Re: [asterisk-users] Asterisk 1.6.2.0 Now Available!
At 11:39 PM 12/18/2009, you wrote: I know it's not free, I want it for 1.6.2 which I've been running for quite a while and it's not there for download yet. I've been running SFA on 1.6.2.x-rcX for quite some time with no stability issues at all. Did you give it a try? What version of SFA? I thought I tried to install the early release and it said it wouldn't work. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HOW to record saynumber output
ok thanks a lot Danny. for your helpfull advice. your are with Steve my technical guardian angels :) 2009/12/18 Danny Nicholas da...@debsinc.com Saynumber just does an “EXECUTE BACKGROUND” on the files in /var/lib/asterisk/sounds/digits. So to “record” a saynumber output of 23 to a moh file, you would do sox /var/lib/asterisk/sounds/digits/20.gsm /var/lib/asterisk/sounds/digits/3.gsm /var/lib/asterisk/moh/23.wav. If your moh processes randomly, the 23 would come up every x times. If you use classes to control moh, you can make it come up each time. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *mickael ropars *Sent:* Friday, December 18, 2009 11:30 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] HOW to record saynumber output Hi Danny, I've already have a look to those tools, but I cannot see how I can record the saynumber output audio into a file Mickael 2009/12/18 Danny Nicholas da...@debsinc.com If you have SOX, LAME and time, you can do about anything you want. The default moh files are wav, but a lot of folks use mp3 with the mpg123 player. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *mickael ropars *Sent:* Friday, December 18, 2009 11:05 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] HOW to record saynumber output Hi all, the aims of this mail is to use saynumber fonctionality during Music On Hold while dialing. Music On Hold can only play a music file So Is there a way to pre-record the saynumber output and other .gsm file and then play the record file during Music On Hold ? all solutions are welcome regards Mickael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nortel BCM - Call Accounting Interface?
Dear List, Need to know if anyone on this list has had any experience with using the Nortel BCM 50 for Call Account Reporting using an IP connection to a Linux / Asterisk interface? Presently, I have a BCM 50 installed that uses a local Lenova Small Form Factor PC with a windows XP / os that quit reporting because of a up-grade to the Nortel reporting software. Nortel support is now telling me that my PC needs to be up-graded for it to work with the newly patched reporting software. Nortel as far as I know is a Windows only shop and uses IIS and a utility to pull the data stream out of the BCM and into the local pc that uses MSSql. What I'm looking to accomplish is use a virtual machine running * and MySql to pull the call data. I'm aware that Nortel use proprietary sw, but was wondering if any Nortel Guru on the list has had any luck using * and a 3rd party call accounting software to share? Thanks for any help, suggestions or directions on where to find help you can provide. Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.0 Now Available!
Ira schrieb: At 11:39 PM 12/18/2009, you wrote: I know it's not free, I want it for 1.6.2 which I've been running for quite a while and it's not there for download yet. I've been running SFA on 1.6.2.x-rcX for quite some time with no stability issues at all. Did you give it a try? What version of SFA? I thought I tried to install the early release and it said it wouldn't work. asterisk0*CLI core show version Asterisk 1.6.2.0~rc7-1 built by root @ asterisk0 on a x86_64 running Linux on 2009-12-09 11:24:08 UTC asterisk0*CLI skype show version Skype For Asterisk Components: Channel Driver: 1.6.1_1.0.6 Library: 1.6.1_1.0.6 asterisk0*CLI hth, Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Connect my Asterisk to external sip?
Dear All
I have an application that calls for my Asterisk sip to be connected to an
external sip server for voip routing . Please be informed that my Asterisk
sip is at @192.168.0.2 and the external sip is at @192.168.0.139 . To this
end , I modified my sip.conf extensions.conf as the followings :
Under sip.conf :
-
[general]
register = toronto:welc...@192.168.0.139/osaka
[osaka]
type=friend
secret=welcome
context=osaka_incoming
host=dynamic
disallow=all
allow=alaw
[6672019]
type=friend
host=dynamic
context=phones
Under extensions.conf :
-
[osaka_incoming]
include=local-lines
[local-lines]
exten = _XXX,n,Dial(SIP/osaka/${EXTEN})
Please find attached the log captured when making calls (the call cannot get
through) .Can you please do me favor and let me know what is wrong in my sip
configuration ?
