[asterisk-users] Best way to recording the hold time for a Queue agent or an extension
Hi Everyone, We are using Queuemetrics but it doesn't Record the Hold Time as it's never logged on the queue_log file. However, when an agent or an extension presses HOLD button on their phone, asterisk does create an event for Music On Hold which is logged in the /var/log/asterisk/full. I want to record the total hold time for an extension and save it with an epoch time stamp. What is the best approach to this? read and parse /var/log/asterisk/full in a cron job every few seconds? Have a presistent PHP-AGI connection to check for hold time events? As much detail as possible on above approaches or other ideas are most appreciated. Thanks -- _ -- 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] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Hi Everyone, We use the top buttons on Aastra 55i to login and logout from Queues. This is the order: Button 1 = Login to English Queue Button 2 = Login to Spanish Queue Button 3 = Logout of English/Spanish Queues There are indicator LEDs on each of these buttons. Is there anyway we can send a SIP request or some other communication to get the Aastra 6755i phone to keep the LED for login set to ON if agent is logged in and to put it to off when agent logs out? Thanks, Bruce -- _ -- 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] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Amazing. Thank you very much. Unfortunately, the phone type is 53i and not the 55i as I mistakenly noted. It has only 6 buttons on the left side. Is there a workaround for this? Thanks again. -Bruce On Wed, Oct 20, 2010 at 5:12 PM, bakko asannu...@gmail.com wrote: Hello, you can't utilice the same butons to know the state of the agent but you can configure the LEDs in the opposite position (4,5,6) in the dialplan just before the command to login to the queue put this line (for english queue): exten = yourextension,n,Set(DEVSTATE(Custom:agenten)=INUSE) for spanish queue exten = yourextension,n,Set(DEVSTATE(Custom:agentes)=INUSE) in the dialplan part relative to agent logoff (english) exten = yourextension,n,Set(DEVSTATE(Custom:agenten)=NOT_INUSE) spanish exten = yourextension,n,Set(DEVSTATE(Custom:agentes)=NOT_INUSE) then on the Aastra 6755i web page (on the Programmable Keyshttp://192.168.100.100/programkey.html menu): keytypevalueline 4BLFagentenglobal 5BLFagentesglobal Now each time the agent login to english queue the 4 key LED switch to red. The same with key 5 LED Please try and give us a feedback Regards - Bakko -- _ -- 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] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Here is the login for English:
;English-temp LOGIN
exten = 800,1,Answer()
exten = 800,n,AddQueueMember(500|Local/${CALLERID(num)}...@from-internal/n)
exten = 800,n,Set(DEVSTATE(Custom:agenten)=INUSE)
exten = 800,n,Playback(agent-loginok)
exten = 800,n,Hangup()
;English Logout
;All Queues Logout
exten = 802,1,Answer
exten = 802,n,RemoveQueueMember(500|Local/${CALLERID(num)}...@from-internal/n)
exten = 802,n,RemoveQueueMember(499|Local/${CALLERID(num)}...@from-internal/n)
exten = 802,n,Playback(agent-loggedoff)
exten = 802,n,Hangup
The logout logs both English and Spanish (which is just like English for
Login.
Thanks,
Bruce
On Thu, Oct 21, 2010 at 10:01 AM, bakko asannu...@gmail.com wrote:
Hi Bruce,
can you show agent login/logoff diaplan?
Maybe there is a solution but i have to know how yours agents login/logoff.
Regards
- Bakko
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Re: [asterisk-users] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Thanks for the input. By this configuartion you mean by the way I do Add and Remove member from the Queue? Can you please explain by what sort of configuration (what to use instead of Add and Remove queue member) would get this working. I guess I am looking for speedial/BLF on the same key ?!!! Thanks again On Thu, Oct 21, 2010 at 6:36 PM, bakko asannu...@gmail.com wrote: Hi Bruce, with this configuration you can`t control the state of agent. Sorry 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 -- _ -- 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 over TCP 1194 rather than UDP 1194 - Is there an adverse effect when running Asterisk?
Hi Everyone, For some reason a few phones connected to a pfSense box can't make or allow in OpenVPN in port 1194 UDP. So, I established the VPN tunnel on 1194 TCP and it works fine. I would like to know if there is any disadvantages to using TCP over UDP for the tunnel when using Asterisk or is just as reliable and solid as a UDP tunnel? Thanks -- _ -- 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] a2billing muting enter the phone number
If you want to turn off the audio totally you can set audio to NO (it's probably the 4th or 5th in list of Global settings). Otherway is to blank the file responsible to play that file and keeping the settings intact. However, there are numerous options to turn on and off the various announcements which you should look into in the System Settings. -Bruce On Sat, Oct 23, 2010 at 8:31 AM, Baha @ SH i...@saudihome.com wrote: How can I mute the message please enter the number you wish to call and press the # key in a2billing??? I tried use_dnid = YES but still I keep getting the message prompt... thanks -- _ -- 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] Best way to recording the hold time for a Queue agent or an extension
Anything on this guys? I am sure someone had the need to record the HOLD time or maybe it is already being recorded somewhere? Any thoughts are appreciated. Thanks, Bruce On Wed, Oct 20, 2010 at 3:30 AM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, We are using Queuemetrics but it doesn't Record the Hold Time as it's never logged on the queue_log file. However, when an agent or an extension presses HOLD button on their phone, asterisk does create an event for Music On Hold which is logged in the /var/log/asterisk/full. I want to record the total hold time for an extension and save it with an epoch time stamp. What is the best approach to this? read and parse /var/log/asterisk/full in a cron job every few seconds? Have a presistent PHP-AGI connection to check for hold time events? As much detail as possible on above approaches or other ideas are most appreciated. Thanks -- _ -- 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] Best way to recording the hold time for a Queue agent or an extension
Thanks for the feedback. I don't need the Queue times but rather putting ON HOLD times. If you press the HOLD button on your SIP phone, Asterisk records the event Music On HOLD Playing and that is recorded in /var/log/asterisk/full. I want to harvest the ON HOLD time per phone SET. Thanks On Mon, Oct 25, 2010 at 4:51 AM, Antonio Berrios anto...@sheffieldcitytaxis.com wrote: I would probably do this through the AMI, it should spew out the info you require. Timestamp when they entered the queue and timestamp when they get answered. On 10/25/2010 05:01 AM, Bruce B wrote: Anything on this guys? I am sure someone had the need to record the HOLD time or maybe it is already being recorded somewhere? Any thoughts are appreciated. Thanks, Bruce On Wed, Oct 20, 2010 at 3:30 AM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, We are using Queuemetrics but it doesn't Record the Hold Time as it's never logged on the queue_log file. However, when an agent or an extension presses HOLD button on their phone, asterisk does create an event for Music On Hold which is logged in the /var/log/asterisk/full. I want to record the total hold time for an extension and save it with an epoch time stamp. What is the best approach to this? read and parse /var/log/asterisk/full in a cron job every few seconds? Have a presistent PHP-AGI connection to check for hold time events? As much detail as possible on above approaches or other ideas are most appreciated. Thanks ---DISCLAIMER The information contained in this message is private and confidential and intended only for the recipient named above. If you are not the intended recipient you are notified that any communication, circulation or copying of the information contained in this message is strictly prohibited. If you have received this message in error please notify us immediately by telephone in order that we are made aware of this fact and the message can be returned to us at our address as indicated above. Activity and use of the Sheffield City Taxis e-mail service is monitored to secure its effective operation and for other lawful business purposes. Sheffield City Taxis Ltd. Registered Office: 912 City Road, Sheffield, S2 1GQ. Registered in England no: 4674148. Sheffield City Taxis Limited uses regularly updated anti-virus software in an attempt to reduce the possibility of infection. However we do not guarantee that any attachments to this e-mail are virus free. -- _ -- 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] Best way to recording the hold time for a Queue agent or an extension
Thanks for the input. Would I have to process each call through a specific dial-plan for the AMI to be in charge of each call so that it can see the Unique ID of the channel and the Hold event? Because that seems like a lot of work. If AMI (I have no experience with it) allows me to open a socket and just read whatever comes through then it might work for me. Thanks On Mon, Oct 25, 2010 at 10:47 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* Monday, October 25, 2010 9:32 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Best way to recording the hold time for a Queue agent or an extension Thanks for the feedback. I don't need the Queue times but rather putting ON HOLD times. If you press the HOLD button on your SIP phone, Asterisk records the event Music On HOLD Playing and that is recorded in /var/log/asterisk/full. I want to harvest the ON HOLD time per phone SET. Thanks Again, the AMI would be your most likely help here. The information in /v/l/a/full is going to give you a start and stop time for MOH, but not in a format where you can easily tie it back to an extension. You will have a “start moh” and “stop moh” event in the AMI that is tied to an extension by the uniqueid. In “PERL Weenie” world, the way to process this is to pipe the AMI output where it is an input file keyed by the uniqueid. You can find some decent examples on voip-info.org. -- _ -- 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
[asterisk-users] Pop-up for MS Outlook 2007 recommended
Hi Everyone, Which paid or unpaid commercial plugin is available out there for Asterisk that would do Outlook contacts pop-up that is proven to work great with MS Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well through the Outlook. Thanks, Bruce -- _ -- 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] Pop-up for MS Outlook 2007 recommended
Great suggestion but unfortunately for this client a proven technology is needed and we don't mind paying a bit for it so once the time is available we might do this the way you suggested. Thanks On Mon, Oct 25, 2010 at 2:20 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* Monday, October 25, 2010 1:14 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Pop-up for MS Outlook 2007 recommended Hi Everyone, Which paid or unpaid commercial plugin is available out there for Asterisk that would do Outlook contacts pop-up that is proven to work great with MS Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well through the Outlook. Thanks, Bruce Not specifically what you are looking for, but it is very simple to use Apache/Ajax to make AMI links to launch calls from anywhere. I would invest 30-240 minutes into this method before bothering with the other stuff that is out there. Also, will make it easier when you eventually jump to 1.8/1.10. -- _ -- 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
[asterisk-users] Updating asteriskcdrdb with uniqueid field from Master.csv, Master.csv.1....Master.csv.5 - Must I bring all files together first?
