Re: [asterisk-users] Bug in Dial() string processing

On Tuesday 27 October 2020 at 11:00:10, Antony Stone wrote: > Hi. > > I've discovered a bug in the Dial() string processing (for Asterisk 13.14.1 > at least). I've now confirmed that the same bug exists in 16.2.1 A Dial() command containing a SIP username/password combination which has a !

Re: [asterisk-users] bug in pjsip trust_id_outpound?

Hi Gang If anyone else stumbles over the same Problem. This is how I solved it for now: On the IC Trunk: trust_id_inbound=no => Makes sure the CallerID is taken from the From Header. trust_id_outbound=yes => Does nothing useful, maybe a bug? send_pai=no On the incoming call, you have to pull

Re: [asterisk-users] Bug in main/bridge.c:ast_bridge_update_talker_src_video_mode

On Mon, Aug 28, 2017 at 6:35 PM, Richard Kenner wrote: > I've had two Asterisk crashes today that seem to be caused by errors > where chan->tech_pvt is pointing to something that can't be deallocated > and I think I see a reference count bug in the above function. > > It

Re: [asterisk-users] BUG or ???

On Saturday 25 Feb 2017, Антон Сацкий wrote: > Thanks U Richard > i know about this solution > but the main question why "${} substitution containing > the SHELL is evaluated before anything else" For the same reason why you do raising to powers before multiplications and divisions, and all

Re: [asterisk-users] BUG or ???

Thanks U Richard i know about this solution but the main question why "${} substitution containing the SHELL is evaluated before anything else" Can U describe the rules when and why it happens? Thanks 2017-02-24 23:44 GMT+02:00 Richard Mudgett : > > > On Fri, Feb 24, 2017

Re: [asterisk-users] BUG or ???

On Fri, Feb 24, 2017 at 3:30 PM, Антон Сацкий wrote: > Got a strange situation > > [ext-queues] > ... > exten => h,2,ExecIf($[${CALLERID(num)} = ' ']?Set(var29=${SHELL(curl -X > POST --header "Content-Type: application/json" --header "Accept: > application/json" -d

Re: [asterisk-users] Bug in ast_frame_adjust_volume in 12.2.0?

Richard Kenner wrote: I'm getting a SIGSEGV at ast_slinear_saturated_multiply at the line: 351 res = (int) *input * *value; It's called from ast_frame_adjust_volume. The frame looks like: (gdb) print *f $6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 100021, format = {

Re: [asterisk-users] Bug in ast_frame_adjust_volume in 12.2.0?

This is an interpolated frame from func_jitterbuffer. It's part of packet loss concealment. What scenario exposed this? We were testing for clipping by doing Set(VOLUME(RX)=100) but we were connecting to a ConfBridge that had a jitterbuffer. This occurred when the phone (SIP) hung up. --

Re: [asterisk-users] bug? 'dahdi show channel x' HWEC echo cancellation display is incorrect while not on a call

On Thu, Dec 20, 2012 at 03:13:01PM -0800, Justin Killen wrote: When doing a 'dahdi show channel X' from the asterisk console, when the line is not part of a call the echo cancellation line ALWAYS says 'currently OFF'. Once a call is in progress, it will change to 'ON'. Is this a bug, or is the

Re: [asterisk-users] bug? 'dahdi show channel x' HWEC echo cancellation display is incorrect while not on a call

20, 2012 3:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] bug? 'dahdi show channel x' HWEC echo cancellation display is incorrect while not on a call On Thu, Dec 20, 2012 at 03:13:01PM -0800, Justin Killen wrote: When doing a 'dahdi show channel X

Re: [asterisk-users] bug in queuemanager?

] bug in queuemanager? Anyone? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger Sent: dinsdag 1 november 2011 13:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] bug

Re: [asterisk-users] bug in queuemanager?

...@lists.digium.com] On Behalf Of Henry Dogger Sent: Monday, November 07, 2011 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] bug in queuemanager? Perhaps some help on where to look myself? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk

Re: [asterisk-users] bug in queuemanager?