Let me thank you in advance
log-sip
Description: Binary data
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Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @192.168.0.139 . To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E1 ingress to SIP egress problem with 183 response
Hi,
I've looked around the archives and have spent a while on voip-info.org
but not found an answer so forgive me if this is in a FAQ somewhere.
We've got several Asterisk servers with E1 cards (some Digium, some
Sangoma). We provide non geographic numbers for customers and route
calls to their existing phone numbers. Calls come in over the PSTN and
into Asterisk.
This works perfectly if we route calls out via the PSTN but it is
expensive, so we have been trying a different carrier where we egress to
them using SIP and let them break out to the PSTN at their end. However,
when using the SIP carrier, we've had complaints that the caller hears
the remote phone ring for about 15 seconds and if unanswered at this
point, the line hangs up.
The problem seems to be that when the remote phone starts ringing (the
caller can hear the ringing sound) the SIP carrier is sending back a 183
Session Progress SIP message but Asterisk isn't changing the state of
the incoming E1 channel to indicate this. If the called party doesn't
answer within 15 - 20 seconds, the calling equipment assumes that the
line is dead and requests a hangup.
If the called party answers the phone quite quickly, everything works.
If I route out to a different SIP gateway that sends a 180 Ringing back,
then this is OK - the call can ring for a long time and not cut off.
After the 180 Ringing message comes back, the incoming E1 gets a state
change to Call Received:
logger.c: q931.c:2802 q931_alerting: call 16123 on channel 2 enters state 7
(Call Received)
However when we only get a 183 back, this doesn't happen and is causing
us a problem.
I would prefer to solve the problem by changing a configuration option
somewhere but I'm running out of ideas.
I've had a look in chan_sip.c and have seen this:
case 180: /* 180 Ringing */
case 182: /* 182 Queued */
if (!ast_test_flag(req, SIP_PKT_IGNORE) (p-invitestate !=
INV_CANCELLED) sip_cancel_destroy(p))
ast_log(LOG_WARNING, Unable to cancel SIP destruction.
Expect bad things.\n);
if (!ast_test_flag(req, SIP_PKT_IGNORE) p-owner) {
ast_queue_control(p-owner, AST_CONTROL_RINGING);
if (p-owner-_state != AST_STATE_UP) {
ast_setstate(p-owner, AST_STATE_RINGING);
}
}
if (find_sdp(req)) {
if (p-invitestate != INV_CANCELLED)
p-invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) p-owner) {
/* Queue a progress frame only if we have SDP
in 180 or 182 */
ast_queue_control(p-owner,
AST_CONTROL_PROGRESS);
}
}
check_pendings(p);
break;
case 183: /* Session progress */
if (!ast_test_flag(req, SIP_PKT_IGNORE) (p-invitestate !=
INV_CANCELLED) sip_cancel_destroy(p))
ast_log(LOG_WARNING, Unable to cancel SIP destruction.
Expect bad things.\n);
/* Ignore 183 Session progress without SDP */
if (find_sdp(req)) {
if (p-invitestate != INV_CANCELLED)
p-invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) p-owner) {
/* Queue a progress frame */
ast_queue_control(p-owner,
AST_CONTROL_PROGRESS);
}
}
check_pendings(p);
break;
I wondered if I made case 183 be the same as cases 180 and 182 whether
it would solve this problem and if so, whether that would be OK or
whether it might cause other problems.
Or is there a better way to make it work with a 183 response?
For testing, if I specify the r option to the Dial command to tell
Asterisk to generate its own ringing tone, this solves the hangup
problem as well, though we can't use this option because of the other
potentially useful tones it masks. When using the r option, we get
these Q.931 log entries:
logger.c: q931.c:3509 q931_receive: call 22267 on channel 27 enters state 6
(Call Present)
logger.c: q931.c:2774 q931_call_proceeding: call 22267 on channel 27 enters
state 9 (Incoming Call Proceeding)
This is on Asterisk 1.4.22.