Hi Everyone, Just noted that PBXinaFLASH failed me again on yet something else. The uniqueid field didn't update on the asteriskcdrdb in the past few months but it is available in the .csv files in /var/log/asterisk/cdr-csv/*.csv I have already re-did the asterisk-addons install with correct parameters to include future calls uniqueid in the table (I have no clue why these flavours of Asterisk chose to remove neccessary parameters from time to time) Anyhow, the uniqueid field is needed to be there for ARI to work and for recordings to be pulled. So, I am checking the folder for .csv files and I see Master.csv.1 Master.csv.2 Master.csv.3..Master.csv.5. My Questions: 1- Must I bring those files into one big file first before trying to update the MySQL table? 2- Must I remove all data from asteriskcdrdb first to avoid duplication and then do the update through a php script? Thanks -- _ -- 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] Is queue Members priority supposed to show in the queue show command
Hi Everyone,
I am doing a queue show and I can't see any column that shows a queue member
priority. Is there any other command that can show the member priority on
the Asterisk 1.4x CLI?
We are using this format of dialplan to login agents:
exten = 123,Answer()
exten = 123,n,AddQueueMember(500|Local/${CALLERID(num)}...@from-internal/n)
exten = 123,Hangup()
Where 500 is the English queue number and extension 123 is dial to
login into Q-500
The priorities don't seem to work when we use the LEASTRECENT strategy so I
am suspecting my dialplan above or maybe even the priorities are not
supposed to work with LEASTRECENT strategy at all?
Thanks
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Re: [asterisk-users] Is queue Members priority supposed to show in the queue show command
Thanks Warren. That should do.
Regards,
Bruce
On Thu, Nov 4, 2010 at 2:54 PM, Warren Selby wcse...@selbytech.com wrote:
On Thu, Nov 4, 2010 at 12:56 PM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
I am doing a queue show and I can't see any column that shows a queue
member priority. Is there any other command that can show the member
priority on the Asterisk 1.4x CLI?
We are using this format of dialplan to login agents:
exten = 123,Answer()
exten = 123,n,AddQueueMember(500|Local/${CALLERID(num)}...@from-internal/n)
exten = 123,Hangup()
You mean the penalty?
~]# asterisk -rx core show application AddQueueMember
-= Info about application 'AddQueueMember' =-
[Synopsis]
Dynamically adds queue members
[Description]
AddQueueMember(queuename[|interface[|penalty[|options[|membername[|state_interface]):
Dynamically adds interface to an existing queue.
If the interface is already in the queue and there exists an n+101 priority
then it will then jump to this priority. Otherwise it will return an error
The option string may contain zero or more of the following characters:
'j' -- jump to +101 priority when appropriate.
This application sets the following channel variable upon completion:
AQMSTATUSThe status of the attempt to add a queue member as a
text string, one of
ADDED | MEMBERALREADY | NOSUCHQUEUE
If a device is provided in the state_interface parameter, then this will
be the device which will be used to determine the device state of the
added queue member.
Example: AddQueueMember(techsupport|SIP/3000)
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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[asterisk-users] Short rings for extensions when part of the Queue
Hi Everyone, We have three different Queues set to leastrecent strategy and from time to time I hear someone complain that they receive short rings (partial ring cycle) and since it's not their turn even if they pickup the phone the call is not given to them since the Queue is actually hitting someone else at the same time. Is this short ring an indication of some sort for leastrecent strategy. Like maybe you are next in turn or something? It's annoying and odd that I don't see a pattern to it. If it's a feature please let me know. Using Asterisk 1.4x Thanks -- _ -- 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] Short rings for extensions when part of the Queue
Chad, You are absolutely right on this one. I had setup the Queue time out for agent set to 15 seconds and retry to 2 seconds. So, I think during those two seconds Asterisk for some crazy reason hits another extension and then comes back to the same extension to ring again. So, I have setup the agents to ring for ever for this call center since their agents always have to present or logout if not present. I will see the behavior tomorrow as they test it. My issue might be solved but for those call centers where you want the Queue to move onto the next agent or if you don't want to ring for ever and take a Retry break then it will still remain an issue. I will report back if setting to ring Unlimited doesn't work. Warren, The CLI shows the regular stuff. Nothing out of the ordinary but that it move on to the next agent because the first agent has timed-out for two seconds. Regards, Bruce On Thu, Nov 4, 2010 at 9:29 PM, Chad Wallace cwall...@lodgingcompany.comwrote: On Thu, 4 Nov 2010 20:12:54 -0400 Bruce B bruceb...@gmail.com wrote: Hi Everyone, We have three different Queues set to leastrecent strategy and from time to time I hear someone complain that they receive short rings (partial ring cycle) and since it's not their turn even if they pickup the phone the call is not given to them since the Queue is actually hitting someone else at the same time. Is this short ring an indication of some sort for leastrecent strategy. Like maybe you are next in turn or something? It's annoying and odd that I don't see a pattern to it. If it's a feature please let me know. Using Asterisk 1.4x We have the same issue, and I think I've figured out what's causing it, by watching the SIP debug output. What I noticed is that when someone misses a call on the queue, it switches over to the next person, but almost immediately (after a short ring) it breaks off and cancels that call because of a timeout. I think it's a matter of timing between the different timeouts you can set in queues.conf. I haven't had time to tweak it yet, so that's all I can tell you. I don't know how hard it would be to get the timeouts to jive perfectly, but I'm sure it can be done. Mind you, this is based on only one observation, so I may be wrong... It's only a minor annoyance, and I have bigger fish to fry right now. Speaking of which, one of them is almost burnt, so I have to go. ;-) I'd be interested to hear your findings... -- _ -- 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] Short rings for extensions when part of the Queue
Yeah, I think I had it set to 2 seconds and that creates that short ring on another extension. Thanks, On Fri, Nov 5, 2010 at 9:47 AM, Mark Deneen mden...@gmail.com wrote: On Fri, Nov 5, 2010 at 1:18 AM, Bruce B bruceb...@gmail.com wrote: Chad, You are absolutely right on this one. I had setup the Queue time out for agent set to 15 seconds and retry to 2 seconds. So, I think during those two seconds Asterisk for some crazy reason hits another extension and then comes back to the same extension to ring again. So, I have setup the agents to ring for ever for this call center since their agents always have to present or logout if not present. I will see the behavior tomorrow as they test it. My issue might be solved but for those call centers where you want the Queue to move onto the next agent or if you don't want to ring for ever and take a Retry break then it will still remain an issue. I will report back if setting to ring Unlimited doesn't work. Warren, The CLI shows the regular stuff. Nothing out of the ordinary but that it move on to the next agent because the first agent has timed-out for two seconds. Regards, Bruce Have you considered setting the queue timeout to 14 or 16 seconds and retry to 2 seconds? This way the timeout and the retry should line up better. -- _ -- 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] Short rings for extensions when part of the Queue
Sorry, I am not following. If an extension rings for 15 or 16 seconds and then waits for 2 or three seconds what difference does the being divisible make? Is there something internal to Asterisk that makes the Retry time dependent on Time Out (also known as Ring Time)? P.S. I think the 15 seconds is just three rings complete. Thanks, Bruce On Fri, Nov 5, 2010 at 11:31 AM, Mark Deneen mden...@gmail.com wrote: On Fri, Nov 5, 2010 at 10:38 AM, Bruce B bruceb...@gmail.com wrote: Yeah, I think I had it set to 2 seconds and that creates that short ring on another extension. Thanks, The point was that 14 and 16 are divisible by 2 (evenly) while 15 is not. -- _ -- 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
[asterisk-users] Polycom WEB UI configuration - What needs to be put in for basic SIP registration?
Hi Everyone, Configuring a Polycom conference bridge IP 5000 to connect to Asterisk. For some reason I don't see any SIP packets coming in to Asterisk at all. I don't want to use XML or ftp etc for now and just use the Web Interface to get it going with basic features. But the Web UI is a bit confusing with SIP and Line tabs. I have put this on the web interface: SIP Outbound Proxy: Address = 192.168.0.2 Port = 5060 Server 1: Address = 192.168.0.2 Port = 5060 Transport = DNSnaptr Expires = 300 Register = 1 Line: Display Name = 100 Address = 192.168.0.2 Authentication User ID = 100 Authentication Password = * Label = 100 Server 1: Address = 192.168.0.2 Port = 5060 Transport = DNSnaptr Expires = 300 Register = 1 I don't see any registration attempts but Snom phones on the same network can register to Asterisk just fine. Thanks -- _ -- 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] Alternative to Proxmox
Hi Everyone, Is there other comparable products to Proxmox to be used for Asterisk instances? Ease of use, web interface, and Asterisk/CentOS support would be ideal. Thanks -- _ -- 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] Alternative to Proxmox
Thanks. OpenNode seems promising and neat. Proxmox is disappointing when it comes to their forums and documentation. Only few videos listedanyhow. OpenNode is promising but doesn't have a Web UI yet. Is there anything else as well? Thanks On Fri, Nov 5, 2010 at 4:06 PM, Tim Nelson tnel...@rockbochs.com wrote: - Tim Nelson tnel...@rockbochs.com wrote: Hi Everyone, Is there other comparable products to Proxmox to be used for Asterisk instances? Ease of use, web interface, and Asterisk/CentOS support would be ideal. There is OpenNode: http://opennode.activesys.org/ I've heard good things thus far but have not had time nor need to test it myself. Oh, and I meant to ask why you're looking for an alternative to Proxmox. Have you had problems with it that cannot be solved? --Tim -- _ -- 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
[asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Hi Everyone, Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Thanks -- _ -- 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] Any good guides for installing Asterisk on Embedded systems like Alix boards?
