Re: [asterisk-users] Call List Campaign to an IVR

2017-02-02 Thread Pete Mundy
On 2/02/2017, at 9:52 pm, A J Stiles wrote: > > but in simple solidarity with everyone who has ever > been pissed off by a machine-initiated spam marketing phone call at an > inappropriate moment, I am not going to tell you how to do it. > Hat-tip to you, AJ :)

[asterisk-users] Problem with rport (CGNAT) going from Linux kernel 3.16 to 4.9

2017-02-02 Thread martin f krafft
Hello, I operate an Asterisk server (v11.13.1) on Debian stable, and it's rock-solid. The other day, however, I accidentally upgraded the kernel from the stable 3.16.0 to 4.9.0. Subsequently, audio stopped working. Below you can find my analysis while running the 4.9.0 kernel. 888 is a simply

[asterisk-users] asterisk 13.13.1 Everyone is busy-congested at this time (1:1/0/0)

2017-02-02 Thread Motty Cruz
Hi, my server is running a fresh install of Asterisk 13.13.1 on CentOS 7. My extensions.conf file was mostly copied from server running Asterisk 1.8. That being said! If I dial a number and get a busy signal I get the following error: -- SIP/voipeer-084b redirecting info has changed,

Re: [asterisk-users] asterisk callerid issue PJSIP Realtime

2017-02-02 Thread George Joseph
On Thu, Feb 2, 2017 at 4:06 AM, Zakir Mahomedy wrote: > Yes, from_user was set, removing those entries solved the problem. > > Can someone please explain to me the correct use for fromuser field? > from_user forces the user portion of the From header to a specific value on

Re: [asterisk-users] PJSIP hangupcause how to

2017-02-02 Thread Joshua Colp
On Thu, Feb 2, 2017, at 08:11 AM, Saint Michael wrote: > if a PJSIP call fails, how can I capture SIP code, like 503,603 etc? > in old SIP channel, we had ${HASH(SIP_CAUSE,)} > but in PJSIP it has to be the outbound channel, which is gone when the > control returns to the calling channel. This

[asterisk-users] PJSIP hangupcause how to

2017-02-02 Thread Saint Michael
if a PJSIP call fails, how can I capture SIP code, like 503,603 etc? in old SIP channel, we had ${HASH(SIP_CAUSE,)} but in PJSIP it has to be the outbound channel, which is gone when the control returns to the calling channel. --

Re: [asterisk-users] asterisk callerid issue PJSIP Realtime

2017-02-02 Thread Zakir Mahomedy
Yes, from_user was set, removing those entries solved the problem. Can someone please explain to me the correct use for fromuser field? thanksZakir On Wednesday, February 1, 2017 8:00 PM, "asterisk-users-requ...@lists.digium.com" wrote: Send

[asterisk-users] asterisk13+app_queue scalability

2017-02-02 Thread marek cervenka
hi, i have similar problem to https://issues.asterisk.org/jira/browse/ASTERISK-25806 do you know about some workarounds/patches for better scalability? thanks marek -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Call List Campaign to an IVR

2017-02-02 Thread A J Stiles
On Thursday 02 Feb 2017, Amelye Chatila wrote: > Hi, > I need to make calls to a list of numbers one at a time and once the user > pick the phone connects to an IVR where I can get few data, after a call > finishes the 2nd number get called and so forth. > > I'm familiar with Asterisk/Elastix

Re: [asterisk-users] Call List Campaign to an IVR

2017-02-02 Thread Amelye Chatila
To be more clear I need unattended campaign On Feb 2, 2017 11:26 AM, "Amelye Chatila" wrote: > Hi, > > > I need to make calls to a list of numbers one at a time and once the user > pick the phone connects to an IVR where I can get few data, after a call > finishes the 2nd

[asterisk-users] Call List Campaign to an IVR

2017-02-02 Thread Amelye Chatila
Hi, I need to make calls to a list of numbers one at a time and once the user pick the phone connects to an IVR where I can get few data, after a call finishes the 2nd number get called and so forth. I'm familiar with Asterisk/Elastix but the Campaign feature on Elastix does not seem to fill