[asterisk-users] Asterisk 11 - CONNECTEDLINE and Asterisk applications
Hello, I recently gave CONNECTEDLINE function a try on an Asterisk 11 setup with a couple of SIP phones. When a SIP phone dials an other one, with a CONNECTEDLINE statement in its dialplan, I noticed that Asterisk update caller's information using a Remote-Party-ID header in 180 Ringing message. For instance: Alice Asterisk ---Bob --- INVITE -- ... -180 Ringing--- (*) (*) Ringing message includes Remote-Party-ID header with which Asterisk can update dialed name or number. Now, I'm trying to let Asterisk update dialed name or number when caller simply dials an Asterisk application such as VoiceMail ou Playback. At the moment, I can't get what I'm looking for. Specifically, I tried with: Set(CONNECTEDLINE(name)=Foo); Ringing(); Playback(tt-monkeys); HangUp(); As you may guess, I want to have Foo displayed on caller's phone screen. I can see that 180 Ringing message sent back by Asterisk following Ringing() statement doesn't hold any Remote-Party-ID header. What would you suggest ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OpenVPN Clients Intermittently Cannot Call In
Hello,
I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal
phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP
phones, which appear to be working correctly. I have a few external phones
(Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an
OpenVPN client configured on them that connects back to the LAN network through
a pfSense gateway with OpenVPN configured on it.
Asterisk server LAN IP: 10.10.32.10
My internal test phone: 146 at 10.10.32.96
My external test phone: 265 at 192.168.32.10
My sip.conf for these external users is as follows:
http://pastebin.com/2b9YE7Dz
The dialplan uses this Dial() invocation when dialing either an internal or
external phone. Note that the max timeout is 12 seconds:
exten = _[12]XX,1,Dial(SIP/${EXTEN},12)
These external phones register correctly, and internal users can call these
external users, the phones ring immediately, and the call is normal. However,
if the external users try to dial an internal phone, I've observed some
different failure modes:
* operating normally: sometimes the call rings immediately, the internal user
answers, and the audio is present immediately
* ringing delay and no connection even after pickup: sometimes there's a
significant delay between when the call starts ringing on the external side
and when it actually starts ringing on the internal user's phone. Consequently,
the internal user only has 1 or 2 rings to answer. Even if they do answer
during this time, the line is dead and it goes to voicemail (the next step in
the dialplan)
* delay before audio is connected after answer: sometimes the internal user
answers, but there's a delay of 3-10 seconds before either party can hear audio
I've enabled rtp and sip debug for this particular external phone
(192.168.32.10) and attached console logs from both types of these failures:
* ringing delay and no connection even after pickup:
http://pastebin.com/fe1khEmF
* delay before audio is connected after answer: http://pastebin.com/uZSMKczk
What else can I try to debug these problems? Since it is intermittent, I am not
always able to reproduce (sometimes the calls work just fine).
Thanks,
Andrew Martin
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Re: [asterisk-users] Asterisk 11 - CONNECTEDLINE and Asterisk applications [SOLVED]
2015-04-30 17:45 GMT+02:00 Richard Mudgett rmudg...@digium.com: On Thu, Apr 30, 2015 at 4:50 AM, Olivier oza.4...@gmail.com wrote: Hello, I recently gave CONNECTEDLINE function a try on an Asterisk 11 setup with a couple of SIP phones. When a SIP phone dials an other one, with a CONNECTEDLINE statement in its dialplan, I noticed that Asterisk update caller's information using a Remote-Party-ID header in 180 Ringing message. For instance: Alice Asterisk ---Bob --- INVITE -- ... -180 Ringing--- (*) (*) Ringing message includes Remote-Party-ID header with which Asterisk can update dialed name or number. Now, I'm trying to let Asterisk update dialed name or number when caller simply dials an Asterisk application such as VoiceMail ou Playback. At the moment, I can't get what I'm looking for. Specifically, I tried with: Set(CONNECTEDLINE(name)=Foo); Ringing(); Playback(tt-monkeys); HangUp(); As you may guess, I want to have Foo displayed on caller's phone screen. I can see that 180 Ringing message sent back by Asterisk following Ringing() statement doesn't hold any Remote-Party-ID header. What would you suggest ? You usually need to set a number to go with a name. Depending upon the channel driver protocol, a name with no number may not be enough to send out on its own. Simply adding a CONNECTEDLINE(num) statement made it ! Thank you very much for helping ! Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
Le 30/04/2015 19:18, Andrew Martin a écrit : Hello, Hello I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP phones, which appear to be working correctly. I have a few external phones (Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an OpenVPN client configured on them that connects back to the LAN network through a pfSense gateway with OpenVPN configured on it. I faced problems with pfsense -no VPN involved- and finally installed siproxd on it. Also set the firewall mode to conservative. [...] -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
- Original Message - From: Administrator TOOTAI ad...@tootai.net To: asterisk-users@lists.digium.com Sent: Thursday, April 30, 2015 4:43:33 PM Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP phones, which appear to be working correctly. I have a few external phones (Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an OpenVPN client configured on them that connects back to the LAN network through a pfSense gateway with OpenVPN configured on it. I faced problems with pfsense -no VPN involved- and finally installed siproxd on it. Also set the firewall mode to conservative. Daniel, Thanks for the information. Do you have an example or documentation on the siproxd configuration that you used? Thanks, Andrew -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 - CONNECTEDLINE and Asterisk applications
On Thu, Apr 30, 2015 at 4:50 AM, Olivier oza.4...@gmail.com wrote: Hello, I recently gave CONNECTEDLINE function a try on an Asterisk 11 setup with a couple of SIP phones. When a SIP phone dials an other one, with a CONNECTEDLINE statement in its dialplan, I noticed that Asterisk update caller's information using a Remote-Party-ID header in 180 Ringing message. For instance: Alice Asterisk ---Bob --- INVITE -- ... -180 Ringing--- (*) (*) Ringing message includes Remote-Party-ID header with which Asterisk can update dialed name or number. Now, I'm trying to let Asterisk update dialed name or number when caller simply dials an Asterisk application such as VoiceMail ou Playback. At the moment, I can't get what I'm looking for. Specifically, I tried with: Set(CONNECTEDLINE(name)=Foo); Ringing(); Playback(tt-monkeys); HangUp(); As you may guess, I want to have Foo displayed on caller's phone screen. I can see that 180 Ringing message sent back by Asterisk following Ringing() statement doesn't hold any Remote-Party-ID header. What would you suggest ? You usually need to set a number to go with a name. Depending upon the channel driver protocol, a name with no number may not be enough to send out on its own. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
