Re: [asterisk-users] Rate sheet normalization
could MS-Excel possibly be the easiest way to do that normalization ! just merge two rate sheets put some formulas in there and use it in your A2billing or XYZ tool ! On Thu, Mar 15, 2012 at 10:07 AM, Ast Coder asteriskcod...@gmail.comwrote: A2Billing doesn't do that. A2Billing in fact has a lot of shortcomings one of which is this exact issue. I would suggest running rate sheets against each other for finding true LCR and then only uploading the rates that are cheaper into the system. In most cases there are not such high differences but if there are then this is the only way. I know rate normalization talk comes up all the time on FreeSwitch Freenode channel and it probably does on OpenSIPs as well. Check there for some good advice. On Wed, Mar 14, 2012 at 7:35 PM, Benny Amorsen benny+use...@amorsen.dkwrote: Markus unive...@truemetal.org writes: Does such a thing exist? How does a2billing do it? It should be pretty easy in an AGI. If you can afford a linear lookup per call, just grep through the array of prefixes to find the ones matching a particular call, then pick the cheapest from the results. If you need something faster than linear it gets tricky. It would be tempting to preprocess the list to say 5 digits, do a hash lookup on those, and then use the process above. /Benny -- _ -- 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] Getting Mac Address on connected IP phones
2012/3/13, resea...@businesstz.com resea...@businesstz.com: I am struggling to get the mac-addresses of IP phones that are connected to asterisk as the phone are in different VLAN with * and they were manually configured. I want to centralize their configuration using res_phoneprov or tftp I have tried nmap and arp in vain. Have you tried sip show peer and its Useragent field ? Any idea? Sam -- _ -- 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] Getting Mac Address on connected IP phones
I understand you want to choose the easy way but I really think you should not be lazy and go phone by phone and write down the Mac address. Of course if that's ever possible... For future ease of administering those phones, like if you want to do provisioning, troubleshooting etc etc. Better go make one round and than have far more work later on. This is my honest opinion only. Good luck! Sent from my iPhone On Mar 15, 2012, at 6:05 PM, Olivier oza_4...@yahoo.fr wrote: 2012/3/13, resea...@businesstz.com resea...@businesstz.com: I am struggling to get the mac-addresses of IP phones that are connected to asterisk as the phone are in different VLAN with * and they were manually configured. I want to centralize their configuration using res_phoneprov or tftp I have tried nmap and arp in vain. Have you tried sip show peer and its Useragent field ? Any idea? Sam -- _ -- 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 10 and AMR?
On 15-03-12 01:54, Jan Blom wrote: Hello, Is anyone aware of an AMR-NB codec for Asterisk 10? The old patch floating around for older versions of Asterisk doesn’t seem to work anymore. The only patch I have seen is the one for 1.8 which is on sourceforge (search for asterisk-amr). I did a quick test and it compiles fine against 1.8.10. I have not seen a patch for 10. Iirc the 1.8 patch was created by PrivateWave. Perhaps you can ask (hire) them for a 10 version of the patch? 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] Capacity of single instance of Asterisk
Hi, Appreciate everyone for your valuable inputs. All these inputs provided by you are really useful. Thanks Regards, Amit Patkar -- _ -- 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] Caller ID not working in DAHDI 2.6.0
I have a TDM410 with one FXO and one FXS. I've been running dahdi 2.5.0.2 without any problems. A couple of weeks ago I upgraded to 2.6.0 and found that caller ID was no long working for me. All calls came in with a blank caller id. I reverted back to 2.5.0.2 and everything was happy again. I tried upgrading to 2.6.0 again this morning and got the same results. I'm compiling from source on an Ubuntu 10.04.4 box. I was very careful when I merged my settings from my old 2.5.0.2 setup with the new 2.6.0 configuration files. I've searched the bugs and read through the Changes file but didn't see anything obvious. Should I file a bug? -- Chris -- _ -- 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] Caller ID not working in DAHDI 2.6.0
On Thu, Mar 15, 2012 at 10:04:56AM -0500, Chris Gentle wrote: I have a TDM410 with one FXO and one FXS. I've been running dahdi 2.5.0.2 without any problems. A couple of weeks ago I upgraded to 2.6.0 and found that caller ID was no long working for me. All calls came in with a blank caller id. I reverted back to 2.5.0.2 and everything was happy again. I tried upgrading to 2.6.0 again this morning and got the same results. I'm compiling from source on an Ubuntu 10.04.4 box. I was very careful when I merged my settings from my old 2.5.0.2 setup with the new 2.6.0 configuration files. I've searched the bugs and read through the Changes file but didn't see anything obvious. Should I file a bug? Hi Chris, I believe this is fixed in the head of the 2.6 branch. We're prepping a 2.6.0.1 release now... [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10481 If you could try out the branch and let me know if it *doesn't* work for you, I would be appreciative. Thanks, 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
[asterisk-users] External callerid issues using Q931 against Toshiba Strata
Hi Guys,
I currently have an Asterisk 1.6.2.18 server running a patched (see
below) libpri 1.4.10.2 connected to a Toshiba Strata CTX670. All
external calls come in via the Strata and then are routed to the
Asterisk server over a single PRI link using Q931. This setup is
working and has been working for some time (with various earlier
versions of Asterisk) and with a patch (read hack) to libpri I've
managed to successfully pass through the numerical portion of the
callerid from the Strata.
I would like to upgrade to Asterisk 1.8 or 10 and use libpri 1.4.12
but am having difficulties picking up the callerid from the Strata and
due to significant changes in libpri my patch no longer applies.
Below is a pri intense debug capturing the Strata sending through the
callerid with libpri upgraded to 1.4.12 (running Dahdi 2.4.1). Libpri
obviously receives the callerid information, but I am unsure of how to
actually access it in Asterisk. I expect that if the callerid
information is properly acquired and recognized in libpri it would
simply be accessible in Asterisk in the 'CALLERID(all)' variable, but
it is always empty. Internal calls from an extension on the Strata to
an Asterisk extension show the callerid as expected.
Does anyone have any tips on how to get Asterisk to use the callerid
passed through by the Strata?
Thanks!
