Re: [asterisk-users] Asterisk simply stops call processing
On Wednesday 22 February 2023 at 15:29:38, John Harragin wrote:
> If there are multiple connections that the utilize the same driver, try
> putting:
>
> Threading = 2
>
> in the appropriate driver section of
> /etc/odbcinst.ini
I'll give that a go, however I doubt that it is the problem, since I see the
correct result from the ODBC query recorded in the assignment verbose log
output, therefore the query is done and the result has been used by the time
Asterisk freezes.
> ...this would be a possibility if the problem is intermittent.
It's actually extremely repeatable - I have not seen call processing proiceed
beyond this stage once so far.
> Also can you successfully execute the same SQL from the cli?
Yes, and as I say, they query is working fine and Asterisk is correctly using
the returned value in the assignment.
The further detail which I think I added in a later post is that this is
actually in a context which gets called using a Gosub() from two different
places in the dialplan.
From one, it works fine; from the other, it gets stuck. Completely consistent.
> By the way, what driver is asterisk using?
You mean ODBC? That's connected to MariaDB.
> On Mon, Feb 20, 2023 at 11:12 PM Antony Stone wrote:
> > Hi.
> >
> > I have a strange problem and I'm looking for suggestions on how to
> > investigate it.
> >
> > I have a dialplan which is processing a call, and Asterisk simply stops
> > doing anything for that call.
> >
> > I have verbose and debug logging turned on.
> >
> > There are two steps at a particular point in the dialplan:
> > Set(UserCredit=${ODBC_GENERIC(select Credit('${DDI}'))})
> >
> > Verbose(6,Current credit level for user ${DDI} is ${UserCredit}
> > pence)
> >
> >
> > Everything gets processed up to and including the first line - the
> > verbose log file shows me:
> >
> > pbx.c:2946 in pbx_extension_helper: Executing
> > [0044509903@DialBleg:46]
> > Set("SIP/TrunkTwo-1184", "UserCredit=999") in new stack
> >
> > (Phone number obscured here for anonymity).
> >
> > Then, that is it. Nothing further happens with call processing (until
> > one of the parties hangs up) and the second dialplan command above never
> > appears in the verbose log file. I have several other Verbose(6,.)
> > commands preceding this, and they all output into the log file as expected.
> >
> > If another call arrives on the same server, Asterisk quite happily starts
> > processing it and records what it's doing in the log files.
> >
> >
> > Can anyone suggest how I can investigate what Asterisk is doing at the
> > point where it "gets stuck", and how to find out why it simply stops
> > processing the call and doesn't continue with the dialplan commands?
> >
> >
> > Thanks,
> >
> >
> > Antony.
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Re: [asterisk-users] RTP address learning and timing problem
On Tue, Feb 28, 2023 at 9:50 AM David Cunningham wrote: > Hello, > > Does anyone know if one of the "strictrtp" options disables RTP learning? > As far as I can tell from the documentation the values "no" and "seqno" are > more permissive in allowing other sources rather than less, but I thought > I'd check. > Setting it to "no" disables the learning. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- 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: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 5s delays before executing the dialplan
On Tue, Feb 28, 2023 at 9:48 AM Kingsley Tart wrote: > Hi, > > We've recently hit an issue with Asterisk 18.8.0 where a call comes in > via SIP (using pjsip) but it can take 5 seconds before starting to > execute the dialplan. > > This was intermittent, but frequent (eg approx half of the calls). > > We have verbose logging on, but I didn't see any errors. > > Running asterisk -r - and then watching SIP traffic in another > window showed the INVITE coming in, then a good 5 seconds before > dialplan execution started showing within the Asterisk console. > > We've never seen this before, but it affected 6 our of 8 of our > Asterisk servers. They're running in debian 11 VMS, with the VMs > running under KVM on the host OS which is also debian 11. > > Any suggestions where to look next? We've been running Asterisk for > years but never seen this issue before. > > FWIW, the match= lines for the SIP proxies sending to Asterisk are > configured by IPv4 address, not host name. > Is the local hostname configured in /etc/hosts and not reliant on an outside DNS server? Are you using ICE or STUN at all? -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- 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: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ping
My last post did not make it back or to the archive... testing... Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- 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: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk PJSIP setting don't fragment bit on UDP