Discussion' Subject: Re: [asterisk-users] bug in queuemanager? Have you posted this to the forum Asterisk Support on asterisk.org? One thing I see is that you are doing an attended transfer (*2) vs a blind transfer (#1); that could be causing some sort of problem. From: asterisk-users-boun

Re: [asterisk-users] bug in queuemanager?

...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger Sent: Monday, November 07, 2011 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] bug in queuemanager? Nope, I encounter this with blind transfer as well as attended

Re: [asterisk-users] bug in queuemanager?

2011 16:46 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] bug in queuemanager? Do you have an isolated environment where you can do a core show channels verbose after the transfer, but before the end of the call? My suspicion is that you are spawning

Re: [asterisk-users] bug in queuemanager?

Of Danny Nicholas Sent: maandag 7 november 2011 16:46 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] bug in queuemanager? Do you have an isolated environment where you can do a core show channels verbose after the transfer, but before the end of the call

Re: [asterisk-users] bug in queuemanager?

2011 17:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] bug in queuemanager? Call 1 was from 346 to 900. The log in the link provided shows it correctly being in local/901 (line 8) from the queue and redialed (line 9). Line 12 seems to be in sync

Re: [asterisk-users] bug in queuemanager?

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger Sent: Monday, November 07, 2011 11:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] bug in queuemanager? Hm, I see what you mean. What I must add, the phones are registered on a different

Re: [asterisk-users] bug in queuemanager?

Anyone? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger Sent: dinsdag 1 november 2011 13:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] bug in queuemanager? Sorry it took

Re: [asterisk-users] bug in queuemanager?

Discussion Subject: Re: [asterisk-users] bug in queuemanager? On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger h.dog...@telecats.nl wrote: Customer 200 calls to queue 900, Agent 300 answers but tells Customer 200 that he should be at Queue 901 and transfers Customer 200 (using *2) to Queue 901

Re: [asterisk-users] Bug with Realtime?

I checked the bug reports and all I could find was similar issues with the Asterisk 1.6 which (according to the reports) have been resolved. I couldnt find anyone talking about 1.4 so I created a new issue and someone closed the case and added this note:- This does not appear to be a bug, but

Re: [asterisk-users] Bug with Realtime?

On Tue, 2010-09-21 at 19:04 -0400, Dan Journo wrote: I checked the bug reports and all I could find was similar issues with the Asterisk 1.6 which (according to the reports) have been resolved. I couldnt find anyone talking about 1.4 so I created a new issue and someone closed the case and

Re: [asterisk-users] Bug with Realtime?

I use realtime on 1.4 and 1.6 servers but always with rtcachefriends=yes in sip.conf I already use that and it doesnt seem to re-register when a call comes in. Only when the registration period expires, or the peer dials out. --

Re: [asterisk-users] Bug with Realtime?

Check the SIP debug and see what is going on. Leif. Hi, I checked the SIP debug. As soon as I issue the RELOAD command, no SIP data gets transferred to the phone. Asterisk output: http://pastebin.com/FB675N16 Any ideas how I can do a SIP reload without losing the Sip Phones registration?

Re: [asterisk-users] Bug with Realtime?

-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Monday, September 20, 2010 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bug with Realtime? Check the SIP debug and see what is going

Re: [asterisk-users] Bug with Realtime?

Can we not do pastebin any more? I just received this:- [PASTEBIN URL REMOVED] has been detected as suspicious URLs,and Quarantine to user's spam folder has been taken on 9/20/2010 8:24:38 AM. Message details: Server: MADRID Sender: d...@keshrcommunications.com; Recipient:

Re: [asterisk-users] Bug with Realtime?

My question is if you are using realtime, why are you doing a sip reload? I said previously:- Let's say I add a new provider to my service and therefore have to add another register= command into sip.conf, I'd have to issue a sip reload which would kill off all the realtime sip phones.

Re: [asterisk-users] Bug with Realtime?

On Mon, Sep 20, 2010 at 11:59:16AM -0400, Dan Journo wrote: Can we not do pastebin any more? No, it's just one user with an excessively paranoid and chatty mailfilter. -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Bug with Realtime?