Justin
chan_dahdi.conf (group 2 is used outgoing only):
[trunkgroups]
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
transfer=yes
cancallforward=yes
echocancel=128
echocancelwhenbridged=yes
echotraining=no
rxgain=0.0
txgain=-10
context=from-toshiba
overlapdial=no
facilityenable=yes
switchtype=qsig
signalling=pri_net
group=1
channel = 1-23
switchtype=national
signalling=pri_cpe
group=2
channel = 25-47
pri intense debug:
TEI: 0 State 7(Multi-frame established)
V(A)=31, V(S)=31, V(R)=42
K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
T200_id=0, N200=3, T203_id=8192
[ 00 01 54 3e 08 02 01 b3 62 1c 66 9f aa 06 80 01 00 82 01 00 a1 31
02 02 01 3a 02 01 0c 30 28 0a 01 01 a0 0f 80 0a 35 35 35 35 35 35 31
36 33 31 0a 01 00 80 0f 41 41 41 20 49 54 2d 44 41 54 41 00 00 00 00
0a 01 01 a1 28 02 02 01 3b 02 01 55 30 1f 86 01 00 a7 1a 06 0a 31 33
31 32 32 31 35 35 35 35 30 0c 81 01 07 8c 04 39 34 31 31 95 01 00 ]
Informational frame:
SAPI: 00 C/R: 0 EA: 0
TEI: 000EA: 1
N(S): 042 0: 0
N(R): 031 P: 0
109 bytes of data
Protocol Discriminator: Q.931 (8) len=109
TEI=0 Call Ref: len= 2 (reference 435/0x1B3) (Sent from originator)
Message Type: FACILITY (98)
[1c 66 9f aa 06 80 01 00 82 01 00 a1 31 02 02 01 3a 02 01 0c 30 28
0a 01 01 a0 0f 80 0a 35 35 35 35 35 35 31 36 33 31 0a 01 00 80 0f 41
41 41 20 49 54 2d 44 41 54 41 00 00 00 00 0a 01 01 a1 28 02 02 01 3b
02 01 55 30 1f 86 01 00 a7 1a 06 0a 31 33 31 32 32 31 35 35 35 35 30
0c 81 01 07 8c 04 39 34 31 31 95 01 00]
Facility (len=104, codeset=0) [ 0x9F, 0xAA, 0x06, 0x80, 0x01, 0x00,
0x82, 0x01, 0x00, 0xA1, '1', 0x02, 0x02, 0x01, ':', 0x02, 0x01, 0x0C,
'0(', 0x0A, 0x01, 0x01, 0xA0, 0x0F, 0x80, 0x0A, '551631', 0x0A,
0x01, 0x00, 0x80, 0x0F, 'AAA IT-DATA', 0x00, 0x00, 0x00, 0x00, 0x0A,
0x01, 0x01, 0xA1, '(', 0x02, 0x02, 0x01, ';', 0x02, 0x01, 'U0', 0x1F,
0x86, 0x01, 0x00, 0xA7, 0x1A, 0x06, 0x0A, '1312210', 0x0C, 0x81,
0x01, 0x07, 0x8C, 0x04, '9411', 0x95, 0x01, 0x00 ]
-- Got ACK for N(S)=31 to (but not including) N(S)=31
-- T200 requested to stop when not started
T203 requested to start without stopping first
-- Starting T203 timer
Received message for call 0x7f5020283b80 on link 0xb2f010 TEI/SAPI 0/0
-- Processing IE 28 (cs0, Facility)
-- Delayed processing IE 28 (cs0, Facility)
ASN.1 dump
Context Specific/C [10 0x0A] AA Len:6 06
Context Specific [0 0x00] 80 Len:1 01
00 - ~
Context Specific [2 0x02] 82 Len:1 01
00 - ~
Context Specific/C [1 0x01] A1 Len:49 31
Integer(2 0x02) 02 Len:2 02
01 3A - ~:
Integer(2 0x02) 02 Len:1 01
0C - ~
Sequence/C(48 0x30) 30 Len:40 28
Enumerated(10 0x0A) 0A Len:1 01
01 - ~
Context Specific/C [0 0x00] A0 Len:15 0F
Context Specific [0 0x00] 80 Len:10 0A
35 35 35 35 35 35 31 36-33 31 - 551631
Enumerated(10 0x0A) 0A Len:1 01
00 - ~
Context Specific [0 0x00] 80 Len:15 0F
41 41 41 20 49 54 2D 44-41 54 41 00 00 00 00 - AAA IT-DATA
Enumerated(10 0x0A) 0A Len:1 01
01 - ~
Context Specific/C [1 0x01] A1 Len:40 28
Integer(2 0x02) 02 Len:2 02
01 3B - ~;
Integer(2 0x02) 02 Len:1 01
55 - U
Sequence/C(48 0x30) 30 Len:31 1F
Context Specific [6 0x06] 86 Len:1 01
00 - ~
Context Specific/C [7 0x07] A7 Len:26 1A
OID(6 0x06) 06 Len:10 0A
31 33 31 32 32 31 35 35-35 35 - 131221
Sequence/C(48 0x30) 30 Len:12 0C
Context Specific [1 0x01] 81 Len:1 01
07 - ~
Context Specific [12 0x0C] 8C Len:4 04
39 34
Re: [asterisk-users] Caller ID not working in DAHDI 2.6.0
On Thu, Mar 15, 2012 at 10:08 AM, Shaun Ruffell sruff...@digium.com wrote: Hi Chris, I believe this is fixed in the head of the 2.6 branch. We're prepping a 2.6.0.1 release now... Hey Shaun. Thanks for the quick reply. I applied the patch for the bug to my 2.6.0 and it works fine. I've made five test calls and the caller ID came through fine. It wasn't coming through at all before, not even intermittently. Thanks for the help! I'll be watching for the 2.6.0.1 release. -- Chris -- _ -- 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] Reliable SIP Trunk Provider
I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. I've worked with Coredial, Broadvox, and Broadvoice and have had some bad experiences with each of these providers. Broadvoice offers low cost service, however I have constant issues with Broadvoice blocking my customers due to Asterisk registering too often. Support either does not respond to e-mails, hangs up on phone calls, or gives me the we don't support Asterisk and we can use your account no problem using the SIP phone on our desk line. Coredial resigned me into a two year agreement after making a change to my SIP trunk configuration without my knowledge, then demanded two years of the full monthly charge when I tried to cancel over a dispute regarding services that I did not order. Check out coredialhorrorstory.com for the whole story. While the service is decent, the customer service leaves much to be desired. Broadvox has been the best provider that I have found so far, however I initially had a lot of issues with sales quoting a product which could not be provisioned and also not being able to deliver service on a timely schedule. I also was given the run around by customer service recently on a simple request to add a DID number to an account. Thanks for your input! -- _ -- 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] Reliable SIP Trunk Provider
