I took a look through Asterisk 11 and 13 change logs but didn't see any mention of that patch/fix. Am I missing something?
Derek B > On May 4, 2016, at 8:50 AM, "[email protected]" > <[email protected]> wrote: > > Send asterisk-users mailing list submissions to > [email protected] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [email protected] > > You can reach the person managing the list at > [email protected] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. Re: Asterisk 13 Realtime Voicemail frustrating issue > (John Kiniston) > 2. Re: Migrating asterisk 11 to 13: some callers get no ringback > tone any more (Michael Maier) > 3. Re: Migrating asterisk 11 to 13: some callers get no ringback > tone any more (Joshua Colp) > 4. Re: Migrating asterisk 11 to 13: some callers get no ringback > tone any more (Eric Wieling) > 5. Re: Migrating asterisk 11 to 13: some callers get no ringback > tone any more (Joshua Colp) > 6. Call a subroutine via Originate? (John Kiniston) > 7. Re: Call a subroutine via Originate? (Bruce Ferrell) > 8. Double queue calls being delivered to agents (Derek Bolichowski) > 9. Execute an app on the master channel from inside a Macro on > the called channel (Saint Michael) > 10. Re: Double queue calls being delivered to agents (Richard Mudgett) > 11. Re: Migrating asterisk 11 to 13: some callers get no ringback > tone any more (Michael Maier) > 12. Re: T.38 with Audiocodes gateway [SOLVED] (Olivier) > 13. Asterisk registers with TLS, but sends out calls via UDP > (Sebastian Damm) > 14. Compatibilty between agi for asterisk 13.8.0 and php5.6 > (Mamadou NGOM) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 3 May 2016 11:39:44 -0700 > From: John Kiniston <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] Asterisk 13 Realtime Voicemail > frustrating issue > Message-ID: > <cafjqogc8syl_fsl8pmr+p6f6p1-nzk-_3rayrakw4kzjev8...@mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Have you tried using the table definition that comes with the Asterisk > source? > > the file mysql_config.sql is located in contrib/realtime/mysql and defines > a very different voicemail table than what you have in your configuration. > > On Tue, May 3, 2016 at 3:10 AM, Michele Pinassi <[email protected]> > wrote: > >> Hi all, >> >> i'm experiencing a really frustrating issue with my Asterisk 13.7.2 with >> realtime configuration on MySQL and Voicemail. >> >> Here's res_config_mysql.conf: >> >> *[default]* >> *dbhost = 192.168.1.1* >> *dbname = asterisk* >> *dbuser = asterisk* >> *dbpass = [xxxxx]* >> *dbport = 3306* >> *requirements=warn ; or createclose or createchar* >> >> extconfig.conf: >> >> *[settings]* >> *sipusers => mysql,default,sipusers* >> *sippeers => mysql,default,sipusers* >> *sipregs => mysql,default,sipregs* >> *voicemail => mysql,default,vmusers* >> *meetme => mysql,default,meetme* >> >> on Asterisk console: >> >> *asterisk*CLI> realtime mysql status * >> *default connected to [email protected] <[email protected]>, port >> 3306 with username asterisk for 56 minutes.* >> *asterisk*CLI> * >> >> "vmusers" table on MySQL: >> >> >> uniqueid >> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60uniqueid%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> >> customer_id >> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60customer_id%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> >> context >> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60context%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> >> mailbox >> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60mailbox%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> >> password >> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60password%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> >> fullname >> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60fullname%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> >> email >> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60email%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> >> pager >> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60pager%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> >> stamp >> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60stamp%60+DESC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> >> 5002 5002 default 5002 xxxx AAA >> >> *NULL* 0000-00-00 00:00:00 >> 5005 5005 default 5005 xxxx bbb >> *NULL* 0000-00-00 00:00:00 >> 5018 5018 default 5018 xxxx ccc >> *NULL* 0000-00-00 00:00:00 >> 5007 5007 default 5007 xxxx sdddd >> *NULL* 0000-00-00 00:00:00 >> *BUT* when i type, on Asterisk console: >> >> *asterisk*CLI> voicemail show zones * >> *There are no voicemail zones currently defined* >> *Command 'voicemail show zones ' failed.