[asterisk-users] Pattern matching repeating digits
'lo, all,
Is there some (possibly undocumented?) way that I can pattern-match on a
specified number of repeating digits? (Something similar to regular
expressions' {})
Here's an example: let's say I have a string of things that need to be done for
both extensions 233 and 255. I can either...
A) Repeat the exact same code for both extensions, like so:
exten = 233,1,DoStuff()
exten = 233,n,AndMoreStuff()
exten = 233,n,Dial(something)
exten = 255,1,DoStuff()
exten = 255,n,AndMoreStuff()
exten = 255,n,Dial(something)
...which is stupid, or...
B) I can attempt code reuse for similar cases (a Good Thing[tm]), and make as
specific of a match as possible, like so:
exten = _2[35][35],1,DoStuff()
exten = _2[35][35],n,AndMoreStuff()
exten = _2[35][35],n,Dial(something)
...but this will not only match 233 and 255, but 235 and 253 as well.
It'd be nice if there was a substitute character that meant a character that
is exactly the same as the preceding one; for example, if R was meant to
represent such a concept, then this would do what I want:
exten = _2[35]R,1,DoStuff()
exten = _2[35]R,n,AndMoreStuff()
exten = _2[35]R,n,Dial(something)
You could even do crazy things like chain them together (this would match 2
and 2 and nothing else);
exten = _2[35]RRR,1,DoStuff()
exten = _2[35]RRR,n,AndMoreStuff()
exten = _2[35]RRR,n,Dial(something)
Am I missing something or does this really not exist?
Thanks,
--
Nathan Anderson
First Step Internet, LLC
nath...@fsr.com
--
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Re: [asterisk-users] WARNING[28151] from CLI
What hardware do you use? Do your have some E1 or T1 Ports? Maybe one or more of this ports is down. Am 26.03.2013 17:57, schrieb Salaheddine Elharit: Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! -- _ -- 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] WARNING[28151] from CLI
Am 26.03.2013 17:57, schrieb Salaheddine Elharit: Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! this can have different causes... mostly a wrong setting in your zaptel configuration file... this could be e.g. mixing american / european settings (e1/t1), wrong timing settings, wrong master / source clock setting, [...] post more details... what span (e1 or t1), which hardware, driver version, asterisk version, config files... regards, yves -- _ -- 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] Fundemental changes to CDR within single asterisk family
On Tue, 2013-03-26 at 07:26 -0500, Matthew Jordan wrote: On 03/26/2013 05:22 AM, Ishfaq Malik wrote: Hi In asterisk 1.8.7.0, an inbound call that was transferred to another peer would have 2 cdr entries. In asterisk 1.8.18.0 this same activity has a single cdr entry. This is a rather large and fundamental change to be enacting halfway through a single family branch, was there any reason why this happened? It means we can't upgrade without doing significant extra development and testing. This was most likely an unintended consequence of some other change (most likely dealing with masquerades). Is 1.8.18.0 the exact version when the behaviour changed? Just so I'm clear on the scenario, what are the channel technologies involved? Is the transfer initiated via a protocol message or via a DTMF feature? Thanks, Matt Hi Matt I couldn't say for sure which version between 1.8.7.0 and 1.8.18.0 the change happened in. The call comes in via SIP and the transfer is done using the dedicated transfer button on a hard phone (so not from the features.conf method) Regards Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] WARNING[28151] from CLI
Hi i use 2 digium cards 1 card with 2 ports and the second card with 4 ports but actually i use just the span 1 and span 6 Asterisk 1.4-r110474M i use E1 ports zaptel.conf # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 # span=3,3,0,ccs,hdb3 # termtype: te # bchan=63-77,79-93 # dchan=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # span=4,4,0,ccs,hdb3 # termtype: te # bchan=94-108,110-124 # dchan=109 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1 span=5,5,0,ccs,hdb3 # termtype: te bchan=125-139,141-155 dchan=140 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2 span=6,6,0,ccs,hdb3 # termtype: te bchan=156-170,172-186 dchan=171 # Global data loadzone = us defaultzone = us etc/asterisk/zapata.conf [channels] context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=mycallerid immediate=no channel = 156-170 channel = 172-176 channel = 125-139 channel = 141-155 thanks and regards 2013/3/27 Yves A. yves...@gmx.de Am 26.03.2013 17:57, schrieb Salaheddine Elharit: Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! this can have different causes... mostly a wrong setting in your zaptel configuration file... this could be e.g. mixing american / european settings (e1/t1), wrong timing settings, wrong master / source clock setting, [...] post more details... what span (e1 or t1), which hardware, driver version, asterisk version, config files... regards, yves -- _ -- 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] WARNING[28151] from CLI