I've had pretty good experience with VoicePulse. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jake Wicke Sent: Thursday, March 15, 2012 10:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Reliable SIP Trunk Provider I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. I've worked with Coredial, Broadvox, and Broadvoice and have had some bad experiences with each of these providers. Broadvoice offers low cost service, however I have constant issues with Broadvoice blocking my customers due to Asterisk registering too often. Support either does not respond to e-mails, hangs up on phone calls, or gives me the we don't support Asterisk and we can use your account no problem using the SIP phone on our desk line. Coredial resigned me into a two year agreement after making a change to my SIP trunk configuration without my knowledge, then demanded two years of the full monthly charge when I tried to cancel over a dispute regarding services that I did not order. Check out coredialhorrorstory.com for the whole story. While the service is decent, the customer service leaves much to be desired. Broadvox has been the best provider that I have found so far, however I initially had a lot of issues with sales quoting a product which could not be provisioned and also not being able to deliver service on a timely schedule. I also was given the run around by customer service recently on a simple request to add a DID number to an account. Thanks for your input! -- _ -- 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] Reliable SIP Trunk Provider
You might check flowroute. We have been with them for over a year now and have been spot on with service and support. www.flowroute.com and they are one of the cheapest providers we have found for our needs. Regards, James Miller Agent Black Web Hosting I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 On Thu, Mar 15, 2012 at 11:45, Jake Wicke j...@nxtphase.net wrote: I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. I've worked with Coredial, Broadvox, and Broadvoice and have had some bad experiences with each of these providers. Broadvoice offers low cost service, however I have constant issues with Broadvoice blocking my customers due to Asterisk registering too often. Support either does not respond to e-mails, hangs up on phone calls, or gives me the we don't support Asterisk and we can use your account no problem using the SIP phone on our desk line. Coredial resigned me into a two year agreement after making a change to my SIP trunk configuration without my knowledge, then demanded two years of the full monthly charge when I tried to cancel over a dispute regarding services that I did not order. Check out coredialhorrorstory.com for the whole story. While the service is decent, the customer service leaves much to be desired. Broadvox has been the best provider that I have found so far, however I initially had a lot of issues with sales quoting a product which could not be provisioned and also not being able to deliver service on a timely schedule. I also was given the run around by customer service recently on a simple request to add a DID number to an account. Thanks for your input! -- _ -- 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] Reliable SIP Trunk Provider
On 15/3/12 3:45 pm, Jake Wicke wrote: I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. You should probably let the list know what region/country you're in, as you'll want to be as close (i.e. low latency) to your trunk provider as possible. shameless plugIf you're in the UK, we (Minotaur IT) are a SIP trunk provider, and I'd like to think we support Asterisk and offer decent support :-) /shameless plug Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Rate sheet normalization
With like 10 different ratesheets from 10 different providers, of which many change their rates every few days, manually doing it in Excel is too time consuming... Am 15.03.2012 07:26, schrieb SamyGo: could MS-Excel possibly be the easiest way to do that normalization ! just merge two rate sheets put some formulas in there and use it in your A2billing or XYZ tool ! On Thu, Mar 15, 2012 at 10:07 AM, Ast Coder asteriskcod...@gmail.com mailto:asteriskcod...@gmail.com wrote: A2Billing doesn't do that. A2Billing in fact has a lot of shortcomings one of which is this exact issue. I would suggest running rate sheets against each other for finding true LCR and then only uploading the rates that are cheaper into the system. In most cases there are not such high differences but if there are then this is the only way. I know rate normalization talk comes up all the time on FreeSwitch Freenode channel and it probably does on OpenSIPs as well. Check there for some good advice. On Wed, Mar 14, 2012 at 7:35 PM, Benny Amorsen benny+use...@amorsen.dk mailto:benny%2buse...@amorsen.dk wrote: Markus unive...@truemetal.org mailto:unive...@truemetal.org writes: Does such a thing exist? How does a2billing do it? It should be pretty easy in an AGI. If you can afford a linear lookup per call, just grep through the array of prefixes to find the ones matching a particular call, then pick the cheapest from the results. If you need something faster than linear it gets tricky. It would be tempting to preprocess the list to say 5 digits, do a hash lookup on those, and then use the process above. /Benny -- _ -- 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] External callerid issues using Q931 against Toshiba Strata
I currently have an Asterisk 1.6.2.18 server running a patched (see
below) libpri 1.4.10.2 connected to a Toshiba Strata CTX670. All
external calls come in via the Strata and then are routed to the
Asterisk server over a single PRI link using Q931. This setup is
working and has been working for some time (with various earlier
versions of Asterisk) and with a patch (read hack) to libpri I've
managed to successfully pass through the numerical portion of the
callerid from the Strata.
I would like to upgrade to Asterisk 1.8 or 10 and use libpri 1.4.12
but am having difficulties picking up the callerid from the Strata
and
due to significant changes in libpri my patch no longer applies.
Below is a pri intense debug capturing the Strata sending through the
callerid with libpri upgraded to 1.4.12 (running Dahdi 2.4.1). Libpri
obviously receives the callerid information, but I am unsure of how
to
actually access it in Asterisk. I expect that if the callerid
information is properly acquired and recognized in libpri it would
simply be accessible in Asterisk in the 'CALLERID(all)' variable, but
it is always empty. Internal calls from an extension on the Strata to
an Asterisk extension show the callerid as expected.
Does anyone have any tips on how to get Asterisk to use the callerid
passed through by the Strata?
Thanks!