Check the SIP debug and see what is going on. Alternatively you could turn off the qualify option with qualify=no. I'll take a look at the sip debug, but qualify needs to stay on, so thats not an option. -- _ -- Bandwidth

Re: [asterisk-users] Bug with Realtime?

That's not a bug. Only when the phone registers or performs some sort of action (such as placing a call, etc...) does Asterisk query the database. If your phones have a short re-registration time this becomes less of a problem. How do you explain that as soon as I issue a reload command,

Re: [asterisk-users] Bug with Realtime?

...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, September 16, 2010 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bug with Realtime? That's not a bug. Only when the phone registers or performs

Re: [asterisk-users] Bug with Realtime?

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, September 16, 2010 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bug with Realtime

Re: [asterisk-users] Bug with Realtime?

A reload flushes the SIP registration database, so once you do a reload, that phones reg is gone. Finally an answer that seemed more realistic. But it doesnt explain why the phones that are hard coded in the sip.conf file don't lose registration. Any ideas? Thanks Dan --

Re: [asterisk-users] Bug with Realtime?

As someone else said, the answer is don't do a 'reload', do an extensions reload or whatever it is specific to your changes. You are correct. I'm just being lazy. But I'm just worried that some time in the future, I'll have to reload the sip config, and therefore flush out all the realtime

Re: [asterisk-users] Bug with Realtime?

Danny Nicholas wrote: snip If your clients can't take 2 minutes of downtime on a phone, they don't need to be on VOIP. If VOIP ( and Asterisk ) ever really expect to be the future of Telephony this ( attitude ) has to change 90 percent availability is unacceptable, even 95 percent,

Re: [asterisk-users] Bug with Realtime?

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, September 16, 2010 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bug with Realtime

Re: [asterisk-users] Bug with Realtime?

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Thursday, September 16, 2010 9:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bug with Realtime

Re: [asterisk-users] Bug with Realtime?

it is specific to your changes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-bo... Sent: Thursday, September 16, 2010 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussi... Subject: Re: [asterisk-users] Bug with Realtime? That's

Re: [asterisk-users] Bug with Realtime?

Noted - but if OP does a reload once a day, 120 seconds (2 minutes) out of 1 day (14400 seconds) is 99.17% uptime; close enough to 99.999 percent in most folks books. What percentage of businesses use their phones 24/7? Even if its once a month, it's still too much in my book. No wonder

Re: [asterisk-users] Bug with Realtime?

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, September 16, 2010 10:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bug with Realtime

Re: [asterisk-users] Bug with Realtime?

But it doesnt explain why the phones that are hard coded in the sip.conf file don't lose registration. On a reload, it re-reads the sip.conf config file and sees the users in there, so it doesn't flush them. It doesn't pull down the whole SIP table on a reload, it only loads a realtime peer

Re: [asterisk-users] Bug with Realtime?

Have you checked the Issue Tracker Not yet. I wanted to see if it's just me before searching through/raising a bug report. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Bug with Realtime?

Is there any development work being done on the realtime addon? Theres been no updates since April. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Bug with Realtime?

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, September 16, 2010 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bug with Realtime

Re: [asterisk-users] Bug with Realtime?

On Thursday 16 September 2010 11:23:37 Dan Journo wrote: Is there any development work being done on the realtime addon? Theres been no updates since April. Realtime is integrated into the core; it is not an addon. Perhaps you're referring to the mysql realtime driver? The driver modules tend

Re: [asterisk-users] Bug with Realtime?

On 10-09-16 09:43 AM, Dan Journo wrote: That's not a bug. Only when the phone registers or performs some sort of action (such as placing a call, etc...) does Asterisk query the database. If your phones have a short re-registration time this becomes less of a problem. How do you explain that

Re: [asterisk-users] Bug with Realtime?

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, September 15, 2010 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bug with Realtime? Hi, I think ive found a

Re: [asterisk-users] Bug with Realtime?

By reload you mean sip reload or just any reload in general? Reload in general. It might be an issue only with the Polycom sip phones. Not been able to test any others. I'll try a software phone tomorrow. -- _ -- Bandwidth

Re: [asterisk-users] Bug with Realtime?

