Re: [asterisk-users] Question about null routing calls to DIDs we don't handle
Why not setup a default catch-all route that goes to either your main line (to
drive sales) or a pre-recorded message (the number you dialed is
disconnected...etc), and then setup more specific pattern matches for assigned
numbers? I've done this before for clients that have large blocks of did's
assigned to them but only a small number of extensions that need direct dial
capabilities.
Thanks,
--Warren Selby, dCAP
On Jun 3, 2011, at 2:34 PM, Jesse Thompson jes...@gmail.com wrote:
(reposted with correct subject line, I think messing up the subject
line last time prevented my question from being read. Cheers :)
On Thu, Jun 2, 2011 at 12:27 PM, Jesse Thompson jes...@gmail.com wrote:
Letting a carrier use you as a carrier seems like quite a bad idea
generally..
I think I would agree. :)
_NXXNXX = Dial(SIP/${EXTEN}@upstream,120); // numbers not handled here
get routed upstream
in the 'local' context instead of the other one?
So here is where the finer points of Asterisk pattern matching must
come into play.
All of the customer DID's match the pattern _NXXNXX. If we put
that pattern in the local context, then wouldn't that mean that calls
from a local customer to another local customer would match the
_NXXNXX pattern before even trying to match against the specific
patterns in the clients context? We need to be able to route
local-to-local calls without using two trunks to go back and forth
through the upstream provider.
Thank you for your input. I know this is a problem most operators can
get past, so there's got to be just something not lining up quite
right in my mental model. :)
- - Jesse
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Re: [asterisk-users] Question about null routing calls to DIDs we don't handle
On 1 Jun 2011, at 22:50, Jesse Thompson wrote:
We are managing an Asterisk installation for residential VOIP service, and we
are having a problem where all inbound calls to DIDs which are assigned to us
by our wholesaler but not yet assigned to a downstream customer get caught in
a routing loop.
Put this line:
_NXXNXX = Dial(SIP/${EXTEN}@upstream,120); // numbers not handled here get
routed upstream
in the 'local' context instead of the other one? Letting a carrier use you
as a carrier seems like quite a bad idea generally..
S
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[asterisk-users] Question about null routing calls to DIDs we don't handle
Hello, this is Jesse with Webformix.
We are managing an Asterisk installation for residential VOIP service, and
we are having a problem where all inbound calls to DIDs which are assigned
to us by our wholesaler but not yet assigned to a downstream customer get
caught in a routing loop.
** For example: we get assigned DID block 1230-1239 and only 1230-1233 are
given to our customers, then our routing logic sends inbound calls to 1234
back to upstream, which bounce back to us again, tying up all our trunks
before emitting an error.
When we recognized the problem I automatically added ael commands to route
all known unassigned numbers to a friendly error message, but that is a
clumsy approach and the larger issue remains that if we get routed a call
for a number we didn't expect — due to provisioning mistake, premature
porting, or mis-routing at our wholesaler — we still need to avoid a routing
loop.
** For example, we have block 1230-1239, and upstream sends us a call for
4321 for no reason at all.. we're not specifically failing that DID so we
still get a routing loop.
What I would like is to somehow identify all inbound calls (originating from
upstream) that are not terminated inside our network, and give them the
friendly error. Calls originating from our customers should get routed
upstream, but calls originating upstream should not bounce back upstream
without an intervening new Dial command.
** For example: we want...
upstream calls customer @ 1230, rings customer
upstream sends us call for 4321, we play an error
customer calls customer @ 1230, rings customer without bothering upstream
customer calls 4321, call bubbles upstream.
anyone calls customer @ 1233, which has unconditional forwarding set to
4322. Via new Dial command, call should route upstream.
The trouble is that, to our knowledge, all calls from local clients and all
calls we get from upstream have to pass through a context we call clients
that routes calls to specific local clients. Local calls can't be sent
upstream until after they've run this gauntlet, and inbound calls can't be
failed until they have been matched against same patterns, but after those
matches it's no longer clear how to separate the remotely sourced calls into
an Unavailable() bin and send the locally sourced calls upstream.
Here is a simplified version of our configs to give you an idea of the tack
we are presently taking.
Thanks guys! :)
- - Jesse Thompson
Webformix Telephone Services
== sip.conf
[general]
context=clients; default context for all calls
register = skd...@peer.upstreamvoip.com
[upstream]
type=peer
host=peer.upstreamVOIP.com
username=dfjhjkb
secret=redacted
context=clients; context for all inbound calls
call-limit=8
[residential]
type=friend
host=dynamic
context=local
(SIPcust1)[residentia] ; SIP customer name
username=SIPcust1
mailbox=SIPcust1
secret=redacted
; and other sip customers
== extentions.ael
context local
{
// Does some local cleanup, strips leading 1 off number, _N11, star codes,
fun stuff
includes { clients; };
}
context clients
{
custDID1 = Dial(SIP/SIPcust1,35);
custDID2 = Dial(SIP/SIPcust2,35);
custDID3 = Dial(SIP/UNCforwardNumber@upstream,120); // inbound calls to
forwarding customers should route back upstream
unassignedDID = Unassigned();
_NXXNXX = Dial(SIP/${EXTEN}@upstream,120); // numbers not handled
here get routed upstream
// ... which is a problem if those calls originally came *from* upstream.
:(
}
macro Unassigned() // Audio message played for unavailable numbers, not sure
if there's a machine-friendlier error approach or not. :)
{
PlayBack(ro_sit);
PlayBack(unassigned);
PlayBack(5413);
PlayBack(ro_sit);
PlayBack(unassigned);
PlayBack(5413);
HangUp();
}
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[asterisk-users] Question on AMI
I am using asterisk 1.4.41 and the AMI I am trying to execute a command over AMI, specifically core show channels concise sometimes I get this back: asterisk_command_show_channels() execute failed. 'Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ]OutgoingSpoolFailed!smvoice-dialout!failed!1!Down!AGI!smvoice|-digium_failed!!!3!0!(None)[LF ]' I'm not expecting to see that... My manager.conf file section is: [] secret=YES permit=127.0.0.1/255.255.255.0 read = system,call,command,agent,user write = system,call,command,agent,user,originate ;read = system,call,log,verbose,command,agent,user ;write = system,call,log,verbose,command,agent,user Is there something not configured right? At the time I get the above error I had PREVIOUSLY tryied to dispatch a call to an invalid extension. Which is correct (I am testing that). however I am not expecting to see the above error over the interface when I am asking for show channels. Certainly the AGI smvoice will execute about that is fine. I just dont want to see that particular message come over the AMI when I am asking for show channels result. How do I disable that? Thanks, Jerry -- _ -- 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] Question on AMI
Jerry Geis wrote: I am using asterisk 1.4.41 and the AMI I am trying to execute a command over AMI, specifically core show channels concise sometimes I get this back: asterisk_command_show_channels() execute failed. 'Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ]OutgoingSpoolFailed!smvoice-dialout!failed!1!Down!AGI!smvoice|-digium_failed!!!3!0!(None)[LF ]' I'm not expecting to see that... My manager.conf file section is: [] secret=YES permit=127.0.0.1/255.255.255.0 read = system,call,command,agent,user write = system,call,command,agent,user,originate ;read = system,call,log,verbose,command,agent,user ;write = system,call,log,verbose,command,agent,user Is there something not configured right? At the time I get the above error I had PREVIOUSLY tryied to dispatch a call to an invalid extension. Which is correct (I am testing that). however I am not expecting to see the above error over the interface when I am asking for show channels. Certainly the AGI smvoice will execute about that is fine. I just dont want to see that particular message come over the AMI when I am asking for show channels result. How do I disable that? Thanks, Jerry One thing I forgot to add was that when I login the AMI I have Events: off. jerry -- _ -- 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] question on digium repo
I an running centos 5. I added this to the digium.repo file in /etc/yum.repos.d directory. [digium-current] name=CentOS-$releasever - Digium - Current baseurl=http://packages.digium.com/centos/$releasever/current/$basearch/ enabled=1 gpgcheck=0 #gpgkey=http://packages.digium.com/RPM-GPG-KEY-Digium Then I did yum install asterisk14 addons | 951 B 00:00 base | 2.1 kB 00:00 base/primary_db | 2.2 MB 00:03 digium-current | 1.1 kB 00:00 digium-current/primary | 33 kB 00:00 digium-current 260/260 extras | 2.1 kB 00:00 extras/primary_db | 244 kB 00:00 updates | 1.9 kB 00:00 updates/primary_db | 544 kB 00:01 Setting up Install Process No package asterisk14 available. What did I miss? jerry -- _ -- 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] question on digium repo
On 05/16/2011 08:36 AM, Jerry Geis wrote: I an running centos 5. I added this to the digium.repo file in /etc/yum.repos.d directory. [digium-current] name=CentOS-$releasever - Digium - Current baseurl=http://packages.digium.com/centos/$releasever/current/$basearch/ enabled=1 gpgcheck=0 #gpgkey=http://packages.digium.com/RPM-GPG-KEY-Digium Then I did yum install asterisk14 addons | 951 B 00:00 base | 2.1 kB 00:00 base/primary_db | 2.2 MB 00:03 digium-current | 1.1 kB 00:00 digium-current/primary | 33 kB 00:00 digium-current 260/260 extras | 2.1 kB 00:00 extras/primary_db | 244 kB 00:00 updates | 1.9 kB 00:00 updates/primary_db | 544 kB 00:01 Setting up Install Process No package asterisk14 available. What did I miss? jerry You missed the Asterisk repo. Replace all instances of digium.com with asterisk.org (and then Digium with Asterisk). packages.digium.com is Digium modules, such as FaxForAsterisk, whereas packages.asterisk.org is Asterisk, DAHDI, libpri, etc. -- _ -- 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] question on ways to activate voicemail light on polycom
On 11-05-06 02:56 PM, Watkins, Bradley wrote: Yes, use the MinivmMWI application. That's how I've done it in the past as well. Leif. -- _ -- 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] question on ways to activate voicemail light on polycom
Jerry Geis wrote: Is there a way in asterisk to Activate/Clear the blinking light on polycom phones I've always just dumped a msg000.txt in the voicemail directory of that phone and removed it when not needed. Under 1.4, the Polycoms act on it. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] question on ways to activate voicemail light on polycom