John, AstLinux seems promising. Have you used this flavor in production environment? Paul, So, don't use the Yum repositoy?! And, are you sure that is the only thing needs to be done. I am thinking there is more tweaking need to be done. I am not looking to just install Asterisk but it should be production ready as well. Meaning solid, reliable machine. Thanks On Sun, Nov 7, 2010 at 12:28 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sun, Nov 7, 2010 at 11:23 AM, Bruce B bruceb...@gmail.com wrote: Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Only compile the modules you need. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Thanks for the input. I am looking to use it as a DHCP server as well. And I also I want it as a VPN server so that I can securely log in to it from time to time to monitor it's state. The Alix board with pfSense can nicely do VPN and DHCP (no Asterisk). Wondering if those two service would play nice along with Asterisk. Thanks, On Mon, Nov 8, 2010 at 3:04 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Sun, Nov 07, 2010 at 01:10:01PM -0500, Paul Belanger wrote: Most desktop distros are just too bloated for an embedded solution. I use Debian on an Alix system as my home router. It runs Asterisk as well. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Thanks. I think I would still need a firewall. Maybe a 1u rack double enclosure for two Alix boards - one as firewall - and one as PBX would do better. Anyhow, I don't want to open the box if I don't have to. Is there any way I can push the .gz file over console cable rather than putting the CF in a reader? Thanks On Mon, Nov 8, 2010 at 1:06 PM, Darrick Hartman (lists) dhart...@djhsolutions.com wrote: Bruce, AstLinux supports dhcp and dns as well as several vpn options including openvpn. You can download a live ISO image to test. http://www.astlinux.org Darrick On 11/08/2010 08:34 AM, Bruce B wrote: Thanks for the input. I am looking to use it as a DHCP server as well. And I also I want it as a VPN server so that I can securely log in to it from time to time to monitor it's state. The Alix board with pfSense can nicely do VPN and DHCP (no Asterisk). Wondering if those two service would play nice along with Asterisk. Thanks, On Mon, Nov 8, 2010 at 3:04 AM, Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com wrote: On Sun, Nov 07, 2010 at 01:10:01PM -0500, Paul Belanger wrote: Most desktop distros are just too bloated for an embedded solution. I use Debian on an Alix system as my home router. It runs Asterisk as well. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- 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] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Yes, it is a small office. I am familiar with pfSense. I am not sure if firewall on Astlinux is as versatile and flexible. But also, I am wondering if with all those attacks around now-a-days if the box will be able to handle 5 extensions, voicemail, IVR, firewall, DHCP, openvpn all together. Thanks On Mon, Nov 8, 2010 at 7:24 PM, John Novack jnov...@stromberg-carlson.orgwrote: Bruce B wrote: Thanks. I think I would still need a firewall. Maybe a 1u rack double enclosure for two Alix boards - one as firewall - and one as PBX would do better. Anyhow, I don't want to open the box if I don't have to. Is there any way I can push the .gz file over console cable rather than putting the CF in a reader? DO you mean once you have built the system? AstLinux has an upgrade facility built into the system, with the ability to revert to the previous version, all built into the web interface. To initially build the system, it seems to me you would need to put the first OS on the CF card to get the board alive. And if you have an Alix with 2 Ethernet ports, why a second one as a firewall? AstLinux has a built in firewall You did say a SMALL office, didn't you? John Novack Thanks On Mon, Nov 8, 2010 at 1:06 PM, Darrick Hartman (lists) dhart...@djhsolutions.com wrote: Bruce, AstLinux supports dhcp and dns as well as several vpn options including openvpn. You can download a live ISO image to test. http://www.astlinux.org Darrick On 11/08/2010 08:34 AM, Bruce B wrote: Thanks for the input. I am looking to use it as a DHCP server as well. And I also I want it as a VPN server so that I can securely log in to it from time to time to monitor it's state. The Alix board with pfSense can nicely do VPN and DHCP (no Asterisk). Wondering if those two service would play nice along with Asterisk. Thanks, On Mon, Nov 8, 2010 at 3:04 AM, Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com wrote: On Sun, Nov 07, 2010 at 01:10:01PM -0500, Paul Belanger wrote: Most desktop distros are just too bloated for an embedded solution. I use Debian on an Alix system as my home router. It runs Asterisk as well. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] Is this a DDoS to reach Asterisk?
Hi Everyone, I have pfSense running which supplies Asterisk with DHCP. I had some testing ports opened for a web server which I have totally closed now but when I chose option 10 (filter log) on pfSense I get all of this type of traffic (note that it was only 1 single IP and once I blocked that one it was like opening a can full of bees with all different IPs): tcpdump: WARNING: pflog0: no IPv4 address assigned tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on pflog0, link-type PFLOG (OpenBSD pflog file), capture size 96 bytes 00 rule 70/0(match): block in on vr1: 221.132.34.165.33556 69.90.78.53.52229: tcp 20 [bad hdr length 0 - too short, 20] 6. 239658 rule 70/0(match): block in on vr1: 121.207.254.227.6667 69.90.78.38.3072: tcp 24 [bad hdr length 0 - too short, 20] 7. 986724 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 867707 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 799337 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 931814 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 574556 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 956066 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 598334 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 20 [bad hdr length 8 - too short, 20] 072759 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 109451 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 731065 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 16 [bad hdr length 12 - too short, 20] 159413 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 374293 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 16 [bad hdr length 12 - too short, 20] 10. 234202 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 2. 985558 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 13. 236084 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 2. 982122 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 18. 493312 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 2. 477084 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 9. 92 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: tcp 16 [bad hdr length 12 - too short, 20] 1. 216002 rule 70/0(match): block in on vr1: 172.168.0.4.1568 69.90.78.49.445: [|tcp] 321600 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: tcp 20 [bad hdr length 8 - too short, 20] 1. 383839 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: [|tcp] 1. 466115 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: [|tcp] 7. 977140 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 2. 920013 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 29. 032839 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 2. 996906 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 62. 079279 rule 70/0(match): block in on vr1: 82.165.131.28.6005 69.90.78.47.1024: [|tcp] 34. 224871 rule 67/0(match): block in on vr1: 77.34.234.241.1899 69.90.78.43.445: [|tcp] 3. 006367 rule 67/0(match): block in on vr1: 77.34.234.241.1899 69.90.78.43.445: [|tcp] 20. 274886 rule 67/0(match): block in on vr1: 66.211.120.62.1132 69.90.78.55.445: [|tcp] 2. 893859 rule 67/0(match): block in on vr1: 66.211.120.62.1132 69.90.78.55.445: [|tcp] 28. 739620 rule 67/0(match): block in on vr1: 117.197.247.151.1042 69.90.78.55.445: [|tcp] 2. 936286 rule 67/0(match): block in on vr1: 117.197.247.151.1042 69.90.78.55.445: [|tcp] 1. 207250 rule 67/0(match): block in on vr1: 118.171.176.188.42965 69.90.78.43.445: [|tcp] 3. 015370 rule 67/0(match): block in on vr1: 118.171.176.188.42965 69.90.78.43.445: [|tcp] 7. 088359 rule 67/0(match): block in on vr1: 61.130.103.10 69.90.78.42: [|icmp] 11. 825521 rule 67/0(match): block in on vr1: 71.100.221.211.4521 69.90.78.33.445: [|tcp] 2. 316564 rule 67/0(match): block in on
Re: [asterisk-users] Is this a DDoS to reach Asterisk?
And that's the problem. There is no such service running or such port is not open. They only keep trying this for no reason. It might cost us bandwidth for no reason. In fact there is no open ports on our network whatsoever. Thanks On Mon, Nov 8, 2010 at 9:50 PM, Lyle Giese l...@lcrcomputer.net wrote: Bruce B wrote: Hi Everyone, I have pfSense running which supplies Asterisk with DHCP. I had some testing ports opened for a web server which I have totally closed now but when I chose option 10 (filter log) on pfSense I get all of this type of traffic (note that it was only 1 single IP and once I blocked that one it was like opening a can full of bees with all different IPs): tcpdump: WARNING: pflog0: no IPv4 address assigned tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on pflog0, link-type PFLOG (OpenBSD pflog file), capture size 96 bytes 00 rule 70/0(match): block in on vr1: 221.132.34.165.33556 69.90.78.53.52229: tcp 20 [bad hdr length 0 - too short, 20] 6. 239658 rule 70/0(match): block in on vr1: 121.207.254.227.6667 69.90.78.38.3072: tcp 24 [bad hdr length 0 - too short, 20] 7. 986724 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 867707 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 799337 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 931814 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 574556 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 956066 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 598334 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 20 [bad hdr length 8 - too short, 20] 072759 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 109451 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 731065 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 16 [bad hdr length 12 - too short, 20] 159413 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 374293 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 16 [bad hdr length 12 - too short, 20] 10. 234202 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 2. 985558 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 13. 236084 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 2. 982122 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 18. 493312 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 2. 477084 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 9. 92 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: tcp 16 [bad hdr length 12 - too short, 20] 1. 216002 rule 70/0(match): block in on vr1: 172.168.0.4.1568 69.90.78.49.445: [|tcp] 321600 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: tcp 20 [bad hdr length 8 - too short, 20] 1. 383839 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: [|tcp] 1. 466115 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: [|tcp] 7. 977140 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 2. 920013 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 29. 032839 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 2. 996906 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 62. 079279 rule 70/0(match): block in on vr1: 82.165.131.28.6005 69.90.78.47.1024: [|tcp] 34. 224871 rule 67/0(match): block in on vr1: 77.34.234.241.1899 69.90.78.43.445: [|tcp] 3. 006367 rule 67/0(match): block in on vr1: 77.34.234.241.1899 69.90.78.43.445: [|tcp] 20. 274886 rule 67/0(match): block in on vr1: 66.211.120.62.1132 69.90.78.55.445: [|tcp] 2. 893859 rule 67/0(match): block in on vr1: 66.211.120.62.1132 69.90.78.55.445: [|tcp] 28. 739620 rule 67/0(match): block in on vr1: 117.197.247.151.1042 69.90.78.55.445: [|tcp] 2. 936286 rule 67/0(match): block in on vr1: 117.197.247.151.1042
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Thanks for input. Great info. Good to know all this about the router. I see you use a 256MB CF card there. Do you use a USB key stick for storage? Thanks, Bruce On Tue, Nov 9, 2010 at 4:09 AM, Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net wrote: On Mon, 8 Nov 2010, Bruce B wrote: Yes, it is a small office. I am familiar with pfSense. I am not sure if firewall on Astlinux is as versatile and flexible. But also, I am wondering if with all those attacks around now-a-days if the box will be able to handle 5 extensions, voicemail, IVR, firewall, DHCP, openvpn all together. I've benchmarked an Alix board with a 500MHz processor to 80 concurrent calls handling media. They're the mainstay of my small office VoIp only PBX range right now where I limit them to 60 extensions. (the real limitation on number of calls is their broadband bandwidth). Storing voicemail and call recording won't be an issue for you - but do get a fast CF card. http://unicorn.drogon.net/cutie.jpg What you need to do is learn Linux networking and iptables - then you won't need pfsense, etc. Install a good text-only distribution and you're done. e.g. Debian Lenny in text-only mode. To get he best from the hardware then you'll need a custom kernel, but that's no big deal. However the thing that will kill it is multiple VPN terminations - unless you can persuade the system to use the on-board AES crypto engine, but I regularly use ssh into my systems without any detriment, so you could use OpenVPN, etc. I am considering making my boxes into a router and handle PPPoE too, then they can do proper traffic shaping, etc. They're more than capable. Gordon -- _ -- 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
[asterisk-users] eSXI and Asterisk?