Justin
chan_dahdi.conf (group 2 is used outgoing only):
[trunkgroups]
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
transfer=yes
cancallforward=yes
echocancel=128
echocancelwhenbridged=yes
echotraining=no
rxgain=0.0
txgain=-10
context=from-toshiba
overlapdial=no
facilityenable=yes
switchtype=qsig
signalling=pri_net
group=1
channel = 1-23
switchtype=national
signalling=pri_cpe
group=2
channel = 25-47
pri intense debug:
TEI: 0 State 7(Multi-frame established)
V(A)=31, V(S)=31, V(R)=42
K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
T200_id=0, N200=3, T203_id=8192
[ 00 01 54 3e 08 02 01 b3 62 1c 66 9f aa 06 80 01 00 82 01 00 a1 31
02 02 01 3a 02 01 0c 30 28 0a 01 01 a0 0f 80 0a 35 35 35 35 35 35 31
36 33 31 0a 01 00 80 0f 41 41 41 20 49 54 2d 44 41 54 41 00 00 00 00
0a 01 01 a1 28 02 02 01 3b 02 01 55 30 1f 86 01 00 a7 1a 06 0a 31 33
31 32 32 31 35 35 35 35 30 0c 81 01 07 8c 04 39 34 31 31 95 01 00 ]
Informational frame:
SAPI: 00 C/R: 0 EA: 0
TEI: 000EA: 1
N(S): 042 0: 0
N(R): 031 P: 0
109 bytes of data
Protocol Discriminator: Q.931 (8) len=109
TEI=0 Call Ref: len= 2 (reference 435/0x1B3) (Sent from originator)
Message Type: FACILITY (98)
[1c 66 9f aa 06 80 01 00 82 01 00 a1 31 02 02 01 3a 02 01 0c 30 28
0a 01 01 a0 0f 80 0a 35 35 35 35 35 35 31 36 33 31 0a 01 00 80 0f 41
41 41 20 49 54 2d 44 41 54 41 00 00 00 00 0a 01 01 a1 28 02 02 01 3b
02 01 55 30 1f 86 01 00 a7 1a 06 0a 31 33 31 32 32 31 35 35 35 35 30
0c 81 01 07 8c 04 39 34 31 31 95 01 00]
Facility (len=104, codeset=0) [ 0x9F, 0xAA, 0x06, 0x80, 0x01, 0x00,
0x82, 0x01, 0x00, 0xA1, '1', 0x02, 0x02, 0x01, ':', 0x02, 0x01, 0x0C,
'0(', 0x0A, 0x01, 0x01, 0xA0, 0x0F, 0x80, 0x0A, '551631', 0x0A,
0x01, 0x00, 0x80, 0x0F, 'AAA IT-DATA', 0x00, 0x00, 0x00, 0x00, 0x0A,
0x01, 0x01, 0xA1, '(', 0x02, 0x02, 0x01, ';', 0x02, 0x01, 'U0', 0x1F,
0x86, 0x01, 0x00, 0xA7, 0x1A, 0x06, 0x0A, '1312210', 0x0C, 0x81,
0x01, 0x07, 0x8C, 0x04, '9411', 0x95, 0x01, 0x00 ]
-- Got ACK for N(S)=31 to (but not including) N(S)=31
-- T200 requested to stop when not started
T203 requested to start without stopping first
-- Starting T203 timer
Received message for call 0x7f5020283b80 on link 0xb2f010 TEI/SAPI
0/0
-- Processing IE 28 (cs0, Facility)
-- Delayed processing IE 28 (cs0, Facility)
ASN.1 dump
Context Specific/C [10 0x0A] AA Len:6 06
Context Specific [0 0x00] 80 Len:1 01
00 - ~
Context Specific [2 0x02] 82 Len:1 01
00 - ~
Context Specific/C [1 0x01] A1 Len:49 31
Integer(2 0x02) 02 Len:2 02
01 3A - ~:
Integer(2 0x02) 02 Len:1 01
0C - ~
Sequence/C(48 0x30) 30 Len:40 28
Enumerated(10 0x0A) 0A Len:1 01
01 - ~
Context Specific/C [0 0x00] A0 Len:15 0F
Context Specific [0 0x00] 80 Len:10 0A
35 35 35 35 35 35 31 36-33 31 - 551631
Enumerated(10 0x0A) 0A Len:1 01
00 - ~
Context Specific [0 0x00] 80 Len:15 0F
41 41 41 20 49 54 2D 44-41 54 41 00 00 00 00 - AAA
IT-DATA
Enumerated(10 0x0A) 0A Len:1 01
01 - ~
Context Specific/C [1 0x01] A1 Len:40 28
Integer(2 0x02) 02 Len:2 02
01 3B - ~;
Integer(2 0x02) 02 Len:1 01
55 - U
Sequence/C(48 0x30) 30 Len:31 1F
Context Specific [6 0x06] 86 Len:1 01
00 - ~
Context Specific/C [7 0x07] A7 Len:26 1A
OID(6 0x06) 06 Len:10 0A
31 33 31 32 32 31 35 35-35 35 - 131221
Sequence/C(48 0x30) 30 Len:12 0C
Re: [asterisk-users] Reliable SIP Trunk Provider
I'm a fan of Vitelity. They are no-frills, but they work well for my very low usage. I think their web portal is ugly, not all that intuitive, but it does work. I've been with them since early 2006 for my few low usage DIDs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jake Wicke Sent: Thursday, March 15, 2012 11:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Reliable SIP Trunk Provider I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. I've worked with Coredial, Broadvox, and Broadvoice and have had some bad experiences with each of these providers. Broadvoice offers low cost service, however I have constant issues with Broadvoice blocking my customers due to Asterisk registering too often. Support either does not respond to e-mails, hangs up on phone calls, or gives me the we don't support Asterisk and we can use your account no problem using the SIP phone on our desk line. Coredial resigned me into a two year agreement after making a change to my SIP trunk configuration without my knowledge, then demanded two years of the full monthly charge when I tried to cancel over a dispute regarding services that I did not order. Check out coredialhorrorstory.com for the whole story. While the service is decent, the customer service leaves much to be desired. Broadvox has been the best provider that I have found so far, however I initially had a lot of issues with sales quoting a product which could not be provisioned and also not being able to deliver service on a timely schedule. I also was given the run around by customer service recently on a simple request to add a DID number to an account. Thanks for your input! -- _ -- 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] Reliable SIP Trunk Provider
On Thu, Mar 15, 2012 at 9:02 AM, Chris Bagnall aster...@lists.minotaur.ccwrote: On 15/3/12 3:45 pm, Jake Wicke wrote: I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. You should probably let the list know what region/country you're in, as you'll want to be as close (i.e. low latency) to your trunk provider as possible. Also the amount of traffic and number of DIDs, since some of us serve specific size ranges. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Rate sheet normalization
On Thursday 15 Mar 2012, Markus wrote: With like 10 different ratesheets from 10 different providers, of which many change their rates every few days, manually doing it in Excel is too time consuming... Is it possible to get samples? I'd be interested in looking into developing a script that can handle this problem generically, and presumably you're available to alpha- and beta-test in any case :) Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] Reliable SIP Trunk Provider
+1 for flowroute. very cheap and their support has been top notch when any issues have come up On Thu, Mar 15, 2012 at 12:15 PM, Carlos Alvarez car...@televolve.comwrote: On Thu, Mar 15, 2012 at 9:02 AM, Chris Bagnall aster...@lists.minotaur.cc wrote: On 15/3/12 3:45 pm, Jake Wicke wrote: I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. You should probably let the list know what region/country you're in, as you'll want to be as close (i.e. low latency) to your trunk provider as possible. Also the amount of traffic and number of DIDs, since some of us serve specific size ranges. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Reliable SIP Trunk Provider