You do use only span 1 and 6? So the other ports are not plugged? That is the cause for the warnings. I use a Sangoma E1-Card. The configure script gives me the option unused for any port. Maybe your configure script offers you the same option. Am 27.03.2013 11:54, schrieb Salaheddine Elharit: Hi i use 2 digium cards 1 card with 2 ports and the second card with 4 ports but actually i use just the span 1 and span 6 Asterisk 1.4-r110474M i use E1 ports zaptel.conf # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 # span=3,3,0,ccs,hdb3 # termtype: te # bchan=63-77,79-93 # dchan=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # span=4,4,0,ccs,hdb3 # termtype: te # bchan=94-108,110-124 # dchan=109 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1 span=5,5,0,ccs,hdb3 # termtype: te bchan=125-139,141-155 dchan=140 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2 span=6,6,0,ccs,hdb3 # termtype: te bchan=156-170,172-186 dchan=171 # Global data loadzone= us defaultzone= us etc/asterisk/zapata.conf [channels] context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=mycallerid immediate=no channel = 156-170 channel = 172-176 channel = 125-139 channel = 141-155 thanks and regards 2013/3/27 Yves A. yves...@gmx.de mailto:yves...@gmx.de Am 26.03.2013 17:57, schrieb Salaheddine Elharit: Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! this can have different causes... mostly a wrong setting in your zaptel configuration file... this could be e.g. mixing american / european settings (e1/t1), wrong timing settings, wrong master / source clock setting, [...] post more details... what span (e1 or t1), which hardware, driver version, asterisk version, config files... 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] WARNING[28151] from CLI
thank you for your help ,but which configure script and when i can find this script ? in etc/asterisk best regards 2013/3/27 Thorsten Göllner t...@ovm-group.com You do use only span 1 and 6? So the other ports are not plugged? That is the cause for the warnings. I use a Sangoma E1-Card. The configure script gives me the option unused for any port. Maybe your configure script offers you the same option. Am 27.03.2013 11:54, schrieb Salaheddine Elharit: Hi i use 2 digium cards 1 card with 2 ports and the second card with 4 ports but actually i use just the span 1 and span 6 Asterisk 1.4-r110474M i use E1 ports zaptel.conf # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 # span=3,3,0,ccs,hdb3 # termtype: te # bchan=63-77,79-93 # dchan=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # span=4,4,0,ccs,hdb3 # termtype: te # bchan=94-108,110-124 # dchan=109 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1 span=5,5,0,ccs,hdb3 # termtype: te bchan=125-139,141-155 dchan=140 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2 span=6,6,0,ccs,hdb3 # termtype: te bchan=156-170,172-186 dchan=171 # Global data loadzone = us defaultzone = us etc/asterisk/zapata.conf [channels] context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=mycallerid immediate=no channel = 156-170 channel = 172-176 channel = 125-139 channel = 141-155 thanks and regards 2013/3/27 Yves A. yves...@gmx.de Am 26.03.2013 17:57, schrieb Salaheddine Elharit: Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! this can have different causes... mostly a wrong setting in your zaptel configuration file... this could be e.g. mixing american / european settings (e1/t1), wrong timing settings, wrong master / source clock setting, [...] post more details... what span (e1 or t1), which hardware, driver version, asterisk version, config files... 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] WARNING[28151] from CLI
you have already listed the two config files for using zaptel. on first sight, they look ok to me (did not use zaptel for years now) maybe you should definitely comment out any span that is not in use... or do the opposite. i´ve seen this warning several times, but i cant remember it had anything to do with spans being configured but not used. it always had something to do with timing or even defective cards or cabling or even wrong settings on providers´ site. what changes were made to the system so that these warnings occur? or have they been visible from the very start? do they affect telefony (e.g. loss of calls, one side audio only etc.)