Re: [asterisk-users] Asterisk crashing on AAAA lookup

On Tue, Jun 26, 2018 at 6:15 PM, Dovid Bender wrote: > I have Asterisk running on a Ubuntu 18.0.4 on Digital Ocean. Every so > often asterisk crashes and then restarts. I am not seeing any core dumps on > the box. The only I thing I see every time is a second before Asterisk > crashes there is a

Re: [asterisk-users] Asterisk not matching longest prefix with include

On Tue, Jun 26, 2018 at 7:40 PM, Joshua Colp wrote: > On Tue, Jun 26, 2018, at 8:31 PM, Dovid Bender wrote: > > On Tue, Jun 26, 2018 at 7:28 PM, Doug Lytle wrote: > > > > > On 06/26/2018 07:20 PM, Dovid Bender wrote: > > > > > > Doug, > > > > > > I tried that as well. Even with my dialplan

Re: [asterisk-users] Asterisk not matching longest prefix with include

On Tue, Jun 26, 2018, at 8:31 PM, Dovid Bender wrote: > On Tue, Jun 26, 2018 at 7:28 PM, Doug Lytle wrote: > > > On 06/26/2018 07:20 PM, Dovid Bender wrote: > > > > Doug, > > > > I tried that as well. Even with my dialplan looking like this: > > > > > > > > Ordering by includes works for me

Re: [asterisk-users] Asterisk not matching longest prefix with include

On Tue, Jun 26, 2018 at 7:28 PM, Doug Lytle wrote: > On 06/26/2018 07:20 PM, Dovid Bender wrote: > > Doug, > > I tried that as well. Even with my dialplan looking like this: > > > > Ordering by includes works for me under Asterisk 11 and 13 > > What does the output of the below show? > >

Re: [asterisk-users] Asterisk not matching longest prefix with include

On 06/26/2018 07:20 PM, Dovid Bender wrote: Doug, I tried that as well. Even with my dialplan looking like this: Ordering by includes works for me under Asterisk 11 and 13 What does the output of the below show? dialplan show from-external Doug --

Re: [asterisk-users] Asterisk not matching longest prefix with include

On Tue, Jun 26, 2018 at 7:06 PM, Doug Lytle wrote: > On 06/26/2018 06:57 PM, Dovid Bender wrote: > >> Hi, >> >> My dialplan looks like this: >> [from-external] >> >> Exten => _X.,1,Noop(CALL IS COMING INTO FROM EXTERNAL CONTEXT) >> Exten => _X.,n,Noop(IP: ${CHANNEL(recvip)}) >> Exten =>

[asterisk-users] Asterisk crashing on AAAA lookup

I have Asterisk running on a Ubuntu 18.0.4 on Digital Ocean. Every so often asterisk crashes and then restarts. I am not seeing any core dumps on the box. The only I thing I see every time is a second before Asterisk crashes there is a lookup for the boxes hostname. As soon as it gets the

Re: [asterisk-users] Asterisk not matching longest prefix with include

On 06/26/2018 06:57 PM, Dovid Bender wrote: Hi, My dialplan looks like this: [from-external] Exten => _X.,1,Noop(CALL IS COMING INTO FROM EXTERNAL CONTEXT) Exten => _X.,n,Noop(IP: ${CHANNEL(recvip)}) Exten => _X.,n,Noop(CALLED NUMBER: ${EXTEN}) Exten => _X.,n,Ringing Exten =>

[asterisk-users] Asterisk not matching longest prefix with include

Hi, My dialplan looks like this: [from-external] Exten => _X.,1,Noop(CALL IS COMING INTO FROM EXTERNAL CONTEXT) Exten => _X.,n,Noop(IP: ${CHANNEL(recvip)}) Exten => _X.,n,Noop(CALLED NUMBER: ${EXTEN}) Exten => _X.,n,Ringing Exten => _X.,n,WaitExten(15) Exten => _X.,n,Congestion() Exten =>

[asterisk-users] Asterisk receiving 415 Unsupported Media Type upon T.38 invite behaving absolutely weird.

