80) NOT NULL default '', src varchar(80) NOT NULL default '', dst varchar(80) NOT NULL default '', dcontext varchar(80) NOT NULL default '', channel varchar(80) NOT NULL default '', dstchannel varchar(80) NOT NULL default '', lastapp varchar(80) NOT NULL default '', lastdata varchar(80) NOT NULL default '', duration int(11) NOT NULL default '0', billsec int(11) NOT NULL default '0', disposition varchar(45) NOT NULL default '', amaflags int(11) NOT NULL default '0', accountcode varchar(20) NOT NULL default '', uniqueid varchar(32) NOT NULL default '', userfield varchar(255) NOT NULL default '' ); Just curious if anyone has successfully patched cdr_addon_mysql to use accept the latest cdr fields from 1.4 ... namely: 'start', 'answer', 'end'? Seems logical that the cdr_mysql addon should be updated to reflect the current cdr. And for backwards compatibility it can still accept 'calldate'. Thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR_MYSQL 1.4 Database Structure
On Wednesday 30 December 2009 10:52:48 Robert Broyles wrote: So I'm noticing from the docs/ on Asterisk Addons 1.4.10 that the database structure for cdr_mysql is: CREATE TABLE cdr ( calldate datetime NOT NULL default '-00-00 00:00:00', clid varchar(80) NOT NULL default '', src varchar(80) NOT NULL default '', dst varchar(80) NOT NULL default '', dcontext varchar(80) NOT NULL default '', channel varchar(80) NOT NULL default '', dstchannel varchar(80) NOT NULL default '', lastapp varchar(80) NOT NULL default '', lastdata varchar(80) NOT NULL default '', duration int(11) NOT NULL default '0', billsec int(11) NOT NULL default '0', disposition varchar(45) NOT NULL default '', amaflags int(11) NOT NULL default '0', accountcode varchar(20) NOT NULL default '', uniqueid varchar(32) NOT NULL default '', userfield varchar(255) NOT NULL default '' ); Just curious if anyone has successfully patched cdr_addon_mysql to use accept the latest cdr fields from 1.4 ... namely: 'start', 'answer', 'end'? Seems logical that the cdr_mysql addon should be updated to reflect the current cdr. And for backwards compatibility it can still accept 'calldate'. The MySQL driver contains all of the same information, albeit in a slightly different form. Calldate is the same as start, calldate plus duration minus billsec is the same as answer, and calldate plus duration is the same as end. Generally, we do not make design changes in the middle of a release cycle, especially given that such changes would break a great many existing systems. Given that there's no security reason why we would need to make such a change, it is out of the question. While you're certainly welcome to make such a change on your own systems, such a change will not be committed in the 1.4 addons. In the 1.6 series and forward, we've changed the mysql driver to scan the table metadata and adapt the queries to the table structure. Therefore, you could, in fact, use 'start', 'answer', and 'end' in the 1.6 series, as you suggested, above, and it would work perfectly well. On the other hand, if you kept the legacy structure, that would work, too. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR_MYSQL 1.4 Database Structure
Tilghman Lesher wrote: On Wednesday 30 December 2009 10:52:48 Robert Broyles wrote: So I'm noticing from the docs/ on Asterisk Addons 1.4.10 that the database structure for cdr_mysql is: CREATE TABLE cdr ( calldate datetime NOT NULL default '-00-00 00:00:00', clid varchar(80) NOT NULL default '', src varchar(80) NOT NULL default '', dst varchar(80) NOT NULL default '', dcontext varchar(80) NOT NULL default '', channel varchar(80) NOT NULL default '', dstchannel varchar(80) NOT NULL default '', lastapp varchar(80) NOT NULL default '', lastdata varchar(80) NOT NULL default '', duration int(11) NOT NULL default '0', billsec int(11) NOT NULL default '0', disposition varchar(45) NOT NULL default '', amaflags int(11) NOT NULL default '0', accountcode varchar(20) NOT NULL default '', uniqueid varchar(32) NOT NULL default '', userfield varchar(255) NOT NULL default '' ); Just curious if anyone has successfully patched cdr_addon_mysql to use accept the latest cdr fields from 1.4 ... namely: 'start', 'answer', 'end'? Seems logical that the cdr_mysql addon should be updated to