Hi Everyone, I don't have much experience with eSXI. I can really use some advise on how to run it without any trouble with Asterisk on CentOS VMs. First of all, is it a good option to run multiple hosted Asterisk instances on a VMware eSXI? or would you rather prefer something like Xen, proxmox, opennode, etc? (All SIP trunking, no PRI or Analogue) If there are limitations such as timing, reliability, I/O access, bad voice quality due to sharing resources please let me know. If you have experience working with it in production with Asterisk please let me know what type of fine tuning you do to get this running. Thanks -- _ -- 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] Using AMI to harvest / record HOLD time - Using FreePBX
Hi Everyone, I am looking into AMI (using PHP) to record every instance of HOLD that is generated by putting a caller on HOLD (press hold button on the phone set). There is no HOLD in Asterisk but the event Music on Hold is generated when HOLD is pressed. The complexity is that all of the the calls are handled by FreePBX so I don't have the channel IDs etc... Can someone please point out how I can have an AMI session connected at all times (if that is wise) to harvest these Music on Hold events and to record the duration of the HOLD? I would be able to place it in the asteriskcdrdb then for reporting purposes. Thanks, -- _ -- 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] Why doesn't Asterisk project document certain important features of Asterisk officially?
Hi Everyone, I am wondering why documentation of some of the vital parts of Asterisk is hosted on voipinfo.org (unreliable is some parts) and not on asterisk.org? For example the list of AMI events are not well documented and one has to guess which version supports which event. The documentation file for AMI for Asterisk 1.4 is really only a startup guide and it doesn't even provide a full list available events. I am wondering if this is a labor tedious job for the programmers or is it a tactical move in part of a party with some sort of interest? Because the programmer(s) who does the programming for something like the AMI events may as well do something known as Copy Paste (very easy process of pressing keys: Ctrl+C + Ctrl+V) of what s/he has changed in the source code and make it available in a text document without much fancy editing even. This is not to bash the Asterisk project or Digium. Don't respond if you have a difference of opinion as I am not looking for personal opinions but rather JUST WONDERING THE TECHNICAL/TACTICAL CAUSE of not documenting thing that are really hidden from the community and one has to go through a much manual process to find out about. Thanks for the input. -Bruce -- _ -- 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] Why doesn't Asterisk project document certain important features of Asterisk officially?
To be honest this is the first time I see this wiki mentioned. It doesn't even come up in talks on this list. The wiki should be advertised often and there should be some sort of active monitoring and supervision of the contents as well as some serious ongoing official contributions. All this well help get it running otherwise voipinfo.org still stays as the main (not so trusted and partially documented) site to be. Just checking for AMI events for Asterisk 1.8 on the wiki and there is no mention of it. That is an example of what I think Digium can fill in rather than outsider contributers. Thanks for the inputs guys. On Wed, Nov 24, 2010 at 9:19 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Wed, Nov 24, 2010 at 8:06 PM, Paul Belanger pabelan...@digium.com wrote: On 10-11-24 08:34 PM, Sherwood McGowan wrote: True, but then some of us registered on that site and still don't have the ability to edit...I thought it was a community effort? Maybe I was wrong Once registered you will be able to post comments, not edit. If you would like to become part of the documentation process, I would recommend talking to some on #asterisk-dev. It is my understanding there will be workflows around the wiki content and a reviews process for new submissions. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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 Hey thanks for the reply mate! I'll pop by the -dev channel soon and see about getting in on the project. INSERTDEITYNAMEHERE knows I've been in the community long enough that I should warrant at least a probational add to the process... Slainte, Sherwood McGowan That guy who bugged Murf like CRAZY about the Macro iteration bug, and several AEL items ;-) -- _ -- 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
[asterisk-users] How to quickly move on to Dahdi channels when SIP provider fails?
Hi Everyone, There are situations when internet connection is lost, SIP provider fails, or even authentication to SIP provider fails, and we want to use the backup Dahdi channels (PSTN). As simple as it may sound but with the many different situations and error messages it seems like it's not so easy to predict all the errors. Is there any single parameter value that can be changed to send the call to Dahdi instead of SIP in-case for any reason the SIP provider fails? Because I see a server with this need waits for long time for DNS to resolve when there is no internet even. Thanks -- _ -- 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] How to quickly move on to Dahdi channels when SIP provider fails?
Thanks for the input guys. I really appreciate all the input and I am sure they work but I thought there would be a much better way to do this. Sounds like patching things to me. Why doesn't Asterisk take advantage of the qualify values to make sure if the SIP connection is up or not? Shouldn't this become a native feature of the PBX rather than trying to do work-around? Thanks On Wed, Dec 8, 2010 at 12:20 PM, klitz...@pool.informatik.rwth-aachen.dewrote: Hi! There are situations when internet connection is lost, SIP provider fails, or even authentication to SIP provider fails, and we want to use the backup Dahdi channels (PSTN). As simple as it may sound but with the manydifferentsituations and error messages it seems like it's not so easy to predict all the errors. Is there any single parameter value that can be changed to send the call to Dahdi instead of SIP There is nothing available out-of-the-box. You need to include your own IP SIP tests in the dialplan before dialing out to a SIP channel. Useful for this purpose are - ping and host or wget, - GROUP() and GROUP_COUNT(), - SIPPEER(xxx:status), - CHANISAVAIL(), - dial timeouts and - post-dial error handling (see DIALSTATUS and HANGUPCAUSE as well as Asterisk 1.8 with its ability to act directly upon the SIP response code). Philipp -- _ -- 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
[asterisk-users] Why does sip show peers show my router/gateway address as the client IP address?
Hi Everyone, I am using pfSense to do firewall and NAT on an Asterisk server. I have ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local IP 192.168.5.5. However, when a user from outside using Linksys WRP400 ata connects to the Asterisk server and registers I see them as 192.168.1.1 in the sip show peers command. In face, all many different of the Linksys WRP400 show the same. It seems that pfsense does something to the packets that when they reach Asterisk it thinks they are sent from the Gateway rather than the actual endpoint hence the calls are not reaching the other side but registration is made. Any experience with this? Thanks -- _ -- 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] Why does sip show peers show my router/gateway address as the client IP address?
Thanks for the feedback Ryan. Siproxd is not installed. I think Siproxd like you said just does the reverse meaning if phones are part of pfSense subnet then it connects to outside world. But in my case they are coming into Asterisk which is on pfSense subnet. I do have a static IP and it's set like: externip=34.34.34.34 localnet=192.168.5.0/255.255.255.0 Do you use pfSense for this same situation? Can you do a sip show peers and let me know if you actually see the outside public IP addresses for the clients? Also how is your outbound NAT setup? AON? Thanks On Sat, Dec 11, 2010 at 10:15 AM, Ryan Wagoner rswago...@gmail.com wrote: On Sat, Dec 11, 2010 at 3:06 AM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am using pfSense to do firewall and NAT on an Asterisk server. I have ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local IP 192.168.5.5. However, when a user from outside using Linksys WRP400 ata connects to the Asterisk server and registers I see them as 192.168.1.1 in the sip show peers command. In face, all many different of the Linksys WRP400 show the same. It seems that pfsense does something to the packets that when they reach Asterisk it thinks they are sent from the Gateway rather than the actual endpoint hence the calls are not reaching the other side but registration is made. Any experience with this? Thanks Do you have the siproxd package installed on pfsense? It is suspossed to handle registrations from multiple phones behind NAT. In your case since the phones are external I would probably remove it if installed. I haven't needed siproxd. Also on Asterisk set externip to your static IP in sip.conf. Or if you don't have a static IP set externhost. You also need to configure localnet. Ryan -- _ -- 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] Why does sip show peers show myrouter/gateway address as the client IP address?
Hi Wang, Did you mean to write a feedback? You sent an empty message. Regards, On Sat, Dec 11, 2010 at 11:56 AM, w...@pythian.com wrote: Sent from my “contract free” BlackBerry® smartphone on the WIND network. -Original Message- From: Bruce B bruceb...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Sat, 11 Dec 2010 11:45:15 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Why does sip show peers show my router/gateway address as the client IP address? -- _ -- 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 -- _ -- 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] Why does sip show peers show my router/gateway address as the client IP address?
Hi Again, Here is what I see which is wrong for AddrIP and is fine for Reg. Contact parameter - In fact both parameters should show the public IP address: ** DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-IP : 192.168.0.1 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: SIP Options : (none) Codecs : 0xe (gsm|ulaw|alaw) Codec Order : (ulaw:20,alaw:20,gsm:20) Auto-Framing : No 100 on REG : No Status : OK (14 ms) Useragent: Linksys/WRP400-1.01.00 Reg. Contact : sip:5...@45.45.45.45:5060 Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs ** Regards, Bruce On Sat, Dec 11, 2010 at 10:15 AM, Ryan Wagoner rswago...@gmail.com wrote: On Sat, Dec 11, 2010 at 3:06 AM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am using pfSense to do firewall and NAT on an Asterisk server. I have ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local IP 192.168.5.5. However, when a user from outside using Linksys WRP400 ata connects to the Asterisk server and registers I see them as 192.168.1.1 in the sip show peers command. In face, all many different of the Linksys WRP400 show the same. It seems that pfsense does something to the packets that when they reach Asterisk it thinks they are sent from the Gateway rather than the actual endpoint hence the calls are not reaching the other side but registration is made. Any experience with this? Thanks Do you have the siproxd package installed on pfsense? It is suspossed to handle registrations from multiple phones behind NAT. In your case since the phones are external I would probably remove it if installed. I haven't needed siproxd. Also on Asterisk set externip to your static IP in sip.conf. Or if you don't have a static IP set externhost. You also need to configure localnet. Ryan -- _ -- 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] Why does sip show peers show my router/gateway address as the client IP address?