I have been using bandwidth.com since 2006 and have no problems at all. They do not support t38 but have free local termination in a lot of US cities. Tech support is good and they do support asterisk. Jonn On 03/15/2012 10:45 AM, Jake Wicke wrote: I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. I've worked with Coredial, Broadvox, and Broadvoice and have had some bad experiences with each of these providers. Broadvoice offers low cost service, however I have constant issues with Broadvoice blocking my customers due to Asterisk registering too often. Support either does not respond to e-mails, hangs up on phone calls, or gives me the we don't support Asterisk and we can use your account no problem using the SIP phone on our desk line. Coredial resigned me into a two year agreement after making a change to my SIP trunk configuration without my knowledge, then demanded two years of the full monthly charge when I tried to cancel over a dispute regarding services that I did not order. Check out coredialhorrorstory.com for the whole story. While the service is decent, the customer service leaves much to be desired. Broadvox has been the best provider that I have found so far, however I initially had a lot of issues with sales quoting a product which could not be provisioned and also not being able to deliver service on a timely schedule. I also was given the run around by customer service recently on a simple request to add a DID number to an account. Thanks for your input! -- _ -- 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] Reliable SIP Trunk Provider
Jake Wicke wrote: I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. I've worked with Coredial, Broadvox, and Broadvoice and have had some bad experiences with each of these providers. I'm going to assume that you're in the US, since those 3 providers are all based here. I can highly recommend XO Communications. http://www.xo.com/ We currently have 35 SIP trunks with them and will be adding more. Our corp office is on a DS3 SIP trunk with 500 DID's and our stores are on a T1 SIP trunk with 100 DID's. They have several levels of support. We use their upper level support called SNA (I forget what it stands for), which gives us direct access to their upper level engineers when needed. Their front line support people that I deal with are very good and may be VoIP engineers themselves. Ron Bergin Network Operations Administrator Fry's Electronics Inc. -- _ -- 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] disable dahdi pri
Hello, is there a way to disable a span for maintenance purpose (i.e. send yellow alarm) ? What would be the correct ioctl definition ? DAHDI_MAINT seems not to be the right candidate. Would DAHDI_SHUTDOWN send an alarm ? Thanks 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
Re: [asterisk-users] Transfer to fax
On Tuesday, March 13, 2012, Kevin P. Fleming kpflem...@digium.com wrote: On 03/13/2012 05:45 PM, Eric Wieling wrote: The faxdetect option is documented in the 1.8 sip.conf.sample. Right, I forgot about that. Now I've really confused things. /me heads back to his hole It was actually added to chan sip in 1.6.2, I remember that being a selling point on a 1.6.2 upgrade for a client of mine about a year and a half ago. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com 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] disable dahdi pri
On Thu, Mar 15, 2012 at 05:44:53PM +0100, Johann Steinwendtner wrote: is there a way to disable a span for maintenance purpose (i.e. send yellow alarm) ? This could be a good feature to add to the dahdi_maint utility. What would be the correct ioctl definition ? DAHDI_MAINT seems not to be the right candidate. You could add an additional context in the DAHDI_MAINT ioctl handler of your base card to set/unset the yellow alarm bit in the framer. For an example, see the Yellow Alarm handler in t4_check_alarms() and the DAHDI_MAINT ioctl handler in t4_maint() in drivers/dahdi/wct4xxp/base.c Would DAHDI_SHUTDOWN send an alarm ? No. This will shutdown the entire card. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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] Reliable SIP Trunk Provider
On Thu,Mar 15 12:10:PM, Eric Wieling wrote: I'm a fan of Vitelity. They are no-frills, but they work well for my very low usage. I think their web portal is ugly, not all that intuitive, but it does work. I've been with them since early 2006 for my few low usage DIDs. +1 for Vitelity , I like them for recognizing the fact that some people actually prefer to run pure Asterisk (no GUI) . Guy Gold -- _ -- 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] Reliable SIP Trunk Provider
www.didforslae.com have wide range of products to fit low usage to very high usage. Dont want to put too much details here. Check it out let me know if interested, since you are using I will help you waive activation fee. -Jai On Thu, Mar 15, 2012 at 11:30 AM, Guy Gold g...@the-golds.us wrote: On Thu,Mar 15 12:10:PM, Eric Wieling wrote: I'm a fan of Vitelity. They are no-frills, but they work well for my very low usage. I think their web portal is ugly, not all that intuitive, but it does work. I've been with them since early 2006 for my few low usage DIDs. +1 for Vitelity , I like them for recognizing the fact that some people actually prefer to run pure Asterisk (no GUI) . Guy Gold -- _ -- 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] Reliable SIP Trunk Provider
I use flowroute.com. Intuitive GUI, cheap, and good customer service. On Thu, Mar 15, 2012 at 1:30 PM, Guy Gold g...@the-golds.us wrote: On Thu,Mar 15 12:10:PM, Eric Wieling wrote: I'm a fan of Vitelity. They are no-frills, but they work well for my very low usage. I think their web portal is ugly, not all that intuitive, but it does work. I've been with them since early 2006 for my few low usage DIDs. +1 for Vitelity , I like them for recognizing the fact that some people actually prefer to run pure Asterisk (no GUI) . Guy Gold -- _ -- 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 -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- 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 1.8.4 polycom sp650
Hello All, I'm having issues with asterisk 1.8.4 dropping calls during transfer, and transfer to park extension. We're using polycom soundpoint IP 650. when the park button is hit the response is i'm sorry not an extension at the same time number 7 appers on the lcd. Thanks in advance. -motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.4 polycom sp650
I'm having issues with asterisk 1.8.4 dropping calls during transfer, and transfer to park extension. We're using polycom soundpoint IP 650. when the park button is hit the response is i'm sorry not an extension at the same time number 7 appers on the lcd. Please use a newer version of Asterisk. A lot of parking issues have been fixed since that version. Richard -- _ -- 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] how to show used wrong password