* >> *asterisk*CLI> * >> >> the same, of course, for "show users default". And whet i try to access a >> mailbox, i get a "Invalid password". >> >> Any hints ? Please, i'm really frustrated ! >> >> Michele >> >> -- >> Michele Pinassi >> Responsabile Telefonia di Ateneo >> Servizio Reti, Sistemi e Sicurezza Informatica - Universit? degli Studi di >> Siena >> tel: 0577.(23)5000 - [email protected] >> >> Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di >> Ateneo, http://www.faq.unisi.it >> >> >> -- >> _____________________________________________________________________ >> -- 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 > > > > -- > A human being should be able to change a diaper, plan an invasion, butcher > a hog, conn a ship, design a building, write a sonnet, balance accounts, > build a wall, set a bone, comfort the dying, take orders, give orders, > cooperate, act alone, solve equations, analyze a new problem, pitch manure, > program a computer, cook a tasty meal, fight efficiently, die gallantly. > Specialization is for insects. > ---Heinlein > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20160503/8c789a62/attachment-0001.html> > > ------------------------------ > > Message: 2 > Date: Tue, 3 May 2016 20:45:05 +0200 > From: Michael Maier <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some > callers get no ringback tone any more > Message-ID: <[email protected]> > Content-Type: text/plain; charset=windows-1252 > >> On 05/03/2016 at 05:43 PM Joshua Colp wrote: >> Michael Maier wrote: >>>> On 05/03/2016 at 04:50 PM Joshua Colp wrote: >>>> Michael Maier wrote: >>>>> Hello Joshua! >>>>> >>>>> >>>>> I attached the sip debug without the progressinband=never set. The >>>>> caller didn't get a ring back tone as expected. >>>> Please keep this on list so that anyone who may run into a similar >>>> problem in the future has a chance of finding this discussion. >>> >>> You are right - normally I'm going exactly this way. But I don't want >>> the traces to be world wide readable (-> privacy). I will write a >>> summary to the list as far as we know more. >>> >>>> As for your log there's nothing of note really, it's just expecting to >>>> send the ringing as inband audio instead of out of band. Does "rtp set >>>> debug on" show the RTP traffic going to the other side? >>> >>> Yes. I attached it. >>> >>> And no - there isn't any packet blocked by iptables :-). >> >> There is nothing abnormal here and Asterisk appears to be doing the >> correct thing. It's sending an audio stream with early progress to the >> caller. It may be that in a previous FreePBX, or when used with 13, they >> changed the behavior for this to force early media and the provider is >> not allowing it. > > Ok - but this doesn't seem to answer my main question: > > Why must > > progressinband=never > > be applied especially if asterisk uses it by default? The big difference > between w/ and w/o it is: > > w/o the option progrssinband=never just the sip-package > 183 Session Progress > is sent. > > w/ the option set, the additional sip-packages > 100 Trying > 180 Ringing > 180 Ringing > are sent. > > If progrssinband=never is the default, the Ringing package should be > sent always, shouldn't it? > > If I remove the option progrssinband=never via FreePBX, I can't find any > other value provided to progrssinband in /etc/asterisk/*. > > > Why does it work always correctly w/ the second trunk, which is > connected directly to the extension? > > Is it possible to switch off the standard behavior of asterisk / > progrssinband for ring groups only by setting some other options? > > > > Thanks, > kind regards, > Michael > > > > ------------------------------ > > Message: 3 > Date: Tue, 03 May 2016 15:52:05 -0300 > From: Joshua Colp <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some > callers get no ringback tone any more > Message-ID: <[email protected]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Whoops, email client auto-filled dev previously. Let's try this again. > > Michael Maier wrote: > > <snip> > >> Ok - but this doesn't seem to answer my main question: >> >> Why must >> >> progressinband=never >> >> be applied especially if asterisk uses it by default? The big difference >> between w/ and w/o it is: > > The default in 13 is "no" which still allows early media through. That > option has a complicated past. > >> >> w/o the option progrssinband=never just the sip-package >> 183 Session Progress >> is sent. > > Yes, because it's doing inband progress using a media stream. > >> >> w/ the option set, the additional sip-packages >> 100 Trying >> 180 Ringing >> 180 Ringing >> are sent. >> >> If progrssinband=never is the default, the Ringing package should be >> sent always, shouldn't it? > > It's not the default. > >> >> If I remove the option progrssinband=never via FreePBX, I can't find any >> other value provided to progrssinband in /etc/asterisk/*. >> >> >> Why does it work always correctly w/ the second trunk, which is >> connected directly to the extension? > > FreePBX may not use inband progress for that scenario, causing it to > send out of band ringing instead. > >> >> Is it possible to switch off the standard behavior of asterisk / >> progrssinband for ring groups only by setting some other options? > > Asterisk does not have the concept of ring groups, this is a FreePBX > construct. Asterisk itself does allow the option to be set on an > individual basis for the entries in sip.conf. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > > > > ------------------------------ > > Message: 4 > Date: Tue, 3 May 2016 15:07:09 -0400 > From: Eric Wieling <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some > callers get no ringback tone any more > Message-ID: <[email protected]> > Content-Type: text/plain; charset=windows-1252; format=flowed > > I don't know the default setting for progressinband in the code, but it > is documented in Asterisk 11's sip.conf.sample as defaulting to never. > Maybe the docs were fixed since Asterisk 11. > > from 11.21.x sip.conf.sample: > > ;progressinband=never ; If we should generate in-band ringing > always > ; use 'never' to never use in-band > signalling, even in cases > ; where some buggy devices might not > render it > ; Valid values: yes, no, never Default: > never > > >> On 05/03/2016 02:52 PM, Joshua Colp wrote: >> Whoops, email client auto-filled dev previously. Let's try this again. >> >> Michael Maier wrote: >> >> <snip> >> >>> Ok - but this doesn't seem to answer my main question: >>> >>> Why must >>> >>> progressinband=never >>> >>> be applied especially if asterisk uses it by default? The big >> difference >>> between w/ and w/o it is: >> >> The default in 13 is "no" which still allows early media through. That >> option has a complicated past. >> >>> >>> w/o the option progrssinband=never just the sip-package >>> 183 Session Progress >>> is sent. >> >> Yes, because it's doing inband progress using a media stream. >> >>> >>> w/ the option set, the additional sip-packages >>> 100 Trying >>> 180 Ringing >>> 180 Ringing >>> are sent. >>> >>> If progrssinband=never is the default, the Ringing package should be >>> sent always, shouldn't it? >> >> It's not the default. >> >>> >>> If I remove the option progrssinband=never via FreePBX, I can't find >> any >>> other value provided to progrssinband in /etc/asterisk/*. >>> >>> >>> Why does it work always correctly w/ the second trunk, which is >>> connected directly to the extension? >> >> FreePBX may not use inband progress for that scenario, causing it to >> send out of band ringing instead. >> >>> >>> Is it possible to switch off the standard behavior of asterisk / >>> progrssinband for ring groups only by setting some other options? >> >> Asterisk does not have the concept of ring groups, this is a FreePBX >> construct. Asterisk itself does allow the option to be set on an >> individual basis for the entries in sip.conf. > > -- > if at first you don't succeed, skydiving isn't