Re: [asterisk-users] External callerid issues using Q931 against Toshiba Strata
On Thu, Mar 15, 2012 at 12:09 PM, Richard Mudgett rmudg...@digium.com wrote:
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: 000 EA: 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
[asterisk-users] SendText causes Retransmission errors
Hi, I'm using SendText to send a text message when the user picks up a line in a SLA setup (even though I'm not sure the problem is related to SLA). I'm on Asterisk 10.2.1 (same in 1.8.9) [from-office] .. same = n,SendText(hi) same = n,SLAStation(line1234) .. Here is a simplified version of the SIP messages: 1 phone = Asterisk INVITE 2 Asterisk = phone Trying 3 Asterisk = phone MESSAGE 4 Asterisk = phone OK (for the INVITE at 1) 5 phone = Asterisk OK (for the MESSAGE at 3) 6 Asterisk = phone OK (for the INVITE at 1)*** RESEND of 4 7 Asterisk = phone OK (for the INVITE at 1)*** RESEND of 4 .. The text message is sent and the call is connected, but Asterisk keeps resending OK for the INVITE, and eventually drops the call after Transmission timeout. If I insert a WAIT after SendText, the order of the OKs changes, and everything works: same = n,SendText(hi) same = n,Wait(1) same = n,SLAStation(line1234) Here is the SIP message flow with WAIT (4 and 5 above are swapped): 1 phone = Asterisk INVITE 2 Asterisk = phone Trying 3 Asterisk = phone MESSAGE 4 phone = Asterisk OK (for the MESSAGE at 3) 5 Asterisk = phone OK (for the INVITE at 1) Is there anything else I can do other than using WAIT (which might not be a consistent solution anyway)? Thanks, Matt -- _ -- 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 16.03.2012 04:14, schrieb Ast Coder: 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. Ok, but how would this system handle FAS? Call shows as connected but it's still ringing to the caller. And probably won't get connected to the callee ever. I.e. cases where the route is just faulty in some way and it will ring forever but actually no one will ever get connected. How do you differentiate between nobody is home and route is faulty... -- _ -- 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
If we had reports of every call, we could downgrade status of routes that had frequency calls not completed that were outside the norm. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Sent: Friday, March 16, 2012 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Rate sheet normalization Am 16.03.2012 04:14, schrieb Ast Coder: 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. Ok, but how would this system handle FAS? Call shows as connected but it's still ringing to the caller. And probably won't get connected to the callee ever. I.e. cases where the route is just faulty in some way and it will ring forever but actually no one will ever get connected. How do you differentiate between nobody is home and route is faulty... -- _ -- 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] SendText causes Retransmission errors
On 03/16/2012 09:43 AM, Matt Hamilton wrote: Hi, I'm using SendText to send a text message when the user picks up a line in a SLA setup (even though I'm not sure the problem is related to SLA). I'm on Asterisk 10.2.1 (same in 1.8.9) [from-office] .. same = n,SendText(hi) same = n,SLAStation(line1234) .. Here is a simplified version of the SIP messages: 1 phone = Asterisk INVITE 2 Asterisk = phone Trying 3 Asterisk = phone MESSAGE 4 Asterisk = phone OK (for the INVITE at 1) 5 phone = Asterisk OK (for the MESSAGE at 3) 6 Asterisk = phone OK (for the INVITE at 1)*** RESEND of 4 7 Asterisk = phone OK (for the INVITE at 1)*** RESEND of 4 .. Did the phone send an ACK for message 4? If not, that explains why Asterisk is retransmitting the '200 OK'. Posting a packet capture of this problem occurring would probably provide the details necessary to figure out what is going on. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] External callerid issues using Q931 against Toshiba Strata
snip
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: 000 EA: 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 31 31 - 9411
Context Specific [21 0x15] 95 Len:1 01
00 - ~
ASN.1 end
nfe NetworkFacilityExtension Context Specific/C [10 0x0A]
sourceEntity Context Specific [0 0x00] = 0 0x
destinationEntity Context Specific [2 0x02] = 0 0x
INVOKE Component Context Specific/C [1 0x01]
invokeId Integer(2 0x02) = 314 0x013A
operationValue Integer(2 0x02) = 12 0x000C
operationValue = ROSE_QSIG_CallTransferComplete
CallTransferComplete Sequence/C(48 0x30)
endDesignation Enumerated(10 0x0A) = 1 0x0001
redirectionNumber PresentedNumberScreened
presentationAllowedNumber NumberScreened Context Specific/C [0
0x00]
partyNumber PartyNumber
unknownPartyNumber Context Specific [0 0x00] = 551631
screeningIndicator Enumerated(10 0x0A) = 0 0x
redirectionName Name
namePresentationAllowedSimple Context Specific [0 0x00] =
41 41 41 20 49 54 2D 44-41 54 41 00 00 00 00 - AAA
IT-DATA
callStatus Enumerated(10 0x0A) = 1 0x0001
INVOKE Component Context Specific/C [1 0x01]
invokeId Integer(2 0x02) = 315 0x013B
operationValue Integer(2 0x02) = 85 0x0055
operationValue = ROSE_Unknown
Skipping unused constructed component octets!