On 09/15/2010 09:41 PM, Dan Journo wrote: Hi, I think ive found a bug but need someone to double check. Whenever I issue a reload in Asterisk, any realtime extensions stop receiving calls. I have to reboot the sip phones in order to get them to re-register. Can anyone see if they have a

Re: [asterisk-users] Bug with Realtime?

On 10-09-15 03:41 PM, Dan Journo wrote: I think ive found a bug but need someone to double check. Whenever I issue a reload in Asterisk, any realtime extensions stop receiving calls. I have to reboot the sip phones in order to get them to re-register. Can anyone see if they have a similar

Re: [asterisk-users] Bug with Realtime?

You can do 'extensions reload' or 'ael reload' if you don't want to lose real-time sip registrations. I only reload what is needed to be reloaded instead of reloading everything. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-15 4:28 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On

Re: [asterisk-users] bug with Moh on MeetMe ?

Hi Michael, Can you show us the output from: moh show classes and moh show files Command Or try it to set a new exten after setting the language with: exten = 12345,n,Set(CHANNEL(musicclass)=personalised) Daniel Am 13.06.2010 um 12:35 schrieb Mickael Monsieur: Hello, The MeetMe application

Re: [asterisk-users] bug with Moh on MeetMe ?

Look! refuses MusicOnHold personalized -- Started music on hold, class 'personnalised' Can you see it?! Two typos. ;- Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk?

Re: [asterisk-users] bug in asterisk

Execute such commands with cronjob every night: /etc/init.d/asterisk stop sleep 3 killall -9 safe_asterisk killall -9 asterisk /etc/init.d/asterisk start Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com From:

Re: [asterisk-users] bug in asterisk

Hello I think this is not right way :) Look the log's files, find the problem and resolv The cronjob is the way to stay fat always online, until you find the problem :) Vardan Mindaugas Kezys wrote: Execute such commands with cronjob every night: /etc/init.d/asterisk stop sleep 3 killall

Re: [asterisk-users] bug in asterisk

Subject: Re: [asterisk-users] bug in asterisk Hello I think this is not right way :) Look the log's files, find the problem and resolv The cronjob is the way to stay fat always online, until you find the problem :) Vardan Mindaugas Kezys wrote: Execute such commands with cronjob every night: /etc

Re: [asterisk-users] Bug or feature: cdr_odbc.conf.sample

On Sunday 18 April 2010 04:10:11 Olivier wrote: From asterisk-1.6.1.18/configs/cdr_odbc.conf.sample : ; ; cdr_odbc.conf ; ;[global] ;dsn=MySQL-test ;username=username ;password=password ;loguniqueid=yes ;dispositionstring=yes ;table=cdr ;cdr is default table name

Re: [asterisk-users] Bug#557262: 2.6.31+2.6.31.4: XFS - All I/O locks up to D-state after 24-48 hours (sysrq-t+w available) - root cause found = asterisk

Justin Piszcz wrote: Found root cause-- root cause is asterisk PBX software. I use an SPA3102. When someone called me, they accidentally dropped the connection, I called them back in a short period. It is during this time (and the last time) this happened that the box froze under

Re: [asterisk-users] Bug#557262: 2.6.31+2.6.31.4: XFS - All I/O locks up to D-state after 24-48 hours (sysrq-t+w available) - root cause found = asterisk

On Sat, 21 Nov 2009, Faidon Liambotis wrote: Justin Piszcz wrote: Found root cause-- root cause is asterisk PBX software. I use an SPA3102. When someone called me, they accidentally dropped the connection, I called them back in a short period. It is during this time (and the last time)

Re: [asterisk-users] Bug#557262: 2.6.31+2.6.31.4: XFS - All I/O locks up to D-state after 24-48 hours (sysrq-t+w available) - root cause found = asterisk

When this bug occurs, it freezes I/O to all devices and the only way to recover is to reboot the system. Are you running asterisk with realtime priority (-p)? I once managed to take town a box with a dial plan loop; asterisk was taking to 100% CPU and because it had highest priority, nothing