Hello I am hunting Fax Problems. Now I have come across a situation on which, I fear asterisk behaves in a wrong manner.. A T.38 enabled ATA is connected to the asterisk and receiving a call from a non T.38 capable endpoint. The ATA is detecting the CED tone and initiating a T.38 re-invite.

[asterisk-users] Asterisk Realtime PJSIP - slow output on "pjsip show xxxxx" commands

Hi all! I have an Asterisk instance (15.4.1) with a MySQL DB in realtime configuration (using MySQL ODBC connector). I noticed, that when I issue “pjsip show endpoints” in the CLI this takes forever (on average about 1 minute with 1029 entries in ps_endpoints table). If I query the database

[asterisk-users] [Asterisk-video] (no subject)

http://fit.diggitradio.com Pankaj Pandey -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] Asterisk kafka connector feedback

Dear all, I recently developed a connector for Kafka and Asterisk AMI. The purpose is to send Asterisk AMI events, e.g. CDRs to a Kafka topic. This could have many uses, like running billing or analytics through Kafka streams in real time.

Re: [asterisk-users] Asterisk TLS 5061 not listening

On Mon, May 28, 2018 at 09:26:32AM +0200, Benjamin Marty wrote: > I'm currently trying to setup an Asterisk Box with a Let's Encrypt > certificate. > > I merged privatekey, cert and chain to one file: This is not necessary. You could use tlscertfile and tlsprivatekey. > tlsenable=yes >

[asterisk-users] Asterisk TLS 5061 not listening

I'm currently trying to setup an Asterisk Box with a Let's Encrypt certificate. I merged privatekey, cert and chain to one file: cat /etc/letsencrypt/live/domain/privkey.pem > /etc/asterisk/tls/a-keycert.pem cat /etc/letsencrypt/live/domain/fullchain.pem >> /etc/asterisk/tls/a-keycert.pem My

[asterisk-users] asterisk rules

mailman drools -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here:

[asterisk-users] Asterisk 15.4.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 15.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.4.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.21.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.21.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.21.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk Community Services Outages

Dear Asterisk Community, For the past 24 hours or so, Digium’s upstream provider has had a few outages that have affected Asterisk community services, including Asterisk.org, the mailing lists, and potentially other services. We apologize for any inconvenience that it has caused. Hopefully

Re: [asterisk-users] Asterisk behind NAT Early Media Video

On Fri, Apr 13, 2018, at 11:56 AM, Benjamin Marty wrote: > The current behaviour is that Earlymedia video isn't working when NAT's in > between are involved. The source/destination IP's are correct. So the > client is sending Early media video + Early media audio to the Asterisk > Server "in the

Re: [asterisk-users] Asterisk behind NAT Early Media Video

The current behaviour is that Earlymedia video isn't working when NAT's in between are involved. The source/destination IP's are correct. So the client is sending Early media video + Early media audio to the Asterisk Server "in the cloud" and the Asterisk Server "in the cloud" is sending both to

Re: [asterisk-users] Asterisk behind NAT Early Media Video

On Wed, Apr 11, 2018, at 4:33 AM, Benjamin Marty wrote: > I added the bind_rtp_to_media_address=yes on all endpoints but still the > same behaviour. The funny thing is that the G711 audio early media works > and doesn't have that Private IP issue. I was also able to cross check with > chan_sip on

Re: [asterisk-users] Asterisk behind NAT Early Media Video

> > *Von:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] *Im Auftrag von *Benjamin Marty > *Gesendet:* Mittwoch, 11. April 2018 08:55 > *An:* Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com>

Re: [asterisk-users] Asterisk behind NAT Early Media Video

, 11. April 2018 08:55 An: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Video I think I found the root cause. The H264 Early Media video is received successfully on the Asterisk Serve

Re: [asterisk-users] Asterisk behind NAT Early Media Video

t; >>> >>> >>> >>> With best regards >>> >>> Florian Floimair >>> Innovation - Software-Development - VoIP & DevOps >>> >>> COMMEND INTERNATIONAL GMBH >>> A-5020 Salzburg, Saalachstraße 51 >>> Tel:

