similar to: some domains resolving issues

Hello everyone. I allow myself to submit a problem that I can not solve with my VOIP provider Orange in France [2023-06-08 13:19:03] ERROR[185091]: res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error configuring endpoint 'Biv_Sortie' - 'from_user' field contains invalid character '@' [2023-06-08 13:19:03] ERROR[185091]: config_options.c:798 aco_process_var:

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 < >

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

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:

Am 03.03.2015 um 18:16 schrieb James B. Byrne: > CentOS-6.5 (FreePBX-2.6) > Asterisk-11.14.2 (FreePBX) > snom870-SIP 8.7.3.25.5 > > I am having a very difficult time attempting to get TLS and SRTP > working with Asterisk and anything else. At the moment I am trying to > get TLS functioning with our Snom870 desk-sets. And I am not having > much luck. > > Since this

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

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

CentOS-6.5 (FreePBX-2.6) Asterisk-11.14.2 (FreePBX) snom870-SIP 8.7.3.25.5 I am having a very difficult time attempting to get TLS and SRTP working with Asterisk and anything else. At the moment I am trying to get TLS functioning with our Snom870 desk-sets. And I am not having much luck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten

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 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

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 all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem

I can not register via sip to iptel or sipgate. i do not unterstand why.. but i am new to asterisk. Iam behind a susefirewall2 but asterisk even do not register if it shut down. No answer seems coming back. thx for help. nico here is my config if anybody can help: ----------------------------------------- [general] port = 5060?????????????????????; Port to bind to bindaddr =

Bug in the register= code; see details below from the developer of "ser" (SIP Express Router) Apparently, ACKs don't need to be sent on OK's to REGISTERs. Plus, malformed data somewhere... no details on that, though. JT >Date: Sun, 23 Feb 2003 23:54:07 +0100 >To: John Todd <jtodd at loligo.com> >From: Jiri Kuthan <jiri at iptel.org> >Subject: Re:

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

I can not register via sip to iptel or sipgate and do not see sip into ethereal. I do not unterstand why thats Wudu .. but i am new to asterisk and sip. I am behind a susefirewall2 but asterisk even do not register if it is down. The asterisk is running onto the machine witch is connected to the internet. No answer seems coming back from iptel (sip debug in asterisk). Ports are open (5060,

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>