Re: [asterisk-users] Bug or feature: SIP chanvars not overriden

On Wednesday 11 November 2009 14:23:31 Olivier wrote: Hello, Using 1.6.2-rc5, my settings include: [local-phone](!) context=mylocal type=friend nat=no canreinvite=no host=dynamic qualify=yes dtmf=info language=fr call-limit=5

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s![swapper:0]

I'd contact Digium - they're really good with providing support - just add the following line and dial it: Thanks Matt for your suggestion. We despatched a new TE412P card to replace the existing card but the same problem occurred. So, I think it is not the Digium card problem. At the same

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

The error on system crash is: Digum Board: TDM2400P OS: Debian Lenny 5.02 dahdi_ec_chunk 0x3c/0x200 [dahdi] SS:ESP 00068:F747FE38 KERNEL PANIC NOT SYNCING

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

Luis Morales wrote: The error on system crash is: Digum Board: TDM2400P OS: Debian Lenny 5.02 dahdi_ec_chunk 0x3c/0x200 [dahdi] SS:ESP 00068:F747FE38 KERNEL PANIC NOT SYNCING

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

You rite Kevin, We enabeled sec echo canceller. I'll be test now and let's know the results. Regards, On Tue, Aug 18, 2009 at 8:47 AM, Kevin P. Flemingkpflem...@digium.com wrote: Luis Morales wrote: The error on system crash is: Digum Board: TDM2400P OS: Debian Lenny 5.02

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

Luis Morales wrote: You rite Kevin, We enabeled sec echo canceller. I'll be test now and let's know the results. SEC is not a good choice. If you are going to try something other than HPEC, use MG2 or KB1, which are the current best options that are included with DAHDI. You can also stick

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

Keving, We use MG2 and KB1, but the best result was on SEC. How i can do to modify dahsi to include SEC option with generic CPU. Jhon Lee, Take a look on this link, there are an option to solve your issue: http://archives.free.net.ph/message/20080126.111546.a2569851.nl.html Regards, On Tue,

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

Luis Morales wrote: Keving, We use MG2 and KB1, but the best result was on SEC. How i can do to modify dahsi to include SEC option with generic CPU. If you are not using HPEC, then none of this matters; when you use an echo canceller included with DAHDI, it's compiled for your CPU type and

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

On Sat, Aug 15, 2009 at 10:58:07PM +1000, Lee, John (Sydney) wrote: I have this DELL PE2950 running Asterisk 1.4.21.2 on RHEL 5 with no problems since Dec last year. We are using Digium TE412P to connect to [snip] Pid: 0, comm: swapper EIP: 0060:[,C0417911.] CPU: 1 EIP is at

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

On 16/08/09 12:58 AM, Lee, John (Sydney) wrote: I have this DELL PE2950 running Asterisk 1.4.21.2 on RHEL 5 with no problems since Dec last year. We are using Digium TE412P to connect to an E1 ISDN line. Since Dec last year, we did not add or delete any software or hardware. We also did not do

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

I have the same trouble but on hp server. In my case the digium board used is : TE121P. My server is an ML150 G5 on ubuntu 8.04.2.TLS. I'll be appreciate if you can solve it. Regards, On Sun, Aug 16, 2009 at 6:36 PM, Matt Riddellli...@venturevoip.com wrote: On 16/08/09 12:58 AM, Lee, John

Re: [asterisk-users] Bug or Not?

'One touch park' was designed to work around this issue. PaulH Danny Nicholas wrote: Hi gang, When I try to park a call using blind-transfer (#1), the caller hears the lot instead of the transferring party. Attended transfer and blind transfer from the phone buttons (Polycom 501) work

Re: [asterisk-users] Bug or feature in 1.6.1 (Was: How to register with TCP transport) ?

I filed https://issues.asterisk.org/view.php?id=15202 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] bug 14153 and svn checkout.

Jerry Geis wrote: I just did an SVN check out and the fix for bug 14153 was not included in the SVN checkout. Is there something special I need to issue in the SVN checkout to get it? Jerry I did not include the command I used. svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk

Re: [asterisk-users] bug 14153 and svn checkout.