? how much load (concurrent calls) is on the asterisk, does the warning occur periodically or only a few times? these are all questions you should ask yourself to help you find the answer yourself... it can be very frustrating sometimes, but for me, thats all i can tell about. regards, yves Am 27.03.2013 13:06, schrieb Salaheddine Elharit: thank you for your help ,but which configure script and when i can find this script ? in etc/asterisk best regards 2013/3/27 Thorsten Göllner t...@ovm-group.com mailto:t...@ovm-group.com You do use only span 1 and 6? So the other ports are not plugged? That is the cause for the warnings. I use a Sangoma E1-Card. The configure script gives me the option unused for any port. Maybe your configure script offers you the same option. Am 27.03.2013 11:54, schrieb Salaheddine Elharit: Hi i use 2 digium cards 1 card with 2 ports and the second card with 4 ports but actually i use just the span 1 and span 6 Asterisk 1.4-r110474M i use E1 ports zaptel.conf # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 # span=3,3,0,ccs,hdb3 # termtype: te # bchan=63-77,79-93 # dchan=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # span=4,4,0,ccs,hdb3 # termtype: te # bchan=94-108,110-124 # dchan=109 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1 span=5,5,0,ccs,hdb3 # termtype: te bchan=125-139,141-155 dchan=140 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2 span=6,6,0,ccs,hdb3 # termtype: te bchan=156-170,172-186 dchan=171 # Global data loadzone= us defaultzone= us etc/asterisk/zapata.conf [channels] context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=mycallerid immediate=no channel = 156-170 channel = 172-176 channel = 125-139 channel = 141-155 thanks and regards 2013/3/27 Yves A. yves...@gmx.de mailto:yves...@gmx.de Am 26.03.2013 17:57, schrieb Salaheddine Elharit: Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! this can have different causes... mostly a wrong setting in your zaptel configuration file... this could be e.g. mixing american / european settings (e1/t1), wrong timing settings, wrong master / source clock setting, [...] post more details... what span (e1 or t1), which hardware, driver version, asterisk version, config files... 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 -- _ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] WARNING[28151] from CLI
ok thanks for support and help 2013/3/27 Yves A. yves...@gmx.de you have already listed the two config files for using zaptel. on first sight, they look ok to me (did not use zaptel for years now) maybe you should definitely comment out any span that is not in use... or do the opposite. i´ve seen this warning several times, but i cant remember it had anything to do with spans being configured but not used. it always had something to do with timing or even defective cards or cabling or even wrong settings on providers´ site. what changes were made to the system so that these warnings occur? or have they been visible from the very start? do they affect telefony (e.g. loss of calls, one side audio only etc.)? how much load (concurrent calls) is on the asterisk, does the warning occur periodically or only a few times? these are all questions you should ask yourself to help you find the answer yourself... it can be very frustrating sometimes, but for me, thats all i can tell about. regards, yves Am 27.03.2013 13:06, schrieb Salaheddine Elharit: thank you for your help ,but which configure script and when i can find this script ? in etc/asterisk best regards 2013/3/27 Thorsten Göllner t...@ovm-group.com You do use only span 1 and 6? So the other ports are not plugged? That is the cause for the warnings. I use a Sangoma E1-Card. The configure script gives me the option unused for any port. Maybe your configure script offers you the same option. Am 27.03.2013 11:54, schrieb Salaheddine Elharit: Hi i use 2 digium cards 1 card with 2 ports and the second card with 4 ports but actually i use just the span 1 and span 6 Asterisk 1.4-r110474M i use E1 ports zaptel.conf # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 # span=3,3,0,ccs,hdb3 # termtype: te # bchan=63-77,79-93 # dchan=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # span=4,4,0,ccs,hdb3 # termtype: te # bchan=94-108,110-124 # dchan=109 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1 span=5,5,0,ccs,hdb3 # termtype: te bchan=125-139,141-155 dchan=140 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2 span=6,6,0,ccs,hdb3 # termtype: te bchan=156-170,172-186 dchan=171 # Global data loadzone = us defaultzone = us etc/asterisk/zapata.conf [channels] context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=mycallerid immediate=no channel = 156-170 channel = 172-176 channel = 125-139 channel = 141-155 thanks and regards 2013/3/27 Yves A. yves...@gmx.de Am 26.03.2013 17:57, schrieb Salaheddine Elharit: Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! this can have different causes... mostly a wrong setting in your zaptel configuration file... this could be e.g. mixing american / european settings (e1/t1), wrong timing settings, wrong master / source clock setting, [...] post more details... what span (e1 or t1), which hardware, driver version, asterisk version, config files... 