!! ROSE invoke operation not handled on switchtype:Q.SIG switch!
ROSE_Unknown
snip
I'm looking for the information contained in the
ROSE_QSIG_CallTransferComplete section. I have attached a complete
log
of the call (with intense debug on unfortunately). An additional
note:
1. The Strata system initially sends through a virtual internal
Re: [asterisk-users] [Slightly OT] Audiocodes Mediant Failover Routing with Asterisk
- Original Message - Greetings- First off, my apologies for the slightly OT nature of this post. It does involve Asterisk to a degree, but errs a bit on the side of Audiocodes inquiry. I accept all responsibility for my actions and the consequences. :) The scenario is this: I have an Asterisk box connected to a Mediant 2000 (M2K) via T1. Calls made via DAHDI-T1-M2K are then routed by the M2K via SIP using the Audiocodes Tel-to-IP routing tables to a remote IP. This has worked quite well for some time. I now want to add multiple IP's to the routing table for failover. However, when adding additional entries to the Tel-to-IP routing table, and the first entry fails, the other IPs are not attempted. The first IP is attempted, and if it fails, the call fails instead of trying any of the additional IPs in the Tel-to-IP routing table. *BUMP* Any ideas from the brilliant folks here? --Tim -- _ -- 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 had many of the same problems with sip station. If you just need sip termination, Check out flow route. The service just seems to work properly for me, and they respond to tickets. You can open up new cases through their site. On Mar 15, 2012 11:48 AM, 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
Jake, We are DIDForSale support asterisk. We do IP based authentication and do not require registration. You can test our DIDs without paying anything. I am sending you the rates just to make it easy to compare apple to apple, no run around for pricing ;) . Let me know if I can have an opportunity to earn your business. We have been in business for 5 year and handle traffic for few of top calling card companies. So have very stable environment. Thanks for your interest in our service. Our Product offerings: We sell DIDs all over US. For the list of all the available rate centers please visit us at http://www.didforsale.com/moreinfo.php?help=ratecenter. Please contact Sangeeta at +19494568787 for details. We have inbound DIDs in 4 different configurations. 1) Flat Rate DID with 20 channels ($8.99 per DID per month + $5 Activation per DID)* #of DIDs Rate (USD)/DID 1-30 $8.99 31-100 $8.75 101-200 $8.50 201+ Contact us http://www.didforsale.com/contactus.php - NOTE: - Average Limit of 8000 inbound minutes, per DID per month. - Overage will be charged at $0.008 per minute. 2) DID with metered inbound (Pay as you go) $1 per DID + $5 activation per DID and 0.4 cents ($0.004) per minute for all incoming calls. 3) Toll free numbers $3 per month and 1.9 cents per minute. 4) Channelized Option Inbound: For high usage customers, like Calling Card, Call Centers, Conferences etc. #of Channel’s/Month Rate (USD)/Channel Rate (USD)/DID Rate (USD)/DID 1-30 $8.99 1-200 $0.99 31-100 $8.75 201-500 $0.85 101-200 $8.50 501-1000 $0.70 201+ Contact us http://www.didforsale.com/contactus.php 1000+http://www.didforsale.com/contactus.php Contact us. http://www.didforsale.com/contactus.php - Note: - DID/Channel ratio can not be more that 25. Example, for Every 25 DIDs you must have 1 channel. - Activation fee $1/DID. 5) Outbound is all metered The rates depends on the volume of commitment. For US rates start from 1.9 cents per minute. 