Thanks for the confirmation. Do you have both LAN and WAN as outbound AON like this: WAN any * * * * * YES LAN any * * * * * YES ??? I am stumped as to why pfSense behaves like this in this instance. Thanks again. On Sat, Dec 11, 2010 at 12:34 PM, Ryan Wagoner rswago...@gmail.com wrote: On Sat, Dec 11, 2010 at 11:45 AM, Bruce B bruceb...@gmail.com wrote: Thanks for the feedback Ryan. Siproxd is not installed. I think Siproxd like you said just does the reverse meaning if phones are part of pfSense subnet then it connects to outside world. But in my case they are coming into Asterisk which is on pfSense subnet. I do have a static IP and it's set like: externip=34.34.34.34 localnet=192.168.5.0/255.255.255.0 Do you use pfSense for this same situation? Can you do a sip show peers and let me know if you actually see the outside public IP addresses for the clients? Also how is your outbound NAT setup? AON? Thanks Yep I am using pfSense 1.2.3 with a static IP. I have port forwarded UDP SIP and the UDP RTP port range to the private IP of the Asterisk box. I have enabled manual outbound nat and configured the static port option. If you use the automatic outbound nat it will randomize the ports, which you don't want. My sip.conf looks like yours with the externip and localnet set. When I do sip show peers I see the external IP. Ryan -- _ -- 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
[asterisk-users] What to check for when there are sound interference using SIP channels only? standard debug methods?
Hi Everyone, I ocassionally hear echo, static, and garbled voice when calling extension to extension between two office (different geographic locations connected using OpenVPN - 1 with DSL and other with T1 - 1500 KM apart). I am guessing it's a bandwidth or jitter issue that is giving me faint problem in playback of prompts when I call in to an echo() test. However, to prove my theory I need to gather some sample of data, network stats, and sound samples that correspond to the network status. Can you please explain the methods and tools used to do this and please show me the simple easy ways rather than the complex detailed ways as I would rather spend really little time on this. If you are going to mention things like wire-shark, I would appreciate it if you dig your notes and send me sample commands and detailed instructions of how packets can be obtained and analyzed. Thanks a lot for the feedback. -- _ -- 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] Wireless Desktop VoIP Phone?
Nortel 1535. Does video as well. On Fri, Dec 17, 2010 at 10:40 AM, Matt mhop...@gmail.com wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? -- _ -- 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
[asterisk-users] How to install the new cdr-stats?
Hi Everyone, I am trying to install the new cdr-stats from http://www.cdr-stats.org/ for Asterisk 1.6 but it's installation instructions are all garbled. It mentions both sqlite and mysql and there are no organized documentation. Not to mention that the apache port 8000 and port 9000 are also confusing and I don't know why is it not easy enough as adding a /var/www/html/cdr-stats rather than tampering with apache settings. I have a standard install of Asterisk 1.6 with addons and I already have the asteriskcdrdb database and everything is logged fine. I would appreciate it if someone can provide their commands for the install or step-by-step instructions. Thanks -- _ -- 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] What is equivalent function to mv command in php for Asterisk Spool directory usage?
Hi Everyone, I understand that there are a few warnings about using cp to move .call files to /var/spool/asterisk/outgoing as they might acted upon before copy is done. So, using PHP, What is the equivalent of mv command? Would it be rename() in php or is there a better method? Thanks, -- _ -- 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 hangs up call after 20s
This is a NAT issue like noted before. Try: localnet=192.168.0.0/ http://192.168.0.0/24255.255.255.0 instead of: localnet=192.168.0.0/24 http://192.168.0.0/24Also, make sure you have all your VPN connections as localnet and other side subnet as localnet as well if you are using VPN. Otherwise, open the neccessary ports needed for SIP and RTP. If you note your router type someone might be able to help more specifically. -Bruce On Wed, Dec 22, 2010 at 12:27 PM, Gilles codecompl...@free.fr wrote: On Wed, 22 Dec 2010 13:18:38 +, Steve Davies davies...@gmail.com wrote: Look in the XLite advanced network settings and disable the 2 timeout settings (RTP and RTCP?). This is not always necessary, but there are sufficient cases where the packets XLite expects appear early on, but do not persist, thus causing a hangup. I think the default timeout is 20 seconds. Thanks for the tip, but I get the same problem with SJPhone and PhonerLite, so it looks like a problem in Asterisk. -- _ -- 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] How to install the new cdr-stats?
Thanks for looking into it. Yes, it missed up and not worth looking at it. Unfortuantly, so are a few products from the same company (probably trying to make money of support which I understand)but it seems they released an install script which is here for CentOS: https://github.com/Star2Billing/cdr-stats/tree/master/scripts/ https://github.com/Star2Billing/cdr-stats/tree/master/scripts/Regards, On Fri, Dec 24, 2010 at 1:29 PM, Doug Lytle supp...@drdos.info wrote: Doug Lytle wrote: I'll let you know what I come up with, hopefully before the weekend ends. Bruce, I gave it a shot this weekend. It's very specific to whatever distro they were using, most of the path information and program location weren't found under Mandriva. The area where they were talking about myql and sqlite, were just the databases that they supported. I find that it's not worth the effort, for such a poorly documented project, to go any further. I'd suggest Asterisk-Stat, it hasn't been updated in a few years, but still works well. It can be had at: http://www.areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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
[asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?
Hi Everyone, I use Asterisk for regularPBX use it's made for. But I want to take it a bit further and use it at cmmand level to be able to send SIP notifies to restart a phone or take advantage of a phone's UPnP capabilities. Is Asterisk capable of that? If so, what is a simple SIP reboot message like and how can I invoke it from a Asterisk CLI? If Asterisk is not the best tool for this purpose what is a very simple to implement SIP stack out there that can do this? Thanks, -- _ -- 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] Using SIP stack within Asterisk to reboot phones - Possible?
Thanks Kai-Uwe and everyone else. I have seen all those examples and I am exploring the sip_notify.conf file now which makes things more clear to me. However, when sending a SIP notify to a phone that is not registered to Asterisk via it's IP address should I expect to receive a success of fail packet back or that is not how SIP Notify works? *sip notify aastra-check-cfg 192.168.0.5* *Sending NOTIFY of type 'aastra-check-cfg' to '192.168.0.5'* * * That is all I see and the phone is not restarted. There might be a few things different about Aastra phones to get them accept SIP Notifies and I would like to hear your experience about it and what features and notifies are available to me as it pretains particulary to Aastra phones. P.S. Are these SIP notifies anything different than simple HTTP get or XML push and receive or do they require a sip stack or a program like Asterisk and it's much more complicated than I think? I want to get a simple page where some phone controls can be done without relying on a heavy program like Asterisk but again if it get's too complicated I won't mind using Asterisk for this purpose. Just want to know my options. Thanks again, On Mon, Dec 27, 2010 at 1:59 PM, Kai-Uwe Jensen kujen...@gmail.com wrote: Lots of good info and pointers so far. But do keep in mind that not all phones will automatically reboot just because you sent it a check-sync or resync event with the sip notify command. I vaguely remember that for e.g. the Polycoms some other condition had to be true: either the phone's config file on the ftp/tftp server had to have a newer time-stamp than the one that was downloaded during the phone's last boot, or a config option had to be set to a non-default value to make the phone reboot unconditionally upon receiving the SIP notify, regardless of the config file's modification date. -- _ -- 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
[asterisk-users] OutCall for Outlook only shows Name from CLID and not Number - hence not pulling contact
Hi Everyone, I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can originate calls see the program login nicely but when a call comes in it only shows the Name portion of the CLID and not the number hence it pulls up a new contact on Outlook. The new contact only show name and last name and no CLID Number again. So, this repeats every-time I call even if I manually enter a number and save the contact or save it without a number. Seems to me that Outcall is not harvesting the CLID number as it should or maybe it's not passing it to outlook so that the old contact which already exists for that number to be pulled. I am wondering if anyone else has experienced this or if you guys think OutCall is really not reliable and I should look for an alternative. Please let me know if there is a solid alternative out there. Thanks -- _ -- 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] OutCall for Outlook only shows Name from CLID and not Number - hence not pulling contact
Thanks for feedback. I am looking mainly for pop-up of Outlook and don't need outgoing call at all but it would be nice to have. Regards, On Tue, Dec 28, 2010 at 4:01 AM, Stefan Schmidt s...@sil.at wrote: Am 28.12.10 07:26, schrieb Bruce B: Hi Everyone, I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can originate calls see the program login nicely but when a call comes in it only shows the Name portion of the CLID and not the number hence it pulls up a new contact on Outlook. The new contact only show name and last name and no CLID Number again. So, this repeats every-time I call even if I manually enter a number and save the contact or save it without a number. Seems to me that Outcall is not harvesting the CLID number as it should or maybe it's not passing it to outlook so that the old contact which already exists for that number to be pulled. I am wondering if anyone else has experienced this or if you guys think OutCall is really not reliable and I should look for an alternative. Please let me know if there is a solid alternative out there. Thanks Hello, I dont know outcall but i can show you an alternative which really works nice for my needs. http://sourceforge.net/projects/siptapi/ this small program use the windows TAPI interface and dials out via sip. But it only works for outgoing calls in the freeware version. For inbound you have to buy it. best regards Stefan -- _ -- 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 -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // v...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - -- _ -- 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] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?