Re: [asterisk-users] Asterisk behind NAT Early Media Video

ONAL GMBH >> A-5020 Salzburg, Saalachstraße 51 >> Tel: +43-662-85 62 25 >> Fax: +43-662-85 62 26 >> http://www.commend.com >> >> Security and Communication by Commend >> >> FN 178618z | LG Salzburg >> >> -Ursprüngliche Nachricht- >> Von

Re: [asterisk-users] Asterisk behind NAT Early Media Video

Ursprüngliche Nachricht- > Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] Im Auftrag von Joshua Colp > Gesendet: Montag, 9. April 2018 18:15 > An: asterisk-users@lists.digium.com > Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Vi

Re: [asterisk-users] Asterisk behind NAT Early Media Video

I applied the patch to my Asterisk 13.20. But it seems that it still doesn't forward the early media video stream. Do I need to put something special into the extensions.conf? I basically just make a Dial. The calling Client sends the 183 protocol. [public] exten => 6001,1,Dial(SIP/${EXTEN})

Re: [asterisk-users] Asterisk behind NAT Early Media Video

: Re: [asterisk-users] Asterisk behind NAT Early Media Video On Mon, Apr 9, 2018, at 1:05 PM, Benjamin Marty wrote: > wohoo, so if I unterstand it correctly with that patch early media > video works over the Asterisk server? In other words the Asterisk > server get's able to (process/)forward th

Re: [asterisk-users] Asterisk behind NAT Early Media Video

On Mon, Apr 9, 2018, at 1:05 PM, Benjamin Marty wrote: > wohoo, so if I unterstand it correctly with that patch early media video > works over the Asterisk server? In other words the Asterisk server get's > able to (process/)forward the early media video stream with that patch? The patch forwards

Re: [asterisk-users] Asterisk behind NAT Early Media Video

wohoo, so if I unterstand it correctly with that patch early media video works over the Asterisk server? In other words the Asterisk server get's able to (process/)forward the early media video stream with that patch? 2018-04-09 17:57 GMT+02:00 Joshua Colp : > On Mon, Apr 9,

Re: [asterisk-users] Asterisk behind NAT Early Media Video

On Mon, Apr 9, 2018, at 12:04 PM, Benjamin Marty wrote: > My understanding based on Wireshark analysis is that the signaling works > (also the recipent phone is displaying the video frame before accepting the > call), also the calling phone send video (i see that also via Wireshark) > but the

Re: [asterisk-users] Asterisk behind NAT Early Media Video

My understanding based on Wireshark analysis is that the signaling works (also the recipent phone is displaying the video frame before accepting the call), also the calling phone send video (i see that also via Wireshark) but the recipent phone doesn't get any video from the Asterisk before the

Re: [asterisk-users] Asterisk behind NAT Early Media Video

On Mon, Apr 9, 2018, at 11:53 AM, Benjamin Marty wrote: > Yes, media is flowing through Asterisk because both client's are behind > different NAT's. This doesn't answer the question of what is ACTUALLY happening in the scenario you describe which is very important. > Do I need to do something

Re: [asterisk-users] Asterisk behind NAT Early Media Video

Yes, media is flowing through Asterisk because both client's are behind different NAT's. Do I need to do something special in the Call Flow? Or anything additional to the pjsip.conf? 2018-04-09 16:50 GMT+02:00 Joshua Colp : > On Mon, Apr 9, 2018, at 11:42 AM, Benjamin Marty

Re: [asterisk-users] Asterisk behind NAT Early Media Video

On Mon, Apr 9, 2018, at 11:42 AM, Benjamin Marty wrote: > Hello, > > I have an Asterisk 15 with PJSIP behind NAT (Amazon EC2). > > Now I would like to get Early Media Video working between clients in > different NATed networks. The 183 signalling goes trough perfectly, but > asterisk doesn't