Hi Gang I noticed, that when I enable multiple codecs and rtp encrypting (generating a large SDP) invites with credentials do not get through anymore. So sniffed the connection and found that the IP packets have the don't fragment bit set, causing a VDSL router with 1472 MTU in the path to reject them. So it asterisk or the underlying OS the culpit? nping --udp -g 5070 -p 5060 registrarIP --data-length 1472 sniffing both sides. nping issues packets without don't fragment bit. Router fragments them, registrar receives two fragmented packets per one sent packet. So I guess it's asterisk which forcefully is setting don't fragment. But I could not find any such setting regarding SIP. Did I miss something? How do I make asterisk not set the don't fragment bit on UDP? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- 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: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP address learning and timing problem
Hello, Does anyone know if one of the "strictrtp" options disables RTP learning? As far as I can tell from the documentation the values "no" and "seqno" are more permissive in allowing other sources rather than less, but I thought I'd check. Thanks. On Thu, 23 Feb 2023 at 12:13, David Cunningham wrote: > Hello, > > We have a system that interoperates with an external service, so that the > basic call flow is: > > PSTN origination -> Asterisk A -> External service -> Asterisk B > > Initially the SDP from the external service tells the two Asterisks to > send RTP directly to each other. Part way through the call the external > service sends re-INVITEs both Asterisks to change the address for audio to > itself, but this fails to work intermittently. The problem seems to be one > of timing. > > If there's no RTP between the two re-INVITEs then it works fine, and both > Asterisks send future RTP to the external service as instructed. > > The problem is if RTP is transmitted/received in the fraction of the > second between the two re-INVITEs. If Asterisk A receives the re-INVITE > first, and then receives RTP from Asterisk B (which hasn't yet received its > re-INVITE), then it re-learns the media address of Asterisk B and sends > audio there instead of the new address. Asterisk B gets the second > re-INVITE with the new media address, but soon re-learns the media address > of Asterisk A because it's getting RTP from it. > > Note we have "canreinvite = no" in sip.conf, but I don't think that's > relevant to the problem. > > Can anyone suggest how to prevent this problem? Is it possible to turn off > learning the media address per call or per peer? > > Thanks for your help. > > -- > David Cunningham, Voisonics Limited > http://voisonics.com/ > USA: +1 213 221 1092 > New Zealand: +64 (0)28 2558 3782 > -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 5s delays before executing the dialplan
Hi, We've recently hit an issue with Asterisk 18.8.0 where a call comes in via SIP (using pjsip) but it can take 5 seconds before starting to execute the dialplan. This was intermittent, but frequent (eg approx half of the calls). We have verbose logging on, but I didn't see any errors. Running asterisk -r - and then watching SIP traffic in another window showed the INVITE coming in, then a good 5 seconds before dialplan execution started showing within the Asterisk console. We've never seen this before, but it affected 6 our of 8 of our Asterisk servers. They're running in debian 11 VMS, with the VMs running under KVM on the host OS which is also debian 11. Any suggestions where to look next? We've been running Asterisk for years but never seen this issue before. FWIW, the match= lines for the SIP proxies sending to Asterisk are configured by IPv4 address, not host name. Cheers, Kingsley. -- _ -- 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: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP address learning and timing problem
Hello, We have a system that interoperates with an external service, so that the basic call flow is: PSTN origination -> Asterisk A -> External service -> Asterisk B Initially the SDP from the external service tells the two Asterisks to send RTP directly to each other. Part way through the call the external service sends re-INVITEs both Asterisks to change the address for audio to itself, but this fails to work intermittently. The problem seems to be one of timing. If there's no RTP between the two re-INVITEs then it works fine, and both Asterisks send future RTP to the external service as instructed. The problem is if RTP is transmitted/received in the fraction of the second between the two re-INVITEs. If Asterisk A receives the re-INVITE first, and then receives RTP from Asterisk B (which hasn't yet received its re-INVITE), then it re-learns the media address of Asterisk B and sends audio there instead of the new address. Asterisk B gets the second re-INVITE with the new media address, but soon re-learns the media address of Asterisk A because it's getting RTP from it. Note we have "canreinvite = no" in sip.conf, but I don't think that's relevant to the problem. Can anyone suggest how to prevent this problem? Is it possible to turn off learning the media address per call or per peer? Thanks for your help. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk simply stops call processing
If there are multiple connections that the utilize the same driver, try
putting:
Threading = 2
in the appropriate driver section of
/etc/odbcinst.ini
...this would be a possibility if the problem is intermittent.
