Re: [asterisk-users] Asterisk w/ Nokia "e" Series Handsets
On Mon, May 11, 2009 at 11:24:36AM -0400, Cory Andrews wrote: > Anyone using Nokia "E" Series handsets with Asterisk? I'm trying to > deploy some e71's and am having an issue. I can get a single handset > working, but when I try to create a SIP profile on the second phone, it > won't allow me to save the profile, saying that devices in the same > "realm" must have identical username and password. > > Anyone have a workaround for this to add a second Nokia phone under the > Asterisk "realm" with a different userid and PW? I have an E71 and E61i working with asterisk in my office network without problems. I did have a different problem when I used a second SIP profile on the E61i with my home asterisk server and it would not register. Turned out I could not have the same realm name with the same username and password on the two profiles. Changing the realm name on my home profile and asterisk fixed the problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-user] Which policy for ISDN BRI support in NT/PtMP ?
On Tue, May 12, 2009 at 08:05:49AM +0200, Olivier wrote: > 2009/5/12 Kristijan Vrban > > > For those also need NT over PtMP, i started a initial patch for it. Very > > limited at the moment, only one incoming call to chan_dahdi from one > > device is possible. But i was pleasantly surprised that NT-ptmp is working > > anyway > > > > Get the patch here: http://bugs.digium.com/view.php?id=15048 Or rather: works in one direction: calls from a phone to the NT work. Calls in the other way don't make it. I believe it's much better than nothing, though, and I'm testing this patch in the new debs I have. > That is great news !!! > > How best can we contribute to make this happen ? Test this. Report how it (mis)behaves. And maybe try to trace why calls from the NT side don't get through. > Will the output most probably be a new libpri 1.4.X (or 1.6.X) or will it > also include a new Asterisk version ? For starters there will likely be some changes required in libpri . There is no libpri 1.6.x and not likely to be one in the near future. The "trunk" of libpri is branches/1.4 . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk w/ Nokia "e" Series Handsets
On Mon, May 11, 2009 at 11:24, Cory Andrews wrote: > Anyone using Nokia “E” Series handsets with Asterisk? I’m trying to deploy > some e71’s and am having an issue. I can get a single handset working, but > when I try to create a SIP profile on the second phone, it won’t allow me to > save the profile, saying that devices in the same “realm” must have > identical username and password. > > > > Anyone have a workaround for this to add a second Nokia phone under the > Asterisk “realm” with a different userid and PW? If you do mean a 2nd phone, it should not have anything to do with the 1st phone. Make sure your settings in the proxy server menu and registrar server menu are the same. Another thing, it is practically required to use the Nokia SIP VoIP settings program (download from europe.nokia.com, it is not listed at nokiausa.com). Unless you use that program to set count of voip digits (I set to 11) and ignoring domain section to digits only the call logs are a mess -- by default if you dial a VoIP call in the call log it shows it with @server at the end. If you then go out of WLAN coverage you can not redial that call. Overall, given the limitations of WiFi, it works rather well. I've never had to reboot my E71 or play with the settings after it was setup. Something I can't say about other WiFi (only) phones I have used. And VoIP on Windows mobile phones is crap. On Mon, May 11, 2009 at 11:27, Steve J. Douglas wrote: > In my case, I was trying to > add more than one SIP profiles for the same user account, but with > different access point. In that case, just setup the 1st profile and then select "WLAN found" (or "WLAN scanning off, twice) from the home screen, select search for WLAN and once you are connected you will see the option "Connect to PROFILE" in that same menu and it will automatically create it for you. ___ -- 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] DTMF received twice
hi
there is a file wich describe the dtmf duration it depends on every country.
you should look for that file and be sure you put you are on france
it works for inband but look at it...
Davod
2009/5/11 Administrator TOOTAI
> David fire a écrit :
> > out there is a file to change the dtmf duration
> > where are you?
> France
> > [...]
> > from other phones like lkand lines it works well?
> >
> No, the same. The called number is a number received by a trunk SIP, the
> GW is also setted as dtmfmode=auto. Calling from mobile phone or
> landline to other services using DTMF -like banks- is OK.