reflect the current cdr. And for backwards compatibility it can still accept 'calldate'. The MySQL driver contains all of the same information, albeit in a slightly different form. Calldate is the same as start, calldate plus duration minus billsec is the same as answer, and calldate plus duration is the same as end. Generally, we do not make design changes in the middle of a release cycle, especially given that such changes would break a great many existing systems. Given that there's no security reason why we would need to make such a change, it is out of the question. While you're certainly welcome to make such a change on your own systems, such a change will not be committed in the 1.4 addons. In the 1.6 series and forward, we've changed the mysql driver to scan the table metadata and adapt the queries to the table structure. Therefore, you could, in fact, use 'start', 'answer', and 'end' in the 1.6 series, as you suggested, above, and it would work perfectly well. On the other hand, if you kept the legacy structure, that would work, too. Thanks for the reply. So my next question is could I take the cdr_mysql from 1.6's addons and use it in 1.4? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR_MYSQL 1.4 Database Structure
Wednesday, December 30, 2009, 6:48:37 PM, Robert wrote: Tilghman Lesher wrote: On Wednesday 30 December 2009 10:52:48 Robert Broyles wrote: Just curious if anyone has successfully patched cdr_addon_mysql to use accept the latest cdr fields from 1.4 ... namely: 'start', 'answer', 'end'? Seems logical that the cdr_mysql addon should be updated to reflect the current cdr. And for backwards compatibility it can still accept 'calldate'. The MySQL driver contains all of the same information, albeit in a slightly different form. Calldate is the same as start, calldate plus duration minus billsec is the same as answer, and calldate plus duration is the same as end. Generally, we do not make design changes in the middle of a release cycle, especially given that such changes would break a great many existing systems. Given that there's no security reason why we would need to make such a change, it is out of the question. While you're certainly welcome to make such a change on your own systems, such a change will not be committed in the 1.4 addons. In the 1.6 series and forward, we've changed the mysql driver to scan the table metadata and adapt the queries to the table structure. Therefore, you could, in fact, use 'start', 'answer', and 'end' in the 1.6 series, as you suggested, above, and it would work perfectly well. On the other hand, if you kept the legacy structure, that would work, too. Thanks for the reply. So my next question is could I take the cdr_mysql from 1.6's addons and use it in 1.4? I don't think so. But you can define more columns, and an insert trigger which calculates the missing fields as defined in Tilghman's reply. -- Best regards, Gergomailto:csi...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR_MYSQL 1.4 Database Structure
On Wednesday 30 December 2009 11:48:37 Robert Broyles wrote: So my next question is could I take the cdr_mysql from 1.6's addons and use it in 1.4? No. The APIs are significantly different enough that a backport would require a good amount of modification. However, there is a backport of cdr_adaptive_odbc to 1.4: http://svnview.digium.com/community/tilghman/branches/1.4/ -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New 1.4 system: registered, but not responding to invite?
At 23:33 12/21/2009, Doug wrote: At 00:46 12/21/2009, Alex Balashov wrote: A packet capture would be needed to illuminate the source of the problem. Thanks, Alex for your suggestion. I just don't see where the extension responds to the INVITE. What would prevent that? Problem solved: Each peer in sip.conf needs: qualify=yes By the way, I have a bunch of phones behind this same router that work just fine on our old v1.2 system. On 12/21/2009 01:39 AM, Doug wrote: I've turned on NAT everywhere I can think, but even though I hear ringing on the calling phone (different system) the called phone does not ring. Has anyone bumped into this lately? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New 1.4 system: registered, but not responding to invite?