On Wed, Mar 14, 2012 at 1:36 PM, Randall rand...@songshu.org wrote: all works as expected only there is 1 extension that is trying to register with a wrong password causing fail2ban to block the IP address, normally that is ok behaviour but i have several extensions on that IP address. First of all, white list the IP in fail2ban and you won't accidentally ban the whole office. This can be done by following this guide: http://www.fail2ban.org/wiki/index.php/Whitelist Second, this is kind of outside the box thinking, so it may not work at all, but try setting the NAT on that peer to no, and then tcpdump the incoming registration attempts and see if you can see the internal private IP address of the packet. If there's a SIP helper on the far end, this may not help. Possibly, remove the secret= line from that peer in sip.conf and see if it successfully registers. Again, with the right nat= setting, you may be able to tcpdump the communication with that peer and get the private IP address so that you can then attempt narrow it down. This is not a long term solution, obviously, as it would create a gaping security hole, but it's worth a shot. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com 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
[asterisk-users] AST-2012-003: Stack Buffer Overflow in HTTP Manager
Asterisk Project Security Advisory - AST-2012-003 Product Asterisk Summary Stack Buffer Overflow in HTTP Manager Nature of Advisory Exploitable Stack Buffer Overflow Susceptibility Remote Unauthenticated Sessions SeverityCritical Exploits Known No Reported On 03/15/2012 Reported By Russell Bryant Posted On03/15/2012 Last Updated On March 15, 2012 Advisory ContactMatt Jordan mjordan AT digium DOT com CVE Name Description An attacker attempting to connect to an HTTP session of the Asterisk Manager Interface can send an arbitrarily long string value for HTTP Digest Authentication. This causes a stack buffer overflow, with the possibility of remote code injection. Resolution Upgrade to one of the versions of Asterisk listed in the Corrected In section, or apply a patch specified in the Patches section. Affected Versions Product Release Series Asterisk Open Source1.8.x All versions Asterisk Open Source 10.x All versions Corrected In Product Release Asterisk Open Source 1.8.10.1 Asterisk Open Source10.2.1 Patches SVN URL Revision http://downloads.asterisk.org/pub/security/AST-2012-003-1.8.diff v1.8 http://downloads.asterisk.org/pub/security/AST-2012-003-10.diff v10 Links https://issues.asterisk.org/jira/browse/ASTERISK-19542 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/.pdf and http://downloads.digium.com/pub/security/.html Revision History Date Editor Revisions Made 03-15-2012 Matt Jordan Initial release Asterisk Project Security Advisory - AST-2012-003 Copyright (c) 2012 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-2012-002: Remote Crash Vulnerability in Milliwatt Application
Asterisk Project Security Advisory - AST-2012-002 ProductAsterisk SummaryRemote Crash Vulnerability in Milliwatt Application Nature of Advisory Exploitable Stack Buffer Overflow with locally defined data SusceptibilityRemote Unauthenticated Sessions Severity Minor Exploits KnownNo Reported On 03/14/2012 Reported By Russell Bryant Posted On 03/15/2012 Last Updated OnMarch 15, 2012 Advisory Contact Matt Jordan mjordan AT digium DOT com CVE Name Description An attacker can cause Asterisk to crash in one of two ways: 1. A dialplan uses the Milliwatt application with 'o' option 2. The internal_timing opion in asterisk.conf is off 3. The attacker sends a large audio packet. The number of samples in the audio packet determines the number of internal data samples that are copied into the buffer. This overruns the buffer, potentially causing a crash. OR 1. A diaplan uses the Milliwatt application with the 'o' option 2. The attacker negotiates a media format with a sampling rate greater than 32kHz. The application will attempt to generate an audio packet using the sample rate of the negotiated format, where the sample rate will require a number of data points greater then the size of the buffer. Again, the the application copies a number of internal data samples into the buffer that are greater then the size of the buffer, potentially causing a crash. Note that the latter attack vector is only possible in Asterisk 10, as it supports codecs with a sample rate greater then 32kHz. Resolution Upgrade to one of the versions of Asterisk listed in the Corrected In section, or apply a patch specified in the Patches section. Affected Versions Product Release Series Asterisk Open Source1.4.x All Versions Asterisk Open Source 1.6.2.x All Versions Asterisk Open Source1.8.x All Versions Asterisk Open Source 10.x All Versions Corrected In Product Release Asterisk Open Source1.4.44 Asterisk Open Source 1.6.2.23 Asterisk Open Source 1.8.10.1 Asterisk Open Source10.2.1 Patches SVN URL Revision http://downloads.asterisk.org/pub/security/AST-2012-002-1.4.diff v1.4 http://downloads.asterisk.org/pub/security/AST-2012-002-1.6.2.diff v1.6.2 http://downloads.asterisk.org/pub/security/AST-2012-002-1.8.diff v1.8 http://downloads.asterisk.org/pub/security/AST-2012-002-10.diffv10 Links https://issues.asterisk.org/jira/browse/ASTERISK-19541 Asterisk
[asterisk-users] Asterisk 1.4.44, 1.6.2.23, 1.8.10.1, 10.2.1 Now Available (Security Releases)
The Asterisk Development Team has announced security releases for Asterisk 1.4, 1.6.2, 1.8, and 10. The available security releases are released as versions 1.4.44, 1.6.2.23, 1.8.10.1, and 10.2.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of Asterisk 1.4.44 and 1.6.2.23 resolve an issue wherein app_milliwatt can potentially overrun a buffer on the stack, causing Asterisk to crash. This does not have the potential for remote code execution. The release of Asterisk 1.8.10.1 and 10.2.1 resolve two issues. First, they resolve the issue in app_milliwatt, wherein a buffer can potentially be overrun on the stack, but no remote code execution is possible. Second, they resolve an issue in HTTP AMI where digest authentication information can be used to overrun a buffer on the stack, allowing for code injection and execution. These issues and their resolution are described in the security advisory. For more information about the details of these vulnerabilities, please read the security advisories AST-2012-002 and AST-2012-003, 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/asterisk/releases/ChangeLog-1.4.44 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.23 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.10.1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.2.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2012-002.pdf * http://downloads.asterisk.org/pub/security/AST-2012-003.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
Re: [asterisk-users] Rate sheet normalization
Am 15.03.2012 17:20, schrieb Raj Mathur (राज माथुर): On Thursday 15 Mar 2012, Markus wrote: With like 10 different ratesheets from 10 different providers, of which many change their rates every few days, manually doing it in Excel is too time consuming... Is it possible to get samples? I'd be interested in looking into developing a script that can handle this problem generically, and presumably you're available to alpha- and beta-test in any case :) Most definitely! I'll get in touch off-list. :) -- _ -- 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] Rate sheet normalization