Is there a way in asterisk to Activate/Clear the blinking light on polycom phones indicating VM. Either from an AGI or some way in the dialplan? I want to be able to control this light for from my application. I have an AGI to do something similiar to VM and want to light /clear the light myself. Thanks, Jerry -- _ -- 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] question on ways to activate voicemail light on polycom
On Fri, May 6, 2011 at 2:14 PM, Jerry Geis ge...@pagestation.com wrote: Is there a way in asterisk to Activate/Clear the blinking light on polycom phones indicating VM. Either from an AGI or some way in the dialplan? I want to be able to control this light for from my application. I have an AGI to do something similiar to VM and want to light /clear the light myself. Thanks, Jerry I don't think there is a way to do it natively inside of asterisk, but I control it from a shell script. The shell script parses the output of sip show peers, crafts an application/simple-message-summary SIP message and then uses netcat to send it to the appropriate IP address / port. -M -- _ -- 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] question on ways to activate voicemail light on polycom
The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Friday, May 06, 2011 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] question on ways to activate voicemail light on polycom Is there a way in asterisk to Activate/Clear the blinking light on polycom phones indicating VM. Either from an AGI or some way in the dialplan? I want to be able to control this light for from my application. I have an AGI to do something similiar to VM and want to light /clear the light myself. Thanks, Jerry Yes, use the MinivmMWI application. - Brad -- _ -- 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] question on register and dnsmgr_lookup
I thought I has everything using IP addresses. I am not making outside calls this is all internal. I have a connection between two machines both running asterisk. I am using 1.8.3 and I see a lot of dnsmgr_lookup's for mymachine. I have a register line in sip.conf that is the only place mymachine is referenced. the actual definition for host= is the IP address. the register line is something like: register = jerry_asterisk:password@mymachine I presume the mymachine is causing the dnsmgr_lookup's Can I just replace the @mymachine with the machines IP address? Will this work for 1.8 and 1.4? Thanks, Jerry -- _ -- 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] Question About Codecs
On Wednesday 06 April 2011 09:49:17 Jon Farmer wrote: Hi I have a call into a MeetMe conference that when I do a core show channel returns NativeFormats: 0x4 (ulaw) WriteFormat: 0x1000 (g722) ReadFormat: 0x1000 (g722) Can someone explain what the differences between Native, Wite and Read are? Your native format is the format that the phone actually uses (on the wire). The read and write formats are what Asterisk expects to send to and receive from the application, because Asterisk has set up a translation path to ensure that the application gets a format that is more conducive to its purpose. Internally to Asterisk, when you ast_read() a frame from the channel, you should expect that, when the frame is a voice frame, the frame will be in the ReadFormat. And, when you ast_write() a voice frame to that channel, it should be in the WriteFormat. -- Tilghman -- _ -- 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] Question on Asterisk 1.8 and Wait()
When I switched to 1.8 from 1.4 I am getting this error pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension (default, s, 1) http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands This page says its in 1.0 and I dont think has been removed. Did I do something wrong? Everything else seems to be ok. Thanks, Jerry -- _ -- 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] Question on Asterisk 1.8 and Wait()
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Wednesday, March 02, 2011 3:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question on Asterisk 1.8 and Wait() When I switched to 1.8 from 1.4 I am getting this error pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension (default, s, 1) http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+c ommands This page says its in 1.0 and I dont think has been removed. Did I do something wrong? Everything else seems to be ok. Thanks, Jerry Best guess is that syntax changed from 1.4 to 1.8. Change line to Exten = s,1,Wait(1) -- _ -- 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] Question on Asterisk 1.8 and Wait()
On Wed, 2011-03-02 at 16:33 -0500, Jerry Geis wrote: When I switched to 1.8 from 1.4 I am getting this error pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension (default, s, 1) http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands This page says its in 1.0 and I dont think has been removed. Did I do something wrong? Everything else seems to be ok. Thanks, Jerry Could you post the dialplan usage? Maybe you have a typo. It should be something like: exten = s,1,Wait(1) https://wiki.asterisk.org/wiki/display/AST/Application_Wait -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] Question on Asterisk 1.8 and Wait()
Best guess is that syntax changed from 1.4 to 1.8. Change line to Exten = s,1,Wait(1) Danny Your correct. it was a syntax change. the above works. jerry -- _ -- 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] Question on Asterisk 1.8 and Wait()
Jerry Geis wrote: Your correct. it was a syntax change. the above works. I've always used Wait(#) in my 1.4.x dial plans. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Question about how traffic passes from phones
Hopefully this is a simple question. How does a non-secure phone that is on a PBX connected to an asterisk over a SIP trunk communicate with a secure phone connected to the Asterisk server? I like to think that the secure call terminates on the Asterisk and the non-secure call is somehow joined to that call on the Asterisk. Same with the voice traffic, I'm once again thinking that the Asterisk proxies the calls. Just not sure. Thanks, Mitch -- _ -- 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] Question about how traffic passes from phones
On Mon, Feb 28, 2011 at 8:49 PM, Mitch Johnson mitch.johns...@gmail.comwrote: How does a non-secure phone that is on a PBX connected to an asterisk over a SIP trunk communicate with a secure phone connected to the Asterisk server? I think, although I'm not positive, that if either leg of the call doesn't offer SRTP, then there won't be any SRTP in the call path. -- Thanks, --Warren Selby, dCAP 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] Question about EuroBRI final 2 digits
On 2/10/11 5:54 AM, Christian Gansberger christian.gansber...@accm.at wrote: Hello, Maybe try that: In your incoming isdn context: [isdn-incoming] exten = s,1,Set(TIMEOUT(digits)=3) exten = s,2,WaitExten(2) exten = s,3,Dial(SIP/operator...) exten = 10,1,Dial(SIP/10) exten = 20,1,Dial(SIP/20) So if a call comes in Asterisk waits, 2 seconds for further digits dialed and if so jumps to the right extension in the context. Overlapdial should be yes. yours christian gansberger www.accm.at Many thanks for this idea, Christian I have put this equivalent into the dialplan And when the Austria team gets to the office in the morning they will test it. (BTW changed TIMEOUT(digits) to TIMEOUT(digit)). Cassius On 3 February 2011 20:45, Cassius Smith cass...@cassius.org wrote: Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call before they finish dialing all 8 digits. Is there a way to prevent this, or to catch the additional 2 digits somewhere in the stream? The receptionist is unhappy because she gets all the 6-digit calls and must then transfer. Is this a p2p vs p2mp issue? Thanks in advance, Cassius Smith -- _ -- 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] Question about EuroBRI final 2 digits
This sounds like a job for DISA. http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA to see if DISA helps. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: 03 February 2011 19:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question about EuroBRI final 2 digits Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call before they finish dialing all 8 digits. Is there a way to prevent this, or to catch the additional 2 digits somewhere in the stream? The receptionist is unhappy because she gets all the 6-digit calls and must then transfer. Is this a p2p vs p2mp issue? Thanks in advance, Cassius Smith If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Question about EuroBRI final 2 digits
Hello, Maybe try that: In your incoming isdn context: [isdn-incoming] exten = s,1,Set(TIMEOUT(digits)=3) exten = s,2,WaitExten(2) exten = s,3,Dial(SIP/operator...) exten = 10,1,Dial(SIP/10) exten = 20,1,Dial(SIP/20) So if a call comes in Asterisk waits, 2 seconds for further digits dialed and if so jumps to the right extension in the context. Overlapdial should be yes. yours christian gansberger www.accm.at On 3 February 2011 20:45, Cassius Smith cass...@cassius.org wrote: Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call before they finish dialing all 8 digits. Is there a way to prevent this, or to catch the additional 2 digits somewhere in the stream? The receptionist is unhappy because she gets all the 6-digit calls and must then transfer. Is this a p2p vs p2mp issue? Thanks in advance, Cassius Smith -- _ -- 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] Question about EuroBRI final 2 digits
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian
Gansberger
Sent: Thursday, February 10, 2011 5:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about EuroBRI final 2 digits
Hello,
Maybe try that:
In your incoming isdn context:
[isdn-incoming]
exten = s,1,Set(TIMEOUT(digits)=3)
exten = s,2,WaitExten(2)
exten = s,3,Dial(SIP/operator...)
exten = 10,1,Dial(SIP/10)
exten = 20,1,Dial(SIP/20)
So if a call comes in Asterisk waits, 2 seconds for further digits
dialed and if so jumps to the right extension in the context.
Overlapdial should be yes.
yours
christian gansberger
www.accm.at
On 3 February 2011 20:45, Cassius Smith cass...@cassius.org wrote:
Hello,
I have an installation in Austria; ISDN service provided by Austria
Telekom.
The main number of the service is 6 digits. Incoming calls may contain 2
additional digits, which I then use to route the call to the correct
extension. Telekom sends me all the digits.
My problem is that when someone tries to dial an 8 digit number to an
extension from an outside analog phone, AT sends the call before they
finish
dialing all 8 digits. Is there a way to prevent this, or to catch the
additional 2 digits somewhere in the stream? The receptionist is unhappy
because she gets all the 6-digit calls and must then transfer.
Is this a p2p vs p2mp issue?
Thanks in advance,
Cassius Smith
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You should be able to query the length of ${CALLERID(num)} and process the
full 8 digits that way. Telekom sends me all the digits tells me that the
number dialed to get to the extension arrives intact and that your dialplan
is truncating it.