6) Porting a Number We can port a number if it exists in the rate centers. It costs additional $10 fee. Before you buy our DID you can test our service for free. Free trial does not require you make any payment or purchase. Please follow these steps to reserve a test DID for free trial for 6 hours:- a) Signup to create an account on our website. Your login id is the email id that you created the account with. b) Login into your account and Click on Testing Center link on the Left hand side menu. Select the DID to test. To know what our customer are saying about us, please click on the link, http://www.didforsale.com/blog/2011-didforsale-customer-satisfaction-survey-resultshttp://www.didforsale.com/blog/?p=103 Please let me know in case you need any more information. Thanks Regards, www.didforsale.com On Fri, Mar 16, 2012 at 10:10 AM, white hat whitehat...@gmail.com wrote: I had many of the same problems with sip station. If you just need sip termination, Check out flow route. The service just seems to work properly for me, and they respond to tickets. You can open up new cases through their site. On Mar 15, 2012 11:48 AM, 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 --
[asterisk-users] a2billing script
hi folks, i was wondering if some one has a2billing script,which can be used to install a2billing easly ? thanks in advance -- * HARAZ Tahar * *Engineering Student at the National Institute for Posts and Telecommunications (INPT) * * Phone: +212 6 78030050 E-mail: harazta...@gmail.com ouabimedcha...@gmail.com * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to show used wrong password
On 03/15/2012 02:38 PM, Warren Selby wrote: 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. There's an interesting option in there: if you remove the 'secret', then the peer will be able to register. Once it is registered, you can call it, and the user/owner/etc. will hopefully be there so you can tell them to fix their endpoint. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing script
We have had success with this: https://sites.google.com/site/a2billing2asterisk/ On Fri, Mar 16, 2012 at 1:28 PM, Tahar .H harazta...@gmail.com wrote: hi folks, i was wondering if some one has a2billing script,which can be used to install a2billing easly ? thanks in advance -- * HARAZ Tahar * *Engineering Student at the National Institute for Posts and Telecommunications (INPT) * * Phone: +212 6 78030050 E-mail: harazta...@gmail.com ouabimedcha...@gmail.com * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] wct4xxp Interrupts not detected with dahdi 2.6, but working ok with 2.5
Hi, I've tried upgrading one of my servers with yum update to the latest dahdi/asterisk, and found out that my 4th gen TE410P is failing the dahdi init with Running dahdi_cfg: DAHDI startup failed: Input/output error Rolling back to 2.5 restores the normal operation, and reading the dahdi 2.6 change log I think I'm hitting this bug fix with my mobo/card combo? 2011-12-14 19:02 + [r10379-10380] Shaun Ruffell sruff...@digium.com With dahdi 2.6 I'm getting this: #cat /proc/interrupts 209: 1 0 IO-APIC-level wct4xxp No interrupts?! #dmesg kernel: ACPI: PCI Interrupt :02:01.0[A] - GSI 24 (level, low) - IRQ 209 kernel: wct4xxp :02:01.0: Firmware Version: c01a016c kernel: wct4xxp :02:01.0: FALC Framer Version: 2.1 or earlier kernel: wct4xxp :02:01.0: Found a Wildcard: Wildcard TE410P (4th Gen) kernel: VPM450: echo cancellation for 128 channels kernel: wct4xxp :02:01.0: VPM450: hardware DTMF disabled. kernel: wct4xxp :02:01.0: VPM450: Present and operational servicing 4 span(s) kernel: wct4xxp :02:01.0: TE4XXP: Span 1 configured for CCS/HDB3/CRC4 kernel: wct4xxp :02:01.0: RCLK source set to