Doug, It doesn't respond to the INVITE - the trace says No response to the INVITE?. If the phone doesn't even ring it's probably not getting anything, which points to a problem with the router it's behind. How is the router set up to deliver SIP and RTP to the phone? On Tue, Dec 22, 2009 at 5:33 AM, Doug d...@natel.net wrote: At 00:46 12/21/2009, Alex Balashov wrote: A packet capture would be needed to illuminate the source of the problem. Thanks, Alex for your suggestion. Here is a link for the packet capture: http://www.A7H.com/~stuph/TCPdump-2009-Dec-21-2304.txthttp://www.A7H.com/%7Estuph/TCPdump-2009-Dec-21-2304.txt I just don't see where the extension responds to the INVITE. What would prevent that? By the way, I have a bunch of phones behind this same router that work just fine on our old v1.2 system. On 12/21/2009 01:39 AM, Doug wrote: I've turned on NAT everywhere I can think, but even though I hear ringing on the calling phone (different system) the called phone does not ring. Has anyone bumped into this lately? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New 1.4 system: registered, but not responding to invite?
At 00:46 12/21/2009, Alex Balashov wrote: A packet capture would be needed to illuminate the source of the problem. Thanks, Alex for your suggestion. Here is a link for the packet capture: http://www.A7H.com/~stuph/TCPdump-2009-Dec-21-2304.txt I just don't see where the extension responds to the INVITE. What would prevent that? By the way, I have a bunch of phones behind this same router that work just fine on our old v1.2 system. On 12/21/2009 01:39 AM, Doug wrote: I've turned on NAT everywhere I can think, but even though I hear ringing on the calling phone (different system) the called phone does not ring. Has anyone bumped into this lately? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New 1.4 system: registered, but not responding to invite?
I've turned on NAT everywhere I can think, but even though I hear ringing on the calling phone (different system) the called phone does not ring. Has anyone bumped into this lately? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New 1.4 system: registered, but not responding to invite?
A packet capture would be needed to illuminate the source of the problem. On 12/21/2009 01:39 AM, Doug wrote: I've turned on NAT everywhere I can think, but even though I hear ringing on the calling phone (different system) the called phone does not ring. Has anyone bumped into this lately? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 and kernel panic and IRQ interrupts
Hi, I'm having trouble with one machine that kernel panics with Asterisk 1.4. The motherboard is an Asus P5W Deluxe. I reported the kernel panic here: http://lists.digium.com/pipermail/asterisk-users/2009-November/241006.html I'm now trying to understand if the problem can be an IRQ issue or not. I disabled APIC in the BIOS because I thought that maybe it could be buggy (not sure though). My interrupts are now as follows: # more /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 93 0 0 0XT-PIC-XTtimer 1: 1531 0 0 0XT-PIC-XTi8042 2: 0 0 0 0XT-PIC-XTcascade 3: 0 0 0 0XT-PIC-XTuhci_hcd:us b3 5:4012524 0 0 0XT-PIC-XTehci_hcd:us b1, uhci_hcd:usb2 6: 3 0 0 0XT-PIC-XTfloppy 7:4341422 0 0 0XT-PIC-XTahci, HFC-multi 8: 2 0 0 0XT-PIC-XTrtc 9: 1 0 0 0XT-PIC-XTacpi 10: 10306916 0 0 0XT-PIC-XTeth1, eth2 11: 30845499 0 0 0XT-PIC-XTeth0, wcte12xp0 12: 3137 0 0 0XT-PIC-XTi8042 14:213 0 0 0XT-PIC-XTide0 NMI: 0 0 0 0 LOC:3049870304985930498553049853 ERR: 0 MIS: 0 This doesn't look good for 3 reasons (I think): 1. only one core out of a quad-core CPU handles the interrupts 2. the telephony cards share IRQs with other devices (HFC-multi and wcte12xp0) 3. wcte12xp0 and eth0 are sharing the same IRQ and eth0 is particularly active on this system Note that on another system (Asus P5B motherboard with APIC enabled) I have a very stable Asterisk 1.2 and the IRQs are as follows: # cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0:104 0 0 0 IO-APIC-edge timer 1: 1558 0 0 0 IO-APIC-edge i8042 6: 3 0 0 0 IO-APIC-edge floppy 8: 2 0 0 0 IO-APIC-edge rtc 9: 1 0 0 0 IO-APIC-fasteoi acpi 16: 50387 0 0 0 IO-APIC-fasteoi ahci 17:4710977 11071200 164252433430308 IO-APIC-fasteoi ide0, eth0 18: 64081335 65109221 31317172 33363907 IO-APIC-fasteoi ahci, eth1 20: 114824294 87784625 79980388 99000512 IO-APIC-fasteoi wcte12xp0 21: 645793 0 0 0 IO-APIC-fasteoi eth2 22:94996127398138 108897606725944 IO-APIC-fasteoi HFC-multi NMI: 0 0 0 0 LOC: 37865864 37865853 37857176 37857173 ERR: 0 MIS: 0 Each telephony card is on its own IRQ. Can IRQ sharing actually cause a kernel panic? or does it usually only cause voice distortion, ticks, etc.? What do you suggest I should try? Should I enable APIC again and try to get each card on a different IRQ? Is anyone using an Asus P5W Deluxe? If so, could you please share your /proc/interrupts and BIOS settings? Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 DISA is jumping after one digit in the DISA context