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Re: [asterisk-users] Question about EuroBRI final 2 digits
On Thu, Feb 10, 2011 at 6:08 AM, Andrew Thomas a...@datavox.co.uk wrote: This sounds like a job for DISA. http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA to see if DISA helps. If OP is using Asterisk18, perhaps we should direct him to look here? https://wiki.asterisk.org/wiki/display/AST/Application_DISA cheers, -- -Bob -- _ -- 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] Question about EuroBRI final 2 digits
...or there :) Anyway AT sends the call before they finish dialling all 8 digits means that they don't send all the digits. Conflicting sentence in OP. Perhaps it would help if the OP could determine if AT actually send 6 or 8 digits in the signalling (I reckon it's 6). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Beers Sent: 10 February 2011 14:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about EuroBRI final 2 digits On Thu, Feb 10, 2011 at 6:08 AM, Andrew Thomas a...@datavox.co.uk wrote: This sounds like a job for DISA. http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA to see if DISA helps. If OP is using Asterisk18, perhaps we should direct him to look here? https://wiki.asterisk.org/wiki/display/AST/Application_DISA cheers, -- -Bob -- _ -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Question about EuroBRI final 2 digits
2011/2/5 Roberto Piola roberto.pi...@visiant.it In Italy, you must enable overlapdial=yes Is this relevant for incoming calls, as OP asked ? -- _ -- 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] Question about EuroBRI final 2 digits
In Italy, you must enable overlapdial=yes On Thu, Feb 3, 2011 at 7:45 PM, Cassius Smith cass...@cassius.org wrote: Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call before they finish dialing all 8 digits. Is there a way to prevent this, or to catch the additional 2 digits somewhere in the stream? The receptionist is unhappy because she gets all the 6-digit calls and must then transfer. Is this a p2p vs p2mp issue? -- _ -- 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] Question about EuroBRI final 2 digits
Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call before they finish dialing all 8 digits. Is there a way to prevent this, or to catch the additional 2 digits somewhere in the stream? The receptionist is unhappy because she gets all the 6-digit calls and must then transfer. Is this a p2p vs p2mp issue? Thanks in advance, Cassius Smith -- _ -- 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] Question About Conferencing Capabilities
Hi Siobhan, Asterisk is all capacity to work-on but you need to find out some way of handling conference system through WEB part , also one more thing on last point for switching between conference i am not much sure about it but i think it is possible if i will look into code implementation. regards dhaval On Tue, Jan 4, 2011 at 10:34 AM, Siobhan Hamilton siobhan.plugge...@gmail.com wrote: My company is building a VOIP application, and initially were just using a barebones OpenSIPS implementation to host one-on-one calls; however, we want to expand the functionality to conferencing (which, of course, OpenSIPS doesn't handle) and was looking into Asterisk (the other option being Freeswitch). I've been poring through the docs, and have even set up a test server myself, but there are some very specific things we are looking for that I can't figure out if Asterisk can do or not. We want to be able to do the following: - Create dynamic, on-the-fly conferences that can remain active even when initiating user leaves - Within a conference, give users the ability to mute and/or deaf individual users - Give users the ability to enter a whisper mode with another user - where they are holding a private conversation that can only be heard by the two of them ( It sounds like the Meetme module has a functionality like this, but it is a little vague in the documentation) - Allow users to be in two conferences at once; the user would most likely have one muted at any given time so as to hear the other one, but we want them to be able to switch back and forth easily Could anyone advise me on whether Asterisk can accomplish these needs, or perhaps what it might take to do so? We are not averse to doing some customization if we can find the people who know how to make it happen! Thanks, Siobhan Hamilton -- _ -- 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] Question About Conferencing Capabilities
Anyone else know about the holding concurrent conferences (and switching back and forth) issue ? Is it possible? And can you set up dynamic conferences that continue even when the initiator leaves? Thanks! On Tue, Jan 4, 2011 at 7:11 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.comwrote: Hi Siobhan, Asterisk is all capacity to work-on but you need to find out some way of handling conference system through WEB part , also one more thing on last point for switching between conference i am not much sure about it but i think it is possible if i will look into code implementation. regards dhaval On Tue, Jan 4, 2011 at 10:34 AM, Siobhan Hamilton siobhan.plugge...@gmail.com wrote: My company is building a VOIP application, and initially were just using a barebones OpenSIPS implementation to host one-on-one calls; however, we want to expand the functionality to conferencing (which, of course, OpenSIPS doesn't handle) and was looking into Asterisk (the other option being Freeswitch). I've been poring through the docs, and have even set up a test server myself, but there are some very specific things we are looking for that I can't figure out if Asterisk can do or not. We want to be able to do the following: - Create dynamic, on-the-fly conferences that can remain active even when initiating user leaves - Within a conference, give users the ability to mute and/or deaf individual users - Give users the ability to enter a whisper mode with another user - where they are holding a private conversation that can only be heard by the two of them ( It sounds like the Meetme module has a functionality like this, but it is a little vague in the documentation) - Allow users to be in two conferences at once; the user would most likely have one muted at any given time so as to hear the other one, but we want them to be able to switch back and forth easily Could anyone advise me on whether Asterisk can accomplish these needs, or perhaps what it might take to do so? We are not averse to doing some customization if we can find the people who know how to make it happen! Thanks, Siobhan Hamilton -- _ -- 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
[asterisk-users] Question About Conferencing Capabilities
My company is building a VOIP application, and initially were just using a barebones OpenSIPS implementation to host one-on-one calls; however, we want to expand the functionality to conferencing (which, of course, OpenSIPS doesn't handle) and was looking into Asterisk (the other option being Freeswitch). I've been poring through the docs, and have even set up a test server myself, but there are some very specific things we are looking for that I can't figure out if Asterisk can do or not. We want to be able to do the following: - Create dynamic, on-the-fly conferences that can remain active even when initiating user leaves - Within a conference, give users the ability to mute and/or deaf individual users - Give users the ability to enter a whisper mode with another user - where they are holding a private conversation that can only be heard by the two of them ( It sounds like the Meetme module has a functionality like this, but it is a little vague in the documentation) - Allow users to be in two conferences at once; the user would most likely have one muted at any given time so as to hear the other one, but we want them to be able to switch back and forth easily Could anyone advise me on whether Asterisk can accomplish these needs, or perhaps what it might take to do so? We are not averse to doing some customization if we can find the people who know how to make it happen! Thanks, Siobhan Hamilton -- _ -- 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] Question About Conferencing Capabilities
My company is building a VOIP application, and initially were just using a barebones OpenSIPS implementation to host one-on-one calls; however, we want to expand the functionality to conferencing (which, of course, OpenSIPS doesn't handle) and was looking into Asterisk (the other option being Freeswitch). I've been poring through the docs, and have even set up a test server myself, but there are some very specific things we are looking for that I can't figure out if Asterisk can do or not. We want to be able to do the following: - Create dynamic, on-the-fly conferences that can remain active even when initiating user leaves - Within a conference, give users the ability to mute and/or deaf individual users - Give users the ability to enter a whisper mode with another user - where they are holding a private conversation that can only be heard by the two of them ( It sounds like the Meetme module has a functionality like this, but it is a little vague in the documentation) - Allow users to be in two conferences at once; the user would most likely have one muted at any given time so as to hear the other one, but we want them to be able to switch back and forth easily Could anyone advise me on whether Asterisk can accomplish these needs, or perhaps what it might take to do so? We are not averse to doing some customization if we can find the people who know how to make it happen! Thanks, Siobhan Hamilton -- _ -- 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] question on asterisk 1.8 meetme
Currently using 1.4.X and looking to JUMP to 1.8 was reading the docs and have a question. in 1.4 I could do: /usr/sbin/asterisk -rx meetme to see all the current meetme's. I dont see what this is now in 1.8? Thanks Jerry -- _ -- 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] Question regarding SMS(), SMSQ, SMSC
Our SMS-gateway is not PSTN accessible. On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese l...@lcrcomputer.net wrote: AMARDEEP SINGH wrote: Hello All, Scenario: -We use asterisk as voicemail server for our cellular network. Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip. -Extensions in * are virtual, just for leaving and accessing voicemail. Requirement: Asterisk to send SMS to cell switch(running SMSC) on reception of new voicemail. Pointers required from Maillist users: -How can I make * talk to SMSC(ip address:port). -Anyone using similar topology? -there are not enough examples/man/maillist of using app_sms(), smsq. Thanks: -A qpage? -- _ -- 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] Question regarding SMS(), SMSQ, SMSC
Maybe you need to read the man page for qpage. The qpage client can send the page to an SNPP server over TCP/IP. Lyle AMARDEEP SINGH wrote: Our SMS-gateway is not PSTN accessible. On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese l...@lcrcomputer.net mailto:l...@lcrcomputer.net wrote: AMARDEEP SINGH wrote: Hello All, Scenario: -We use asterisk as voicemail server for our cellular network. Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip. -Extensions in * are virtual, just for leaving and accessing voicemail. Requirement: Asterisk to send SMS to cell switch(running SMSC) on reception of new voicemail. Pointers required from Maillist users: -How can I make * talk to SMSC(ip address:port). -Anyone using similar topology? -there are not enough examples/man/maillist of using app_sms(), smsq. Thanks: -A qpage? -- _ -- 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] Question regarding SMS(), SMSQ, SMSC
Do you have working script? On Fri, Jul 23, 2010 at 10:14 AM, Lyle Giese l...@lcrcomputer.net wrote: Maybe you need to read the man page for qpage. The qpage client can send the page to an SNPP server over TCP/IP. Lyle AMARDEEP SINGH wrote: Our SMS-gateway is not PSTN accessible. On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese l...@lcrcomputer.net mailto:l...@lcrcomputer.net wrote: AMARDEEP SINGH wrote: Hello All, Scenario: -We use asterisk as voicemail server for our cellular network. Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip. -Extensions in * are virtual, just for leaving and accessing voicemail. Requirement: Asterisk to send SMS to cell switch(running SMSC) on reception of new voicemail. Pointers required from Maillist users: -How can I make * talk to SMSC(ip address:port). -Anyone using similar topology? -there are not enough examples/man/maillist of using app_sms(), smsq. Thanks: -A qpage? -- _ -- 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] Question regarding SMS(), SMSQ, SMSC
qpage -s snppserver.example.com -p lyle -f lyle test page AMARDEEP SINGH wrote: Do you have working script? On Fri, Jul 23, 2010 at 10:14 AM, Lyle Giese l...@lcrcomputer.net mailto:l...@lcrcomputer.net wrote: Maybe you need to read the man page for qpage. The qpage client can send the page to an SNPP server over TCP/IP. Lyle AMARDEEP SINGH wrote: Our SMS-gateway is not PSTN accessible. On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese l...@lcrcomputer.net mailto:l...@lcrcomputer.net mailto:l...@lcrcomputer.net mailto:l...@lcrcomputer.net wrote: AMARDEEP SINGH wrote: Hello All, Scenario: -We use asterisk as voicemail server for our cellular network. Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip. -Extensions in * are virtual, just for leaving and accessing voicemail. Requirement: Asterisk to send SMS to cell switch(running SMSC) on reception of new voicemail. Pointers required from Maillist users: -How can I make * talk to SMSC(ip address:port). -Anyone using similar topology? -there are not enough examples/man/maillist of using app_sms(), smsq. Thanks: -A qpage? -- _ -- 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