I would be more interested in a system where quality routes are tested with different providers because rate really doesn't matter if a call can't be placed or if a destination is a fake one. We have seen many fake destinations with top tier providers but they had the best rates so the strategy to pick them first really didn't work. So, maybe a subscription service where a dialler system continuously tests routes with a list of 10 providers so that it's established which routes actually work and then allow that data to be downloaded for usage. On Thu, Mar 15, 2012 at 8:42 PM, Markus unive...@truemetal.org wrote: Am 15.03.2012 17:20, schrieb Raj Mathur (राज माथुर): On Thursday 15 Mar 2012, Markus wrote: With like 10 different ratesheets from 10 different providers, of which many change their rates every few days, manually doing it in Excel is too time consuming... Is it possible to get samples? I'd be interested in looking into developing a script that can handle this problem generically, and presumably you're available to alpha- and beta-test in any case :) Most definitely! I'll get in touch off-list. :) -- __**__**_ -- 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] Rate sheet normalization
Our system just rolls over until it finds a carrier that will take it. Up to 30 different routes are supported, and rollover is pretty instantaneous in most cases. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Atlanta, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com On Mar 15, 2012, at 11:14 PM, Ast Coder asteriskcod...@gmail.com wrote: I would be more interested in a system where quality routes are tested with different providers because rate really doesn't matter if a call can't be placed or if a destination is a fake one. We have seen many fake destinations with top tier providers but they had the best rates so the strategy to pick them first really didn't work. So, maybe a subscription service where a dialler system continuously tests routes with a list of 10 providers so that it's established which routes actually work and then allow that data to be downloaded for usage. On Thu, Mar 15, 2012 at 8:42 PM, Markus unive...@truemetal.org wrote: Am 15.03.2012 17:20, schrieb Raj Mathur (राज माथुर): On Thursday 15 Mar 2012, Markus wrote: With like 10 different ratesheets from 10 different providers, of which many change their rates every few days, manually doing it in Excel is too time consuming... Is it possible to get samples? I'd be interested in looking into developing a script that can handle this problem generically, and presumably you're available to alpha- and beta-test in any case :) Most definitely! I'll get in touch off-list. :) -- _ -- 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] Rate sheet normalization
So, maybe a subscription service where a dialler system continuously tests routes with a list of 10 providers so that it's established which routes actually work and then allow that data to be downloaded for usage. I think that it may not be humanly possible and also not possible to have a separate automated setup to ring destinations and prioritize according to Quality or Least-rates. BUT I am sure that real-time call success rate(or ASR) via multiple providers and sorting providers accordingly for particular destinations is possible or maybe available. Provided a good piece of code is written which analyses the call status / quality and then picks favourite carrier/provider for any destination. !! Not sure if anyone can understand it completely ;) I am thinking in terms of DynamicRouting or LCR modules from Kamailio or OpenSIPS. Regards, Sammy On Fri, Mar 16, 2012 at 8:14 AM, Ast Coder asteriskcod...@gmail.com wrote: I would be more interested in a system where quality routes are tested with different providers because rate really doesn't matter if a call can't be placed or if a destination is a fake one. We have seen many fake destinations with top tier providers but they had the best rates so the strategy to pick them first really didn't work. So, maybe a subscription service where a dialler system continuously tests routes with a list of 10 providers so that it's established which routes actually work and then allow that data to be downloaded for usage. On Thu, Mar 15, 2012 at 8:42 PM, Markus unive...@truemetal.org wrote: Am 15.03.2012 17:20, schrieb Raj Mathur (राज माथुर): On Thursday 15 Mar 2012, Markus wrote: With like 10 different ratesheets from 10 different providers, of which many change their rates every few days, manually doing it in Excel is too time consuming... Is it possible to get samples? I'd be interested in looking into developing a script that can handle this problem generically, and presumably you're available to alpha- and beta-test in any case :) Most definitely! I'll get in touch off-list. :) -- __**__**_ -- 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 -- _ -- 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] Rate sheet normalization
How about a central coop that manages the “normalized” rate sheet and distributes it with “unknown” call quality metrics for each route. Coop members report call quality for all calls/routes so the call quality metrics can be updated in the rate sheet and distributed to members. --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of SamyGo Sent: Thursday, March 15, 2012 11:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Rate sheet normalization So, maybe a subscription service where a dialler system continuously tests routes with a list of 10 providers so that it's established which routes actually work and then allow that data to be downloaded for usage. I think that it may not be humanly possible and also not possible to have a separate automated setup to ring destinations and prioritize according to Quality or Least-rates. BUT I am sure that real-time call success rate(or ASR) via multiple providers and sorting providers accordingly for particular destinations is possible or maybe available. Provided a good piece of code is written which analyses the call status / quality and then picks favourite carrier/provider for any destination. !! Not sure if anyone can understand it completely ;) I am thinking in terms of DynamicRouting or LCR modules from Kamailio or OpenSIPS. Regards, Sammy On Fri, Mar 16, 2012 at 8:14 AM, Ast Coder asteriskcod...@gmail.com wrote: I would be more interested in a system where quality routes are tested with different providers because rate really doesn't matter if a call can't be placed or if a destination is a fake one. We have seen many fake destinations with top tier providers but they had the best rates so the strategy to pick them first really didn't work. So, maybe a subscription service where a dialler system continuously tests routes with a list of 10 providers so that it's established which routes actually work and then allow that data to be downloaded for usage. On Thu, Mar 15, 2012 at 8:42 PM, Markus unive...@truemetal.org wrote: Am 15.03.2012 17:20, schrieb Raj Mathur (राज माथुर): On Thursday 15 Mar 2012, Markus wrote: With like 10 different ratesheets from 10 different providers, of which many change their rates every few days, manually doing it in Excel is too time consuming... Is it possible to get samples? I'd be interested in looking into developing a script that can handle this problem generically, and presumably you're available to alpha- and beta-test in any case :) Most definitely! I'll get in touch off-list. :) -- _ -- 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] Rate sheet normalization