[asterisk-users] Asterisk behind NAT Early Media Video

Hello, I have an Asterisk 15 with PJSIP behind NAT (Amazon EC2). Now I would like to get Early Media Video working between clients in different NATed networks. The 183 signalling goes trough perfectly, but asterisk doesn't forward the Early Media RTP stream from the caller to the recipent. I

Re: [asterisk-users] Asterisk / PRI and Outbound Overlap Dialing

Thanks Richard, That did the trick. I had the 'overlapdial=yes' line farther down in the chan_dahdi.conf file. Moving up before the channel line worked. Appreciate the help. matt On Thu, Apr 5, 2018 at 12:37 PM Richard Mudgett wrote: > On Thu, Apr 5, 2018 at 7:20 AM, Mtt

Re: [asterisk-users] Asterisk / PRI and Outbound Overlap Dialing (Mtt Cannon)

According to Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching#PatternMatching-SpecialCharactersUsedinPatternMatching The exclamation mark (!) character is similar to the period and matches zero or more remaining characters. It is used in overlap dialing to dial through

Re: [asterisk-users] Asterisk / PRI and Outbound Overlap Dialing

On Thu, Apr 5, 2018 at 7:20 AM, Mtt Cannon wrote: > > > I am trying to setup Asterisk to act like a PBX connected via a PRI > gateway to a voice netowrk where Asterisk is doing outbound > overlap dialing for calls that terminate via that PRI. AFter researching > through

[asterisk-users] Asterisk / PRI and Outbound Overlap Dialing

I am trying to setup Asterisk to act like a PBX connected via a PRI gateway to a voice netowrk where Asterisk is doing outbound overlap dialing for calls that terminate via that PRI. AFter researching through the archives and online dcocs, I thought I had everyting setup right, dialplan configured

[asterisk-users] Asterisk Local channel Earlymedia

Hello, I noticed that Earlymedia isn't possible as soon as the Media Stream has to pass through Asterisk. In my Scenario I need Local Channels and Earlymedia at the same time. So the Media Stream needs to flow through Asterisk. Is there a Workaround to get Earlymedia with Local Channels working?

Re: [asterisk-users] Asterisk as gateway

This is a setup of Asterisk as extension to an existing Asterisk PBX. It has to be that way and not IAX. Simply we need to an extension number with DIDs to an external PBX which is a helper to our office. This has to be done for the second PBX as well. On Thu, Mar 22, 2018 at 2:18 PM, Atux Atux

[asterisk-users] Asterisk as gateway

i would like to ask how to connect 2 systems. I would like to have an asterisk where it will have all the connections to the outside world (sip trunks) and it will called the gateway. This asterisk will have extension numbers of 3XX. In the LAN there will be 2 other asterisk boxes (A & B) where A

[asterisk-users] Asterisk 15.3.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 15.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.3.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.20.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.20.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.20.0 resolves several issues reported by the community and would have not been

Re: [asterisk-users] Asterisk server as TLS/SRTP

Hi! I have used this document https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport You can specify transport=tls and encryption=yes for those peers which need to use encryption. пн, 5 мар. 2018 г. в 14:20, Antony Stone < antony.st...@asterisk.open.source.it>: > On Monday 05 March 2018 at

Re: [asterisk-users] Asterisk server as TLS/SRTP

On Monday 05 March 2018 at 12:06:51, Atux Atux wrote: > Hi. I have an Asterisk Server (A) where it acts as the main gateway to > offer services. > There are different asterisk servers (B -D) that connect as extensions to > the Server A. Why not use IAX? > I would like to implement TLS and SRTP

[asterisk-users] Asterisk server as TLS/SRTP

Hi. I have an Asterisk Server (A) where it acts as the main gateway to offer services. There are different asterisk servers (B -D) that connect as extensions to the Server A. I would like to implement TLS and SRTP for these extensions, but have the non secure as well for other extensions. for

[asterisk-users] Asterisk 15 autocreate peer

Hi, Is it possible to use autocreate peer with webrtc in asterisk 15? With kind regards, Marcus Kvarsell [www.fogwise.se] marcus.kvars...@fogwise.se | www.fogwise.se | +46 (0)766 350384 --

