Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

Ok, I also tried to hangup directly through dialplan, it doesn't work. == Using SIP RTP CoS mark 5 -- Executing [12023300643@default:1] Hangup("SIP/66.226.76.70-d0b0", "41") in new stack == Spawn extension (default, 12023300643, 1) exited non-zero on 'SIP/66.226.76.70-d0b0' ==

Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

On Mon, Feb 13, 2017, at 08:58 PM, Anas Moiz wrote: > Yes Joshua, Its SIP and but the problem is I have tried everything but it > doesn't seem to work. > > In the SIP Trace I can see that I am sending 503 Service Unavailable as a > response. > > You can check the SIP trace attached below: > >

Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

Yes Joshua, Its SIP and but the problem is I have tried everything but it doesn't seem to work. In the SIP Trace I can see that I am sending 503 Service Unavailable as a response. You can check the SIP trace attached below: 162.243.107.173:5060 -> 66.226.76.70:5060 SIP/2.0 503 Service

Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

On Mon, Feb 13, 2017, at 05:46 PM, Anas Moiz wrote: > Hi Everyone, > > I am dealing with a problem for now and its really annoying. > > I want to hangup calls from AGI but it seems that my AGI is not rejecting > the calls properly. > > { >

[asterisk-users] Certified Asterisk 13.13-cert1 Now Available

The Asterisk Development Team has announced the release of Certified Asterisk 13.13-cert1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 13.13-cert1 resolves several issues reported by the

[asterisk-users] Asterisk 14.3.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 14.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.3.0 resolves several issues reported by the community and would have not been possible

[asterisk-users] Asterisk 13.14.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 13.14.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.14.0 resolves several issues reported by the community and would have not been possible

[asterisk-users] CALLS NOT HANGING UP THROUGH AGI

Hi Everyone, I am dealing with a problem for now and its really annoying. I want to hangup calls from AGI but it seems that my AGI is not rejecting the calls properly. { $agi->verbose("number-not-in-service"); $agi->exec("Congestion","1");

Re: [asterisk-users] First SIP-registering succeeds, second doesn't

+1! This sounds an awful lot like an ALG doing it best to 'help'... > On 14/02/2017, at 6:38 am, Israel Gottlieb wrote: > > Disable all sip alg/helpers in the router smime.p7s Description: S/MIME cryptographic signature --

Re: [asterisk-users] ATA Adapter YGW30 - manual

Behold: The Wayback Machine. Link to manual: http://web.archive.org/web/20070224144946/http://www.yntx.com/files/YGW30en.rar Manual says user/pass is root/test. -- Nathan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]

Re: [asterisk-users] First SIP-registering succeeds, second doesn't

Disable all sip alg/helpers in the router   Original Message   From: andregronwal...@gmail.com Sent: February 13, 2017 6:40 PM To: asterisk-users@lists.digium.com Reply-to: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] First SIP-registering succeeds, second doesn't Some further

Re: [asterisk-users] Disallow CALLS without registry

sorry NoOP(${DB_EXISTS(SIP/Registry/${CHANNEL(peername)})}); 2017-02-13 19:31 GMT+02:00 Антон Сацкий : > THINK i found a solution > > NoOP(${DB_EXISTS(/SIP/Registry/${CHANNEL(peername)})}); > > THANKS TO ALL > > 2017-02-12 12:34 GMT+02:00 Frank Vanoni

Re: [asterisk-users] Disallow CALLS without registry

THINK i found a solution NoOP(${DB_EXISTS(/SIP/Registry/${CHANNEL(peername)})}); THANKS TO ALL 2017-02-12 12:34 GMT+02:00 Frank Vanoni : > On Sat, 2017-02-11 at 12:25 +1300, Pete Mundy wrote: > > > > sip.conf configuration > > > In the [general] section, define: > >

[asterisk-users] ATA Adapter YGW30 - manual

I have one: YGW30 1FXS,1FXO SIP ATA unit it was made by company Yuxin I think they are no longer in business. I forgot the default user name / password for log-in. Does anybody know what was the default login or have a manual? -- Regards, Joseph --

Re: [asterisk-users] First SIP-registering succeeds, second doesn't

Some further information: asterisk version: 13.13.1, pjsip (pjproject) 2.5.5 regards, andre Am 13.02.2017 um 17:32 schrieb Andre Gronwald: Hi all, I have a strange issue, with a some kind complicate architecture... A router of our internet provider is in front of another bintec rs353j

[asterisk-users] First SIP-registering succeeds, second doesn't

Hi all, I have a strange issue, with a some kind complicate architecture... A router of our internet provider is in front of another bintec rs353j router, at which my freepbx installation is located. However, NAT etc. seems to work fine. BUT: Something is not working...: When registering my