On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote:
>
> 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 the info at the last
On 06/14/2017 at 05:53 PM Joshua Colp wrote:
> On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote:
>
>
>
>>
>> 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 19:44:16.0
>>
Thank you - At first glance it seems to have done the trick.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
Sent: June 14, 2017 10:41
To: Asterisk Users Mailing List - Non-Commercial
Does anyone on this list know how to make the Snom870 with FW:8.7.5.35
display the Caller ID in the display field while the ringing either
together with, or instead of, the topmost virtual key in the info
column? I realise that the purpose of having the virtual key display
the caller ID so as to
On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote:
>
> 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 19:44:16.0
> +0200
> +++ a/res/res_pjsip/pjsip_distributor.c 2017-06-13
On 06/11/2017 at 06:51 PM Joshua Colp wrote:
> On Sun, Jun 11, 2017, at 01:47 PM, Joshua Colp wrote:
>> The distributor is in res/res_pjsip/pjsip_distributor.c, the distributor
>> function being the entry point. That function returning PJ_TRUE
>> indicates to PJSIP that it has been handled and no
On 12.06.2017 17:00, Hans-Peter Jansen wrote:
>
> * zero prefix for outside calls
> * zero zero or plus prefix for international calls
> * handle emergency calls
>
> With ISDN, one was able to just forward the called number, but with sip, one
> has to normalize the dialed pattern in order to
On Wed, Jun 14, 2017 at 10:18:19AM -0400, Mike wrote:
> I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP.
> PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has
> its own callerid values and presence. I pass on those calls to PBX_B via
> SI, and I'm trying
Actually, a correction: the callerid isn't passed on properly either: on
SIP_B I get "Anonymous " instead of " <514-555-1234>" that my
dial app is sending.
The exact dial command that is used, once variables are evaluated, is
this:
Dial(SIP/pbx3/555,,f(""
Hi,
I've run into a minor snag trying to pass on CALLERID presence from one
Asterisk to another via SIP (both running 13.16.0)
I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP.
PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has
its own callerid
I am in the process of configuring my systems to store voicemail in a mysql
databse as opposed to on the filesystem, as it is now.
My backup server is currently configured for db storage, while my production
server is still using the filesystem during testing.
When I record a vm message on my
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