Re: [asterisk-users] Asterisk 15.3.0-rc1 regression

On Mon, Feb 26, 2018 at 8:22 AM, Dmitriy Serov wrote: > Asterisk 15.2.0, pjsip > > Yesterday I installed 15.3.0-rc1 (after 15.2.0). Today I had to rollback > to 15.2.0 (not 15.2.2). > > The reason is: the loaded server very often hung on the same places: > SUBSCRIBE/NOTIFY

[asterisk-users] Asterisk 15.3.0-rc1 regression

Asterisk 15.2.0, pjsip Yesterday I installed 15.3.0-rc1 (after 15.2.0). Today I had to rollback to 15.2.0 (not 15.2.2). The reason is: the loaded server very often hung on the same places: SUBSCRIBE/NOTIFY processing. Unloading modules was the solution at that time. ; Off SUBSCRIBE +

Re: [asterisk-users] Asterisk crash on core show channel

Wakano Skickat: den 21 februari 2018 23:34 Till: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Ämne: Re: [asterisk-users] Asterisk crash on core show channel Thanks for you answer Marcus, So maybe this means some bug was fixed? Anyone aware of som

Re: [asterisk-users] Asterisk crash on core show channel

Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Marcus Kvarsell Skickat: den 22 februari 2018 09:14 Till: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Ämne: Re: [asterisk-users] Asterisk

Re: [asterisk-users] Asterisk crash on core show channel

, Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Marcus Kvarsell Skickat: den 22 februari 2018 08:29 Till: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Ämne: Re: [asterisk-users] Asterisk

Re: [asterisk-users] Asterisk crash on core show channel

List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > *Ämne:* [asterisk-users] Asterisk crash on core show channel > > > > Hello Asterisk list, > > I am facing some Asterisk crashes which are consistently pointing to the > same backtrace

[asterisk-users] Asterisk 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3 Now Available (Security)

The Asterisk Development Team would like to announce security releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3. These releases are available for immediate download at

Re: [asterisk-users] Asterisk crash on core show channel

com> Ämne: [asterisk-users] Asterisk crash on core show channel Hello Asterisk list, I am facing some Asterisk crashes which are consistently pointing to the same backtrace, which is the following (using DONT_OPTIMIZE, BETTER_BACKTRACES and MALLOC_DEBUG): Thread 1 (Thread 0x7f1f08be8700 (LWP 1767

[asterisk-users] Asterisk crash on core show channel

Hello Asterisk list, I am facing some Asterisk crashes which are consistently pointing to the same backtrace, which is the following (using DONT_OPTIMIZE, BETTER_BACKTRACES and MALLOC_DEBUG): Thread 1 (Thread 0x7f1f08be8700 (LWP 1767)): #0 0x7f1f9bed3395 in __strcasecmp_l_sse42 () from

[asterisk-users] Asterisk 13.19.1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.19.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.19.1 resolves an issue reported by the community and would have not been

[asterisk-users] Asterisk 15.2.1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 15.2.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.2.1 resolves an issue reported by the community and would have not been

[asterisk-users] asterisk not transcoding on a re-invite codec change

I have an old setup based in Asterisk 1.8. The carrier is accepting Ulaw in the initial invite, but immediately the call is established they send a re-invite to change to Alaw. This doesn't get transcoded and the user gets no audio from after the re-invite Is/was this a problem for asterisk

Re: [asterisk-users] asterisk mysql contacts

ailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Betreff: Re: [asterisk-users] asterisk mysql contacts On Wed, 17 Jan 2018 09:26:28 -0700 John Kiniston <johnkinis...@gmail.com> wrote: > use func_odbc, create a new function that does a lookup. > > [CALL

Re: [asterisk-users] asterisk mysql contacts

On Wed, 17 Jan 2018 09:26:28 -0700 John Kiniston wrote: > use func_odbc, create a new function that does a lookup. > > [CALLERID] > prefix=LOOKUP > dsn=MyDB > readsql=SELECT CALLERID from MyNames where CallerIdNum = > '${SQL_ESC(${ARG1})}' > > exten =>