for you > > > > > ------------------------------ > > Message: 5 > Date: Tue, 03 May 2016 16:16:04 -0300 > From: Joshua Colp <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some > callers get no ringback tone any more > Message-ID: <[email protected]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Eric Wieling wrote: >> I don't know the default setting for progressinband in the code, but it >> is documented in Asterisk 11's sip.conf.sample as defaulting to never. >> Maybe the docs were fixed since Asterisk 11. > > The behavior change to actually do what the option was documented to do. > As part of that the default was changed to reflect the past behavior, > thus why it was changed to no. The commit itself: > > chan_sip: make progressinband default to no > > After the "progressinband" value setting of "never" was updated to never > send a 183 this separated its use from the "no" value. Since "never" was > the default, but most users probably expect "no" this patch updates the > default for the "progressinband" setting to "no." > > This was tracked under ASTERISK-24835[1]. > > [1] https://issues.asterisk.org/jira/browse/ASTERISK-24835 > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > > > > ------------------------------ > > Message: 6 > Date: Tue, 3 May 2016 14:24:25 -0700 > From: John Kiniston <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: [asterisk-users] Call a subroutine via Originate? > Message-ID: > <cafjqogf1kx+yfp+0xk1j8klmqwtnxq5r_zkrvuspb4vfuq6...@mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Howdy everyone, > > I'm writing a little click to dial type tool and I've run into a snag where > my Originate command needs to call a Sub routine to do a database lookup > and some other stuff. > > I can't seem to get the syntax right to call Gosub with Originate > > Just testing with the command line I've been unable to make it work with > any of these attempts: > > originate PJSIP/johntest application Gosub sub-callout s,1 > > originate PJSIP/johntest application Gosub sub-callout(s,1) > > originate PJSIP/johntest application Gosub (sub-callout,s,1) > > What Syntax should I be using? > > And if it helps I'll be calling this via AMI over https. > > Thanks! > > -- > A human being should be able to change a diaper, plan an invasion, butcher > a hog, conn a ship, design a building, write a sonnet, balance accounts, > build a wall, set a bone, comfort the dying, take orders, give orders, > cooperate, act alone, solve equations, analyze a new problem, pitch manure, > program a computer, cook a tasty meal, fight efficiently, die gallantly. > Specialization is for insects. > ---Heinlein > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20160503/6596facc/attachment-0001.html> > > ------------------------------ > > Message: 7 > Date: Tue, 3 May 2016 14:33:32 -0700 > From: Bruce Ferrell <[email protected]> > To: [email protected] > Subject: Re: [asterisk-users] Call a subroutine via Originate? > Message-ID: <[email protected]> > Content-Type: text/plain; charset="windows-1252"; Format="flowed" > > use the macro construct and return from the macro > >> On 5/3/16 2:24 PM, John Kiniston wrote: >> Howdy everyone, >> >> I'm writing a little click to dial type tool and I've run into a snag >> where my Originate command needs to call a Sub routine to do a >> database lookup and some other stuff. >> >> I can't seem to get the syntax right to call Gosub with Originate >> >> Just testing with the command line I've been unable to make it work >> with any of these attempts: >> >> originate PJSIP/johntest application Gosub sub-callout s,1 >> >> originate PJSIP/johntest application Gosub sub-callout(s,1) >> >> originate PJSIP/johntest application Gosub (sub-callout,s,1) >> >> What Syntax should I be using? >> >> And if it helps I'll be calling this via AMI over https. >> >> Thanks! >> >> -- >> A human being should be able to change a diaper, plan an invasion, >> butcher a hog, conn a ship, design a building, write a sonnet, balance >> accounts, build a wall, set a bone, comfort the dying, take orders, >> give orders, cooperate, act alone, solve equations, analyze a new >> problem, pitch manure, program a computer, cook a tasty meal, fight >> efficiently, die