Re: [asterisk-users] Delay when dialing...
On Fri, 23 Jul 2021, Jeff LaCoursiere wrote: Are you sure the call has been sent? Some phones have odd dialplans installed, and may not send the call to the SIP relay until you meet the dialplan reqs, press #, or otherwise wait the inter-digit timeout before the call is actually placed. If you enable SIP debugging (and bump up debug and verbose), is the delay between when you dial and the INVITE is displayed or is the delay between the INVITE and subsequent steps in your dialplan. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Delay when dialing...
Are you sure the call has been sent? Some phones have odd dialplans installed, and may not send the call to the SIP relay until you meet the dialplan reqs, press #, or otherwise wait the inter-digit timeout before the call is actually placed. If this is the case you need to take a hard look at the dialplan string for the endpoint, and edit accordingly. j On 7/23/21 9:35 AM, Carlos Chavez wrote: Thank you. The server is running dnsmasq locally for DNS resolution and all queries resolve properly. I just added the hostname to /etc/hosts and restarted but the delay persists. On 7/23/2021 1:41 AM, Jean Aunis wrote: Le 22/07/2021 à 18:32, Carlos Chavez a écrit : I started noticing a few days ago that whenever I dial any number or extension there is a delay of 5 to 10 seconds before Asterisk reacts. I see nothing on the CLI for that time and then the call goes through. I have checked my network to make sure there is nothing slowing down packets between the phones and the server. Any settings I should check on the Asterisk side? This is happening with all phones (several brands). Hi, I've seen this problem several times when there is no DNS resolution of Asterisk's hostname. Try to add your hostname to /etc/hosts and check if it's better. Regards, Jean -- Jeff LaCoursiere StratusTalk, Inc. 703 496 4990 x108 815 546 6599 cell -- _ -- 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] Delay when dialing...
Thank you. The server is running dnsmasq locally for DNS resolution and all queries resolve properly. I just added the hostname to /etc/hosts and restarted but the delay persists. On 7/23/2021 1:41 AM, Jean Aunis wrote: Le 22/07/2021 à 18:32, Carlos Chavez a écrit : I started noticing a few days ago that whenever I dial any number or extension there is a delay of 5 to 10 seconds before Asterisk reacts. I see nothing on the CLI for that time and then the call goes through. I have checked my network to make sure there is nothing slowing down packets between the phones and the server. Any settings I should check on the Asterisk side? This is happening with all phones (several brands). Hi, I've seen this problem several times when there is no DNS resolution of Asterisk's hostname. Try to add your hostname to /etc/hosts and check if it's better. Regards, Jean -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 -- _ -- 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] AST-2021-008: Remote crash when using IAX2 channel driver
On Fri, Jul 23, 2021 at 6:12 AM Doug Lytle wrote: > >>> Asterisk Project Security Advisory - AST-2021-008 > > Downloading asterisk-16-current.tar.gz is still showing Asterisk 16.19.0 > Links should be fixed now. > > Doug > > -- > _ > -- 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 -- _ -- 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] problems with natted phones
I achieved a partial success adding --use-compact-form option. Marek 2021-07-23 13:47 GMT+02:00, Marek Greško : > Hello, > > your suggestion to turn off SIP ALG on provider's router was probably > correct. no problem until now. Thank you very much. > > I just found out another issue. I had a pjsue client in that network > which called specific number when turned on. It was working perfectly > with the old provider with working SIP ALG. But now with this provider > and SIP ALG disabled I am not able to make the call using pjsua > client. > > My pjsua config looks like this: > --id sip:ext@asterisk.domain > --registrar sip:asterisk.domain > --proxy sip:asterisk.domain > --outbound sip:asterisk.domain > --realm * > --username username > --password password > --null-audio > --no-tcp > --max-calls=1 > --no-vad > > The pjsua client successfully registers but is unable to call. > > I see the following: > IP address change detected for account 1 > (localip:5060-->nattedip:newport). Updating registration (using method > 4) > Temporary failure in sending Request msg INVITE/cseq=, will try > next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT) > > What could be the problem? How can I convince pjsue to work correctly > behind nat? > > Thanks > > Marek > > > > > > 2021-07-10 11:08 GMT+02:00, Marek Greško : >> Hello, >> >> I just disabled. Currently it is working. I shloud give it some time >> to confirm the problem has gone. Maybe one month would be enough to >> confirm. >> >> Thanks >> >> Marek >> >> >> 2021-07-09 20:11 GMT+02:00, Abdenasser Ghomri : >>> Yes just disable the SIP ALG and see if it helps, Thanks. >>> >>> Best Regards, >>> >>> On Fri, Jul 9, 2021, 09:10 Antony Stone < >>> antony.st...@asterisk.open.source.it> wrote: >>> On Friday 09 July 2021 at 08:47:46, Marek Greško wrote: > Hello, > > yes SIP ALG are anbled on the router. Should I disable? In my opinion, always. Antony. -- I don't know, maybe if we all waited then cosmic rays would write all our software for us. Of course it might take a while. - Ron Minnich, Los Alamos National Laboratory 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 >>> >> > -- _ -- 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] AST-2021-008: Remote crash when using IAX2 channel driver
>>> Asterisk Project Security Advisory - AST-2021-008 Downloading asterisk-16-current.tar.gz is still showing Asterisk 16.19.0 Doug -- _ -- 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] pjsip in 18.5.X
Is there a way to "not" compile/configure pjsip in 18 ? I am still using the older SIP channel driver and have not converted over just yet. Thanks, Jerry -- _ -- 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] problems with natted phones
Hello, your suggestion to turn off SIP ALG on provider's router was probably correct. no problem until now. Thank you very much. I just found out another issue. I had a pjsue client in that network which called specific number when turned on. It was working perfectly with the old provider with working SIP ALG. But now with this provider and SIP ALG disabled I am not able to make the call using pjsua client. My pjsua config looks like this: --id sip:ext@asterisk.domain --registrar sip:asterisk.domain --proxy sip:asterisk.domain --outbound sip:asterisk.domain --realm * --username username --password password --null-audio --no-tcp --max-calls=1 --no-vad The pjsua client successfully registers but is unable to call. I see the following: IP address change detected for account 1 (localip:5060-->nattedip:newport). Updating registration (using method 4) Temporary failure in sending Request msg INVITE/cseq=, will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT) What could be the problem? How can I convince pjsue to work correctly behind nat? Thanks Marek 2021-07-10 11:08 GMT+02:00, Marek Greško : > Hello, > > I just disabled. Currently it is working. I shloud give it some time > to confirm the problem has gone. Maybe one month would be enough to > confirm. > > Thanks > > Marek > > > 2021-07-09 20:11 GMT+02:00, Abdenasser Ghomri : >> Yes just disable the SIP ALG and see if it helps, Thanks. >> >> Best Regards, >> >> On Fri, Jul 9, 2021, 09:10 Antony Stone < >> antony.st...@asterisk.open.source.it> wrote: >> >>> On Friday 09 July 2021 at 08:47:46, Marek Greško wrote: >>> >>> > Hello, >>> > >>> > yes SIP ALG are anbled on the router. Should I disable? >>> >>> In my opinion, always. >>> >>> Antony. >>> >>> -- >>> I don't know, maybe if we all waited then cosmic rays would write all >>> our >>> software for us. Of course it might take a while. >>> >>> - Ron Minnich, Los Alamos National Laboratory >>> >>>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 >> > -- _ -- 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