>
> I make further tests and so that setting dtmfmode=info for this GW make
> DTMF working correctly! Is this the normal behaviour?
>
> Our dialplan works great for others GW's, if this is normal we have to
> adapt it in case of dtmfmode=info. From where can we get the dtmf type?
> For me it looks like a bug.
>
> Thanks for your help.
>
> > 2009/5/11 Administrator TOOTAI
> >
> >
> >> Hi all,
> >>
> >> I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from
> >> my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so
> >> I can use the GW. For this I use:
> >>
> >> exten => s,1,NoOp(One of our workers (${CALLERID(number)}) is calling
> >> office) ;callerID is the one of the calling mobile phone
> >> exten => s,n,Background(silence/1)
> >>
> >> ; Nokia E65 send digits in DTMF mode, no need to take care about input
> >> corrections
> >> ;
> >> exten => s,n(enterDigits),Read(myExten,pls-entr-num-uwish2-call,0,,,3)
> >> exten => s,n,GotoIf($["${myExten}"=""]?enterDigits)
> >> [...]
> >>
> >> Problem is that received DTMF digits in ${myExten} are received twice eg
> >> for 1234 ${myExten} has 11223344. I correct the extension by dialplan
> >> but I think it's not really a solution.
> >>
> >> In sip.conf, the dtmfmode is set to auto. If I set it to rfc2833, the
> >> same behaviour.
> >>
> >> Can somebody confirm this before I open a bug, thanks.
> >>
> >> Regards
> >> --
> >> Daniel
>
>
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Re: [asterisk-users] Asterisk Manager API Action Originate
On 12/05/2009 3:44 p.m., Nicholas Blasgen wrote: > Has anyone else had issues with Originate returning the wrong error > code? According to the docs, the following errors are supposed to be > returned: > > 0 = no such extension or number > 1 = no answer > 4 = answered > 8 = congested or not available Are you referring to the originateresponse event? Which version of Asterisk? -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] DTMF received twice
On 12/05/2009 12:12 a.m., Administrator TOOTAI wrote: > Hi all, > > I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from > my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so Maybe one of these: http://bugs.digium.com/view.php?id=14815 http://bugs.digium.com/view.php?id=15024 -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] Wanting to manipulate SIP response headers
My boss has asked me if there is a way to send back a 503 response to a request at will. I don't see anything in the documents that would allow for manipulation in asterisk at that low of a level. Am I wrong? Bruce Ferrell ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager API Action Originate
Matt, Oh, I thought it was Asterisk 1.4.23 like I wrote in my first email, but turns out to be Asterisk SVN-branch-1.4-r191778. But yes, I am talking about originateresponse. I'm going to do some more debugging today to see if I can get the more information about the issue. When I either Originate from the CLI or from AMI, I don't get anything on the console for either the errors or the initial connection. I've had a lot of issues trying to debug Originate as a result. And no CDR logs are being recorded. On Tue, May 12, 2009 at 5:36 AM, Matt Riddell wrote: > On 12/05/2009 3:44 p.m., Nicholas Blasgen wrote: > > Has anyone else had issues with Originate returning the wrong error > > code? According to the docs, the following errors are supposed to be > > returned: > > > > 0 = no such extension or number > > 1 = no answer > > 4 = answered > > 8 = congested or not available > > Are you referring to the originateresponse event? > > Which version of Asterisk? > ___ -- 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] Wanting to manipulate SIP response headers
Bruce Ferrell schrieb: > My boss has asked me if there is a way to send back a 503 response to a > request at will. > > I don't see anything in the documents that would allow for manipulation > in asterisk at that low of a level. > > Am I wrong? Depends. Hangup(34) would send a 503 Service Unavailable on the SIP channel. See hangup_cause2sip() in chan_sip.c But in a way you are right. Hangup() works on the "call" level so you can't simply tell Asterisk to send a 503 response to an arbitrary request. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany -> http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Is anyone keeping up with the versions?
We are still using 1.4 and were going to start testing with 1.6.0, but then 1.6.1 was released and now 1.6.2 is already in beta 2. That seems like a lot of independent releases to maintain. I read about all the regressions ans hurried dot releases, makes us nervous. How is everyone doing their testing? -Matt ___ -- 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] Is anyone keeping up with the versions?