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 -- _ -- 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 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones, 11.2.2 Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones, and 11.2.2. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these versions resolve the following issues: * A possible buffer overflow during H.264 format negotiation. The format attribute resource for H.264 video performs an unsafe read against a media attribute when parsing the SDP. This vulnerability only affected Asterisk 11. * A denial of service exists in Asterisk's HTTP server. AST-2012-014, fixed in January of this year, contained a fix for Asterisk's HTTP server for a remotely-triggered crash. While the fix prevented the crash from being triggered, a denial of service vector still exists with that solution if an attacker sends one or more HTTP POST requests with very large Content-Length values. This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11 * A potential username disclosure exists in the SIP channel driver. When authenticating a SIP request with alwaysauthreject enabled, allowguest disabled, and autocreatepeer disabled, Asterisk discloses whether a user exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11 These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-001, AST-2013-002, and AST-2013-003, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.20.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.2-digiumphones http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.2.2 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-001.pdf * http://downloads.asterisk.org/pub/security/AST-2013-002.pdf * http://downloads.asterisk.org/pub/security/AST-2013-003.pdf Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2013-001: Buffer Overflow Exploit Through SIP SDP Header
Asterisk Project Security Advisory - AST-2013-001 Product Asterisk Summary Buffer Overflow Exploit Through SIP SDP Header Nature of Advisory Exploitable Stack Buffer Overflow Susceptibility Remote Unauthenticated Sessions SeverityMajor Exploits Known No Reported On 6 January, 2013 Reported By Ulf Ha:rnhammar Posted On27 March, 2013 Last Updated On March 27, 2013 Advisory ContactJonathan Rose jrose AT digium DOT com CVE NameCVE-2013-2685 Description The format attribute resource for h264 video performs an unsafe read against a media attribute when parsing the SDP. The vulnerable parameter can be received as strings of an arbitrary length and Asterisk attempts to read them into limited buffer spaces without applying a limit to the number of characters read. If a message is formed improperly, this could lead to an attacker being able to execute arbitrary code remotely. Resolution Attempts to read string data into the buffers noted are now explicitly limited by the size of the buffers. Affected Versions Product Release Series Asterisk Open Source 11.x All Versions Corrected In Product Release Asterisk Open Source11.2.2 Patches SVN URL Revision Http://downloads.asterisk.org/pub/security/AST-2013-001-11.diff Asterisk 11 Links https://issues.asterisk.org/jira/browse/ASTERISK-20901 Asterisk Project Security Advisories are posted at http://www.asterisk.org/security This document may be superseded by later versions; if so, the latest version will be posted at http://downloads.digium.com/pub/security/AST-2013-001.pdf and http://downloads.digium.com/pub/security/AST-2013-001.html Revision History Date Editor Revisions Made February 11, 2013 Jonathan Rose Initial Draft March 27, 2013 Matt Jordan CVE Added Asterisk Project Security Advisory - AST-2013-001 Copyright (c) 2013 Digium, Inc. All Rights Reserved. Permission is hereby granted to distribute and publish this advisory in its original, unaltered form. -- _ -- 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] AST-2013-002: Denial of Service in HTTP server
Asterisk Project Security Advisory - AST-2013-002 Product Asterisk Summary Denial of Service in HTTP server Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated Sessions SeverityMajor Exploits Known None Reported On January 21, 2013 Reported By Christoph Hebeisen, TELUS Security Labs Posted OnMarch 27, 2013 Last Updated On March 27, 2013 Advisory ContactMark Michelson mmichelson AT digium DOT com CVE NameCVE-2013-2686 Description AST-2012-014 [1], fixed in January of this year, contained a fix for Asterisk's HTTP server since it was susceptible to a remotely-triggered crash. The fix put in place fixed the possibility for the crash to be triggered, but a possible denial of service still exists if an attacker sends one or more HTTP POST