gallantly. Specialization is for insects. >> ---Heinlein > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20160503/80a6e460/attachment-0001.html> > > ------------------------------ > > Message: 8 > Date: Tue, 3 May 2016 23:15:53 +0000 > From: Derek Bolichowski <[email protected]> > To: "[email protected]" > <[email protected]> > Subject: [asterisk-users] Double queue calls being delivered to agents > Message-ID: <[email protected]> > Content-Type: text/plain; charset="utf-8" > > I posted this over in asterisk-dev, realized I probably should have put it > here. > > Hi there, > We?ve been having a strange issue with a customer?s queues where a queued > call will ring an available agent, agent answers, then a second or two later > the agent is offered a second call which they cannot answer, since they?re > already speaking with a client. > > This in turn causes a few issues: > - Agent stats are no longer accurate, as it gets marked down as a ?missed > call?. > - Cannot use ?autopause? feature any longer, as the second queue call goes > unanswered and pauses the agents. > > The basic queue setup is as follows: > Autofill = yes > Ringinuse = no > Wrapuptime = 5 > Strategy = fewestcalls (tried ?random? also) > Timeout = 15 > > We?re on Asterisk 11.21.2 currently. > > In talking to a few colleagues, they seem to recall there being an old patch > for the Asterisk queues application that inserted a short 100ms delay between > delivering first and second calls. I?ve scoured the web today, and found > some old forums posts of people looking for something exactly like this, but > haven?t found the actual patch, if one even exists. > > I?m hoping someone may have some suggestions on some options we can try to > eliminate this issue. > > Thanks for taking the time to read this. > > -Derek B > > ------------------------------ > > Message: 9 > Date: Tue, 3 May 2016 19:48:25 -0400 > From: Saint Michael <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: [asterisk-users] Execute an app on the master channel from > inside a Macro on the called channel > Message-ID: > <cac9csobwvp0gyibm+sktwnt3yg6othv8wbwqlglgjawm2ux...@mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > ?While I am executing a Macro on the called channel, right after the call > connects?, I need to execute an app on the master channel, from inside that > macro, specifically, SendDTMF. If I execute it now, it send a text message > to the Callee, when my app needs to send it to the caller. > > I could use > set(master_channel(variable)=XXX), but then how do I execute some code on > the master channel. > Note that I could send the name of the master channels to the Macro > M(Name^parameter), but then how do I execute SendDtmf on the identified > Master Channel? > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20160503/b0f382fd/attachment-0001.html> > > ------------------------------ > > Message: 10 > Date: Tue, 3 May 2016 20:59:14 -0500 > From: Richard Mudgett <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] Double queue calls being delivered to > agents > Message-ID: > <cald46g0muwkryxcokm7kf2nzo+rn3gzawcscnk_sirjbpr2...@mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski <[email protected]> > wrote: > >> I posted this over in asterisk-dev, realized I probably should have put it >> here. >> >> Hi there, >> We?ve been having a strange issue with a customer?s queues where a queued >> call will ring an available agent, agent answers, then a second or two >> later the agent is offered a second call which they cannot answer, since >> they?re already speaking with a client. >> >> This in turn causes a few issues: >> - Agent stats are no longer accurate, as it gets marked down as a ?missed >> call?. >> - Cannot use ?autopause? feature any longer, as the second queue call goes >> unanswered and pauses the agents. >> >> The basic queue setup is as follows: >> Autofill = yes >> Ringinuse = no >> Wrapuptime = 5 >> Strategy = fewestcalls (tried ?random? also) >> Timeout = 15 >> >> We?re on Asterisk 11.21.2 currently. >> >> In talking to a few colleagues, they seem to recall there being an old >> patch for the Asterisk queues application that inserted a short 100ms delay >> between delivering first and second calls. I?ve