--
Cheers,
Kingsley.
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Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @192.168.0.139. To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing for incoming call
I have a strange suggestion -- have one extension answer the call and dial the extension you want -- then it should ring before dialing the second one. Bob Smither smit...@c-c-i.com wrote: On Fri, 2009-12-18 at 19:58 -0600, Steve Johnson wrote: Try putting the wait before the Answer. ... exten = s,n,Wait(10) exten = s,n,Answer ... Thanks Steve. I tried that: On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither smit...@c-c-i.com wrote: Dear All, snip ... exten = s,n,Answer exten = s,n,Ringing() exten = s,n,Wait(10) exten = s,n,BackGround(sound file) ... I have also tried moving the Answer app to right before the BackGround app. snip i.e., after the Wait, but still no joy. Anything else I need to look at? Thanks, -- Bob Smither smit...@c-c-i.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @192.168.0.139 . To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance [general] ;register = toronto:welc...@osaka [osaka] type=friend ;username=toronto ;authname=toronto ;secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing for incoming call
On Fri, 2009-12-18 at 23:56 -0600, Steve Johnson wrote: If you try just this, what does the caller hear? It should be ringing for the first 20 sec, and then maybe the congestion tone afterwards. exten = s,1,Wait(20) exten = s,n,Hangup Dialplan: [cci] exten = s,1,Wait(10) exten = s,n,Hangup() When the number is dialed, here is the CLI output: Connected to Asterisk 1.4.21.1 currently running on k6-2 (pid = 6283) Verbosity was 0 and is now 3 -- Executing [8772709...@inbound:1] Goto(SIP/smither-03390860, cci|s|1) in new stack -- Goto (cci,s,1) -- Executing [...@cci:1] Wait(SIP/smither-03390860, 10) in new stack -- Executing [...@cci:2] Hangup(SIP/smither-03390860, ) in new stack == Spawn extension (cci, s, 2) exited non-zero on 'SIP/smither-03390860' The caller hears silence for 10 seconds. When the Hangup is executed, as reported on the CLI, the caller _then_ hears ringing (!?) which continues until the caller hangs up. Here is the entry in sip.conf (Asterisk registers with the provider): [vitel-inbound-cci] type=friend dtmfmode=auto host=provider host context=inbound username=user name secret=my secret allow=all insecure=very nat=yes Context in extensions.conf: [inbound] exten = 8772709688,1,Goto(cci,s,1) The context [cci] is shown above. I appreciate the help, as I am confused! -- Bob Smither, PhD Circuit Concepts, Inc. = There are only 10 kinds of people in the world --Those who understand binary, and those who don't... = smit...@c-c-i.com http://www.C-C-I.Com 281-331-2744(office) -4616(fax) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting the phone number an SIP extention is dialing
This is the first time i face this issue.. i have an extension 100 .. calling 0018001234567 is there a way in Asterisk to get info that 100 is calling that number? sorry for the lame question but i never had to know such info on my system. -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ Hotmail: Powerful Free email with security by Microsoft. http://clk.atdmt.com/GBL/go/171222986/direct/01/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PAP2 Dialing Delay
Possibly OT? I've hooked up a Linksys PAP2 to my Asterisk 1.6 fairly painlessly. The only issue I can't beat with it is the dial delay when calling internal or external numbers. No matter what it seems to take 10 -15 seconds to actually dial. I've altered the device removing all *xx combos and unnecessary waffle and cut the dialplan string to (x.S0) but the problem persists. Anyone else seen this issue? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing for incoming call
On Sat, 2009-12-19 at 08:26 -0500, cov...@ccs.covici.com wrote:
I have a strange suggestion -- have one extension answer the call and
dial the extension you want -- then it should ring before dialing the
second one.
