Re: [asterisk-users] AMI Originate not working

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 =

[asterisk-users] codec negotiation or transcoding issue

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

Re: [asterisk-users] Asterisk 13 externip

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

Re: [asterisk-users] chan_pjsip ignoring endpoint device state (qualify) on dial

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

Re: [asterisk-users] PJSIP_DIAL_CONTACTS issue

Thanks Richord and Carlos. On Wednesday, 20 July 2016, Carlos Chavez <cur...@telecomabmex.com> 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

[asterisk-users] PJSIP_DIAL_CONTACTS issue

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)*".

Re: [asterisk-users] Authentication header in BYE packets

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

Re: [asterisk-users] Delay after Answer

blem solved. Ported everything over 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 re

Re: [asterisk-users] Delay after Answer

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

Re: [asterisk-users] Want to detect sound

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

Re: [asterisk-users] variable to get waittime of caller exiting queue

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 =

Re: [asterisk-users] Is MixMonitor command is blocking ?

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

Re: [asterisk-users] I want to store cdr into database

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,

Re: [asterisk-users] AgentLogin() on the multiple servers?

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