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after upgrade from 1.4.26.2 to 1.8.11.0 my provider gets rport instead of port

if you check out your sip.conf.

On Jun 29, 2012, at 5:54 PM, gincantalupo wrote:

> Hi all,
>
> after upgrading my Asterisk 1.4.26.2 to 1.8.11.0 I cannot register to my VoIP provider because it says I’m trying to connect to port 55150 (that’s what the call center guy told me)…but I’m not. In my sip I’ve set port=5060, not 55150.
> The strange thing is that the rport inside SIP packets (“sip set debug”) coming back from my provider is set to 55150…..seen on both Asterisk 1.4 and 1.8
>
> Does anybody have any idea?
>
> Thank you.
>
> Giorgio
>
> –
> _____________________________________________________________________
> — Bandwidth and Colocation Provided by http://www.api-digital.com
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> http://www.asterisk.org/hello
>
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FSK ETSI or FSK US

Thanks Mitul :)

The patch on the link is so old (2006-2007) so I think it’s already
implemented in the newest version. Honestly to say, I already try any
combitions but still the caller id doesn’t work :(

cidsignalling=bell,dtmf,v23
cidstart=ring,polarity,dtmf

with some parameter if we set it to dtmf

Hopeless :((

On Sun, Jun 3, 2012 at 4:51 PM, Mitul Limbani wrote:

> Welcome to da Matrix :)
>
> Look at this issue : https://issues.asterisk.org/view.php?id=6683
>
> And try different combinations suggested over there, you might get lucky :)
>
> Regards,
> Mitul Limbani,
> Chief Architech & Founder,
> Enterux Solutions Pvt. Ltd.
> 110 Reena Complex, Opp. Nathani Steel,
> Vidyavihar (W), Mumbai – 400 086. India
> http://www.enterux.com/
> http://www.entvoice.com/
> email: mitul@enterux.in
> DID: +91-22-61447605
> Cell: +91-9820332422
>
>
>
>
> On Sun, Jun 3, 2012 at 3:08 PM, Satria Anamarta wrote:
>
>> Hello, All :)
>>
>> Regarding to incoming caller ID on PSTN line, which one is best supported
>> by asterisk: is it FSK ETSI or FSK US?
>> I bought some caller ID converter hardware (convert DTMF to FSK and vice
>> versa) but still asterisk can not detect it.
>> The converter has a switch FSK ETSI or FSK US
>>
>> This is what I put in /etc/asterisk/chan_dahdi.conf
>> …
>> cidsignalling=bell
>> cidstart=ring
>> …
>>
>> If after buying this converter hardware and upgrade to dahdi 2.6.1 still
>> not solve my caller id problem, I really dont know what to do and feel
>> hopeless :(
>>
>> Thanks,
>> Anam.
>>
>>
>> –
>> _____________________________________________________________________
>> — 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
>

Queue callers with Callback option without lose their place

Known as Virtual Hold, you’ll have to program inside asterisk to achieve
that.

El 31/05/12 10:48, equis software escribió:
> Is there any option in Asterisk distribution of this?
>
> Thanks.
>
>
> –
> _____________________________________________________________________
> — 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

extension status using AMI

Why don’t you use AMI? There’s are phpami project if you google.

Sent from my iPhone

On May 25, 2012, at 1:51 AM, Kamlesh Kumar wrote:

> Hi,
>
> I’m using AMI to get the extension status but always get -1 i.e. extension not found.
>
> #!/usr/bin/php -q
> < ?php
> include_once (“phpagi-2.14/phpagi.php”);
> include_once (“/phpagi-2.14/phpagi-asmanager.php”);
> $agi = new AGI();
> $as = new AGI_AsteriskManager();
> $exten = $agi->request['agi_extension'];
> $as->connect(“localhost”, “user”, “passwd”);
> $status = $as->ExtensionState($exten,’context’,1);
> $status1 = $status['Status'];
> $agi->verbose(“Extension status is “.$status1);
> ?>
>
> Always return Extension status is -1
>
> Thanks,
> Kamlesh
>
> –
> _____________________________________________________________________
> — 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

Deleting OLD Voicemails

Thanks Jason,

But how to delete them? there are a lot of old voicemails, but i don’t want
to break the app_voicemail.

2012/5/22 Jason Parker

> On 05/22/2012 04:54 PM, Danny Dias wrote:
> > There are 4 files for each voicemail:
> >
> > msg0000.gsm
> > msg0000.txt
> > msg0000.wav
> > msg0000.WAV
> >
>
> That is perfectly normal. The .txt file is metadata that contains things
> like
> caller ID and duration. Asterisk will also save voicemails into every
> format
> you have specified in voicemail.conf.
>
> –
> _____________________________________________________________________
> — 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
>

50% of time SendDTMF failed

I suggest you put in dtmfmode=auto (so rfc2833 / inband ) can be selected
dynamically.

Wanted to check with the community if this feature holds true on latest
versions of Asterisk ?

Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai – 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mitul@enterux.in
DID: +91-22-61447605
Cell: +91-9820332422

On Fri, May 18, 2012 at 10:11 PM, Ing CIP. Alejandro Celi > wrote:

> **
>
> Did you try putting inband parameter in dtmfmode and dtmf of your sip.conf?
>
> Regards,
>
>
> –
> Ing CIP. Alejandro Celi Mariátegui
>
> http://cipher.pe/web/asterisk.html
>
>
> El mié, 16-05-2012 a las 16:07 +0100, Shahid H escribió:
>
> I am having a problem with SendDTMF() – 50% of time it did not succeed.
>
>
>
> I suspect it is not sending clear DTMF tones to the IVR.
>
>
>
> For example:
>
>
>
> SendDTMF(wwwww3wwwww2wwwwww1wwwww4)
>
>
>
> Sometime digit 3 and 2 work, and failed to do digit 1.
>
> Sometime digit 3 work and failed to do number 2.
>
> Sometime all went through fine.
>
>
>
> dtmfmode=rfc2833 are set in the sip.conf file
>
>
>
> How do I debug to see what went wrong and how to fix?
>
>
>
> Asterisk 1.8.12.0
>
> Installed on VPS (XEN, CentOS 5.x, 768 MB Ram, 1000 GB B/W – Located in
> UK)
>
> VOIP Provider in UK.
>
>
>
> Thanks
>
> –_____________________________________________________________________– 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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-use http://lists.digium.com/mailman/listinfo/asterisk-usersrs
>
>
>
>
> –
> _____________________________________________________________________
> — 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
>