requests with very large Content-Length values. [1] http://downloads.asterisk.org/pub/security/AST-2012-014.html Resolution Content-Length is now capped at a maximum value of 1024 bytes. Any attempt to send an HTTP POST with content-length greater than this cap will not result in any memory allocated. The POST will be responded to with an HTTP 413 Request Entity Too Large response. Affected Versions Product Release Series Asterisk Open Source 1.8.x1.8.19.1, 1.8.20.0, 1.8.20.1 Asterisk Open Source 10.x 10.11.1, 10.12.0, 10.12.1 Asterisk Open Source 11.x 11.1.2, 11.2.0, 11.2.1 Certified Asterisk 1.8.151.8.15-cert1 Asterisk Digiumphones 10.x-digiumphones 10.11.1-digiumphones, 10.12.0-digiumphones, 10.12.1-digiumphones Corrected In Product Release Asterisk Open Source 1.8.20.2, 10.12.2, 11.2.2 Certified Asterisk 1.8.15-cert2 Asterisk Digiumphones10.12.2-digiumphones Patches SVN URL Revision http://downloads.asterisk.org/pub/security/AST-2012-014-1.8.diff Asterisk 1.8 http://downloads.asterisk.org/pub/security/AST-2012-014-10.diff Asterisk 10 http://downloads.asterisk.org/pub/security/AST-2012-014-11.diff Asterisk 11 http://downloads.asterisk.org/pub/security/AST-2012-014-1.8.15-cert.diff Certified Asterisk 1.8.15 ++ | Links | https://issues.asterisk.org/jira/browse/ASTERISK-20967 | | | http://telussecuritylabs.com/threats/show/TSL20130327-01| ++ Asterisk Project Security Advisories are posted at http://www.asterisk.org/security This document may be superseded by later versions; if so, the latest version will be posted at http://downloads.digium.com/pub/security/AST-2013-002.pdf and
[asterisk-users] AST-2013-003: Username disclosure in SIP channel driver
Asterisk Project Security Advisory - AST-2013-003 Product Asterisk Summary Username disclosure in SIP channel driver Nature of Advisory Unauthorized data disclosure Susceptibility Remote Unauthenticated Sessions SeverityModerate Exploits Known No Reported On January 30, 2013 Reported By Walter Doekes, OSSO B.V. Posted OnFebruary 21, 2013 Last Updated On March 27, 2013 Advisory ContactKinsey Moore kmo...@digium.com CVE NameCVE-2013-2264 Description When authenticating via SIP with alwaysauthreject enabled, allowguest disabled, and autocreatepeer disabled, Asterisk discloses whether a user exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. This information was disclosed: * when a 407 Proxy Authentication Required response was sent instead of 401 Unauthorized response. * due to the presence or absence of additional tags at the end of 403 Forbidden such as (Bad auth). * when a 401 Unauthorized response was sent instead of 403 Forbidden response after a retransmission. * when retransmissions were sent when a matching peer did not exist, but were not when a matching peer did exist. Resolution This issue can only be mitigated by upgrading to versions of Asterisk that contain the patch or applying the patch. Affected Versions ProductRelease Series Asterisk Open Source 1.8.xAll Versions Asterisk Open Source 10.x All Versions Asterisk Open Source 11.x All Versions Certified Asterisk 1.8.15All Versions Asterisk Business EditionC.3.xAll Versions Asterisk Digiumphones10.x-digiumphones All Versions Corrected In Product Release Asterisk Open Source 1.8.20.2, 10.12.2, 11.2.2 Asterisk Digiumphones10.12.2-digiumphones Certified Asterisk 1.8.15-cert2 Asterisk Business Edition C.3.8.1 Patches SVN URL Revision http://downloads.asterisk.org/pub/security/AST-2013-003-1.8.diff Asterisk 1.8 http://downloads.asterisk.org/pub/security/AST-2013-003-10.diff Asterisk 10 http://downloads.asterisk.org/pub/security/AST-2013-003-11.diff Asterisk 11 http://downloads.asterisk.org/pub/security/AST-2013-003-1.8.15-cert.diff Certified Asterisk 1.8.15 http://downloads.asterisk.org/pub/security/AST-2013-003-C.3.diff Asterisk BE C.3 Links https://issues.asterisk.org/jira/browse/ASTERISK-21013 Asterisk Project Security Advisories are posted at http://www.asterisk.org/security
[asterisk-users] chan_mobile: FXS
Hi all, Finally i got hold of some bt-dongles that seems p[retty stable, the asus-bt211. After installing them, i rebuild 11.3-rc1 added mobile.conf (bt-addres and blackberry address) and mobile show devices is showing me that the BT-link is up, and remains stable up. Seems good, but it looks like asterisk is seeing the BB as a trunk/FXO. However, i want to use the phone as an FXS. Before ending up in trying something that was never foreseen and perhaps even impossible, i was hoping that i could use the BB as an oridinary audio device and still use the keys on the phone for starting/ending calls, and the dialpad for selecting phone numbers. And having the connections go (via BT) through asterisk instead of GSM. Is this possible at all, or am i embarking on a mission impossible ;-) Hans -- _ -- 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