[asterisk-users] Question regarding SMS(), SMSQ, SMSC
Hello All, Scenario: -We use asterisk as voicemail server for our cellular network. Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip. -Extensions in * are virtual, just for leaving and accessing voicemail. Requirement: Asterisk to send SMS to cell switch(running SMSC) on reception of new voicemail. Pointers required from Maillist users: -How can I make * talk to SMSC(ip address:port). -Anyone using similar topology? -there are not enough examples/man/maillist of using app_sms(), smsq. Thanks: -A -- _ -- 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] Question regarding SMS(), SMSQ, SMSC
AMARDEEP SINGH wrote: Hello All, Scenario: -We use asterisk as voicemail server for our cellular network. Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip. -Extensions in * are virtual, just for leaving and accessing voicemail. Requirement: Asterisk to send SMS to cell switch(running SMSC) on reception of new voicemail. Pointers required from Maillist users: -How can I make * talk to SMSC(ip address:port). -Anyone using similar topology? -there are not enough examples/man/maillist of using app_sms(), smsq. Thanks: -A qpage? -- _ -- 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] question on nortel sip connection
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Friday, June 18, 2010 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] question on nortel sip connection I am using asterisk 1.4.32 and wish to connect using SIP to a nortel 1000 switch with the ability to have 90 calls at a one time outgoing or incoming. the nortel reseller is asking me what to do. I dont know nortel at all. I thought I just needed a SIP trunk and IP address of the their server and an account name, and provide her my IP address. They didn't know what to do with that. What do I tell them? I've successfully set up SIP connectivity to a Nortel CS1000, but it required a SIP proxy in between. The major issue I came across is that Nortel (at least in Succession 4.0 and 4.5, not sure about later versions) uses the maddr URI parameter in an RFC-compliant but otherwise unseen (at least insofar as I've come across) way that Asterisk does not handle gracefully. In order for this to be successful, you'll definitely need to determine what version of Succession they're using and, if it's 5.0 or later, if they are using the newer COTS-based servers with the SIP proxy functionality. You'll probably still need your own proxy, but but some initial testing I did when I had the time indicated that some features (transfers, in particular) may work a lot better in the never version(s). You'll also need to figure out exactly what will be handled by the Asterisk system, because call routing can kind of weird with these boxes. At least in the older versions of Succession, they tended to treat SIP trunks as second-class citizens. As a result, you may end up needing to configure the Nortel to think of the Asterisk box as a trunk of last resort. One other thing: were you planning on using voicemail on the Nortel (i.e., CallPilot)? That *can* work if you want it to, but it's yet another can of worms in setting this up. Also, when I've done it in the past I have had precisely ZERO assistance from any Nortel reseller. So expect to end up learning far more about that side of this setup that you had wanted to. Feel free to ask questions about the particulars, but that's the quick lay of the land. Regards, - Brad -- _ -- 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] question on nortel sip connection
I am using asterisk 1.4.32 and wish to connect using SIP to a nortel 1000 switch with the ability to have 90 calls at a one time outgoing or incoming. the nortel reseller is asking me what to do. I dont know nortel at all. I thought I just needed a SIP trunk and IP address of the their server and an account name, and provide her my IP address. They didn't know what to do with that. What do I tell them? Thanks, Jerry -- _ -- 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] Question about 1.6: multiple IP on a single Asterisk box / multi ISP routing
Hi, I'm still on 1.4 and am wondering if 1.6 would fix an issue for me. Specifically, I have been given the impression that, in contrast to 1.4 which always sends packet from the default IP (if the server has multiple IPs), 1.6 sends packets back from the IP address that was used by the peers. i.e.: On a server with two NICs: eth0 192.168.1.2 eth1 192.168.1.3 If a peer registers to 192.168.1.3, he will get packets back from 192.168.1.3. If a peer uses 192.168.1.2 instead, he will get packets back from 192.168.1.2. .In effect, hiding the fact that this is the same system (And more to the point, allowing easy outgoing routing based on which NIC was used). Am I correct? Bonus question if I am indeed correct: how stable is 1.6 right now, compared to the latest 1.4 (1.4.31)? Mike -- _ -- 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] Question about MaxRetries in the Asterisk Outgoing folder
I'm doing some automated calling by putting .call files in the Outgoing folder of Asterisk. I'm concerned this might be a stupid question, but I'm pretty sure I've done my research well and I'm unable to come up with an answer on my own. I want to know: what happens to the .call files after the MaxRetries number has been reached? In my experience, they stay in the Outgoing folder, but are never deleted. Instead, Asterisk keeps processing them, but never actually making a call. In my mind, once the MaxRetries number has been met, Asterisk should do something to get rid of the files, whether moving them to another failed folder or just deleting them. You can see an example of my problem below. The Yellow Highlighted remarks are my own for clarification and are not in the actual .call file. -- Channel: SIP/8644161...@vitel-outbound MaxRetries: 9 = Set to retry 9 times RetryTime: 120 = Retrys after 120 seconds Context: autodial Extension: s Priority: 1 CallerID: 8645553190 Set: USERNUMBER=8644161809-JohnTimms Set: DIGITS=8644161809 = I've tried the Archive: Yes option here, but had no change in behavior StartRetry: 2397 1 (1270149233)= This line the following are all added by Asterisk EndRetry: 2397 1 (1270149158) StartRetry: 2397 2 (1270149399) EndRetry: 2397 2 (1270149324) StartRetry: 2397 3 (1270149565) EndRetry: 2397 3 (1270149490) StartRetry: 2397 4 (1270149731) EndRetry: 2397 4 (1270149656) StartRetry: 2397 5 (1270149897) EndRetry: 2397 5 (1270149822) StartRetry: 2397 6 (1270150063) EndRetry: 2397 6 (1270149988) StartRetry: 2397 7 (1270150229) EndRetry: 2397 7 (1270150154) StartRetry: 2397 8 (1270150395) EndRetry: 2397 8 (1270150320) StartRetry: 2397 9 (1270150561) DelayedRetry: 2397 8 (1270151821) DelayedRetry: 2397 8 (1270151942) DelayedRetry: 2397 8 (1270152063) DelayedRetry: 2397 8 (1270152184) DelayedRetry: 2397 8 (1270152305) DelayedRetry: 2397 8 (1270152426) DelayedRetry: 2397 8 (1270152547) DelayedRetry: 2397 8 (1270152668) DelayedRetry: 2397 8 (1270152789) DelayedRetry: 2397 8 (1270152910) DelayedRetry: 2397 8 (1270153031) DelayedRetry: 2397 8 (1270153152) DelayedRetry: 2397 8 (1270153273) DelayedRetry: 2397 8 (1270153394) DelayedRetry: 2397 8 (1270153515) DelayedRetry: 2397 8 (1270153636) -- If anyone can help me out, that would be much appreciated. -- John Timms (864) 416-1809 johngti...@gmail.com -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question
Ok so a while back I found an example for having a number dial multiple
numbers and then whoever answers and confirms gets the call. (don't
recall who the example was from, but thank you!)
But Now today I've been playing with TTS and STT and came across the
BackgroundDetect command. Now If I use this allow it works fine. But
when I try and use it with this it never actually detects me talking -
or if it does it doesn't connect the caller so that the Wait time
expires and it goes on.
So my question is how can I make this work to where you can talk and it
will connect you to the caller or press 1. Not now where you just press
1. Which a lot of the time I can't get my phone out of my pocket,
unlocked, and press 1 before it is sent to VM
[default]
exten =
_XX,1,Monitor(wav,/var/store/calls/PersonalLine-${STRFTIME($
{EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)}-${EXTEN},mb)
exten = _XX,2,dial(${bellbu}/${EXTEN:4},40,rM(screen)) ;
without r it seems to pass a second or two of audio first
exten = _XX,4,Hangup ; You can also substitute this with a
Voicemail destination or other alternative destination
[macro-screen]
;exten = s,1,Wait(1)
;exten = s,n,Background(/var/lib/asterisk/sounds/press1) ; substitute a
different playback file if you need to
;exten = s,n,WaitExten(5) ; the value is the Wait time before we assume
the call is not accepted
;exten = 1,1,NoOp(Caller accepted) ; Do not set MACRO_RESULT to
anything to connect the caller
;exten = i,1,Set(MACRO_RESULT=CONTINUE)
;exten = t,1,Set(MACRO_RESULT=CONTINUE)
exten = s,1,Wait(1)
exten = s,n,BackgroundDetect(/var/lib/asterisk/sounds/press1)
exten = s,n,WaitExten(10) ; the value is the Wait time before we assume
the call is not accepted
exten = 1,1,NoOp(Caller accepted) ; Do not set MACRO_RESULT to anything
to connect the caller
exten = i,1,Set(MACRO_RESULT=CONTINUE)
exten = t,1,Set(MACRO_RESULT=CONTINUE)
exten = talk,1,NoOp(Caller accepted)
[Inbound]
exten = 4095551212,1,NoOP()
exten = 4095551212,n,Dial(LOCAL/111222LOCAL/222333,40)
exten = 4095551212,n,Voicemail(1...@default)
James Shigley
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Re: [asterisk-users] Question about Presence and IM feature
15 jan 2010 kl. 08.23 skrev Yuji Kondo: I have two questions for Asterisk feature. 1. Can Asterisk support presence feature ? Asterisk is a telephony PBX and supports presence subscriptions for extension states - if a phone line is busy or not, over a few different SIP presence formats, like SIP dialog-info and SIMPLE. Is latest version supporting PUBLISH method No. 2. Can Asterisk provide Instant Messaging feature as server ? During a phone call, yes. Not otherwise. If you need a full-blown presence server, you should look at Kamailio/OpenSER at http://www.kamailio.org Best regards, /Olle -- _ -- 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] Question about Presence and IM feature
Dear Team, I have two questions for Asterisk feature. 1. Can Asterisk support presence feature ? I found following information. but it is too old...? ** http://www.voip-info.org/wiki/view/Asterisk+presence Extension and device state combined, no user state! Asterisk - with or without the patch - does not currently support PUBLISH method for publishing presence documents in Presence Information Data Format (PIDF) defined in RFC 3863, but it can generate NOTIFY to SUBSCRIBEd users when REGISTER occurs and also when the client disconnects. Is latest version supporting PUBLISH method ? ** If yes, Is it supported by PIDF ? If no, Do you have a support plan ? 2. Can Asterisk provide Instant Messaging feature as server ? Does Asterisk operate independently the call control and the messaging ? How do we configure hint of extensions.conf ? Best Regards, Yuji Kondo - Get the new Internet Explorer 8 optimized for Yahoo! JAPAN -- _ -- 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] Question about SIP registration
I reply to your question below 1) I don't have a secret for that peer. 2) Obviously, the solution is to make the 'host' field static (in my scenario, because the port is non-standard 5080, so no standard endpoint SIP can register with that IPaddress:port) or specify a secret with 'host=dynamic'. The question I made was a little different: I'm wondering why an external SIP endpoint, which is trying to register on eth0 85.X.Y.Z network, is indeed seen by Asterisk as registered with address 1.1.1.1 (the eth1 IP addresses of the PC). I try to explain better: usually, SIP endpoint with IP address X.Y.Z.T which has registered itself on Asterisk (for example with user 200) is seen as following (sip show peers) 200/200 X.Y.Z.T5060OK(xx ms) So the CLI shows the *endpoint's* IP address. Instead, in my scenario, I see a row like this: 999/999 1.1.1.15060UNREACHABLE (1) And 1.1.1.1 is the eth1 IP address of the PC where Asterisk is installed on. But I haven't any endpoint SIP onto that PC which is trying to register, while I can see one of them OUTSIDE my network (i.e. in the Big Internet) that is trying to register as 999: in fact, if in [999] SIP account I put 'host=1.1.1.1', I can see a row like this on Asterisk log: [Jan 13 11:10:54] ERROR[1834]: chan_sip.c:8718 register_verify: Peer '999' is trying to register, but not configured as host=dynamic [Jan 13 11:10:54] NOTICE[1834]: chan_sip.c:15236 handle_request_register: Registration from '999 sip:9...@85.x.y.z ' failed for '174.129.74.46' - Peer is not supposed to register - while if I put 'host=dynamic' I saw (in sip show peers) the row depicted in (1) and no more errors like above. I suspect there is something wrong with network configuration (firewall, NAT). But this behavior is quite odd to me ... Alberto. PS: the network is at customer's site, so I haven't chance to have a clear look over it... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Lister Sent: martedì 12 gennaio 2010 18.51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about SIP registration On Tue, 2010-01-12 at 18:16 +0100, Aggio Alberto wrote: Then I have configured an account as following: [999] type=friend username=999 You don't appear to have a secret= line in there with a password option... or did you snip it? Can someone explain me this kind of behaviour? Is it normal? Can I restrict registration of 999 peer only to SIP UA from network 1.1.1.X? There is an ACL option for the SIP peer which you can add, http://www.voip-info.org/wiki/index.php?page=Asterisk+sip +permit-deny-mask (although there were some issues with this in earlier versions of asterisk.. it should work properly in recent versions.) Rob -- _ -- 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] Question about SIP registration