Am Friday 06 November 2009 00:17:36 schrieb Marc Lindner: Dear list, I have problems with DISA on an specific server with Asterisk 1.4.26.2. After starting DISA I can only press one key and DISA is jumping direct into the context without waiting for further digits. The reason and solution is: exten = _X!,n,DISA(no-password|calls_disa) [calls_disa] exten = _X.,1,NoOp() exten = _X.,n,HangUp() if context [calls_disa] like this exten = _X!,1,NoOp() exten = _X!,n,HangUp() then DISA function is broken, after entering one digit, dialplan jump to calls_disa. I did not expected this... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 DISA is jumoing after one digit in the DISA context
Dear list, I have problems with DISA on an specific server with Asterisk 1.4.26.2. After starting DISA I can only press one key and DISA is jumping direct into the context without waiting for further digits. In dtmf.log I found this: [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin '7' received on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin passthrough '7' on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF end '7' received on SIP/214-00d92db0, duration 60 ms [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF end accepted with begin '7' on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF end '7' has duration 60 but want minimum 80, emulating on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF end emulation of '7' queued on SIP/214-00d92db0 If Iam using the dialplan on another server there is no problem. If Iam using READ I do not have problems to enter digits by DTMF so I assume its related to DISA. best regards Marc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 remote pickup
On 6/11/09 3:37 AM, Antony Stone wrote: On Thursday 05 November 2009 14:28, Danny Nicholas wrote: Hi. I have several Asterisk 1.4.21 machines, each with ISDN cards in them, and Polycom SIP phones on people's desks. I'm trying to work out how to provide a remote pickup facility along the following lines: The normal (as defined in features.conf) way to pick the call would be *82233. Features.conf defines *8 as a global pickup to be followed by an extension. Thanks, I'll investigate this and see if that works instead. What we do is create an Asterisk database entry: Pickup/NUMBER/GROUP Where NUMBER is the extension, and Group is the Pickup Group. We then set pickup mark variable in the macro that dials the extension. Then if someone dials *79 (or whatever) it picks up the group that the person dialling *79 is in. I.E. * Call goes to Jon (who is in group 3) * He is away from his desk * Jane dials *79 (also in group 3) and picks up the call If Fred (in group 5) were to dial *79 he would not pick up the call. Names have been changed to protect the innocent :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 and Fax
Hi, Does anyone have an up to date guide for setting up fax 2 email with asterisk? Thanks Dan IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from £150 per month, please contact Kesher Communications. Dan Journo IT Business Consultant Kesher Communications Ltd Tel: 07957 233 599 Web: http://www.KesherCommunications.com http://www.keshercommunications.com/ Live Chat/Instant Support: Click Here http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7F508C6B61A91700343lang=ensurpre=PreSurvey This email and any files transmitted with it are confidential and intended solely for the recipient(s). If you are not the named addressee you should not disseminate, copy or alter this email. Under no circumstances may this email be distributed without written permission from the sender. 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. All prices exclude VAT unless otherwise stated. No responsibility is taken for any recommendations made by a sender or by Kesher Communications Ltd. Recipient(s) takes responsibility for any actions taken as a result of advice and recommendations given by Kesher Communications Ltd. image001.jpgimage002.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
On 2 Nov 2009, at 17:22, Dan Journo wrote: Does anyone have an up to date guide for setting up fax 2 email with asterisk? So you can fax them obnoxiously long signatures? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
We want to disconnect our PSTN fax line and transfer the number over to our asterisk server. I need to get it up and running before we can put in the order to transfer the fixed line number over to SIP. Thanks Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 November 2009 17:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 and Fax On 2 Nov 2009, at 17:22, Dan Journo wrote: Does anyone have an up to date guide for setting up fax 2 email with asterisk? So you can fax them obnoxiously long signatures? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