Re: [asterisk-users] asterisk mysql contacts

use func_odbc, create a new function that does a lookup. [CALLERID] prefix=LOOKUP dsn=MyDB readsql=SELECT CALLERID from MyNames where CallerIdNum = '${SQL_ESC(${ARG1})}' exten => s,n,Set(CALLERID(NAME)=LOOKUP_CALLERID(${CALLERID(NUM)})) On Wed, Jan 17, 2018 at 6:16 AM, Atux Atux

Re: [asterisk-users] asterisk mysql contacts

On Wed, Jan 17, 2018 at 03:16:04PM +0200, Atux Atux wrote: [asterisk dialplan mysql] > I would like to ask if there is a way to implement this easily in my > dialplan, please. The answer is: yes If you'd search for "asterisk dialplan mysql", you'get something like

[asterisk-users] asterisk mysql contacts

Hi. i have an asterisk 11 installation that i run in my soho environment. My system has mysql to store all the cdrs. I would like make use of the mysql and store numbers and names. eg +4922123456789 "Atux Null". So when the +4922123456789 calls in my system the name "Atux Null" will pop up next to

Re: [asterisk-users] Asterisk 13.19.0 Now Available

On 12.01.2018 20:27, Joshua Colp wrote: >> I highly respect your work and don't want to steal your time since I >> have probably seriously misunderstood something, but could you please >> shortly explain what the string "Security: " (aka "(Security)" and with >> other wordings) at the beginning

Re: [asterisk-users] Asterisk 13.19.0 Now Available

On Fri, Jan 12, 2018, at 3:02 PM, Binarus wrote: > Thanks for taking the time, but ... > > On 12.01.2018 12:04, Joshua Colp wrote: > > >> Could this be one of the rare cases where 13 and 15 needed security > >> fixes, but 14 didn't? > > > > These are normal bug fix releases, not security

Re: [asterisk-users] Asterisk 13.19.0 Now Available

See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for information regarding the release cycle.   It shows v14 went into security fix only mode on Sept 26 2017. On 01/12/2018 02:02 PM, Binarus wrote: Thanks for taking the time, but ... On 12.01.2018 12:04, Joshua Colp

Re: [asterisk-users] Asterisk 13.19.0 Now Available

Thanks for taking the time, but ... On 12.01.2018 12:04, Joshua Colp wrote: >> Could this be one of the rare cases where 13 and 15 needed security >> fixes, but 14 didn't? > > These are normal bug fix releases, not security releases. As such 14 did not > receive a release. > Interesting. The

Re: [asterisk-users] Asterisk 13.19.0 Now Available

On Fri, Jan 12, 2018, at 2:51 AM, Binarus wrote: > On 11.01.2018 20:51, Asterisk Development Team wrote: > > The Asterisk Development Team would like to announce the release of > > Asterisk 13.19.0. > > This release is available for immediate download at > >

Re: [asterisk-users] Asterisk 13.19.0 Now Available

On 11.01.2018 20:51, Asterisk Development Team wrote: > The Asterisk Development Team would like to announce the release of > Asterisk 13.19.0. > This release is available for immediate download at > http://downloads.asterisk.org/pub/telephony/asterisk > > The release of Asterisk 13.19.0 resolves

Re: [asterisk-users] Asterisk 15.2.0 Now Available

On Thu, Jan 11, 2018, at 4:34 PM, Ira wrote: > Re: [asterisk-users] Asterisk 15.2.0 Now AvailableHello Asterisk, > > Thursday, January 11, 2018, 11:59:28 AM, you wrote: > > > The release of Asterisk 15.2.0 resolves several issues reported by the > community and would h

Re: [asterisk-users] Asterisk 15.2.0 Now Available

Title: Re: [asterisk-users] Asterisk 15.2.0 Now Available Hello Asterisk, Thursday, January 11, 2018, 11:59:28 AM, you wrote: The release of Asterisk 15.2.0 resolves several issues reported by the community and would have not been possible without your participation. For amusement I