span 1 kernel: wct4xxp :02:01.0: System timing mode, RCLK set to span 1 kernel: wct4xxp :02:01.0: TE4XXP: Span 2 configured for CCS/HDB3/CRC4 kernel: wct4xxp :02:01.0: RCLK source set to span 1 kernel: wct4xxp :02:01.0: System timing mode, RCLK set to span 1 kernel: wct4xxp :02:01.0: TE4XXP: Span 3 configured for CCS/HDB3/CRC4 kernel: wct4xxp :02:01.0: RCLK source set to span 3 kernel: wct4xxp :02:01.0: Recovered timing mode, RCLK set to span 3 kernel: wct4xxp :02:01.0: SPAN 3: Primary Sync Source kernel: wct4xxp :02:01.0: Interrupts not detected. With dahdi 2.5 everything is OK: #cat /proc/interrupts 201: 9157 962863 IO-APIC-level wct4xxp #dmesg kernel: ACPI: PCI Interrupt :02:01.0[A] - GSI 24 (level, low) - IRQ 201 kernel: wct4xxp :02:01.0: Found TE4XXP at base address f200, remapped to f887c000 kernel: wct4xxp :02:01.0: Firmware Version: c01a016c kernel: wct4xxp :02:01.0: Burst Mode: On kernel: wct4xxp :02:01.0: Octasic Optimizations: Enabled kernel: wct4xxp :02:01.0: FALC Framer Version: 2.1 or earlier kernel: wct4xxp :02:01.0: Board ID: 00 kernel: wct4xxp :02:01.0: Reg 0: 0x37554400 kernel: wct4xxp :02:01.0: Reg 1: 0x37554000 kernel: wct4xxp :02:01.0: Reg 2: 0x kernel: wct4xxp :02:01.0: Reg 3: 0x kernel: wct4xxp :02:01.0: Reg 4: 0x3101 kernel: wct4xxp :02:01.0: Reg 5: 0x kernel: wct4xxp :02:01.0: Reg 6: 0xc01a016c kernel: wct4xxp :02:01.0: Reg 7: 0x1f00 kernel: wct4xxp :02:01.0: Reg 8: 0x kernel: wct4xxp :02:01.0: Reg 9: 0x00ff0031 kernel: wct4xxp :02:01.0: Reg 10: 0x004a kernel: wct4xxp :02:01.0: Found a Wildcard: Wildcard TE410P (4th Gen) [snip] wct4xxp :02:01.0: TE4XXP: Span 1 configured for CCS/HDB3/CRC4 wct4xxp :02:01.0: 2G: Got interrupt, status = 010c, CIS = 0080 wct4xxp :02:01.0: RCLK source set to span 1 wct4xxp :02:01.0: System timing mode, RCLK set to span 1 wct4xxp :02:01.0: TE4XXP: Span 2 configured for CCS/HDB3/CRC4 wct4xxp :02:01.0: 2G: Got interrupt, status = 010c, CIS = 0080 wct4xxp :02:01.0: RCLK source set to span 1 wct4xxp :02:01.0: System timing mode, RCLK set to span 1 wct4xxp :02:01.0: TE4XXP: Span 3 configured for CCS/HDB3/CRC4 wct4xxp :02:01.0: 2G: Got interrupt, status = 010d, CIS = 0081 wct4xxp :02:01.0: RCLK source set to span 3 wct4xxp :02:01.0: Recovered timing mode, RCLK set to span 3 wct4xxp :02:01.0: 2G: Got interrupt, status = 010a, CIS = 0080 wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: SPAN 3: Primary Sync Source wct4xxp :02:01.0: TE4XXP: Span 4 configured for CCS/HDB3/CRC4 wct4xxp :02:01.0: 2G: Got interrupt, status = 000d, CIS = 0084 wct4xxp :02:01.0: RCLK source set to span 3 wct4xxp :02:01.0: Recovered timing mode, RCLK set to span 3 wct4xxp :02:01.0: 2G: Got interrupt, status = 000b, CIS = 0088 wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got interrupt, status = 000b, CIS = 008a wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got interrupt, status = 000a, CIS = 0080 wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got interrupt, status = 000b, CIS = 0085 wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got interrupt, status = 000a, CIS = 0080 wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got interrupt, status = 000b, CIS = 008a wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got interrupt, status = 000a, CIS = 0080 wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got interrupt, status = 000a, CIS = 0080 wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got
[asterisk-users] Ulaw
List, I am getting this warning message in cli. How to fix this problem. *translate.c:163 framein: no samples for ulawtolin ** *Best Regards, Mahesh Katta http://www.buzzworks.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