Sorry Steve, Forgot to remove it before sending the email. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 November 2009 17:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 and Fax On 2 Nov 2009, at 17:22, Dan Journo wrote: Does anyone have an up to date guide for setting up fax 2 email with asterisk? So you can fax them obnoxiously long signatures? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
Dan Journo wrote: I need to get it up and running before we can put in the order to transfer the fixed line number over to SIP. Faxing over SIP is never a good idea. 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 http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
I've heard mixed reports. Some say they've had no problems, some say that faxes fail most of the time. I want to try it out and see how it goes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: 02 November 2009 18:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 and Fax Dan Journo wrote: I need to get it up and running before we can put in the order to transfer the fixed line number over to SIP. Faxing over SIP is never a good idea. 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 http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. Lee. Dan Journo wrote: I've heard mixed reports. Some say they've had no problems, some say that faxes fail most of the time. I want to try it out and see how it goes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: 02 November 2009 18:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 and Fax Dan Journo wrote: I need to get it up and running before we can put in the order to transfer the fixed line number over to SIP. Faxing over SIP is never a good idea. Doug ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
2009/11/2 Doug Lytle supp...@drdos.info Dan Journo wrote: I need to get it up and running before we can put in the order to transfer the fixed line number over to SIP. Faxing over SIP is never a good idea. And why would that be? I think that faxing over SIP using T.38 is a fantastic idea. Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
Lee Howard wrote: I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can also describe T.38, which is not as much of a gamble as FAX over VOIP :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
Kevin P. Fleming wrote: Lee Howard wrote: I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can also describe T.38, which is not as much of a gamble as FAX over VOIP :-) Does Asterisk 1.4 support T.38? Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
Lee Howard wrote: FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can also describe T.38, which is not as much of a gamble as FAX over VOIP :-) Does Asterisk 1.4 support T.38? Only for passthrough between SIP channels; Asterisk 1.6.0 and later also support T.38 termination and origination. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
Christian Victor wrote: 2009/11/2 Doug Lytle supp...@drdos.info mailto:supp...@drdos.info Faxing over SIP is never a good idea. And why would that be? I think that faxing over SIP using T.38 is a fantastic idea. As far as I know, T.38 isn't supported under 1.4 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 http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
Kevin P. Fleming wrote: Lee Howard wrote: I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can also describe T.38, which is not as much of a gamble as FAX over VOIP :-) True, although I've yet to find a provider in this country (UK) that supports T.38. He may be better off porting the number to a fax2email service (although ime they are worth play testing first before you put any real work on them, eg. recently I've found one that doesn't support Fine Print or higher res faxes). AFAICT, to get a (real) fax machine using T.38, you either need to buy one that already supports it (never seen one, but I am assured they exist), Buy an ATA that supports it, or move to callweaver. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
How do these fax2email providers run their service? Do they all use physical lines rather than use the internet? Thanks Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Kenyon Sent: 02 November 2009 20:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 and Fax Kevin P. Fleming wrote: Lee Howard wrote: I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can also describe T.38, which is not as much of a gamble as FAX over VOIP :-) True, although I've yet to find a provider in this country (UK) that supports T.38. He may be better off porting the number to a fax2email service (although ime they are worth play testing first before you put any real work on them, eg. recently I've found one that doesn't support Fine Print or higher res faxes). AFAICT, to get a (real) fax machine using T.38, you either need to buy one that already supports it (never seen one, but I am assured they exist), Buy an ATA that supports it, or move to callweaver. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
