similar to: Problem with pjsip

Hello. I have two records in dialplan: exten => testA,1,Dial(PJSIP/outgoing/sip:thetestcall at sip.linphone.org) exten => testB,1,Dial(PJSIP/outgoing/sip:thetestcall at iptel.org) Calling testA works fine while testB fails with "CONGESTION". Adding debug for console shows that pjsip_resolver.c does `New queries added, performing parallel resolution again` for linphone after

On Thu, Aug 27, 2020 at 8:58 AM Leonid Fainshtein < leonid.fainshtein at xorcom.com> wrote: > I see that pjsip_resolver.c tries unsuccessfuly to resolve the > hostname each 10 seconds: > > [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Wrapper created > [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Set timer to > 2000 msec > [Aug 27 07:51:36]

I see that pjsip_resolver.c tries unsuccessfuly to resolve the hostname each 10 seconds: [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Wrapper created [Aug 27 07:51:36] DEBUG[595] res_pjsip.c: 0x7f75282eb150: Set timer to 2000 msec [Aug 27 07:51:36] DEBUG[595] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target 'rpi6.in.xorcom.com' [Aug 27 07:51:36] DEBUG[595]

I deleted the res_resolver_unbound.so module, and now it works as expected. So, the problem is related to the 'unbound' resolver? FYI: I'm using Asterisk 16.2 installed from Debian 10 repository. Best regards, Leonid Fainshtein On Thu, Aug 27, 2020 at 3:01 PM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Thu, Aug 27, 2020 at 8:58 AM Leonid Fainshtein < >

Hi, I created an IAX2 trunk between my old Asterisk 1.4 server (A) and my new one with v. 16.10.0 (B). The trunk seems to be up, and the calls are initiated, eg. an extension from A can dial an extension in B which rings. However, as soon as the extension in B answers, the call is terminated. This is what I see in the console of B: -- Called PJSIP/4053 -- PJSIP/4053-00000002 is ringing

3rd attempt to post it to the list, please ignore if it is duplicate I have the following problem When trying to setup asterisk 13.1 with PJSIP to connect to my IPV6 capable SIP provider the registration fails. [code][Dec 22 19:24:24] DEBUG[25247] pjsip: tsx0x110736c .Transaction created for Request msg REGISTER/cseq=36181 (tdta0x721d90) [Dec 22 19:24:24] DEBUG[25247] pjsip:

On Wed, Sep 30, 2020 at 8:47 AM sergio <sergio at outerface.net> wrote: > Hello. > > I have two records in dialplan: > > exten => testA,1,Dial(PJSIP/outgoing/sip:thetestcall at sip.linphone.org) > exten => testB,1,Dial(PJSIP/outgoing/sip:thetestcall at iptel.org) > > Calling testA works fine while testB fails with "CONGESTION". > > Adding debug

I was able to put the crash through the gdb on the original VM that encountered the problem. (Not sure why the VM I initially tried to analyze the crash dump on didn’t do this correctly, but not concerned about it now). It’s providing additional details. Would this be considered a better example of the crash? I will go through the version comparisons and see if there are any updates since

On 06/14/2017 at 10:17 PM, Joshua Colp wrote: > On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote: > > <snip> > >> >> I can now say, that asterisk / pjsip seams to work *mostly* as expected. >> Just one exception - and that's the package in question, which can't be >> seen in tcpdump. >> >> I extended the above patch by adding

Hi, I have Asterisk 16.x with a trunk configured with a hostname in PJSIP AOR. The registration is not required for this trunk. I paid attention that Asterisk performs DNS resolving of the hostname that is configured in the AOR 'contact' parameter only upon the Asterisk start only. Thus, if Asterisk is started when the DNS server is unreachable due to the Internet connection failure then

The Asterisk Development Team has announced the release of Asterisk 13.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

On 06/14/2017 at 05:53 PM Joshua Colp wrote: > On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote: > > <snip> > >> >> I added this patch to see, if really all packages are are freed after >> they have been processed: >> >> --- b/res/res_pjsip/pjsip_distributor.c 2017-05-30 19:44:16.000000000 >> +0200 >> +++

On Wed, Aug 9, 2023 at 3:20 PM Dan Cropp <dcropp at amtelco.com> wrote: > I have a customer who just encountered a crash while running Asterisk > 18.17.1 version. > > > > I’m trying to adapt to the changes so not sure where best to look or how > to possibly report this. > > > > I started by going through >

hi i need to install the api sdk for xen cloud platform can u pls tell me how to install the api sdk and, the there procedure how to develop my own program using python , can u pls tell me the steps for the sample progam -- R.G.BABUKARTHIK _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

The Asterisk Development Team has announced the release of Certified Asterisk 13.8-cert1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 13.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the

The Asterisk Development Team has announced the first beta of Asterisk 14.0.0. This beta is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.0.0-beta1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this beta: New

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.3-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.3-cert1 resolves several issues reported by the community and would have not been possible without your participation. *Thank you!* The following