Hi guys, I recently faced an issue regarding SIP registration: I have a 2-NIC Linux PC, with eth0 set to address 192.168.1.1 (NATted over public network, with address 89.X.Y.Z) and eth1 set to address 1.1.1.1. In [sip.conf] I set general option bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) Then I have configured an account as following: [999] type=friend username=999 host=dynamic port=5080 context=sipfrom nat=no canreinvite=no call-limit=8 videosupport=no disallow=all allow=alaw qualify=15000 So far, so good. Now, I have an internal process (onto Linux PC) which is a SIP endpoint and should register to Asterisk as 1.1.1.1:5080, but an external entity (i.e. a SIP endpoint over public Internet) is trying to register to Asterisk as 9...@89.x.y.zmailto:9...@89.x.y.z:5060 and the registration SUCCEEDS! When I launch the CLI command sip show peers, I see a row like this: 999/9991.1.1.1 5060 OK (3 ms) Can someone explain me this kind of behaviour? Is it normal? Can I restrict registration of 999 peer only to SIP UA from network 1.1.1.X? Thanks for your help! Regards, Alberto Aggio -- _ -- 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] Question about SIP registration
Instead of host=dynamic, use host=1.1.1.1, or host=1.1.1.0/255.255.255.0. Thanks, --Warren Selby On Jan 12, 2010, at 11:16 AM, Aggio Alberto alberto.ag...@loquendo.com wrote: Hi guys, I recently faced an issue regarding SIP registration: I have a 2-NIC Linux PC, with eth0 set to address 192.168.1.1 (NATted over public network, with address 89.X.Y.Z) and eth1 set to address 1.1.1.1. In [sip.conf] I set general option bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) Then I have configured an account as following: [999] type=friend username=999 host=dynamic port=5080 context=sipfrom nat=no canreinvite=no call-limit=8 videosupport=no disallow=all allow=alaw qualify=15000 So far, so good. Now, I have an internal process (onto Linux PC) which is a SIP endpoint and should register to Asterisk as 1.1.1.1:5080, but an external entity (i.e. a SIP endpoint over public Internet) is trying to register to Asterisk as 9...@89.x.y.z:5060 and the registration SUCCEEDS! When I launch the CLI command sip show peers, I see a row like this: 999/9991.1.1.1 5060 OK (3 ms) Can someone explain me this kind of behaviour? Is it normal? Can I restrict registration of 999 peer only to SIP UA from network 1.1.1.X? Thanks for your help! Regards, Alberto Aggio -- _ -- 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] Question about SIP registration
On Tue, 2010-01-12 at 18:16 +0100, Aggio Alberto wrote: Then I have configured an account as following: [999] type=friend username=999 You don't appear to have a secret= line in there with a password option... or did you snip it? Can someone explain me this kind of behaviour? Is it normal? Can I restrict registration of 999 peer only to SIP UA from network 1.1.1.X? There is an ACL option for the SIP peer which you can add, http://www.voip-info.org/wiki/index.php?page=Asterisk+sip +permit-deny-mask (although there were some issues with this in earlier versions of asterisk.. it should work properly in recent versions.) Rob -- _ -- 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] question on makefile
There is a line like in codes/Makefile $(if $(filter codec_lpc10,$(EMBEDDED_MODS)),modules.link,codec_lpc10.so): $(LIBLPC10) What is filter? Where is filter? whereis filter doesnt return anything find . | grep filter in asterisk root directory returns nothing. Thanks, Jerry ___ -- 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] question on makefile
On Wednesday 06 January 2010 13:45:55 Jerry Geis wrote: There is a line like in codes/Makefile $(if $(filter codec_lpc10,$(EMBEDDED_MODS)),modules.link,codec_lpc10.so): $(LIBLPC10) What is filter? Where is filter? whereis filter doesnt return anything find . | grep filter in asterisk root directory returns nothing. It's a Makefile command. See: http://www.gnu.org/software/automake/manual/make/Text-Functions.html#index-filter-554 -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on makefile
It's a Makefile command. See: http://www.gnu.org/software/automake/manual/make/Text-Functions.html#index-filter-554 great - thanks is there no method by the configure command to --disable-FEATURE??? the help says its there but doesnt seem to do anything for me. example: ./configure --disable-codec_lpc10 doesnt seem to do anything. I was trying to find a way without running make menuselect to not compile in certain items. Thanks, jerry ___ -- 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] question on makefile
Jerry Geis wrote: is there no method by the configure command to --disable-FEATURE??? There is not. The Asterisk configure script is used for platform specific settings, locating libraries and header files and the like. It is not used (directly) for controlling which portions of Asterisk are built or are not built. the help says its there but doesnt seem to do anything for me. example: ./configure --disable-codec_lpc10 doesnt seem to do anything. I was trying to find a way without running make menuselect to not compile in certain items. What help says it is there? The help text that documents how '--disable-FEATURE' works assumes there is a list of FEATUREs somewhere... but Asterisk does not have such a list since we don't control features from the configure script. You will have to use menuselect to disable modules from being built, although it can be automated so that you don't have to use the interactive mode to do it. Many Linux distributions (and also AsteriskNOW) do this already in their package building recipes, so you might want to take a look at one of them to see how it is done. In addition, if you are scripting this build, you could just as easily 'rm codecs/codec_lpc10.c' before running the 'make' and 'make install' steps. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about PLC of Asterisk
Hi,I want to know how to do to work PLC of Asterisk. Anyone plz help me. PLC (Packet Loss Concealment) is included in Asterisk,I read at voip-info.org or release note. And I see in codecs.conf, genelicplc setting. So I put codecs.conf in '/etc/asterisk' ,and wrote genericplc = true. And I worked Asterisk and tested. I think PLC like this. When I send 3 packet (1,2,3), and caused loss (if No.2 is lossed), then receive-packet after PLC should be same number 3 packet(1,2,3). No.2 is new one. I don't know 2 is same sound to old 2,or is zero level sound. But same number packets should be resend from Asterisk to receiver. But when I send 3 packet, received-packet is 2 packet(1,3). No.2 has lossed. No interpolation has been done. So I can't understand how to do to work PLC of Asterisk. Please help me. I tested Asterisk 1.4.* and 1.6.0. Use uLaw to send and receive on SIP/RTP. And caused 5-20% loss from send-packet and through Asterisk ,I checked receive-packet. Is there another things to do without to put codecs.conf in /etc/asterisk ? Or Asterisk has no PLC on uLaw ? ___ -- 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] question on register
11 dec 2009 kl. 17.18 skrev Jerry Geis: Where in the code does something like: register = user[:secret[:authuse...@host[:port][/extension] from sip.conf 1) get parsed 2) actually register. I tried looking in channels/chan_sip.c and don't see where that happens. Can someone point me the right file and or function. Search for the function sip_register() in chan_sip.c to find the function that parses and builds a list of services we're about to register for. To find where it all happens, there's a function called sip_send_all_registers() that is a good starting point for exploration. Cheers, /O --- * Olle E. Johansson - o...@edvina.net * Asterisk Training http://edvina.net/training/ ___ -- 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] Question regarding digital card TE412p
Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk: 1.4.22.1, Dahdi: 2.2.0.2, libpri: 1.4.10.2. Thank you very much for your help. Regards Sandesh ___ -- 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] Question regarding digital card TE412p
The calls itselves doesn't take a lot of CPU resources, even more considering you're willing to use hardware echo cancelling. The real CPU hogs are apps like MeetMe() and AGI scripts. Those are no worse than audiotranscoding thought. You also should design the system in such way there's as few transcoding as possible, or no transcoding at all. With that in mind, you can have as many cards/ports as your hardware can physically handle. Vinícius Fontes www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP - das sandesh sandesh...@gmail.com escreveu: Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk: 1.4.22.1, Dahdi: 2.2.0.2, libpri: 1.4.10.2. Thank you very much for your help. Regards Sandesh ___ -- 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] Question regarding digital card TE412p
Hi! Having two TE410P with heavy load in a Pentium4 3,2GHz system running Asterisk 1.2 was no problem. It did only IVR and bridging with no transcoding though. Chris 2009/12/14 das sandesh sandesh...@gmail.com: Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk: 1.4.22.1, Dahdi: 2.2.0.2, libpri: 1.4.10.2. Thank you very much for your help. Regards Sandesh ___ -- 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] question on how to connect 2 boxes
Hello, I would like to connect 2 asterisk boxes together, so this is my scenario: Asterisk Main: it is connected to many sip providers and its main purpose as a call termination forwarder. Asterisk B: its connected to E1, and its purpose to terminate calls. It will receive SIP messages from Asterisk_Main, but there will be no voice traffic going between them, Asterisk_Main will send the provider IP address where both Asterisk_B provider will communicate, at the end of the call, Asterisk_Main will log CDR traffic. Now I can make IAX2, but the problem is, the traffic must go like this: Provider -Asterisk_Main- Asterisk_B This will cause a problem with wasted bandwidth and more latency on the call. I want it to be like this: Asterisk_Main | Asterisk_B -ßà Provider Please let me know your suggestions. Regards. ___ -- 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] Question regarding digital card TE412p
Thanks Victor and Vinícius for the information. I will not be doing any transcoding but using some AGI scripts, I will update the status once I configure and start using them. Thanks Sandesh On Mon, Dec 14, 2009 at 1:59 PM, Christian Victor christ...@victormedia.dewrote: Hi! Having two TE410P with heavy load in a Pentium4 3,2GHz system running Asterisk 1.2 was no problem. It did only IVR and bridging with no transcoding though. Chris 2009/12/14 das sandesh sandesh...@gmail.com: Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk: 1.4.22.1, Dahdi: 2.2.0.2, libpri: 1.4.10.2. Thank you very much for your help. Regards Sandesh ___ -- 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] question on queues
I have been looking for a way from the dialplan to inquire if there are any members in a queue. So what I want to do is if no users are members of a queue then I can send the call to a given extention. I have the queue setup all that is working. Just need to be able to send the call to a certain user if no-one is logged into the queue. How do I do that? Thanks Jerry ___ -- 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] question on queues
On Dec 13, 2009, at 7:20 PM, Jerry Geis wrote: I have been looking for a way from the dialplan to inquire if there are any members in a queue. So what I want to do is if no users are members of a queue then I can send the call to a given extention. I have the queue setup all that is working. Just need to be able to send the call to a certain user if no-one is logged into the queue. How do I do that? Thanks Jerry In queues.conf, you can have joinempty=no for the selected queue. If there's noone logged in, the dialplan will move forward to the next entry. ---fred ___ -- 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] question on queues