Actually, that is pretty close to what I do on a *1.6 box and it works.
Here's what I tried on my *1.4 box (in extensions.conf):
[inbound]
exten = 8772709688,1,Dial(Local/s...@cci,15)
exten = 8772709688,n,Hangup()
[cci]
exten = s,1,Set(CallerContext=${CONTEXT}) ; capture context
; document time of call to console
exten = s,n,NoOp(Time is: ${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
; document caller id to console
exten = s,n,NoOp(CallerID is ${CALLERID(all)})
exten = s,n,Set(TIMEOUT(digit)=3) ; Set Digit Timeout
exten = s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout
; create unique call id for this call
exten = s,n,Set(GLOBAL(cid)=${EPOCH})
;
;exten = s,n,Playtones(ring)
exten = s,n,Wait(10)
;exten = s,n,StopPlaytones()
exten = s,n,Answer()
exten = s,n(start),Wait(0.5)
exten = s,n,BackGround(cci/prompt00)
exten = s,n,WaitExten ; Wait for an extension to be dialed.
I tried both with and without the Playtones(ring) / StopPlaytones()
lines.
Here is what I get from the CLI:
Connected to Asterisk 1.4.21.1 currently running on k6-2 (pid = 8998)
Verbosity was 0 and is now 3
-- Executing [8772709...@inbound:1] Dial(SIP/smither-173b4940,
Local/s...@cci|15) in new stack
-- Called s...@cci
-- Executing [...@cci:1] Set(Local/s...@cci-7c61,2,
CallerContext=cci) in new stack
-- Executing [...@cci:2] NoOp(Local/s...@cci-7c61,2, Time is:
2009-12-19 09:43:10) in new stack
-- Executing [...@cci:3] NoOp(Local/s...@cci-7c61,2, CallerID is
* *) in new stack
-- Executing [...@cci:4] Set(Local/s...@cci-7c61,2, TIMEOUT(digit)=3)
in new stack
-- Digit timeout set to 3
-- Executing [...@cci:5] Set(Local/s...@cci-7c61,2,
TIMEOUT(response)=10) in new stack
-- Response timeout set to 10
-- Executing [...@cci:6] Set(Local/s...@cci-7c61,2,
GLOBAL(cid)=1261237390) in new stack
== Setting global variable 'cid' to '1261237390'
-- Executing [...@cci:7] PlayTones(Local/s...@cci-7c61,2, ring) in
new stack
-- Executing [...@cci:8] Wait(Local/s...@cci-7c61,2, 10) in new stack
-- Executing [...@cci:9] StopPlayTones(Local/s...@cci-7c61,2, ) in
new stack
-- Executing [...@cci:10] Answer(Local/s...@cci-7c61,2, ) in new
stack
-- Executing [...@cci:11] Wait(Local/s...@cci-7c61,2, 0.5) in new
stack
-- Local/s...@cci-7c61,1 answered SIP/smither-173b4940
-- Executing [...@cci:12] BackGround(Local/s...@cci-7c61,2,
cci/prompt00) in
new stack
-- Local/s...@cci-7c61,2 Playing 'cci/prompt00' (language 'en')
This all looks as expected to me, but the caller hears nothing until the
BackGround statement is executed. There still is no ringing back to the
caller.
Thanks!
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[asterisk-users] sendmail
Anyone have a cookbook on configuring sendmail to work with Asterisk? Or,a few config examples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sendmail
You do not need any special sendmail options/settings/configuration to use it with Asterisk. If you are talking about sending voicemail notices then you just need to point mailcmd in voicemail.conf to a command that can send the message. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 19, 2009, at 8:20 AM, Thomas Perron wrote: Anyone have a cookbook on configuring sendmail to work with Asterisk? Or,a few config examples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sendmail
On Sat, 19 Dec 2009, Thomas Perron wrote: Anyone have a cookbook on configuring sendmail to work with Asterisk? Or,a few config examples. You don't configure sendmail to work with Asterisk, you configure Asterisk to work with sendmail. What are you trying to accomplish? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