Pick a release and stick with it as long as you can. Only when you have to jump, pick a new release, test the hell out of it, and then leave it alone. Too many people try to keep on the latest release... _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thermal Wetland Sent: Tuesday, May 12, 2009 2:32 PM To: Asterisk Users List Subject: [asterisk-users] Is anyone keeping up with the versions? We are still using 1.4 and were going to start testing with 1.6.0, but then 1.6.1 was released and now 1.6.2 is already in beta 2. That seems like a lot of independent releases to maintain. I read about all the regressions ans hurried dot releases, makes us nervous. How is everyone doing their testing? -Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager API Action Originate
Matt & Others, So to continue the issue, here's what I've learned. Tested on Asterisk: 1.4.24.1 SVN 193870 SVN 191778 So I think that covers most everything. What I've learned is that any Timeout sends back a response code of ZERO instead of what I would have expected, ONE. Anyone offer any other suggestions to try? My way to test this was to make a simple script to perform an AMI Originate call with a 4 second timeout. I then have a standard tool to display all AMI Events. On every system I tried I would get Response of "Failure" and Error Code of ZERO. On Tue, May 12, 2009 at 10:13 AM, Nicholas Blasgen < nicho...@refractivedialer.com> wrote: > Matt, > > Oh, I thought it was Asterisk 1.4.23 like I wrote in my first email, but > turns out to be Asterisk SVN-branch-1.4-r191778. > > But yes, I am talking about originateresponse. I'm going to do some more > debugging today to see if I can get the more information about the issue. > When I either Originate from the CLI or from AMI, I don't get anything on > the console for either the errors or the initial connection. I've had a lot > of issues trying to debug Originate as a result. And no CDR logs are being > recorded. > > > On Tue, May 12, 2009 at 5:36 AM, Matt Riddell wrote: > >> On 12/05/2009 3:44 p.m., Nicholas Blasgen wrote: >> > Has anyone else had issues with Originate returning the wrong error >> > code? According to the docs, the following errors are supposed to be >> > returned: >> > >> > 0 = no such extension or number >> > 1 = no answer >> > 4 = answered >> > 8 = congested or not available >> >> Are you referring to the originateresponse event? >> >> Which version of Asterisk? >> > > ___ -- 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] Is anyone keeping up with the versions?
On Tue, May 12, 2009 at 2:31 PM, Thermal Wetland wrote: > We are still using 1.4 and were going to start testing with 1.6.0, but then > 1.6.1 was released and now 1.6.2 is already in beta 2. > > That seems like a lot of independent releases to maintain. I read about all > the regressions ans hurried dot releases, makes us nervous. > > How is everyone doing their testing? In my particular organization, I have a mix. * some systems that are about to get replaced are on 1.2 * other systems that have very high usage and we're rolled out about a year ago are on 1.4 * the newest applications that are live are on 1.6.0 series * the applications that are in pilot and will be replacing the 1.2 are on 1.6.1 When systems or applications have been getting scaled or upgraded or released, I've built and tested against whatever was current, or if I needed a future-release feature I grabbed that feature. All the code is there. You can use whatever you want as long as you are keeping up with the security alerts and you know you are mitigated or safe. ___ -- 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] Is anyone keeping up with the versions?