Hi Jerry, I use the built-in function queue_member http://www.asterisk.org/docs/asterisk/trunk/functions/queue_member?type=functionsvalue=QUEUE_MEMBER and check with a GotoIf statement to check if the number is equal to zero. If it is not I send the call to the queue, if it is I pass the call to dial a cell-phone number or go directly to voicemail depending on which queue the call was originally destined for. Travis - Original Message - From: Jerry Geis ge...@pagestation.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, December 13, 2009 4:20:40 PM Subject: [asterisk-users] question on queues I have been looking for a way from the dialplan to inquire if there are any members in a queue. So what I want to do is if no users are members of a queue then I can send the call to a given extention. I have the queue setup all that is working. Just need to be able to send the call to a certain user if no-one is logged into the queue. How do I do that? Thanks Jerry ___ -- 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] question on register
Where in the code does something like: register = user[:secret[:authuse...@host[:port][/extension] from sip.conf 1) get parsed 2) actually register. I tried looking in channels/chan_sip.c and don't see where that happens. Can someone point me the right file and or function. Thanks, Jerry ___ -- 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] Question about g729
However, I've read somewhere that passthrough doesnt require a license. Which means that if your sip clients can transmit in g729 and your voip provider can receive in g729, your asterisk server won't need to do any encoding and therefore doesn't need any licenses. It is simply passing the data through from your sip clients to the voip provider. Not sure what happens if you want to play recorded messages and things. It would probably need licenses then because its encoding. Sent from my Windows Mobile® phone. -Original Message- From: Alex Balashov abalas...@evaristesys.com Sent: 02 December 2009 01:13 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Question about g729 All calls. Landy Landy wrote: You only need to purchase 10 licenses, if all 10 clients will be making calls at the same time. Ok. Does this apply only for outbound calls using a voip provider and/or applies to calls within the lan? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Question about g729
Sorry for the repetition. I didn't see the other responses. -Original Message- From: Thomas Kenyon dig...@sanguinarius.co.uk Sent: 02 December 2009 07:36 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Question about g729 Tilghman Lesher wrote: On Tuesday 01 December 2009 19:10:08 Alex Balashov wrote: All calls. Landy Landy wrote: You only need to purchase 10 licenses, if all 10 clients will be making calls at the same time. Ok. Does this apply only for outbound calls using a voip provider and/or applies to calls within the lan? An additional clarification: it only applies to calls in which codecs need to be transcoded. If you have a g729 call bridged to another g729 call, then no license is used in that call path. Also, the only consideration, isn't the endpoints. If the call is being recorded or you are in a conference, then the call needs to be transcoded for mixing purposes. ___ -- 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] Question about g729
My understanding is that Asterisk will not pass through calls in codecs for which it does not have support and/or licenses; it simply does not advertise them in the SDP negotiation. Perhaps I am wrong. Dan Journo wrote: However, I've read somewhere that passthrough doesnt require a license. Which means that if your sip clients can transmit in g729 and your voip provider can receive in g729, your asterisk server won't need to do any encoding and therefore doesn't need any licenses. It is simply passing the data through from your sip clients to the voip provider. Not sure what happens if you want to play recorded messages and things. It would probably need licenses then because its encoding. Sent from my Windows Mobile® phone. -Original Message- From: Alex Balashov abalas...@evaristesys.com Sent: 02 December 2009 01:13 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Question about g729 All calls. Landy Landy wrote: You only need to purchase 10 licenses, if all 10 clients will be making calls at the same time. Ok. Does this apply only for outbound calls using a voip provider and/or applies to calls within the lan? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about g729
Alex Balashov wrote: My understanding is that Asterisk will not pass through calls in codecs for which it does not have support and/or licenses; it simply does not advertise them in the SDP negotiation. 'support' - yes, 'licenses' - no. Asterisk supports passthrough, recording and playback of quite a few codecs for which there are no transcoding modules available at all (G.723.1, H.263/4, etc). G.729 falls into this category if there is no transcoding module loaded, or if there are no licenses available. The only time that Asterisk will not offer G.729 in an outbound negotiation is if the incoming channel is not in G.729 (so thus would require transcoding) and there is no transcoding path available. The same is true for other codecs that don't have transcoding available... but by definition, this is not 'passthrough'. Also, to clarify an earlier point, Digium makes all the standard Asterisk prompt sets available in G.729 format, so the built-in applications can be used on G.729 channels without requiring transcoding. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about g729
Hello. I am currently testing an asterisk server using the default codecs, I have allow=all, and noticed everytime I test it in a wireless lan the latency rockets off the roof to over 1000ms. I would like to test g729 since it uses less bandwidth but, read somewhere I have to buy a license per every channel I have. Does this means if I have my server connected with 10 sip clients I need to buy a license for 10 or more? ___ -- 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] Question about g729
You pay per channel. Which I believe to mean, if you have 10 sip clients but only 2 clients make calls at the same time, you only need 2 licenses. You only need to purchase 10 licenses, if all 10 clients will be making calls at the same time. I hope that makes sense. Maybe someone can explain it better or correct me if Im wrong. Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: 01 December 2009 23:24 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Question about g729 Hello. I am currently testing an asterisk server using the default codecs, I have allow=all, and noticed everytime I test it in a wireless lan the latency rockets off the roof to over 1000ms. I would like to test g729 since it uses less bandwidth but, read somewhere I have to buy a license per every channel I have. Does this means if I have my server connected with 10 sip clients I need to buy a license for 10 or more? ___ -- 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] Question about g729
You only need to purchase 10 licenses, if all 10 clients will be making calls at the same time. Ok. Does this apply only for outbound calls using a voip provider and/or applies to calls within the lan? ___ -- 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] Question about g729
All calls. Landy Landy wrote: You only need to purchase 10 licenses, if all 10 clients will be making calls at the same time. Ok. Does this apply only for outbound calls using a voip provider and/or applies to calls within the lan? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about g729
On Tuesday 01 December 2009 19:10:08 Alex Balashov wrote: All calls. Landy Landy wrote: You only need to purchase 10 licenses, if all 10 clients will be making calls at the same time. Ok. Does this apply only for outbound calls using a voip provider and/or applies to calls within the lan? An additional clarification: it only applies to calls in which codecs need to be transcoded. If you have a g729 call bridged to another g729 call, then no license is used in that call path. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about g729
Tilghman Lesher wrote: On Tuesday 01 December 2009 19:10:08 Alex Balashov wrote: All calls. Landy Landy wrote: You only need to purchase 10 licenses, if all 10 clients will be making calls at the same time. Ok. Does this apply only for outbound calls using a voip provider and/or applies to calls within the lan? An additional clarification: it only applies to calls in which codecs need to be transcoded. If you have a g729 call bridged to another g729 call, then no license is used in that call path. Also, the only consideration, isn't the endpoints. If the call is being recorded or you are in a conference, then the call needs to be transcoded for mixing purposes. ___ -- 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] question about call transfer
Hi all, Refer to http://www.voip-info.org/wiki/view/Asterisk+config+features.conf. It shows we can use variable BLINDTRANSFER to call back the one who transfer the call. However, in my tests below. The result is not as expected. case 1: A calls B (dial(sip/B||Tt) B answers and connects to A B transfer to C C doesn't answer the call and B ring again case 2: A calls B (dial(sip/B||Tt) B answers and connects to A A transfer to C C doesn't answer the call but B ring instead of A In case 2, the person who transfer the call can't get back the call. Anyone can tell whether there is a way to correct in case 2? Thanks, ango ___ -- 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] Question about OSLEC or HPEC with AsteriskNow
AsteriskNow use CentOS 5 and it comes preinstalled with dahdi and asterisk with the freepbx GUI interface and it seems to be missing all the dev packages Martin On 2009-11-17, at 02:19, Olivier wrote: 2009/11/17 Martin Roy m...@mac.com I was previously using an old computer running Asterisk 1.2 with zaptel. Once the CPU fried I switch to a new computer and I chose AsteriskNow 1.5 running in 64bits to simplify the installation process. I manage to find my way with configuring dahdi instead of zaptel and to switch all my previous config to the new computer. Now everything is fine except that even if I use the md2 echo cancellation it's not perfect I still have echo issue. So I made some search around and found that there's oslec and hpec out there that seems to be better then what I'm currently using. So my question should I use hpec or oslec with my TDM400 card? I also tried to recompile dahdi to use oslec (before I found that Digium had hpec) but then I get an error message that the source of my kernel cannot be found Do you imply you previously installed a Dahdi binary package ? If positive, before compiling Dahdi source code, you need to install Linux header files. On Debian systems, you can get this with something like : apt-get -install linux-headers-2.6.26-2-686 Regards so I can never actually compile a new version of dahdi. Thanks Martin ___ -- 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] Question about OSLEC or HPEC with AsteriskNow
I was previously using an old computer running Asterisk 1.2 with zaptel. Once the CPU fried I switch to a new computer and I chose AsteriskNow 1.5 running in 64bits to simplify the installation process. I manage to find my way with configuring dahdi instead of zaptel and to switch all my previous config to the new computer. Now everything is fine except that even if I use the md2 echo cancellation it's not perfect I still have echo issue. So I made some search around and found that there's oslec and hpec out there that seems to be better then what I'm currently using. So my question should I use hpec or oslec with my TDM400 card? I also tried to recompile dahdi to use oslec (before I found that Digium had hpec) but then I get an error message that the source of my kernel cannot be found so I can never actually compile a new version of dahdi. Thanks Martin ___ -- 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] Question about OSLEC or HPEC with AsteriskNow
2009/11/17 Martin Roy m...@mac.com I was previously using an old computer running Asterisk 1.2 with zaptel. Once the CPU fried I switch to a new computer and I chose AsteriskNow 1.5 running in 64bits to simplify the installation process. I manage to find my way with configuring dahdi instead of zaptel and to switch all my previous config to the new computer. Now everything is fine except that even if I use the md2 echo cancellation it's not perfect I still have echo issue. So I made some search around and found that there's oslec and hpec out there that seems to be better then what I'm currently using. So my question should I use hpec or oslec with my TDM400 card? I also tried to recompile dahdi to use oslec (before I found that Digium had hpec) but then I get an error message that the source of my kernel cannot be found Do you imply you previously installed a Dahdi binary package ? If positive, before compiling Dahdi source code, you need to install Linux header files. On Debian systems, you can get this with something like : apt-get -install linux-headers-2.6.26-2-686 Regards so I can never actually compile a new version of dahdi. Thanks Martin ___ -- 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] Question about callerid?
