[asterisk-users] For a failed retransmission - what were the IP addresses?

2015-07-02 Thread Stefan Viljoen
Hi Guys Given these occassional errors on my Asterisk CLI: [Jul 2 10:23:36] WARNING[2060]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission 17bb3a993ad10f8818970ae952b81e73@192.168.11.31:5060 for seqno 102 (Critical Request) -- See

Re: [asterisk-users] asterisk email to fax

2015-07-02 Thread tux john
being honest with i have been lost on what to do. all i want is sent from my email a pdf file and then the server will sent it as fax. what settings do i have to do regarding emailing to the server? what other settings do i have to do? is there a guide on that? Sent:Friday, June 26, 2015 at

Re: [asterisk-users] Asterisk 11 and pulse

2015-07-02 Thread Rusty Newton
On Wed, Jun 24, 2015 at 7:03 AM, Jerry Geis ge...@pagestation.com wrote: I am looking for some great instructions on using asterisk with pulse. I'm using centos 7 and pulse as a user and not having much luck. I have changed all permissions for the asterisk directories. set asterisk.conf

Re: [asterisk-users] Custom header when busy

2015-07-02 Thread Rusty Newton
On Wed, Jul 1, 2015 at 4:46 AM, r...@yandex.ru wrote: Hi, all Is there someway ability to insert custom Header to SIP 486 message, when HANGUP application is invoked? Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom

Re: [asterisk-users] Custom header when busy

2015-07-02 Thread royj
Is there any chance to create feature request for that useful functionality? 02.07.2015, 14:03, "Rusty Newton" rnew...@digium.com:On Wed, Jul 1, 2015 at 4:46 AM, r...@yandex.ru wrote:Hi, all Is there someway ability to insert custom Header to "SIP 486" message, when HANGUP application is invoked?

Re: [asterisk-users] Custom header when busy

2015-07-02 Thread jg
Thanks for the tip. Our goal is to know that call-limit is triggered. And later analyze this info, maybe do some action. Yes, we can parse CDRs or execute AGI script but we do not want inmplement this logic on Asterisk because it can affect performance. 02.07.2015, 15:31, jg

Re: [asterisk-users] Custom header when busy

2015-07-02 Thread royj
Thanks for the tip. Our goal is to know that call-limit is triggered. And later analyze this info, maybe do some action.Yes, we can parse CDRs or execute AGI script but we do not want inmplement this logic on Asterisk because it can affect performance. 02.07.2015, 15:31, "jg"

Re: [asterisk-users] Custom header when busy

2015-07-02 Thread jg
Is there any chance to create feature request for that useful functionality? 02.07.2015, 14:03, Rusty Newton rnew...@digium.com: On Wed, Jul 1, 2015 at 4:46 AM, r...@yandex.ru mailto:r...@yandex.ru wrote: Hi, all Is there someway ability to insert custom Header to SIP 486 message,

Re: [asterisk-users] Custom header when busy

2015-07-02 Thread royj
* call-limit on PBX is triggered 02.07.2015, 15:49, "r...@yandex.ru" r...@yandex.ru:Thanks for the tip. Our goal is to know that call-limit is triggered. And later analyze this info, maybe do some action.Yes, we can parse CDRs or execute AGI script but we do not want inmplement this logic on

Re: [asterisk-users] asterisk email to fax

2015-07-02 Thread jg
being honest with i have been lost on what to do. all i want is sent from my email a pdf file and then the server will sent it as fax. what settings do i have to do regarding emailing to the server? what other settings do i have to do? is there a guide on that? *Sent:* Friday, June 26, 2015

Re: [asterisk-users] asterisk email to fax

2015-07-02 Thread tux john
At the moment it receives fax and sends it as pdf to my email. i have followed this tutorial http://the-asterisk-book.com/1.6/faxserver-mit-iaxmodem-und-hylafax.html i have setup postfix to send using my gmail account. how do i do the reverse to send from my email? Sent:Thursday, July 02,

[asterisk-users] multiple sip trunks with the same ITSP

2015-07-02 Thread Антон Сацкий
HI LIST CAN U HELP ME If there are multiple sip trunks with the same ITSP then an incoming call is arbitarily matched to the last peer with the same host IP address. This is not a serious problem because the DID is still correct but it does have many insidious effects due to the incorrect channel

Re: [asterisk-users] Asterisk 11 and pulseaudio setup as local user

2015-07-02 Thread Jerry Geis
I'm not sure that your question is clear. You'll probably want to be more specific. What is pulse? You mention as a user, are you talking about voicepulse.com ? What are you trying to do with pulse? What problem are you running into? Sorry Rusty... I am trying to get Asterisk 11 to

Re: [asterisk-users] Asterisk 11 and pulseaudio setup as local user

2015-07-02 Thread Matt Riddell
I am trying to get Asterisk 11 to co-exist with a centos 7 box that has pulse audio running as a local user. Has anyone done that? What is the trick? I changed directories /var/run/asterisk, /var/spool/asterisk, /var/log/asterisk, /usr/lib/asterisk, /etc/dahdi and all that stuff.

Re: [asterisk-users] Custom header when busy$

2015-07-02 Thread Duc Giap Van

[asterisk-users] confbridge play tone before speaking

2015-07-02 Thread Jerry Geis
I use both confbridge to bring several devices into a receive only or listen mode, then allow the one person on the phone to speak live over those devices. Works great. However - now I would like to be able to play a tone into the conference before the person speaks. How might that be