Unless there is a new feature or your making a new system. Don't fix it if it aint broke. BUT do stay current on reading about new feature and things in the releases. James Shigley From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Tuesday, May 12, 2009 1:45 PM To: 'Asterisk Users List' Subject: Re: [asterisk-users] Is anyone keeping up with the versions? Pick a release and stick with it as long as you can. Only when you have to jump, pick a new release, test the hell out of it, and then leave it alone. Too many people try to keep on the latest release... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thermal Wetland Sent: Tuesday, May 12, 2009 2:32 PM To: Asterisk Users List Subject: [asterisk-users] Is anyone keeping up with the versions? We are still using 1.4 and were going to start testing with 1.6.0, but then 1.6.1 was released and now 1.6.2 is already in beta 2. That seems like a lot of independent releases to maintain. I read about all the regressions ans hurried dot releases, makes us nervous. How is everyone doing their testing? -Matt ___ -- 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] Hangup()-command does not hang up the line
When I call my Asterisk-server from my cell phone on one of the PSTN-numbers that terminate in a FXO-module on my TDM410P Digium card, and in the dialplan the end of a context is reached and Asterisk needs to execute the Hangup()-command, I notice the following : - Asterisk tells me that the conversation was hung up (the log files tell me the command was executed) - On my cell phone I hear silence, no special tone on the line that tells me the call was terminated by Asterisk, AND time keeps on counting on my cell phone as if the duration of the conversation continues. I see the following solution : - At the end of my context, I initiate the Congestion()-application to force the caller to hang up. But I think it must be enough just to call the Hangup()-command to make Asterisk terminate the conversation... But as I said : on my cell phone I see that time keeps on counting as if I'm still connected + no tone that the line was hung up. On the other hand : Asterisk detects the other end really good and registers when the caller has put down his phone and the conversation is terminated by the caller. Also a fax and a busy-tone is well detected. The option busydetect=yes is set in my chan_dahdi.conf... But this is not the problem. Is this a bug ?? Greetingz, Jonas. ___ -- 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] Wanting to manipulate SIP response headers
Philipp Kempgen wrote: > Bruce Ferrell schrieb: >> My boss has asked me if there is a way to send back a 503 response to a >> request at will. >> >> I don't see anything in the documents that would allow for manipulation >> in asterisk at that low of a level. >> >> Am I wrong? > > Depends. > Hangup(34) would send a 503 Service Unavailable on the SIP channel. > See hangup_cause2sip() in chan_sip.c > > But in a way you are right. Hangup() works on the "call" level so > you can't simply tell Asterisk to send a 503 response to an arbitrary > request. > > > Philipp Kempgen That will actually serve to do what I need. Thanks! ___ -- 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] Hangup()-command does not hang up the line
I would try hanguponpolarityswitch=yes in my dadhi.conf. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Tuesday, May 12, 2009 3:09 PM To: Asterisk Mailing Subject: [asterisk-users] Hangup()-command does not hang up the line When I call my Asterisk-server from my cell phone on one of the PSTN-numbers that terminate in a FXO-module on my TDM410P Digium card, and in the dialplan the end of a context is reached and Asterisk needs to execute the Hangup()-command, I notice the following : - Asterisk tells me that the conversation was hung up (the log files tell me the command was executed) - On my cell phone I hear silence, no special tone on the line that tells me the call was terminated by Asterisk, AND time keeps on counting on my cell phone as if the duration of the conversation continues. I see the following solution : - At the end of my context, I initiate the Congestion()-application to force the caller to hang up. But I think it must be enough just to call the Hangup()-command to make Asterisk terminate the conversation... But as I said : on my cell phone I see that time keeps on counting as if I'm still connected + no tone that the line was hung up. On the other hand : Asterisk detects the other end really good and registers when the caller has put down his phone and the conversation is terminated by the caller. Also a fax and a busy-tone is well detected. The option busydetect=yes is set in my chan_dahdi.conf... But this is not the problem. Is this a bug ?? Greetingz, Jonas. ___ -- 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] Hangup()-command does not hang up the line
On Tue, 12 May 2009, jonas kellens wrote: > When I call my Asterisk-server from my cell phone on one of the > PSTN-numbers that terminate in a FXO-module on my TDM410P Digium card, > and in the dialplan the end of a context is reached and Asterisk needs > to execute the Hangup()-command, I notice the following : > > - Asterisk tells me that the conversation was hung up (the log files > tell me the command was executed) > - On my cell phone I hear silence, no special tone on the line that > tells me the call was terminated by Asterisk, AND time keeps on counting > on my cell phone as if the duration of the conversation continues. Replace the Asterisk box with a standard analogue phone and see what happens. I suspect you'll see the same. It happens in the UK too. The line will eventually clear, but it may take some time. I used to use it to "transfer" a call - ie. just put the phone back on-hook, then go to another phone and lift it... And you'll see it on old films where the bad guy phones a house and loads up the payphone with lots of money to stop the house being able to hang up the call and dial 999 ... Some exchanges do seem to clear the call much quicker now, but I think it's pot-luck, depending on the exchange and maybe hardware/software they have... Gordon ___ -- 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] enum agi interesting problem
Hi,
I am having a strange problem with enum and AGI.