OK, Now I am responding to myself, because I have figured it out (finally). It turns out it's a feature of asterisk (at least the older versions). This is where I found my answer: https://issues.asterisk.org/view.php?id=9678 So the solution for me was to simply rearrange my sip.conf so my incoming call handling peer is at the very end. Pretty wacky. I am hopefully back on the road though with working caller ID as well. Marty On Nov 14, 2009, at 11:10 AM, Martin Joseph wrote: Ok I am replying to myself, because I still don't have this figured out,, but I think I have more info. On Nov 5, 2009, at 8:57 PM, Martin Joseph wrote: Hello again Asterisk people. I am running Asterisk 1.42 on an old PowerPC ibook. I have had this deployed for several years now, with pretty good results. Recently I added a callerid service to my landline (qwest). I am using the audiocodes MP114 2fxo/2fxs gateway, which is an outstanding piece of hardware once it's configured (lol). I think the issue is related to the fact that the MP114 is in my case a combination device. 2fxo/2fxs setup. It seems like what happens is when a call comes into the fxo it is inviting asterisk with the correct callerid information(sip from). Asterisk attempts to use this invite as a basis for a new call. HOWEVER, for some reason or another (bug?) Asterisk identifies the fxs extension at the same IP address as a peer for the basis of the new call, and since the other peer (friend) is the FXS, the authentication fails, and caller ID is lost. If I remove my FXS (friend) definition from sip.conf then suddenly all is well and the the callerID string is passed aok. Of course then none of the phones attached to the FXS work, which is a problem... I hope someone has some ideas on what I am doing wrong/some way to fix this? Thanks in advance for any help you might offer. Marty Anyhow, I can see that the gateway is passing caller id info to asterisk because the console will display something like: [Nov 4 13:01:19] NOTICE[32]: chan_sip.c:13362 handle_request_invite: Failed to authenticate user SEATTLE SCHOOLS sip:2062524...@89.89.89.253 ;tag=1c492497235 This authentication is failing because of the mismatch of extensions described above. The FXO is ext. 2003 and the FXS is ext. 2005. So the caller ID info is right there. However on my extensions (or softphones) the id shows as the extension # (ie 2003). Is there something I need to do to set the callerid? I can't seem to find this in the examples? Thanks in advance for helping with my (I am sure) stupid question... Marty ___ -- 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
Re: [asterisk-users] Question about callerid?
Ok I am replying to myself, because I still don't have this figured out,, but I think I have more info. On Nov 5, 2009, at 8:57 PM, Martin Joseph wrote: Hello again Asterisk people. I am running Asterisk 1.42 on an old PowerPC ibook. I have had this deployed for several years now, with pretty good results. Recently I added a callerid service to my landline (qwest). I am using the audiocodes MP114 2fxo/2fxs gateway, which is an outstanding piece of hardware once it's configured (lol). I think the issue is related to the fact that the MP114 is in my case a combination device. 2fxo/2fxs setup. It seems like what happens is when a call comes into the fxo it is inviting asterisk with the correct callerid information(sip from). Asterisk attempts to use this invite as a basis for a new call. HOWEVER, for some reason or another (bug?) Asterisk identifies the fxs extension at the same IP address as a peer for the basis of the new call, and since the other peer (friend) is the FXS, the authentication fails, and caller ID is lost. If I remove my FXS (friend) definition from sip.conf then suddenly all is well and the the callerID string is passed aok. Of course then none of the phones attached to the FXS work, which is a problem... I hope someone has some ideas on what I am doing wrong/some way to fix this? Thanks in advance for any help you might offer. Marty Anyhow, I can see that the gateway is passing caller id info to asterisk because the console will display something like: [Nov 4 13:01:19] NOTICE[32]: chan_sip.c:13362 handle_request_invite: Failed to authenticate user SEATTLE SCHOOLS sip:2062524...@89.89.89.253 ;tag=1c492497235 This authentication is failing because of the mismatch of extensions described above. The FXO is ext. 2003 and the FXS is ext. 2005. So the caller ID info is right there. However on my extensions (or softphones) the id shows as the extension # (ie 2003). Is there something I need to do to set the callerid? I can't seem to find this in the examples? Thanks in advance for helping with my (I am sure) stupid question... Marty ___ -- 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] Question about callerid?
On Nov 6, 2009, at 5:14 AM, John A. Sullivan III wrote: On Thu, 2009-11-05 at 20:57 -0800, Martin Joseph wrote: Hello again Asterisk people. I am running Asterisk 1.42 on an old PowerPC ibook. I have had this deployed for several years now, with pretty good results. Recently I added a callerid service to my landline (qwest). I am using the audiocodes MP114 2fxo/2fxs gateway, which is an outstanding piece of hardware once it's configured (lol). Anyhow, I can see that the gateway is passing caller id info to asterisk because the console will display something like: [Nov 4 13:01:19] NOTICE[32]: chan_sip.c:13362 handle_request_invite: Failed to authenticate user SEATTLE SCHOOLS sip:2062524...@89.89.89.253 ;tag=1c492497235 So the caller ID info is right there. However on my extensions (or softphones) the id shows as the extension # (ie 2003). Is there something I need to do to set the callerid? I can't seem to find this in the examples? Thanks in advance for helping with my (I am sure) stupid question... snip I'd like to understand this better myself as I know we don't have this right in our environment. I believe the reason you see that is because Asterisk is providing a B2BUA (I think it's called), i.e., your caller is not actually talking to your phone. Instead, your caller is talking to Asterisk on the inbound SIP ID (whatever that is) and then Asterisk is calling your phone from the extension in the dial plan. At least I think that's why the extension shows up in the callerID. OK, that makes sense. So since Asterisk is a back to to back user agent (ie the call is always going through it) then the Caller ID data isn't magically moved along... Still, the fact that it's showing up there in the console means there should be some way to grab it (the callerID data) and stuff into into the proper place for it to be passed along. I see that the callerid valiable can be set as per: http://www.voip-info.org/wiki/view/Setting+Callerid So that's nice, and the only question is how to I get the callerID info from where it show in the console as failed to authenticate? Either that, or I could reconfigure my audiocodes and my asterisk so that instead of incoming calls dialing my desired extension (ie 2020), asterisk could accept the calls from the domain of the audiocodes (ie it's IP address). Maybe that's how get the CID data. Don't really know, but suspect there are lots of people here who do? Thanks for any help in advance, Marty The identity can be overridden in sip.conf with the fromdomain and fromuser parameters. However, we found this introduced its own problems. I suppose we just need to build more sophisticated logic into our dialplan. The problem is, if we set the fromdomain/user, we now show correct sip sources when we make direct SIP calls and can return those calls from the phone's call history. However, it breaks all the internal dialing which wants to dial to the extension. If we remove fromdomain/user, the internal dialing works but public SIP calls now show the extension as the user rather than the user's public SIP ID. I'm sure as with most things in Asterisk, we can fix it if we just take the time to think through the programming logic. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Question about callerid?
Hello again Asterisk people. I am running Asterisk 1.42 on an old PowerPC ibook. I have had this deployed for several years now, with pretty good results. Recently I added a callerid service to my landline (qwest). I am using the audiocodes MP114 2fxo/2fxs gateway, which is an outstanding piece of hardware once it's configured (lol). Anyhow, I can see that the gateway is passing caller id info to asterisk because the console will display something like: [Nov 4 13:01:19] NOTICE[32]: chan_sip.c:13362 handle_request_invite: Failed to authenticate user SEATTLE SCHOOLS sip:2062524...@89.89.89.253 ;tag=1c492497235 So the caller ID info is right there. However on my extensions (or softphones) the id shows as the extension # (ie 2003). Is there something I need to do to set the callerid? I can't seem to find this in the examples? Thanks in advance for helping with my (I am sure) stupid question... Marty ___ -- 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] question about getting instance ringing member in queue
Hi, I have a queue and 3 agents in the queue like below SIP/1001 SIP/1002 SIP/1003 When I dial the queue number, the agent start to ring. How can I get the instance ringing agent as I want to pause the agent (pausequeuemember) after the queue timeout? Any application or variable can use to get the ringing agent? ango ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on SIP and call manager
On Asterisk 1.4, Call doesn't line Channel: AB. You could put the second dialplan snippet into a context and do your callfile like this: [callccm] exten = s,1,Dial(SIP/CCMMAIN,10,KkTt) Exten = s,n,Dial(SIP/CCMSLAVE,10,KkTt) -- Channel: SIP/104 CallerID: SIP/104 MaxRetries: 1 WaitTime: 60 retryTime: 5 Context: callccm Extension: s -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, October 15, 2009 7:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question on SIP and call manager Here are two ways to address this 1. Dial(SIP/CCMMAINSIP/CCMSLAVE) - this tries both at once 2. exten = s,1,Dial(SIP/CCMMAIN,10,KkTt) Exten = s,n,Dial(SIP/CCMSLAVE,10,KkTt) CCMSLAVE only gets called if no one answers CCMMAIN in 10 seconds (2-3 rings) Danny thats good to know for extensions.conf but I am using call files. echo Channel: SIP/CCMMAIN/5551212 /tmp/call echo Context: smvoice-test /tmp/call Can I do the Channel: SIP/CCMMAIN/5551212SIP/CCMSLAVE/5551212 in the Channel for the call file? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question on SIP and call manager
Customer has 2 call manager systems and I am using asterisk to place calls through the CCM. One for the main use - CCMMAIN and another for disaster CCMSLAVE. Can asterisk be setup in such a way that calls first try to use CCMMAIN and if thats not available use CCMSLAVE. Example if I place a call file that places a call like Dial: SIP/CCMMAIN/5551212 that if CCMMAIN is not available then CCMSLAVE will automatically be used? My application placing calls in the call file doesnt have any knowledge of which context to use. CCMMAIN is the only thing my call file nows about. How do I set up such an arrangement if possible? thanks. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on SIP and call manager