On Mon, Nov 2, 2009 at 12:22 PM, Dan Journo d...@keshercommunications.comwrote: Hi, Does anyone have an up to date guide for setting up fax 2 email with asterisk? You can buy this shrink-wrapped from Cisco if you're willing to pay what they're asking. There are probably other vendors who sell that too. If you insist on doing this yourself, and using asterisk, start by moving to 1.6. The fax support is night and day better in 1.6 than 1.4, using native asterisk app_fax (which depends on SpanDSP from Lee Howard). If you want to go SIP as part of the deployment, I recommend either: 1) terminate PSTN at your premise, and only use SIP internally inside your PSTN gateway 2) if you're going to go with a SIP provider, tunnel them on a dedicated circuit so you're not fighting bandwidth limit in addition to the various problems you'll inevitably face with their implementation of fax over voip. Once you price #2 you'll probably discover that #1 is cheaper, and I've already said it's more likely to be reliable when you can control as much of the voip as possible. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
David Backeberg wrote: On Mon, Nov 2, 2009 at 12:22 PM, Dan Journo d...@keshercommunications.com mailto:d...@keshercommunications.com wrote:asterisk app_fax (which depends on SpanDSP from Lee Howard). SpanDSP was written by Steve Underwood. 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 http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
On Mon, Nov 2, 2009 at 3:38 PM, Dan Journo d...@keshercommunications.com wrote: How do these fax2email providers run their service? Do they all use physical lines rather than use the internet? If you read far enough back in the archives, you'll find somebody who claimed they used asterisk-1.4 (I think hylafax) and voip But that they did so in a colo, one-hop and almost no RTT away from their provider. Again, at which point, you're not saving money compared to an analogue fax over PSTN unless you have a really large volume, and even then you can often get better bulk pricing for PSTN. You know your usage and you know your budget. If you don't have time to fight broken faxes, learn asterisk-1.6, and provision a voip provider, just stick with analogue fax over PSTN. My business situation: channelized DS3, that's 28x 23 voice channels - Cisco voice routers - SIP - asterisk-1.6 app_fax() Working very well for us, but I don't know whether your budget or usage is going to justify something like that. As for what a commercial service uses, they use whatever was the cheapest wherever they host their services. Real modem pools, or real brooktrout modem boards are common. That would have been a better idea for my situation if I wasn't sharing the circuits with other voice services. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
Dan Journo wrote: How do these fax2email providers run their service? I've not the faintest Idea, the provider I use afaict outsource it. Do they all use physical lines rather than use the internet? Thanks Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
2009/11/2 Doug Lytle supp...@drdos.info Christian Victor wrote: 2009/11/2 Doug Lytle supp...@drdos.info mailto:supp...@drdos.info Faxing over SIP is never a good idea. And why would that be? I think that faxing over SIP using T.38 is a fantastic idea. As far as I know, T.38 isn't supported under 1.4 That would be Faxing using Asterisk 1.4 is never a good idea. Sorry for being such a bean counter. ;-) To stay on-topic: Terminating fax over PSTN works quite well in 1.4 but the original poster should be warned of trying to terminate fax over a SIP trunk. Using SIP/G.711 to connect the fax machine to Asterisk over LAN works quite well in my experience but others had worse results. Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
On 11/03/2009 04:25 AM, Thomas Kenyon wrote: Kevin P. Fleming wrote: Lee Howard wrote: I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can also describe T.38, which is not as much of a gamble as FAX over VOIP :-) True, although I've yet to find a provider in this country (UK) that supports T.38. He may be better off porting the number to a fax2email service (although ime they are worth play testing first before you put any real work on them, eg. recently I've found one that doesn't support Fine Print or higher res faxes). AFAICT, to get a (real) fax machine using T.38, you either need to buy one that already supports it (never seen one, but I am assured they exist), Buy an ATA that supports it, or move to callweaver. T.38 FAX machines do exist, although they are rare. A number of high end office machines support T.38, or have a T.38 option. There are small FAX machines from Sagem which support T.38. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 vs 1.6
Hi, I was wondering whether there are any problems with v1.6 which means I should avoid it. What are the advantages of upgrading? And finally, why both versions are available? Why not just scrap 1.4? Many thanks Dan Journo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 vs 1.6
