It was due to a lack of tcp or udp sections with transport declaration
in pjsip.conf.
But it's still unclear,
1. How should I find this? Is a log so poor and needs to be reported, or
am I missing something?
2. Why I need to set bind? I use this transport only for outgoing
connections.
Joshua is lieing ASSHOLE
Sent from my iPhone
> On Sep 30, 2020, at 7:08 AM, Joshua C. Colp wrote:
>
>
>> On Wed, Sep 30, 2020 at 9:06 AM sergio wrote:
>
>> On 30/09/2020 14:59, Joshua C. Colp wrote:
>>
>> > latest version of 16 on Ubuntu
>>
>> 16.12.0~dfsg-1 ?
>
> I don't use packages.
On Wed, Sep 30, 2020 at 9:06 AM sergio wrote:
> On 30/09/2020 14:59, Joshua C. Colp wrote:
>
> > latest version of 16 on Ubuntu
>
> 16.12.0~dfsg-1 ?
>
I don't use packages. I built a fresh copy of 16.13.0 as it is latest and
used that.
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma
On 30/09/2020 14:59, Joshua C. Colp wrote:
latest version of 16 on Ubuntu
16.12.0~dfsg-1 ?
--
sergio.
--
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On Wed, Sep 30, 2020 at 8:47 AM sergio wrote:
> Hello.
>
> I have two records in dialplan:
>
> exten => testA,1,Dial(PJSIP/outgoing/sip:thetestc...@sip.linphone.org)
> exten => testB,1,Dial(PJSIP/outgoing/sip:thetestc...@iptel.org)
>
> Calling testA works fine while testB fails with
Hello.
I have two records in dialplan:
exten => testA,1,Dial(PJSIP/outgoing/sip:thetestc...@sip.linphone.org)
exten => testB,1,Dial(PJSIP/outgoing/sip:thetestc...@iptel.org)
Calling testA works fine while testB fails with "CONGESTION".
Adding debug for console shows that pjsip_resolver.c does
Hi All,
I'm new to Asterisk and I'm trying to manage the calls using the rest API.
I want to play a media file after 5 minutes the call started, how can I
move the call to a Stasis application using the HTTP API in order to play
the media using play API(