[asterisk-users] Asterisk 15.2.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 15.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.2.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.19.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.19.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.19.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2 Now Available (Security)

The Asterisk Development Team would like to announce security releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2. These releases are available for immediate download at

Re: [asterisk-users] Asterisk 13.18.4 - New Error PJLIB_UTIL_EDNS_REFUSED

On Thu, Dec 21, 2017 at 8:18 AM, Bryant Zimmerman wrote: > We just updated from 13.17.1 to 13.18.4 and are noticing a new error > > [2017-12-21 10:12:48] ERROR[32343]: res_pjsip.c:3850 endpt_send_request: > Error 320055 'DNS "Refused" (PJLIB_UTIL_EDNS_REFUSED)' sending

[asterisk-users] Asterisk 13.18.4 - New Error PJLIB_UTIL_EDNS_REFUSED

We just updated from 13.17.1 to 13.18.4 and are noticing a new error [2017-12-21 10:12:48] ERROR[32343]: res_pjsip.c:3850 endpt_send_request: Error 320055 'DNS "Refused" (PJLIB_UTIL_EDNS_REFUSED)' sending OPTIONS request to endpoint 6162480909.8009 The DNS on the system seems to be

[asterisk-users] Asterisk-Kannel integration project version 0.0.1 release notes

Hello, We are proud to announce the first release 0.0.1 version of kannel-asterisk integration project. The goal of this project is to allow asterisk users to use kannel capabilities like SMS sending and receiving. Please visit https://asterisk-kannel.sourceforge.io/ for more information. You can

Re: [asterisk-users] asterisk and Hyper-V

I am not sure if the problem is within the system or within a hypervisor or a vswitch. My test is quite simple - when the traffic is captured from a VM interface, everything is fine, and there is no sound distortion; but when the traffic is captured from the hypervisor host interface, wireshark

Re: [asterisk-users] asterisk and Hyper-V

On Mon, Dec 18, 2017 at 10:22:33AM +, Kseniya Blashchuk wrote: > To be honest we are a bit afraid to set 100% )), but we have tried to set > 90% - no luck. I have also tested with 4.8 and 4.11 kernels - same results. > I will try with Centos 6 and kernel 3.10 to check if something changes. >

Re: [asterisk-users] asterisk queues in off-hook mode ?

thanks Richard, you just solved my problem. On Tue, Dec 19, 2017 at 9:05 PM, Richard Mudgett wrote: > > > On Tue, Dec 19, 2017 at 2:56 PM, Khalil Khamlichi > wrote: >> >> Hi, >> >> I am looking to configure asterisk queues in off-hook mode, that

Re: [asterisk-users] asterisk queues in off-hook mode ?

On Tue, Dec 19, 2017 at 2:56 PM, Khalil Khamlichi < khamlichi.kha...@gmail.com> wrote: > Hi, > > I am looking to configure asterisk queues in off-hook mode, that is, > the agent calls into the system and stays connected to this call, when > new customer calls, he is redirected to the queue which

[asterisk-users] asterisk queues in off-hook mode ?

Hi, I am looking to configure asterisk queues in off-hook mode, that is, the agent calls into the system and stays connected to this call, when new customer calls, he is redirected to the queue which should distribute to connected agents. is this possible on teh actual app_queue or we would need

Re: [asterisk-users] asterisk and Hyper-V

No problem, I will try to test with CentOS if I have time пн, 18 дек. 2017 г. в 13:33, Dmitriy Ermakov : > Ok. I am sorry, I don't have any more ideas( > > Please, send here email with your testing results. > > On 12/18/2017 01:22 PM, Kseniya Blashchuk wrote: > > To be

Re: [asterisk-users] asterisk and Hyper-V

Ok. I am sorry, I don't have any more ideas( Please, send here email with your testing results. On 12/18/2017 01:22 PM, Kseniya Blashchuk wrote: > To be honest we are a bit afraid to set 100% )), but we have tried to > set 90% - no luck. I have also tested with 4.8 and 4.11 kernels - same >