Here is what happens:
I have in my agi something like that:
foreach my $resolver ("e164.arpa", "e164.info", "e164.org") {
my @enums = get_enums($phone, $resolver);
foreach my $enum (@enums) {
$dialstring = $enum . "|90|HL(" .
($maxtime * 60 * 1000) . ":6:3)";
$res = $AGI->exec("DIAL $dialstring");
$answeredtime =
$AGI->get_variable("ANSWEREDTIME");
$dialstatus =
$AGI->get_variable("DIALSTATUS");
print LOGFILE "Dialstring: $dialstr
DIALSTATUS: $dialstatus\n";
$callstart = time();
if ($dialstatus eq "ANSWERED") { last; }
}
}
}
Here's the output from my logfile:
Call 1:
Dialstring:
sip/16416418003569...@tollfree.sip-happens.com|90|HL(576:6:3)
DIALSTATUS:
Dialstring:
sip/16416418003569...@sip.tollfreegateway.com|90|HL(576:6:3)
DIALSTATUS:
Dialstring: sip/18003569...@tf.voipmich.com|90|HL(576:6:3)
DIALSTATUS: ANSWER
Call 2:
Dialstring: sip/18002662...@tf.voipmich.com|90|HL(576:6:3)
DIALSTATUS:
Dialstring:
sip/16416418002662...@sip.tollfreegateway.com|90|HL(576:6:3)
DIALSTATUS:
Dialstring:
sip/16416418002662...@tollfree.sip-happens.com|90|HL(576:6:3)
DIALSTATUS: ANSWER
And so on.
The call gets answered the first time (call 1 - through sip-happens, call 2,
through voipmich).
Problem is that after I hang up , it doesn't return a status, so it cycles
through the loop and dials the rest of the entries. The last one gets
dialstatus.
I believe it's a stupid mistake but I cannot think of anything right now.
Any ideas?
Thanks,
Dan
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Re: [asterisk-users] enum agi interesting problem
Forget the typo (s/ANSWERED/ANSWER/g)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Caescu
Sent: Tuesday, May 12, 2009 7:07 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] enum agi interesting problem
Hi,
I am having a strange problem with enum and AGI.
Here is what happens:
I have in my agi something like that:
foreach my $resolver ("e164.arpa", "e164.info", "e164.org") {
my @enums = get_enums($phone, $resolver);
foreach my $enum (@enums) {
$dialstring = $enum . "|90|HL(" .
($maxtime * 60 * 1000) . ":6:3)";
$res = $AGI->exec("DIAL $dialstring");
$answeredtime =
$AGI->get_variable("ANSWEREDTIME");
$dialstatus =
$AGI->get_variable("DIALSTATUS");
print LOGFILE "Dialstring: $dialstr
DIALSTATUS: $dialstatus\n";
$callstart = time();
if ($dialstatus eq "ANSWERED") { last; }
}
}
}
Here's the output from my logfile:
Call 1:
Dialstring:
sip/16416418003569...@tollfree.sip-happens.com|90|HL(576:6:3)
DIALSTATUS:
Dialstring:
sip/16416418003569...@sip.tollfreegateway.com|90|HL(576:6:3)
DIALSTATUS:
Dialstring: sip/18003569...@tf.voipmich.com|90|HL(576:6:3)
DIALSTATUS: ANSWER
Call 2:
Dialstring: sip/18002662...@tf.voipmich.com|90|HL(576:6:3)
DIALSTATUS:
Dialstring:
sip/16416418002662...@sip.tollfreegateway.com|90|HL(576:6:3)
DIALSTATUS:
Dialstring:
sip/16416418002662...@tollfree.sip-happens.com|90|HL(576:6:3)
DIALSTATUS: ANSWER
And so on.
The call gets answered the first time (call 1 - through sip-happens, call 2,
through voipmich).
Problem is that after I hang up , it doesn't return a status, so it cycles
through the loop and dials the rest of the entries. The last one gets
dialstatus.
