Thomas,
this code block should work for your Originate case.
This code block will dial a local channel where actual leg 1 number is
dialed. On Answer of leg1, the leg2 is called.
-
require_once('phpagi-2.20/phpagi-asmanager.php');
$asm = new
Hi,
I'm facing strange issue while establishing inbound calls from SIP trunks.
Provider A is sending (G729,Alaw,uLaw) offer and asterisk dial the peer
with its preferred codec order(G729,aLaw, uLaw). The peer's phone send the
codec list as (uLaw, speex) in 200 OK replay. The Peer's phone has select
On Wednesday, 14 September 2016, Madushan Geethanga
wrote:
> Hi,
>
> What is the equal option for externip in asterisk 13 with pjsip. I have
> tried
>
> external_media_address=XX.XX.XX.XX
> external_signaling_address=XX.XX.XX.XX
>
> but asterisk 13 writes local ip to the from header. any suggesti
Jacek,
This might be a bug or configuration issue, but you need to understand the
SIP Session Timers. With Session Timers you can control the round trip time
and Call Setup time of SIP Requests.
In case of GSM Network with high delay you need to set the T1 timer a
higher value like 1000ms (500 ms d
Thanks Richord and Carlos.
On Wednesday, 20 July 2016, Carlos Chavez wrote:
> On 7/20/16 9:58 AM, Faheem Muhammad wrote:
>
> Hi,
> I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS.
>
> When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the di
Hi,
I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS.
When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial
command breaks and the call control go to hangup block instead of next
priority. The error in CLI says "*Dial requires an argument
(technology/resource)*".
Strange, A BYE should be replied with 200 OK, 481 (non matching dialogid),
408 request time out or similar responses, but it should never be
challenged. Only INVITE, REGISTER and PUBLISH requests are challenged with
401/407.
As per rfc3261 it should not challenge the BYE Requests.
*The workaround
to PJSip
> and build RDNS records for the phones and the server, but I am still
> experiencing the problem on incoming calls.
>
>
> On 6/7/2016 1:00 PM, Faheem Muhammad wrote:
>
> I've faced the same issue. The issue was related to DNS, the reverse
> lookup query failur
I've faced the same issue. The issue was related to DNS, the reverse lookup
query failure caused the delay around(7-9 seconds). The purpose of reverse
lookup is to block IP Spoofing attacks.
Regards,
Faheem
On Tue, Jun 7, 2016 at 7:48 PM, Brent Davidson
wrote:
> I am having an issue with a coup
Try MixMonitor. Land the call to a local channel and answer it.
This code will record the silence as well.
exten => _X.,1,MixMonitor()
exten => _X.,n,Dial(Local/100@context1)
[context1]
exten => _X.,1,Answer()
exten => _X.,n,Dial(SIP/${EXTEN}
On Tue, Jun 7, 2016 at 2:16 PM, Mamadou NGOM wrote:
Israel,
You can calculate the time diff by this dialplan snippet.
---
exten =
_X.,1,Set(callstarttime=${STRFTIME(${EPOCH},,%Y%m%d)}${STRFTIME(${EPOCH},,%H%M%S)})
exten => _X.,n,Queue(queue1)
exten =
_X.,n,Set(c
MixMonitor() is non blocking command.
It sets recording instructions and jumps to next priority instantly.
On Tue, May 3, 2016 at 4:25 PM, Loic Chabert wrote:
> Hello,
>
> I try to find informations concerning Mixmonitor command, but ... without
> success.
> MixMonitor command take at last par
It is very simple, asterisk can log cdrs automatically by configuring
cdr_mysql.conf.
All you need to create a mysql table along with proper read/write
permissions. You can find the cdr table schema from the below link.
https://wiki.asterisk.org/wiki/display/AST/MySQL+CDR+Backend
Regards,
Muhamma
You can achieve this by choosing one of asterisk server for pins collection
on extension 1234. When any member/extension dial that extension you need
to call a script that will make AMI connection on all servers and do
AgentLogin/QueueAdd Request.
You need to do ami login and call the AMI request Q
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