Jerry Geis schrieb: Jerry Geis wrote: I just did an SVN check out and the fix for bug 14153 was not included in the SVN checkout. Is there something special I need to issue in the SVN checkout to get it? I did not include the command I used. svn checkout

Re: [asterisk-users] bug 14153 and svn checkout.

Philipp Kempgen wrote: Jerry Geis schrieb: Jerry Geis wrote: I just did an SVN check out and the fix for bug 14153 was not included in the SVN checkout. Is there something special I need to issue in the SVN checkout to get it? I did not include the command I used. svn checkout

Re: [asterisk-users] bug 14153 and svn checkout.

Jerry Geis wrote: Jerry Geis wrote: I just did an SVN check out and the fix for bug 14153 was not included in the SVN checkout. Is there something special I need to issue in the SVN checkout to get it? Jerry I did not include the command I used. svn checkout

Re: [asterisk-users] Bug in contact header from Asterisk 1.6.0.3-rc1 ?

Hello, 1) How are you setting the nonstandard bind port? Just with a bindport on the specific peer? 2) The Record-Route header is only for proxies, so it is not relevant here; a B2BUA cannot set one. 3) What happens if you have the proxy append a 'received' parameter to the Contact URI

Re: [asterisk-users] bug in Asterisk 1.4.22?

Abel Monzon wrote: and then in my softphone I call to 1 the asterisk log say this: -- Launched AGI Script /usr/local/share/asterisk/agi-bin/a2billing.php == a2billing.php: Failed to execute '/usr/local/share/asterisk/agi-bin/a2billing.php': No such file or directory --

Re: [asterisk-users] bug in Asterisk 1.4.22?

Hi First off, you replied a previous mail to the list, and hence your message appears as part of a previous thread. To post a new message start a new message. Also, On Sun, Oct 26, 2008 at 01:47:03AM -0400, Abel Monzon wrote: Hello is my idea or this is a bug? The thing is that I have in my

Re: [asterisk-users] bug in Asterisk 1.4.22?

Also check the file permissions and if you are using a RedHat like OS, check the SELinux. And about using a2billing,I recommend you to use version 1.4.21 or less. On Sun, Oct 26, 2008 at 3:30 AM, Tzafrir Cohen [EMAIL PROTECTED]wrote: Hi First off, you replied a previous mail to the list, and

Re: [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18

For more info, I grab the relevant portion of the maillog. It looks like asterisk is trying to send using the right from email, but it's getting changed. This would suggest a sendmail problem, EXCEPT, it works fine when I send mail from the command line. Can anyone offer ideas? Mar 19

Re: [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18

to tell sendmail to trust the asterisk account or voicemail from address From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: March-19-08 9:57 AM To: Asterisk Users List Subject: Re: [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18

Re: [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18

List Subject: Re: [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18 For more info, I grab the relevant portion of the maillog. It looks like asterisk is trying to send using the right from email, but it's getting changed. This would suggest a sendmail problem, EXCEPT, it works

Re: [asterisk-users] bug in 1.2.24

Thank you for the example Isaac. I did as you mentioned and now it seems to be working perfectly.   Saludos   -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Isaac Xiao Sent: jueves, 13 de septiembre de 2007 10:33 p.m. To: asterisk-users@lists.digium.com

Re: [asterisk-users] bug in 1.2.24

On 13:33, Fri 14 Sep 07, Isaac Xiao wrote: Here is our dial plan. You need to avoid double recording as well when you transfer the call to other extension. exten = 7141,3,Set(CALLFILENAME=q${EXTEN}-${TIMESTAMP}-${UNIQUEID}) exten = 7141,4,Set(__FROM-EXT-QUEUES=ext-queues) exten =

Re: [asterisk-users] bug in 1.2.24

Thank Isaac, Ill try it this way.. Im currently using this before entering the queue so calls from the queue are recorded: exten = s,n,SetVar(MONITOR_FILENAME=/var/spool/asterisk/${TIMESTAMP}-${UNIQUEID}-${C ALLERIDNUM}-Queue-Ventas) exten = s,n,SetVar(TRANSFER_CONTEXT=internalphones) So I could

Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?