I believe it's a stupid mistake but I cannot think of anything right now.
Any ideas?
Thanks,
Dan
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[asterisk-users] no source on cdr logs in some cases!!
Hello,
I was experiencing a problem like not seeing source info or caller id on
some calls.
When i make a little reserach i figured out that if my dialin plan is like
this:
"exten = _00.,1,Macro(trunkdial-failover-0.3
${span_1}/9${EXTEN:0},,span_1,span_1)"
I see no source on logfiles.. Sure i have the other fields.
If i change the rule to "exten => _00.,1,Dial(DAHDI/g1/9${EXTEN})"
everythings goes normal.
Whats the difference between them(except macro command for sure) or what
can cause such problem
PS: I also have the exact source info in Channel tab of the database as
SIP/
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Re: [asterisk-users] enum agi interesting problem
Maybe it is something to do with AGI - Dial command.
IFAIK you can't control Dial via AGI script.
>From http://www.voip-info.org/wiki/view/Asterisk+AGI :
Dialing out
If the AGI application dials outward by executing Dial, control over
the call returns to the dialplan and the script loses contact with the
Asterisk server. The script continues to run in the background by
itself and is free to clean up and do post-dial processing.
If you want your application to initiate a call out without being
started through the dialplan:
* Asterisk auto-dial out Move (not copy) a file into an Asterisk
spool directory and a call will be placed
* Asterisk Manager API Use the Originate command
Regards,
Chris
2009/5/13 Dan Caescu :
> Forget the typo (s/ANSWERED/ANSWER/g)
>
>
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Caescu
> Sent: Tuesday, May 12, 2009 7:07 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [asterisk-users] enum agi interesting problem
>
>
>
> Hi,
>
>
>
> I am having a strange problem with enum and AGI.
>
>
>
> Here is what happens:
>
>
>
> I have in my agi something like that:
>
>
>
> foreach my $resolver ("e164.arpa", "e164.info", "e164.org") {
>
> my @enums = get_enums($phone, $resolver);
>
> foreach my $enum (@enums) {
>
> $dialstring = $enum . "|90|HL(" .
> ($maxtime * 60 * 1000) . ":6:3)";
>
> $res = $AGI->exec("DIAL $dialstring");
>
> $answeredtime =
> $AGI->get_variable("ANSWEREDTIME");
>
>
>
> $dialstatus =
> $AGI->get_variable("DIALSTATUS");
>
> print LOGFILE "Dialstring: $dialstr
> DIALSTATUS: $dialstatus\n";
>
>
>
> $callstart = time();
>
> if ($dialstatus eq "ANSWERED") { last; }
>
> }
>
> }
>
> }
>
>
>
> Here’s the output from my logfile:
>
>
>
> Call 1:
>
>
>
> Dialstring:
> sip/16416418003569...@tollfree.sip-happens.com|90|HL(576:6:3)
> DIALSTATUS:
>
> Dialstring:
> sip/16416418003569...@sip.tollfreegateway.com|90|HL(576:6:3)
> DIALSTATUS:
>
> Dialstring: sip/18003569...@tf.voipmich.com|90|HL(576:6:3)
> DIALSTATUS: ANSWER
>
>
>
> Call 2:
>
>
>
> Dialstring: sip/18002662...@tf.voipmich.com|90|HL(576:6:3)
> DIALSTATUS:
>
> Dialstring:
> sip/16416418002662...@sip.tollfreegateway.com|90|HL(576:6:3)
> DIALSTATUS:
>
> Dialstring:
> sip/16416418002662...@tollfree.sip-happens.com|90|HL(576:6:3)
> DIALSTATUS: ANSWER
>
>
>
> And so on.
>
> The call gets answered the first time (call 1 – through sip-happens, call 2,
> through voipmich).
>
> Problem is that after I hang up , it doesn’t return a status, so it cycles
> through the loop and dials the rest of the entries. The last one gets
> dialstatus.
>
>
>
> I believe it’s a stupid mistake but I cannot think of anything right now.
>
>
>
> Any ideas?
>
>
>
> Thanks,
>
> Dan
>
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>
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