On Wed, Oct 14, 2009 at 9:01 AM, Dan Journo d...@keshercommunications.com wrote: I was wondering whether there are any problems with v1.6 which means I should avoid it. Try searching the list for the many times this has been answered. Since this is your choice, you need to set up a parallel instance of your environment and vet your particular usage. At the very least you will need to update your dialplan to the new syntax, and upgrade to DAHDI if you're using hardware phone cards. What are the advantages of upgrading? Features, both to the individual applications and new applications not previously available. Scalability, especially for large dialplans, and a much better SIP stack. More eyes on the code as it's the current track. Try searching the list for the many times this has been answered. And finally, why both versions are available? Why not just scrap 1.4? 1.2 is still available too. Because the choice is yours. Many people are still using 1.2, much less 1.4. Some people call old code 'stable' code, as in the bugs are known or worked around. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 and GUI Configuration Help
Hello, I am trying to setup an asterisk box for a small office that has 4 phone lines and a fax. The fax will not be going through the box. I have Digium TDM410P to take 4 analog lines and I will be using grandstream gxp2000 for our setup. I have read the docs just do not understand the dialplan, incoming calls, routing process. I setup the trunks which is the 4 phone lines so the first two numbers go to a IVR and the other two will be direct lines to gxp 2000. How can I configure a dialplan for this. The gui seems to be messing everything up it seems to not want to update or allow you to make changes. It says it made the change but then when you click apply it does do anything. Can someone share what they've done. I know this works as I used Asterisk 1.2. I just want something real simple incoming calls to ivr, with exception to two lines directly to phone. BTW I don't want to done for me just some example code an experiences if possible. Thanks in advance! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4 segfaults when trying to use mixmonitor
Hi. Using asterisk 1.4 svn 21112, when I try to use the mixmonitor feature I get the following in the log file. [Aug 11 09:22:54] WARNING[32057] file.c: Tried to write non-voice frame [Aug 11 09:22:54] WARNING[32057] channel.c: Failed to write data to channel monitor write stream After several sequences like that one it segfaults, I cannot find the core file, so I am doing something wrong, but I wonder if there is any fix for this in a later build or whatever? Thanks much. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4.X, T.38 and NAT
Hi, I have been trying to get T.38 to work with clients behind NAT for the past week but with no success. I have an asterisk server on the public internet and several Grandstream (I tried Linksys too) HT502 ATAs behind NAT in different locations. I tried every possible combination of NAT, canreinvite, t38pt_usertpsource entries, I even tried asterisk 1.4.19, 1.4.24.1, 1.4.25 all with the same result; Failure. I can see the t38 negotiations, and I think the problem is in the reinvite message after T.38 detection. Only one case is working: Both ATAs are on the same NATed network, Asterisk server is on the public internet in another locations canreinvite = yes for both ATAs NAT = yes t38pt_usertpsource = no The reason I think this case is working is that the reInvite is sent with the private IPs of the ATAs, and since they are on the same network, they can find each other and continue the call successfully. This case leads me to think that the problem is NAT related and has something to do the the reInvite after fax detection, something is being sent incorrectly on the reinvite, either the IPs or the ports. does anyone have an idea about how to solve this problem. your help is much appreciated. Antoine Megalla. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4.X, T.38 and NAT
On Fri, 29 May 2009 01:52:08 Antoine Megalla wrote: Hi, I have been trying to get T.38 to work with clients behind NAT for the past week but with no success. I have an asterisk server on the public internet and several Grandstream (I tried Linksys too) HT502 ATAs behind NAT in different locations. I tried every possible combination of NAT, canreinvite, t38pt_usertpsource entries, I even tried asterisk 1.4.19, 1.4.24.1, 1.4.25 all with the same result; Failure. I can see the t38 negotiations, and I think the problem is in the reinvite message after T.38 detection. T38 is If I see another post about problems with T38 I might want to scream... lol. Yesterday I had a standard POTs line installed and I transfered my fax number back to a PSTN provider (from a T38 provider). I had a uncontended, stable, very low latency fibre link to the upline ISP, direct unNAT'd IP connection, then a short hop skip and jump to the T38 provider, and I still could not get it to work reliably. Much less for those on commodity grade cable/xDSL connections with NAT. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