Sven Jacobs wrote: Dear users, I think I may found a bug in the voicemail module of Asterisk 1.4.2! Outgoing email notifications should use a real existing domain (let's call it domain.real) instead of the local domain (domain.local) so that some mail servers won't reject the mails.

Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?

Per Jessen wrote: Outgoing email notifications should use a real existing domain (let's call it domain.real) instead of the local domain (domain.local) so that some mail servers won't reject the mails. That's why I've set the serveremail option in voicemail.conf to [EMAIL PROTECTED]

Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?

Sven Jacobs wrote: You fix that in your mail-server with aliasing and/or canonicalising. I think the Asterisk behaviour is correct. It is similar to receiving an email from cron or some other daemon. That is sent from [EMAIL PROTECTED], which is fine for your internal purposes, but if you

Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?

As far as I can tell (but I'm on 1.4.1), the serveremail option only sets the From-address, not the envelope-address. The envelope will probably always be asterisk-user@hostname The From-address ist set by the fromstring option - which works btw - so you are wrong :) Unfortunately setting the

Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?

Sven Jacobs wrote: As far as I can tell (but I'm on 1.4.1), the serveremail option only sets the From-address, not the envelope-address. The envelope will probably always be asterisk-user@hostname The From-address ist set by the fromstring option - which works btw - so you are wrong :)

Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?

Maybe I'm misinterpreting things, but this is what I se: fromstring = the From:-text, not the From:-address. I'm just using the default fromstring, but I've set serveremail = asterisk@realdomain With this I get From: Asterisk PBX [EMAIL PROTECTED] Still, the envelope is [EMAIL

Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?

Sven Jacobs wrote: Maybe I'm misinterpreting things, but this is what I se: fromstring = the From:-text, not the From:-address. I'm just using the default fromstring, but I've set serveremail = asterisk@realdomain With this I get From: Asterisk PBX [EMAIL PROTECTED] Still, the envelope is

Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?

Joshua Colp wrote: The voicemail email gets handed off to sendmail for actual sending. It's adding on the envelope above. Yes, but asterisk is writing the From: header. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user -

Re: [asterisk-users] Bug in chan_sip mysql support and canreinvite?

5 jul 2006 kl. 13.46 skrev Roger Schreiter: Hi, I did not yet study the newest chan_sip.c versions, but it seems, that chan_sip treats mysql-peers different from other peers, concerning the variable canreinvite. If this variable is not explicitely set for a peer or user in sip.conf, the

Re: [Asterisk-Users] Bug in asterisk static realtime?

20 jun 2006 kl. 13.33 skrev Andrea Spadaccini: Hi folks, I used the ast2sql.pl script (found on www.voip-info.org) to put into the database a simple sip.conf. Among other entries, you could find: [general] context=sip-in ;incoming sip calls Well, the script put the comment into the database

Re: [Asterisk-Users] Bug in asterisk static realtime?

Ciao Olle, IMHO the comments should be stripped off by asterisk itself!! It should be easy to modify the script, but the problem would remain. Should it be filed as an Asterisk bug? A semicolon in realtime separates multiple values, it is *not* used as a comment. So you should fix

Re: [Asterisk-Users] Bug in Voicemail ??

Hi, I'm still a newbie, but try to help you, my voicemail works ok, I can also record messages ok. My extension part is: exten = s,1,Background(welcome-cisl) exten = 1,1,Dial(Sip/vmoreno,10) exten = 1,2,Voicemail(victor) exten = 2,1,Dial(Sip/juliansip,10) exten = 2,2,Voicemail(aajulian) exten

Re: Re: [Asterisk-Users] Bug in Voicemail ??

Hello Victor, Hi, I'm still a newbie, but try to help you, THX ;-)) And voicemail.conf part is: [general] format=wav49 maxmessage=180 minmessage=2 maxsilence=2 silencethreshold=150 maxlogins=3 [EMAIL PROTECTED] skipms=3000 [victor] victor = 1234, Victor Moreno, [EMAIL PROTECTED]

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