scoured the web today, and >> found some old forums posts of people looking for something exactly like >> this, but haven?t found the actual patch, if one even exists. >> >> I?m hoping someone may have some suggestions on some options we can try to >> eliminate this issue. >> >> Thanks for taking the time to read this. > > This issue has been around a long time and was just recently fixed and I > think > it was just released in the latest v11 version. > See https://issues.asterisk.org/jira/browse/ASTERISK-16115 > > Richard > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20160503/a4d6b44b/attachment-0001.html> > > ------------------------------ > > Message: 11 > Date: Wed, 4 May 2016 09:09:17 +0200 > From: Michael Maier <[email protected]> > To: [email protected] > Subject: Re: [asterisk-users] Migrating asterisk 11 to 13: some > callers get no ringback tone any more > Message-ID: <[email protected]> > Content-Type: text/plain; charset=windows-1252 > >> On 05/03/2016 at 09:16 PM Joshua Colp wrote: >> Eric Wieling wrote: >>> I don't know the default setting for progressinband in the code, but it >>> is documented in Asterisk 11's sip.conf.sample as defaulting to never. >>> Maybe the docs were fixed since Asterisk 11. >> >> The behavior change to actually do what the option was documented to do. >> As part of that the default was changed to reflect the past behavior, >> thus why it was changed to no. The commit itself: >> >> chan_sip: make progressinband default to no >> >> After the "progressinband" value setting of "never" was updated to never >> send a 183 this separated its use from the "no" value. > > But "never" option therefore sends 180 Ringing which I was missing. The > new default "no" doesn't send 180 Ringing any more ... . > >> Since "never" was >> the default, but most users probably expect "no" this patch updates the >> default for the "progressinband" setting to "no." >> >> This was tracked under ASTERISK-24835[1]. >> >> [1] https://issues.asterisk.org/jira/browse/ASTERISK-24835 > > This makes sense! I migrated from > > asterisk11-11.8.1-40_centos6.x86_64, > > which had the default progressinband=never to > > asterisk13-core-13.7.2-1.shmz65.1.94.x86_64 > > which had the new default. > > POTS callers advertise support for early media - mobile callers on the > other hand don't advertise it, therefore mobile wasn't a problem because > early media (183) isn't triggered (and used!) at all. > > > Two strange things being left: > > 1. Why does progressinband=no work, if there is *no* ringgroup between > trunk and extension. This seems to be a "feature" of FreePBX. > > 2. Why is early media used even if the caller doesn't advertise it? Are > there other triggers like P-Early-Media? > > > > > Another basic question: > What do I need early media exactly for? I'm only using SIP phones - > nothing else. Couldn't it be completely disabled for these trunks? Or > would it break things like voice mail service e.g.? How can I disable it > completely even if it is advertised by the caller? > > > Thanks, > Michael > > > > ------------------------------ > > Message: 12 > Date: Wed, 4 May 2016 11:12:27 +0200 > From: Olivier <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] T.38 with Audiocodes gateway [SOLVED] > Message-ID: > <capet9jgowcuoa-jyerr9b1+fwf_-kfxuqfx74emakzuquy+...@mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > 2016-05-03 16:43 GMT+02:00 Matt Fredrickson <[email protected]>: > >>> On Fri, Apr 29, 2016 at 1:34 AM, Olivier <[email protected]> wrote: >>> Hello, >>> >>> I'm helping a colleague (*) which has the following setup: >>> >>> ITSP --- <T.38 capable PJSIP trunk> --- Asterisk 13 --- <PJSIP>-- >>> Audiocodes MP-112 --- <FXO/FXS> --- Fax machine >>> >>> My issue is the following : >>> Audiocodes gateway reject INVITEs with 488 Not Acceptable Here >>> >>> It seems this gateway requires t38 settings to be present in SDP body in >> the >>> very first INVITE. >>> >>> My questions are the following: >>> >>> 1. I expected T.38 to exclusively work with reINVITE where calls are >>> established as normal voice calls (PCMA/PCMU in SDP, for instance) and >> then >>> upgraded to T.38 (when CNG is detected, for instance). >>> Have you ever heard of T.38 sessions