Thanks for the input. I can not replicate the situation as it happens randomely or maybe over the weekend. However I have sent you all the requested command and logs in a separate e-mail for your analyzes. The only thing that stood out at me was the output of lsusb -v at the very end where it timed out. Since all lines didn't work I am to assume that both module went down but per my diagnoses with hwprobe I could see one unit connected and the other was not when the problem happened. Simply connecting/disconnecting that unit or connecting it to another port solved the problem and it showed up in hwprobe This is an Acer Aspire Revo mini PC. I am wondering if the U100s draw too much power? The only other USB connected device is the thumb size wireless connector for the keyboard. Acer computer: http://reviews.cnet.com/desktops/acer-aspire-revo-ar1600/4505-3118_7-33777218.html Looking forward to your analysis. Regards, Bruce On Tue, Dec 28, 2010 at 3:58 PM, Moises Silva moises.si...@gmail.comwrote: On Tue, Dec 28, 2010 at 11:33 AM, Bruce B bruceb...@gmail.com wrote: I appreciate your feedback and let me know what info I can post here that may help resolve the issue (such as output from dmesg or lspci?). Hi Bruce, The following would be useful for starters: 1. cat /etc/wanpipe/*.conf 2. ifconfig -a (from a working and non-working situation) 3. lspci -v and lsusb -v (from a working and non-working situation) 4. wanrouter hwprobe verbose (from a working and non-working situation) 5. /var/log/messages (near the date the problem happened) Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- 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] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?
Thanks for the input. I have the latest drivers but it seems that there is some serious incompatibility issue with the kernel as when the FLASHING happens even if the system is restarted it's still not detected. One has to re-plug it in and then it shows in wanrouter hwprobe. It could also be that the atom board is not compatible with the driver in my case. It your /var/log/messages upon restart do you any line that might match this: * * *cd /var/log/* *grep -o device not accepting address ** *grep -o USB device is disconnected ** *dmesg | egrep device not accepting address * *dmesg | egrep USB device is disconnected * If not then you are not experiencing the same issue. If you have then it's a universal issue and not hardware specific. I would really appreciate it if you look into your logs and let me know. Regards, Bruce On Sat, Jan 1, 2011 at 6:57 PM, Sebastian s...@open-t.co.uk wrote: Hi Bruce, On 12/28/2010 10:51 PM, Bruce B wrote: Thanks for the input. I can not replicate the situation as it happens randomely or maybe over the weekend. However I have sent you all the requested command and logs in a separate e-mail for your analyzes. The only thing that stood out at me was the output of lsusb -v at the very end where it timed out. Since all lines didn't work I am to assume that both module went down but per my diagnoses with hwprobe I could see one unit connected and the other was not when the problem happened. Simply connecting/disconnecting that unit or connecting it to another port solved the problem and it showed up in hwprobe This is an Acer Aspire Revo mini PC. I am wondering if the U100s draw too much power? The only other USB connected device is the thumb size wireless connector for the keyboard. Acer computer: http://reviews.cnet.com/desktops/acer-aspire-revo-ar1600/4505-3118_7-33777218.html Don't know if this will help - but I will butt in with what I have :-) I've been using a Sangoma U100 adapter for about 2 years now. It is connected to a Compaq V2120 laptop (Celeron M 1.4GHz processor) which serves as my home server. It is actually the main reason I went for the U100 - as I couldn't add a PCI or PCIe card to a laptop to get the FXO ports I needed. I have to say I really like the U100 - I believe it is the only low(ish) cost USB based FXO interface on the market. I have had occasional problems with it. I remember it used to just stop working - and the lights would start flashing. If I remember correctly - I went to Sangoma's website and downloaded the latest wanpipe drivers, compiled and installed them - and everything was ok after that. At the moment I'm running Asterisk 1.6.2.9 and wanpipe 3.5.11. I can't remember what version of wanpipe was giving problems, I'm afraid. I also found that mine doesn't really like to be hot-plugged - it just freezes the system with strange characters on the screen. But that was a while ago. Since I've learned it's foibles - it must be at least one year since I had to look at it. Sebastian Looking forward to your analysis. Regards, Bruce On Tue, Dec 28, 2010 at 3:58 PM, Moises Silva moises.si...@gmail.com mailto:moises.si...@gmail.com wrote: On Tue, Dec 28, 2010 at 11:33 AM, Bruce B bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: I appreciate your feedback and let me know what info I can post here that may help resolve the issue (such as output from dmesg or lspci?). Hi Bruce, The following would be useful for starters: 1. cat /etc/wanpipe/*.conf 2. ifconfig -a (from a working and non-working situation) 3. lspci -v and lsusb -v (from a working and non-working situation) 4. wanrouter hwprobe verbose (from a working and non-working situation) 5. /var/log/messages (near the date the problem happened) Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com mailto:m...@sangoma.com -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http
[asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?
Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? Thanks -- _ -- 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] Asterisk Outlook integration
Hi Guys, What is out there other than OutCall that works with M$ Outlook and Asterisk 1.4/1.6 ? I prefer opensource and free (as in free in price) but can consider low price - working - programs as well. OutCall is giving issues with various versions of Outlook and it always brings up NEW CONTACT even if contact exists. Thanks, -- _ -- 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 Outlook integration
Thanks but I doubt it does pop-up of outlook contacts. It probably only does outbound calling. My main need is to have an outlook contact pop-up when a call comes in. I also favor open source if possible. Thanks On Wed, Jan 5, 2011 at 4:02 AM, Giorgio Incantalupo gincantal...@fgasoftware.com wrote: Hi BB, you could try this: http://asterisk-outlook-dialer.voip-singapore.qarchive.org/ Never tested it deeply but apparently seems to work fine. Giorgio Incantalupo Bruce B wrote: Hi Guys, What is out there other than OutCall that works with M$ Outlook and Asterisk 1.4/1.6 ? I prefer opensource and free (as in free in price) but can consider low price - working - programs as well. OutCall is giving issues with various versions of Outlook and it always brings up NEW CONTACT even if contact exists. Thanks, -- _ -- 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 -- _ -- 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] Definations of READ/WRITE parameters of manager.conf contexts?
Hi Everyone, I want to know each and every parameter's detail that can be included in the read= write= in manager.conf Where can I find this? Thanks -- _ -- 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] Definations of READ/WRITE parameters of manager.conf contexts?
Thanks Paul. That is exactly what I was looking for. On Sat, Jan 8, 2011 at 2:07 PM, Paul Belanger pabelan...@digium.com wrote: On 11-01-07 01:33 PM, Bruce B wrote: Where can I find this? manager.conf.sample? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Definations of READ/WRITE parameters of manager.conf contexts?
You are the sort of person who takes the time to write something totally in vain. I never understand the sort of mentality people like you have. No need to respond with an obnoxious comment. ***I was hoping for pointers in asterisk's new wiki or somewhere else that I may not be aware of. On Sat, Jan 8, 2011 at 11:27 AM, Steve Edwards asterisk@sedwards.comwrote: On Fri, 7 Jan 2011, Bruce B wrote: I want to know each and every parameter's detail that can be included in the read= write= in manager.conf Where can I find this? 0) Try and spell check the subject a bit better. It will make it easier for the 'next guy' to search for. 1) Google. 2) The Asterisk source code. Even if you aren't a C programmer grepping through the source code can be very productive. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
[asterisk-users] Do I need a sip proxy?
Hi Everyone, I am running multiple instances of Asterisk in Proxmox and so far I had one central Asterisk feeding all others with trunks from one provider. Now, I want to connect each Asterisk server directly to the provider. Based on my understanding, each connection made to the provider port 5060 would be on a port that is unique to that server. And so other connections made to the same provider will go out through a different port and should receive responses through that different port. At least that is my understanding of NAT. The provider should see me trying to register from the same IP with multiple different ports (high number ports; not talking about 5060 as this is outbound and not inbound) and should be able to differentiate between SIP packets coming from various servers. However, it seems to not happen. There is some sort of clash and only one of the servers shows registered with the provider and other's trunks go down. I have noticed that keeping one server works. It could also be that my Fail2ban kicks in on all servers if the SIP packets received are broadcasted to all servers which shouldn't really happen and router should take of this by sending it to the server that has the established connection through that port. *My equipment:* Asterisk 1.6x Pfsense 1.2.3 Dumb Switch *My questions:* A- What is the rational behind this? B- Do I need a sip proxy server? Something like Siproxd, OpenSIPs, or Kamailio? C- Which one of the above is the easiest to get running given I never tried any of those. D- If I am doing an SIP proxy server then it might have to also be redundant. What options do I have in that and which of above or any other suggested package might be great for future expansions. Clarification on how NAT would work in situations like this would be much appreciated. Thanks -- _ -- 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 + SIP configuration?
Hi, I have OpenVPN and Asterisk working nicely. However, I do use certificates. Though, it shouldn't matter. Can you explain what doesn't work for you? Is the connection not established or is the Asterisk and it's client not communicating? -Bruce On Tue, Jan 11, 2011 at 9:20 AM, Gilles codecompl...@free.fr wrote: Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting through Ethernet in hotels or public wifi hotspots. By any chance, would someone have a working configuration so I can take a look? Thank you. -- _ -- 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] Do I need a sip proxy?