Also can you successfully execute the same SQL from the cli?
By the way, what driver is asterisk using?
On Mon, Feb 20, 2023 at 11:12 PM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:
> Hi.
>
> I have a strange problem and I'm looking for suggestions on how to
> investigate
> it.
>
> I have a dialplan which is processing a call, and Asterisk simply stops
> doing
> anything for that call.
>
> I have verbose and debug logging turned on.
>
> There are two steps at a particular point in the dialplan:
>
>
> Set(UserCredit=${ODBC_GENERIC(select Credit('${DDI}'))})
>
> Verbose(6,Current credit level for user ${DDI} is ${UserCredit}
> pence)
>
>
> Everything gets processed up to and including the first line - the verbose
> log
> file shows me:
>
> pbx.c:2946 in pbx_extension_helper: Executing [0044509903@DialBleg:46]
>
> Set("SIP/TrunkTwo-1184", "UserCredit=999") in new stack
>
> (Phone number obscured here for anonymity).
>
> Then, that is it. Nothing further happens with call processing (until one
> of
> the parties hangs up) and the second dialplan command above never appears
> in
> the verbose log file. I have several other Verbose(6,.) commands
> preceding
> this, and they all output into the log file as expected.
>
> If another call arrives on the same server, Asterisk quite happily starts
> processing it and records what it's doing in the log files.
>
>
> Can anyone suggest how I can investigate what Asterisk is doing at the
> point
> where it "gets stuck", and how to find out why it simply stops processing
> the
> call and doesn't continue with the dialplan commands?
>
>
> Thanks,
>
>
> Antony.
>
> --
> "The future is already here. It's just not evenly distributed yet."
>
> - William Gibson
>
>Please reply to the
> list;
> please *don't* CC
> me.
>
> --
> _
> -- 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:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Changing the contact header
Hi David, I chanced upon your question while I was looking for the same thing myself. I don't know whether this is still relevant to you, given that it's over 2 years ago since you asked the question. There's an option in the global section of pjsip.conf that defaults to "no", but if you set it to yes, the user part of the Contact header will be taken from the value of CALLERID(num): I found this looking at the source code in res_pjsip.c - there's a load of XML documentation at the top. [global] use_callerid_contact=yes Cheers, Kingsley. On Wed, 2020-11-25 at 05:58 -0500, Dovid Bender wrote: > Hi, > > Is anyone aware of any way of changing the contact header on a call? > We are sending 911 calls to a provider and they require that the > contact be the call back number. I tried: > Set(PJSIP_HEADER(update,contact)=) > > But the came back with: > No headers had been previously added to this session. > > If I try to do: > Set(PJSIP_HEADER(add,contact)=) > Then Asterisk just adds a second contact header but does not replace > the original (I also suspect adding a contact breaks the RFC?) > > Is anyone aware of a way to modify the contact header? > > TIA. > > Dovid > -- _ -- 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: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
