similar to: How to send SIP_NOTIFY messages with variable content ?

I would be very interested in using sipsak for something like this. What have you tried so far? -Thufir On Mon, 16 Jan 2017, Olivier wrote: > Thinking over my previous, I wonder if sipsak could be used to send > outgoing SIP NOTIFY messages. > Would both Asterisk and sipsak be able to share networks resources ? > > Thoughts ? > > 2017-01-16 14:10 GMT+01:00 Olivier

I have an AGI that can sometimes take time complete. I don't want the dialplan to be held up by the agi. Is there any way to call it and have Asterisk continue with the dialplan? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190410/4c704231/attachment.html>

In PHP something like: $pid = pcntl_fork(); if ($pid != 0) { // we are the parent // do parent stuff exit; } // we are the child, detatch from terminal $sid = posix_setsid(); if ($sid < 0) { die; } // do child stuff On 04/19/2019 02:00 PM, Mark Wiater wrote: > On 4/19/2019 1:49 PM, Dovid Bender wrote: >> Mark, >> >> I am using PHP agi and when forking

Hi, I'm seeing a trend in which SIP devices such as Yealink SIP phones (with v72 firmware), are dropping support of SNMP in favor of "HTTP eventing" if may call this as such : when configuring the SIP device, you can define a couple of HTTP URL which triggered when some event occur (end of boot, on hook, ...). How do deal with those devices ? Do you still try to monitor them with

Hello all, I've got dialout= and callback= set in my voicemail.conf so that I can have users return calls to folks who have left messages. They really like this feature. But when the callback is over, a normal hangup occurs instead of the caller being put back into voicemail at the next message. Is it possible that the users be returned into the voicemail system where they left off? thanks

Hi, I am facing a problem where for legal obligations (LI) I have to copy/mirror/forward the RTP streams for some selected call to an external address/port and I have not found a way to do it with built-in functionality. Do I miss something? The basic requirements are: * Raw RTP (no transcoding, header and payload as is) * Direction (did it arrive at asterisk or was it sent) * End

Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end. Here are my inbound peer settings: username=<accountnumber> secret=<secret>

Hello, On a Asterisk 11.12.0, I'm studying BLF behaviour with Yealink phones. My ultimate goal is to present Operator the name and number of every incoming call so that he/she can if it's worth to pickup a ringing incoming call. I've discovered notifycid option in sip.conf. When a call comes in, I can see that Asterisk is sending two successive NOTIFY messages while the target is

Hello, I'm trying to use SipSak to check if my Asterisk server is available/running with the following : sipsak -vv -s sip:username at sip.domain.tld -c sip:username at sip.domain.tld --password guessthis --hostname XX.XX.XX.63 The SIP OPTION is received by Asterisk as follow : OPTIONS sip:username at sip.domain.tld SIP/2.0 Via: SIP/2.0/UDP

Good day all What is the file sip_notify.conf for Thanks Altus

Hello list, Does anybody managed to reboot GrandStream phone with sip notify <sip_notify.conf section> <peer> It seems that I need to send a sys-control Event but i suspect that's not enaugh.... my phone just answer me a CSeq: 102 NOTIFY..... Cheers Laurent

Hi, I'm trying to integrate Google cloud speech recognition v2 in it. I can get the audio recorded, have created Service key and API key but whenever I try to access it, I just get 403 access denied. I am at my wits end here. Has anybody tried it ? were you successful ? Could you please guide me how to do it ? I'll be grateful to you if this works ! -- Warm Regds. MathuRahul

Hi, I've tried to use SIPSAK to understand the troubles i'm having about sending my voice to the person I've called (extension), after doing this tests below I always got this error "error: this FQDN or IP is not valid: voicegw". This could cause problems (namely audio problems)? Best regards, Helder voicegw:~# sipsak -C empty -a password -s

try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of registrar>" the trick is to specify the "-O desktop" parameter + the "-H <ip of registrar>" parameter. Sipsak fakes the host-header of the registrar so that the Snom thinks it is coming from your Asterisk server, then lets the message through to the

Hi everybody, I'm seeking information on how to report an IP phone on a call that is occurring on another IP phone. Example: While the A phone is ringing, Asterisk sends a notification to a phone B on the call that is going to A, but this notification is displayed on the B phone display and the user does not need to hit anything to view the information. I'm

I'm using sipsak to send messages to Snoms in my subnet. At work, works fine: sipsak -M -O desktop -B "foo" -s sip:1001@192.168.1.220 -H 192.168.1.46 displays "foo" on the Snom display On my home LAN (AAH 1.5, Snom 190 3.60s, switched 100, no VLAN, no routing) the same command (modified for my LAN) always yields: (type: 3, code: 3): from 192.168.171.8 at the console

On Fri, 20 Feb 2015 15:07:35 -0500, Andres wrote: > This is showing nothing so I don't think your test message even made it > here. I think it looped in the 'doge' server. I was wondering the same thing :) in tleilax, I looked in /var/log/asterisk/messages and see: [Feb 20 15:13:19] VERBOSE[3661] chan_sip.c: [Feb 20 15:13:19] <--- SIP read from UDP:192.168.1.3:38154

On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote: > A "sip set debug on" will give you more info on why you are getting the > 404. It probably has to do something with your context/dialplan. on tleilax: tleilax*CLI> tleilax*CLI> sip set debug on SIP Debugging enabled tleilax*CLI> on doge: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:devries at

I've just started deploying these (well the T28P model) after years of Snom issues and they look pretty good (although the documentation is execrable; if you thought the Snom stuff was obtuse Yealink have got them knocked into a cocked hat!). Anyway, for provisioning I use HTTP with a DHCP entry like:- # # Yealink Phones # group { #

Is there a traceroute equivalent in the VoIP world? I would like to see the route a call takes after it gets to the gateway. Basically showing all the hops until it reaches it's destination or PSTN termination. -Dave