being established right from the >> start >>> (ie with T.38 settings in the first INVITE) ? >> >> No. It would seem to be extremely broken if it denies a call based on >> a lack of T.38 sdp parameters on the initial INVITE. > > OK > >> >>> 2. Is it possible to configure Asterisk to pass T.38 settings in SDP in >> the >>> first INVITE it sends ? >>> >>> 3. Any suggestion with Audiocodes gateway ? >> >> Look for T.38 settings maybe? See if there is something keeping you >> from sending an initial invite with non-T.38 SDP....? > > Yes, I think issue must come from incorrect Audiocodes settings. > Requiring T.38 settings within first INVITE seems very unusual. > > Thank you very much for replying > >> >> -- >> Matthew Fredrickson >> Digium, Inc. | Engineering Manager >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA >> >> -- >> _____________________________________________________________________ >> -- 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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20160504/91fb4657/attachment-0001.html> > > ------------------------------ > > Message: 13 > Date: Wed, 4 May 2016 13:25:53 +0200 > From: Sebastian Damm <[email protected]> > To: Asterisk <[email protected]> > Subject: [asterisk-users] Asterisk registers with TLS, but sends out > calls via UDP > Message-ID: > <cabkwsfwak2jmf08zj-fsgmbgtv48rhksq0lvb-qrudou1go...@mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > Hi, > > I have an Asterisk 13.8.2, which is supposed to be only a client to an > encrypted SIP service. All local phones are connected via UDP. > > Since I can't use PJSIP (see my mailing list post from yesterday), I > tried configuring chan_sip to work that way. My settings are: > > [general] > context=public > allowoverlap=no > udpbindaddr=0.0.0. > tlsbindaddr=0.0.0.0 > tcpenable=yes > tcpbindaddr=0.0.0.0 > tlsenable=yes > transport=udp > srvlookup=yes > tlscafile=/usr/local/etc/asterisk/keys/4cfd3c78.0 > tlscapath=/usr/local/etc/asterisk/keys > tlsclientmethod=tlsv1 > sipdebug = yes > > register => tls://[email protected]:[email protected] > > [devtrunk] > type=peer > host=example.org > defaultuser=1234567 > fromuser=1234567 > remotesecret=foobar > transport=tls > outboundproxy=dev.example.org > context=carrier-in > encryption=yes > > When I start up, I see my Asterisk doing a _sips._tcp SRV lookup, but > that's just for the registration, I guess. I also see it doing > _sip._udp SRV queries. I wouldn't know why it would have to do that. > > The REGISTER packets are sent out via TLS, as I would expect. > > When I issue a "sip show peer devtrunk" command, it tells me this: > > Prim.Transp. : TLS > Allowed.Trsp : TLS > > Looks okay to me. But when I place a call, Asterisk does this: > > Reliably Transmitting (no NAT) to 2.3.4.5:5060: > INVITE sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 9.8.7.6:0;branch=z9hG4bK2974d534 > > It sends the packet out via UDP, and to the wrong host, since it > doesn't use the correct SRV entry and instead sends it to the UDP > server. > > I did not generate a certificate for my Asterisk, because it only acts > as a client. I think, this shouldn't be needed. > > Can anyone point me to where I misconfigured something? Or did I > stumble upon a bug? What would I have to do to make Asterisk use the > open TLS connection used for registering for outbound calls, too? > > Best Regards, > Sebastian > > > > ------------------------------ > > Message: 14 > Date: Wed, 4 May 2016 14:49:34 +0200 (CEST) > From: Mamadou NGOM <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: [asterisk-users] Compatibilty between agi for asterisk 13.8.0 > and php5.6 > Message-ID: > > <1979110061.191209.7cc1d90d-d410-4a02-a619-42e64003d44e.open-xcha...@email.1and1.fr> > > Content-Type: text/plain; charset="us-ascii" > > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20160504/60e1ae38/attachment.html> > > ------------------------------ > > _______________________________________________ > --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 > > End of asterisk-users Digest, Vol 142, Issue 5 > ********************************************** -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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