Here are two ways to address this 1. Dial(SIP/CCMMAINSIP/CCMSLAVE) - this tries both at once 2. exten = s,1,Dial(SIP/CCMMAIN,10,KkTt) Exten = s,n,Dial(SIP/CCMSLAVE,10,KkTt) CCMSLAVE only gets called if no one answers CCMMAIN in 10 seconds (2-3 rings) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, October 15, 2009 4:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] question on SIP and call manager Customer has 2 call manager systems and I am using asterisk to place calls through the CCM. One for the main use - CCMMAIN and another for disaster CCMSLAVE. Can asterisk be setup in such a way that calls first try to use CCMMAIN and if thats not available use CCMSLAVE. Example if I place a call file that places a call like Dial: SIP/CCMMAIN/5551212 that if CCMMAIN is not available then CCMSLAVE will automatically be used? My application placing calls in the call file doesnt have any knowledge of which context to use. CCMMAIN is the only thing my call file nows about. How do I set up such an arrangement if possible? thanks. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on SIP and call manager
Here are two ways to address this 1. Dial(SIP/CCMMAINSIP/CCMSLAVE) - this tries both at once 2. exten = s,1,Dial(SIP/CCMMAIN,10,KkTt) Exten = s,n,Dial(SIP/CCMSLAVE,10,KkTt) CCMSLAVE only gets called if no one answers CCMMAIN in 10 seconds (2-3 rings) Danny thats good to know for extensions.conf but I am using call files. echo Channel: SIP/CCMMAIN/5551212 /tmp/call echo Context: smvoice-test /tmp/call Can I do the Channel: SIP/CCMMAIN/5551212SIP/CCMSLAVE/5551212 in the Channel for the call file? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question on pri intense debug
Running asterisk 1.4.26.2 help pri pri debug span Enables PRI debugging on a span pri intense debug span Enables REALLY INTENSE PRI debugging pri no debug span Disables PRI debugging on a span pri set debug file Sends PRI debug output to the specified file pri show debug Displays current PRI debug settings pri show spans Displays PRI Information pri show span Displays PRI Information pri show version Displays version of libpri pri unset debug file Ends PRI debug output to file then I type the following command: pri intense debug 1 No such command 'pri intense debug 1' (type 'help pri intense' for other possible commands) Why is it not understanding my command? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on pri intense debug
pri intense debug span Enables REALLY INTENSE PRI debugging add span keyword or use a tabulator that will do that for you Martin On Wed, Sep 30, 2009 at 10:08 AM, Jerry Geis ge...@pagestation.com wrote: Running asterisk 1.4.26.2 help pri pri debug span Enables PRI debugging on a span pri intense debug span Enables REALLY INTENSE PRI debugging pri no debug span Disables PRI debugging on a span pri set debug file Sends PRI debug output to the specified file pri show debug Displays current PRI debug settings pri show spans Displays PRI Information pri show span Displays PRI Information pri show version Displays version of libpri pri unset debug file Ends PRI debug output to file then I type the following command: pri intense debug 1 No such command 'pri intense debug 1' (type 'help pri intense' for other possible commands) Why is it not understanding my command? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on pri intense debug
Because you need to type pri intense debug SPAN 1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Wednesday, September 30, 2009 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] question on pri intense debug Running asterisk 1.4.26.2 help pri pri debug span Enables PRI debugging on a span pri intense debug span Enables REALLY INTENSE PRI debugging pri no debug span Disables PRI debugging on a span pri set debug file Sends PRI debug output to the specified file pri show debug Displays current PRI debug settings pri show spans Displays PRI Information pri show span Displays PRI Information pri show version Displays version of libpri pri unset debug file Ends PRI debug output to file then I type the following command: pri intense debug 1 No such command 'pri intense debug 1' (type 'help pri intense' for other possible commands) Why is it not understanding my command? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on pri intense debug
Jerry Geis wrote: Running asterisk 1.4.26.2 help pri pri debug span Enables PRI debugging on a span pri intense debug span Enables REALLY INTENSE PRI debugging pri no debug span Disables PRI debugging on a span pri set debug file Sends PRI debug output to the specified file pri show debug Displays current PRI debug settings pri show spans Displays PRI Information pri show span Displays PRI Information pri show version Displays version of libpri pri unset debug file Ends PRI debug output to file then I type the following command: pri intense debug 1 No such command 'pri intense debug 1' (type 'help pri intense' for other possible commands) Why is it not understanding my command? pri intense debug span span number -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on pri intense debug
pri intense debug span span number Just pointing out that was not clear from the HELP command. I thought span was the span number not span span number Thanks for the direction. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on pri intense debug
On Wednesday 30 September 2009 11:54:11 Jerry Geis wrote: pri intense debug span span number Just pointing out that was not clear from the HELP command. I thought span was the span number not span span number Thanks for the direction. At the list level, we only provide the keywords. If you had explicitly requested the individual syntax, you would have seen the complete command structure: *CLI help pri intense debug span Usage: pri intensive debug span span Enables debugging down to the Q.921 level *CLI -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on pri intense debug
Try 'pri intense debug span 1' Used it last night. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, 1 October 2009 4:09 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] question on pri intense debug Running asterisk 1.4.26.2 help pri pri debug span Enables PRI debugging on a span pri intense debug span Enables REALLY INTENSE PRI debugging pri no debug span Disables PRI debugging on a span pri set debug file Sends PRI debug output to the specified file pri show debug Displays current PRI debug settings pri show spans Displays PRI Information pri show span Displays PRI Information pri show version Displays version of libpri pri unset debug file Ends PRI debug output to file then I type the following command: pri intense debug 1 No such command 'pri intense debug 1' (type 'help pri intense' for other possible commands) Why is it not understanding my command? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question of resiliance
Hi
I am in the process of move a company from pstn to an asterisk setup.
They had 2 pstn lines - only really needed a max of 2 previously.
Now I have installed a tdm410 to handle the cross over from pabx to voip
handset. this has been done, the tdm is now just used to provide a
backup pstn line - only used as a last resort for outgoing calls - as
its shared with a fax line.
I use 2 voip providers one primarily for local and std and the other for
mobile calls, all though they can be backups for each other. and then
the pstn line
originally i had (and still till i get around to changing them) macros
to dial each interface and macro's that handle trying each voip and then
pstn in order, this is based around dialstatus (hackled from the
website).
My problem is when I have had long adsl problems (pain in the back side,
back to base alarm system) eventually asterisk seems to not want to talk
to the voip phone - nor does it allow any calls to be placed.
my guess is this
srvlookup=yes
which I had until recently - decided I don't really need it, my guess it
asterisk has a problem with name resolution and get stuck, but stuck
such that call processing can't happen.
snippet of my macro
[macro-dial-sipmnf-sippt-pstn]
;
; Enter with these
; ARG1 = number to dial
; ARG2 = timeout value
; ARG3 = flag determines if hangup or return on no answer
; HR = hangup and return (default)
; RT = return without hangup (must set)
;
; Returns with FOUNDME = DIALSTATUS
;
;
exten = s,1,Set(GLOBAL(FOUNDME)=ANSWER)
exten = s,2,Dial(SIP/${ARG1}${SIPMNF},${ARG2})
exten = s,3,Set(GLOBAL(FOUNDME)=${DIALSTATUS})
exten = s,4,GotoIf([${DIALSTATUS} = CHANUNAVAIL]?5:12)
;
exten = s,5,Set(GLOBAL(FOUNDME)=ANSWER)
exten = s,6,Dial(SIP/${ARG1}${SIPPT},${ARG2})
exten = s,7,Set(GLOBAL(FOUNDME)=${DIALSTATUS})
exten = s,8,GotoIf([${DIALSTATUS} = CHANUNAVAIL]?9:12)
;
exten = s,9,Set(GLOBAL(FOUNDME)=ANSWER)
exten = s,10,Dial(${PSTN}/${ARG1},${ARG2})
exten = s,11,Set(GLOBAL(FOUNDME)=${DIALSTATUS})
exten = s,12,Goto(s-${DIALSTATUS},1)
;
Alex
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Re: [asterisk-users] Question of resiliance
It's been my experience that when asterisk does a dns lookup, for externhost
or to do a SIP register, it blocks the whole server. Not sure if 1.6 has
that problem or just 1.4 though as my internet has been stable while im
awake these days
On Sun, Aug 30, 2009 at 5:54 PM, Alex Samad a...@samad.com.au wrote:
Hi
I am in the process of move a company from pstn to an asterisk setup.
They had 2 pstn lines - only really needed a max of 2 previously.
Now I have installed a tdm410 to handle the cross over from pabx to voip
handset. this has been done, the tdm is now just used to provide a
backup pstn line - only used as a last resort for outgoing calls - as
its shared with a fax line.
I use 2 voip providers one primarily for local and std and the other for
mobile calls, all though they can be backups for each other. and then
the pstn line
originally i had (and still till i get around to changing them) macros
to dial each interface and macro's that handle trying each voip and then
pstn in order, this is based around dialstatus (hackled from the
website).
My problem is when I have had long adsl problems (pain in the back side,
back to base alarm system) eventually asterisk seems to not want to talk
to the voip phone - nor does it allow any calls to be placed.
my guess is this
srvlookup=yes
which I had until recently - decided I don't really need it, my guess it
asterisk has a problem with name resolution and get stuck, but stuck
such that call processing can't happen.
snippet of my macro
[macro-dial-sipmnf-sippt-pstn]
;
; Enter with these
; ARG1 = number to dial
; ARG2 = timeout value
; ARG3 = flag determines if hangup or return on no answer
; HR = hangup and return (default)
; RT = return without hangup (must set)
;
; Returns with FOUNDME = DIALSTATUS
;
;
exten = s,1,Set(GLOBAL(FOUNDME)=ANSWER)
exten = s,2,Dial(SIP/${ARG1}${SIPMNF},${ARG2})
exten = s,3,Set(GLOBAL(FOUNDME)=${DIALSTATUS})
exten = s,4,GotoIf([${DIALSTATUS} = CHANUNAVAIL]?5:12)
;
exten = s,5,Set(GLOBAL(FOUNDME)=ANSWER)
exten = s,6,Dial(SIP/${ARG1}${SIPPT},${ARG2})
exten = s,7,Set(GLOBAL(FOUNDME)=${DIALSTATUS})
exten = s,8,GotoIf([${DIALSTATUS} = CHANUNAVAIL]?9:12)
;
exten = s,9,Set(GLOBAL(FOUNDME)=ANSWER)
exten = s,10,Dial(${PSTN}/${ARG1},${ARG2})
exten = s,11,Set(GLOBAL(FOUNDME)=${DIALSTATUS})
exten = s,12,Goto(s-${DIALSTATUS},1)
;
Alex
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Re: [asterisk-users] Question of resiliance
On Sun, Aug 30, 2009 at 06:49:06PM -0700, Kyle Kienapfel wrote: It's been my experience that when asterisk does a dns lookup, for externhost or to do a SIP register, it blocks the whole server. Not sure if 1.6 has that problem or just 1.4 though as my internet has been stable while im awake these days just for clarity I am running on 1.6 [snip] Alex signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