Good Idea but that means all the members of the coop use the same vendors and it may not be suitable for servers having a bad network with a premium quality provider and thus mark it as bad, whereas others are marking it as Good. !! On Fri, Mar 16, 2012 at 10:00 AM, Don Kelly d...@donkelly.biz wrote: How about a central coop that manages the “normalized” rate sheet and distributes it with “unknown” call quality metrics for each route. Coop members report call quality for all calls/routes so the call quality metrics can be updated in the rate sheet and distributed to members. --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *SamyGo *Sent:* Thursday, March 15, 2012 11:45 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Rate sheet normalization ** ** So, maybe a subscription service where a dialler system continuously tests routes with a list of 10 providers so that it's established which routes actually work and then allow that data to be downloaded for usage. ** ** I think that it may not be humanly possible and also not possible to have a separate automated setup to ring destinations and prioritize according to Quality or Least-rates. BUT I am sure that real-time call success rate(or ASR) via multiple providers and sorting providers accordingly for particular destinations is possible or maybe available. Provided a good piece of code is written which analyses the call status / quality and then picks favourite carrier/provider for any destination. !! Not sure if anyone can understand it completely ;) I am thinking in terms of DynamicRouting or LCR modules from Kamailio or OpenSIPS. ** ** Regards, Sammy ** ** On Fri, Mar 16, 2012 at 8:14 AM, Ast Coder asteriskcod...@gmail.com wrote: I would be more interested in a system where quality routes are tested with different providers because rate really doesn't matter if a call can't be placed or if a destination is a fake one. We have seen many fake destinations with top tier providers but they had the best rates so the strategy to pick them first really didn't work. ** ** So, maybe a subscription service where a dialler system continuously tests routes with a list of 10 providers so that it's established which routes actually work and then allow that data to be downloaded for usage. ** ** ** ** On Thu, Mar 15, 2012 at 8:42 PM, Markus unive...@truemetal.org wrote:*** * Am 15.03.2012 17:20, schrieb Raj Mathur (राज माथुर): ** ** On Thursday 15 Mar 2012, Markus wrote: With like 10 different ratesheets from 10 different providers, of which many change their rates every few days, manually doing it in Excel is too time consuming... Is it possible to get samples? I'd be interested in looking into developing a script that can handle this problem generically, and presumably you're available to alpha- and beta-test in any case :) ** ** Most definitely! I'll get in touch off-list. :) -- _ -- 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] Rate sheet normalization
So the best fit is to create a piece of code which mixes providers and destinations. 1- Sort those with at first rate and then as time passes by(utilization) it rearranges the carriers according to quality metrics or economical rates or success ratios. Like an AI system which enhances itself with passage of time. On Fri, Mar 16, 2012 at 10:06 AM, SamyGo govoi...@gmail.com wrote: Good Idea but that means all the members of the coop use the same vendors and it may not be suitable for servers having a bad network with a premium quality provider and thus mark it as bad, whereas others are marking it as Good. !! On Fri, Mar 16, 2012 at 10:00 AM, Don Kelly d...@donkelly.biz wrote: How about a central coop that manages the “normalized” rate sheet and distributes it with “unknown” call quality metrics for each route. Coop members report call quality for all calls/routes so the call quality metrics can be updated in the rate sheet and distributed to members. --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *SamyGo *Sent:* Thursday, March 15, 2012 11:45 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Rate sheet normalization ** ** So, maybe a subscription service where a dialler system continuously tests routes with a list of 10 providers so that it's established which routes actually work and then allow that data to be downloaded for usage. ** ** I think that it may not be humanly possible and also not possible to have a separate automated setup to ring destinations and prioritize according to Quality or Least-rates. BUT I am sure that real-time call success rate(or ASR) via multiple providers and sorting providers accordingly for particular destinations is possible or maybe available. Provided a good piece of code is written which analyses the call status / quality and then picks favourite carrier/provider for any destination. !! Not sure if anyone can understand it completely ;) I am thinking in terms of DynamicRouting or LCR modules from Kamailio or OpenSIPS. ** ** Regards, Sammy ** ** On Fri, Mar 16, 2012 at 8:14 AM, Ast Coder asteriskcod...@gmail.com wrote: I would be more interested in a system where quality routes are tested with different providers because rate really doesn't matter if a call can't be placed or if a destination is a fake one. We have seen many fake destinations with top tier providers but they had the best rates so the strategy to pick them first really didn't work. ** ** So, maybe a subscription service where a dialler system continuously tests routes with a list of 10 providers so that it's established which routes actually work and then allow that data to be downloaded for usage. ** ** ** ** On Thu, Mar 15, 2012 at 8:42 PM, Markus unive...@truemetal.org wrote:** ** Am 15.03.2012 17:20, schrieb Raj Mathur (राज माथुर): ** ** On Thursday 15 Mar 2012, Markus wrote: With like 10 different ratesheets from 10 different providers, of which many change their rates every few days, manually doing it in Excel is too time consuming... Is it possible to get samples? I'd be interested in looking into developing a script that can handle this problem generically, and presumably you're available to alpha- and beta-test in any case :) ** ** Most definitely! I'll get in touch off-list. :) -- _ -- 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?