Re: [asterisk-users] asterisk and Hyper-V

To be honest we are a bit afraid to set 100% )), but we have tried to set 90% - no luck. I have also tested with 4.8 and 4.11 kernels - same results. I will try with Centos 6 and kernel 3.10 to check if something changes. VSwitch shows 1-2% load on the interfaces, and this host is not overloaded

Re: [asterisk-users] asterisk and Hyper-V

I am using CentOS 6, kernel 3.10 from elrepo.org kernels (3.10.102-1.el6.elrepo.x86_64). Asterisk version is 11.21.2 and Asterisk 13.X.X (I can't get it's version now). Is it possible that your network switches' interfaces which are connected to Hyper-V Server are 100% busy? It is possible that

Re: [asterisk-users] asterisk and Hyper-V

Dmitry, are you using CentOS? What kernel version are you using? I will try with the same to see if it can be also a kernel-related issue. пн, 18 дек. 2017 г. в 11:35, Kseniya Blashchuk : > Thank you for a quick answer, Dmitry! > > We have tried the settings you suggested but

Re: [asterisk-users] asterisk and Hyper-V

Thank you for a quick answer, Dmitry! We have tried the settings you suggested but nothing helped. The machine is running 4.4.0-104 kernel, 4 cores, Intel(R) Xeon(R) CPU E5-2620 v3 @ 2.40GHz, clocksource is hyperv_clocksource_tsc_page, timing module is res_timing_timerfd.so. We have also tried to

Re: [asterisk-users] asterisk and Hyper-V

Hi, Kseniya! I have three installations of Asterisk (as FreePBX but I think it is not important). They work fine. I have made some settings in Asterisk and Hyper-V: Asterisk: timing interface - timerfd. Hyper-V: 1. Virtual Machine => Network Adapter => Hardware Acceleration => Virtual

[asterisk-users] asterisk and Hyper-V

Hi all! Does anybody have experience with asterisk on Hyper-V? My test setup with Ubuntu 16 and asterisk 13.1 (ubuntu repo) shows sound distortion. I have analyzed the RTP flow with wireshark and I see high skew and delta values when the traffic leaves the hypervisor, however everything is okay

[asterisk-users] Asterisk 13.18.4, 14.7.4, 15.1.4 and Certified Asterisk 13.13-cert9 Now Available

The Asterisk Development Team has announced security releases for Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available security releases are released as versions 13.13-cert9, 13.18.4, 14.7.4 and 15.1.4. These releases are available for immediate download at

Re: [asterisk-users] Asterisk / FreePBX Support / Reseller

https://community.freepbx.org https://www.voip-info.org/wiki/view/Asterisk+Mailing+Lists On 12/12/2017 11:43 AM, Doug Lytle wrote: If you do not get an answer from the user list, you might try a post to the dev list. It is a bit more active and some of the people watching for dev news might

Re: [asterisk-users] Asterisk / FreePBX Support / Reseller

>>> If you do not get an answer from the user list, you might try a post to >>> the dev list. >>> It is a bit more active and some of the people watching for dev news >>> might be able to help you. Surely, you mean the Biz List Doug --

Re: [asterisk-users] Asterisk / FreePBX Support / Reseller

SLA is just one factor in the decision to select a vendor. If you do not get an answer from the user list, you might try a post to the dev list. It is a bit more active and some of the people watching for dev news might be able to help you. I used a consultant of the first go around but took

Re: [asterisk-users] Asterisk / FreePBX Support / Reseller

I know but this is not my sole decision. On 12.12.2017 16:17, Ron Wheeler wrote: > If your phone system goes down and you can not get it back up until > tomorrow afternoon because your support person is on another project, > you may wish you had an SLA. --

Re: [asterisk-users] Asterisk / FreePBX Support / Reseller

If your phone system goes down and you can not get it back up until tomorrow afternoon because your support person is on another project, you may wish you had an SLA. I hope that this extra info helps find a solution. Ron On 12/12/2017 3:41 AM, basti wrote: Size: - one location - 15 IP

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