Thanks a lot for the great input Pan. I think you are right on point with this one. I have STATIC PORT enabled in my outbound WAN. I am not sure if it was set for SIP or OpenVPN use but it is there for a reason. So, I try to mingle a bit with Siproxd package. I am a bit fuzzy on it though. If I have the Siproxd enabled, does it act as a one single server that connects multiple times to my provider or providers and then I connect to the Siproxd in return? Or, I can still register from Asterisk directly with the provider(s) and Siproxd will take care of the SIP packets to be handled nicely? If it's the latter then it sounds fine to use otherwise it would not only be complicated but also a downtime to Siproxd mean downtime to all Asterisk servers. ***In addition I have setup Siproxd according to pfsense guide online but once I save the configurations and return to it there are no configs left. I know this question is for pfsense forum but maybe someone else experienced this? ***And to return to my original question, do I need a SIP proxy and which one would be suit my needs? I still like to get an input on my previous e-mail. I have to stay with pfsense for now as it has proven to be a good router in all other aspect. Thanks, On Tue, Jan 11, 2011 at 7:38 AM, Pan B. Christensen p...@ibidium.no wrote: Hello Bruce, Your understanding of NAT is correct, and your setup should work. I’m not familiar with Pfsense, but I suspected that your problem was due to a SIP ALG. Pfsense seems to have a SIP ALG and other special handling of VoIP traffic. Hence, you are not using plain NAT. Pfsense is probably rewriting the SIP packets in addition to the IP packets. Try reconfiguring Pfsense or swapping it for something else. A good way to troubleshoot your scenario is to compare the traffic in your end to the traffic on your providers end (or on either side of pfsense). Pay attention to the source and destination IP and ports in addition to the contents of the SIP messages. http://doc.pfsense.org/index.php/VoIP_Configuration http://en.wikipedia.org/wiki/Application-level_gateway With kind regards, Pan *From:* Bruce B bruceb...@gmail.com *Sent:* Tuesday, January 11, 2011 8:58 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Subject:* [asterisk-users] Do I need a sip proxy? Hi Everyone, I am running multiple instances of Asterisk in Proxmox and so far I had one central Asterisk feeding all others with trunks from one provider. Now, I want to connect each Asterisk server directly to the provider. Based on my understanding, each connection made to the provider port 5060 would be on a port that is unique to that server. And so other connections made to the same provider will go out through a different port and should receive responses through that different port. At least that is my understanding of NAT. The provider should see me trying to register from the same IP with multiple different ports (high number ports; not talking about 5060 as this is outbound and not inbound) and should be able to differentiate between SIP packets coming from various servers. However, it seems to not happen. There is some sort of clash and only one of the servers shows registered with the provider and other's trunks go down. I have noticed that keeping one server works. It could also be that my Fail2ban kicks in on all servers if the SIP packets received are broadcasted to all servers which shouldn't really happen and router should take of this by sending it to the server that has the established connection through that port. *My equipment:* Asterisk 1.6x Pfsense 1.2.3 Dumb Switch *My questions:* A- What is the rational behind this? B- Do I need a sip proxy server? Something like Siproxd, OpenSIPs, or Kamailio? C- Which one of the above is the easiest to get running given I never tried any of those. D- If I am doing an SIP proxy server then it might have to also be redundant. What options do I have in that and which of above or any other suggested package might be great for future expansions. Clarification on how NAT would work in situations like this would be much appreciated. Thanks -- -- _ -- 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
Re: [asterisk-users] OpenVPN + SIP configuration?
Your network layout is correct. I am still unclear what is not working for you, but I guess you can't establish a connection yet. In the config file server.conf for the server side you will have parameter verb=3 which you can change to like 9 and see what the error message is upon connect. If you are using CentOS as client you can also check /var/log/messages on both client and server to see the error messages. You can also try ifconfig on the server side to make sure a Tun0 or a Tunx appears in your network address. -Bruce On Wed, Jan 12, 2011 at 12:14 PM, Gilles codecompl...@free.fr wrote: On Tue, 11 Jan 2011 10:23:18 -0500, Bruce B bruceb...@gmail.com wrote: I have OpenVPN and Asterisk working nicely. However, I do use certificates. Though, it shouldn't matter. Can you explain what doesn't work for you? Is the connection not established or is the Asterisk and it's client not communicating? It's not working, because I'm stuck at what to put in the two configuration files, on either sides :-) Am I correct in understanding that we need three network addresses: - LAN were the server lives, eg. 192.168.0.0/24 - LAN where the client lives, eg. 192.168.1.0/24 - A third network number for the tunnel, eg. 192.168.2.0/24 ? Thank you. -- _ -- 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
[asterisk-users] Paid or Free software that would do pop-up from Outlook 2007 via Asterisk AMI
Hi Everyone, I am looking for a paid version of a program that has proven to work with Outlook 2007 and Asterisk 1.6 on Windows Vista, XP, and maybe even Windows 7. Outcall is not the answer as it has lots of bugs and doesn't work. Something simple with very simple interface would be preferred. ***The program shall query Outlook contacts based on the Caller ID and open up the existing contact or open a New Contact form from Outlook. P.S. Outlook 2007 and Exchange Server 2003 are used. Thanks, -- _ -- 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 + SIP configuration?
In sip_nat.conf you need to specify 10.8.0.1/24 as your localnet and also make sure you have your externip setup as well. Else you will notice one way audio or cut off after 30 seconds. Rest of your work is all good. For security reasons the workstation that creates the keys is not connected to any network (local or internet) -Bruce On Thu, Jan 13, 2011 at 8:24 AM, Gilles codecompl...@free.fr wrote: On Tue, 11 Jan 2011 15:20:39 +0100, Gilles codecompl...@free.fr wrote: By any chance, would someone have a working configuration so I can take a look? Got it working :-) Thanks much guys for the help. For those interested, here's how I did it. Note that the appliance only has the openvpn server, so I used a Ubuntu workstation to create the certificates + keys: = 1. Install OpenVPN on Asterisk server. On appliance, there's only a single binary /bin/openvpn, and configuration files are in /etc/openvpn/. To be positive SIP/RTP packets go through the OpenVPN tunnel, make sure the firewall in front of the OpenVPN/Asterisk server only has OpenVPN port open (default: UDP 1194). 2. On client, from www.openvpn.net, download and install OpenVPN for Windows, which includes Service + GUI 3. If using an appliance with just the openvpn binary, use a workstation to install the OpenVPN package and create certificates + keys: apt-get install openvpn 4. On workstation, copy programs to create keys and certificates: mkdir /etc/openvpn/easy-rsa cp -R /usr/share/doc/openvpn/examples/easy-rsa/2.0/* /etc/openvpn/easy-rsa 5. Create the CA, and one pair of public/private keys for each host (server, clients) #Always use a unique Common Name vi /etc/openvpn/easy-rsa/vars #export variables . ./vars ./clean-all ./build-ca ./build-dh #keys for server ./build-key-server server #keys for client ./build-key client1 6. Create configuration file for server /var/www/server.ovpn: port 1194 proto udp dev tun ca ca.crt cert server.crt key server.key dh dh1024.pem #server will use this network number for OpenVPN tunnel, server = 10.8.0.1 server 10.8.0.0 255.255.255.0 ifconfig-pool-persist ipp.txt keepalive 10 120 #Uncomment if compiled with compression #comp-lzo persist-key persist-tun status openvpn-status.log verb 3 7. Create configuration file for client /var/www/client1.ovpn: dev tun proto udp remote public IP to reach OpenVPN/Asterisk server 1194 resolv-retry infinite nobind persist-key persist-tun ca ca.crt cert client1.crt key client1.key #comp-lzo verb 3 8. Copy keys/certificates/config files to www so can be downloaded by server and client cd /etc/openvpn/easy-rsa/keys cp ca.crt dh1024.pem server.crt server.key client1.crt client1.key server.ovpn client1.ovpn /var/www #So web server can send files chmod 644 /var/www/server.key chmod 644 /var/www/client1.key 9. On server, download files: Asterisk cd /etc/openvpn Asterisk wget http://workstation/ca.crt Asterisk wget http://workstation/dh1024.pem Asterisk wget http://workstation/server.crt Asterisk wget http://workstation/server.key Asterisk chmod 600 server.key Asterisk wget http://workstation/server.ovpn 10. On client, download files: cd c:\program files\openvpn\config wget http://workstation/ca.crt wget http://workstation/client1.crt wget http://workstation/client1.key wget http://workstation/client.ovpn Launch server: Asterisk /bin/openvpn /etc/openvpn/server.ovpn Launch client: Start OpenVPN Service Start OpenVPN GUI with Admin rights: Right-click on OpenVPN GUI icon Connect ping 10.8.0.1 If ping OK, configure SIP client to connect to Asterisk through the server's private IP used by OpenVPN tunnel, eg. 10.8.0.1, and make a call. = HTH, -- _ -- 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 + SIP configuration?
As I said, your tunnel address should be part of localnet. Otherwise you experience what you did. -Bruce On Thu, Jan 13, 2011 at 10:00 AM, Gilles codecompl...@free.fr wrote: On Thu, 13 Jan 2011 15:55:10 +0100, Gilles codecompl...@free.fr wrote: The only issue I notice, is that Asterisk doesn't tell the other end when the local end has hung up, so the other end either remains online or hangs up after 20-30 seconds. Found it: We must add a localnet directive so that Asterisk hangs up the call OK: externip=public IP #local end-points localnet=192.168.0.0/255.255.255.0 #remote end-points through VPN localnet=10.0.0.0/255.0.0.0 -- _ -- 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] CallerID and URL pop up for windows...
What you need already exists: http://bestof.nerdvittles.com/applications/screenpop/ http://bestof.nerdvittles.com/applications/screenpop/But better thing would be to a have TAPI for outlook to query Outlook contact as well because it allows for making notes on the contact. I am willing to pay for that if it is added to URANG II -Bruce On Thu, Jan 13, 2011 at 5:30 PM, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Thursday, January 13, 2011 4:14 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CallerID and URL pop up for windows... On Thu, 13 Jan 2011 13:06:36 -0600, Danny Nicholas da...@debsinc.com wrote: Unless you need a canned app, this would be an easy program to develop on your own. The easiest way (IMO) to do this would be to put a small instance of Apache on your Asterisk server and run a CGI program that interfaces to the local instance of Asterisk and pops a new window when a call comes in. What about a single-EXE Windows app that would connect to the Asterisk Manager Interface and display CID information when a call comes in? Not a bad idea, but possibly a security hole in that the AMI password would have to be imbedded in the application. -- _ -- 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] CallerID and URL pop up for windows...
I tried a lot of these softwares in the past few days and lots of them are just a pile of .. lots of compatibility issues with various versions of Outlook and Windows or simply don't do either of inbound or outbound. However, I have been testing Ingeniussoftware and their product so far works with Inbound and pulls up Outlook contact. Haven't tried outbound. On Fri, Jan 14, 2011 at 9:19 AM, Gilles codecompl...@free.fr wrote: On Thu, 13 Jan 2011 17:59:10 -0500, Bruce B bruceb...@gmail.com wrote: http://bestof.nerdvittles.com/applications/screenpop/But better thing would be to a have TAPI for outlook to query Outlook contact as well because it allows for making notes on the contact. I am willing to pay for that if it is added to URANG II Has someone tried IdentaPop? www.identafone.com/cidpop.html -- _ -- 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
[asterisk-users] Why are 4 ports used for a single call?
Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks -- _ -- 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] Why are 4 ports used for a single call?
I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks -- _ -- 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] Why are 4 ports used for a single call?
Thanks guys. I am not sure whether that call was asymmetric or not but I saw 4 ports open. It could be that the other two ports were remnant of another channel even though I doubt it. Now, when I tried again, it is only 2 ports that is opened like you mentioned, even RTP port, and RTP port +1. So, does Asterisk usually use the symmetric method or is the asymmetric method used as well by some media servers? The reason why I am asking is because there are many many online responses that there is 4 ports needed per call and make sure you keep enough ports open, blah blah... Thanks again On Fri, Jan 14, 2011 at 2:57 PM, Gary Allen solsta...@gmail.com wrote: RTP always uses a random even numbered port, then RTCP will use the next port, which will always be odd numbered. Symmetric RTP only needs two ports, while asymmetric RTP uses four. http://www.armware.dk/RFC/rfc/rfc4961.html On Fri, Jan 14, 2011 at 12:44 PM, Bruce B bruceb...@gmail.com wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks -- _ -- 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 -- _ -- 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] Why are 4 ports used for a single call?
Also, to get the SIP very well as well, SIP uses both TCP/UDP 5060 right? and why are there recommendations of opening 5000-5082 UDP for SIP along with 5060 TCP? Are there any niceties to that as well? maybe video transmission stuff? Thanks again, On Fri, Jan 14, 2011 at 4:12 PM, Bruce B bruceb...@gmail.com wrote: Got it. Thanks. Makes sense to keep an extra two in mind for conference etc Off topic - what is top post? I am using gmail + chrome - no ugly Outlook. On Fri, Jan 14, 2011 at 3:33 PM, Danny Nicholas da...@debsinc.com wrote: Hurray for Microsoft Outlook (for creating this whole top-post thread). Just my .02; The other two ports must have been a remnant of another channel; as for the 4 ports – I think that the 4 port requirement is probably for “niceties” like conferencing and transfers. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* Friday, January 14, 2011 2:15 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Why are 4 ports used for a single call? Thanks guys. I am not sure whether that call was asymmetric or not but I saw 4 ports open. It could be that the other two ports were remnant of another channel even though I doubt it. Now, when I tried again, it is only 2 ports that is opened like you mentioned, even RTP port, and RTP port +1. So, does Asterisk usually use the symmetric method or is the asymmetric method used as well by some media servers? The reason why I am asking is because there are many many online responses that there is 4 ports needed per call and make sure you keep enough ports open, blah blah... Thanks again On Fri, Jan 14, 2011 at 2:57 PM, Gary Allen solsta...@gmail.com wrote: RTP always uses a random even numbered port, then RTCP will use the next port, which will always be odd numbered. Symmetric RTP only needs two ports, while asymmetric RTP uses four. http://www.armware.dk/RFC/rfc/rfc4961.html On Fri, Jan 14, 2011 at 12:44 PM, Bruce B bruceb...@gmail.com wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks -- _ -- 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 -- _ -- 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] Why are 4 ports used for a single call?
Got it. Thanks. Makes sense to keep an extra two in mind for conference etc Off topic - what is top post? I am using gmail + chrome - no ugly Outlook. On Fri, Jan 14, 2011 at 3:33 PM, Danny Nicholas da...@debsinc.com wrote: Hurray for Microsoft Outlook (for creating this whole top-post thread). Just my .02; The other two ports must have been a remnant of another channel; as for the 4 ports – I think that the 4 port requirement is probably for “niceties” like conferencing and transfers. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* Friday, January 14, 2011 2:15 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Why are 4 ports used for a single call? Thanks guys. I am not sure whether that call was asymmetric or not but I saw 4 ports open. It could be that the other two ports were remnant of another channel even though I doubt it. Now, when I tried again, it is only 2 ports that is opened like you mentioned, even RTP port, and RTP port +1. So, does Asterisk usually use the symmetric method or is the asymmetric method used as well by some media servers? The reason why I am asking is because there are many many online responses that there is 4 ports needed per call and make sure you keep enough ports open, blah blah... Thanks again On Fri, Jan 14, 2011 at 2:57 PM, Gary Allen solsta...@gmail.com wrote: RTP always uses a random even numbered port, then RTCP will use the next port, which will always be odd numbered. Symmetric RTP only needs two ports, while asymmetric RTP uses four. http://www.armware.dk/RFC/rfc/rfc4961.html On Fri, Jan 14, 2011 at 12:44 PM, Bruce B bruceb...@gmail.com wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks -- _ -- 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 -- _ -- 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] Why are 4 ports used for a single call?
So, simply pressing Reply and typing in the first line (using gmail webmail without any clients) is a sin here? How is that top posting??? probably your clients reading that way? On Fri, Jan 14, 2011 at 5:13 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote: On Friday 14 January 2011 15:12:29 Bruce B wrote: Off topic - what is top post? I am using gmail + chrome - no ugly Outlook. http://www.justfuckinggoogleit.com/search.pl?query=top+posting It's why most of the experts in here ignore your posts. If you haven't got the good sense to follow etiquette, the Delete key becomes the first line of defense. -- Tilghman -- _ -- 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] Why are 4 ports used for a single call?
You really want to read the LONG LONG signature from some people before you read the actual latest message? I don't know about thatI guess it's a preference. Back to my other questions, now that UDP is clear for me, what ports does SIP require? TCP/UDP 5060 ? and why are there recommendations of opening 5000-5082 UDP for SIP along with 5060 TCP? Are there any niceties to that as well? maybe video transmission stuff? Thanks On Fri, Jan 14, 2011 at 6:32 PM, Tom Rymes try...@rymes.com wrote: On Jan 14, 2011, at 5:24 PM, Bruce B wrote: So, simply pressing Reply and typing in the first line (using gmail webmail without any clients) is a sin here? How is that top posting??? probably your clients reading that way? It may be a sin here, but it is certainly impolite many places, and illogical everywhere. This is because we normally read top to bottom, but top-posting forces you to read bottom to top. Tom -- _ -- 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
[asterisk-users] Tools to Monitor Asterisk Servers and VMs
Hi Everyone, Are there any generally accepted and widely used tools made and tailored to be used for purpose of monitoring Asterisk servers? I am wondering if there is anything that the Asterisk community mostly uses or are there lots of manual scripts written and nothing really exists that every one kind of uses (e.g. Fail2ban for security). Thanks -- _ -- 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] Why are 4 ports used for a single call?
On Fri, Jan 14, 2011 at 6:53 PM, Tom Rymes try...@rymes.com wrote: On Jan 14, 2011, at 6:45 PM, Bruce B wrote: You really want to read the LONG LONG signature from some people before you read the actual latest message? I don't know about thatI guess it's a preference. Suffice it to say, Bruce, this subject has been hashed over thousands, nay, hundreds of thousands of times, and I doubt anything new can be had from doing it again. FYI, It is also considered good etiquette to remove any non-relevant information from the quoted text to keep it short and easy to parse, especially removing the automatically generated footers from the list. As for your question about ports (see, I can stay on topic occasionally!), someone already mentioned something about some equipment using 5004 for RTP, IIRC, and I mentioned the common use of 5061, 5062, 5063, etc for multiple SIP clients behind NAT. There may be other reasons, too. Tom Thanks. That is in both TCP and UDP for SIP right? or simply UDP would do it as well? I am talking strictly in case of Asterisk. -Bruce -- _ -- 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] Top Posting
It was only the people who ONLY asked in a response to go to Google to find answers that annoyed me but slowly posting preference adds up as well. As long as the subject header is not changed all e-mail clients (no matter how stupid they are), now-a-days, create a nice tree. Even so does Hotmail, the worst out there. So I totally don't get what the fuss is. And I was being sarcastic about what top-posting is because I believe it's a stupid personal preference rule that someone made it. I don't care if one is pre-dinosaur age or not. In addition, if all one has to say is Search google or oh my god, my eyes hurt, due to your WHATEVER WAY you post, I see nothing but whining. What is the list for then? If you are smart to answer or have your news up on the air you don't have to bother to answer. I personally never respond to a post if I am going to say use Google or unless I am sure I know what I am talking about. So, in order to be courteous to each other, please move on if you don't want to respond. I can't believe there are people who are setting behind their desks all day waiting to moderate the Asterisk user list while there is no moderation on to be done on this list. I suggest play a game of pacman rather than smart alek responses. At the end I also want to give credit for many smart people out here who without any prejudice do respond and do understand what they are talking about. But there are also the occasional whiners.meh who cares... Thanks for bringing this up. On Fri, Jan 14, 2011 at 8:30 PM, Andrew Latham lath...@gmail.com wrote: Seconded. Although I've succumbed to bottom posting on occasion when following the convention of the ongoing thread. On 01/14/2011 07:42 PM, Don Kelly wrote: Bruce et al… I’m posting a new thread with the “Top Posting” subject so I won’t draw complaints about “hijacking” the 4-port thread. snip When I post (which is rarely, as I have little to offer the list), I top post and explain that it’s my preference and I don’t know how to do it effectively otherwise. This gives everyone fair warning to delete my posts before reading them. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax As mentioned in the past, trimming your post is the best first step on mailing lists. Many of the top post vs bottom post comments happen on the 5+ post on a thread when the size of the email becomes an issue. I have blindly replied in the past and was unable to understand my own email. Take a moment and trim out the messy bits. Use a (snip) or snip to note huge missing areas. As you will note in Don's post there is a history to the argument. Also note Don's multiple signatures which I think he will review after he sees it in action. :) Above all, be polite... ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- -- _ -- 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] Top Posting
Since I don't want anyone bitch at my spelling again: news up = nose up :-) -Bruce On Fri, Jan 14, 2011 at 8:55 PM, Bruce B bruceb...@gmail.com wrote: It was only the people who ONLY asked in a response to go to Google to find answers that annoyed me but slowly posting preference adds up as well. As long as the subject header is not changed all e-mail clients (no matter how stupid they are), now-a-days, create a nice tree. Even so does Hotmail, the worst out there. So I totally don't get what the fuss is. And I was being sarcastic about what top-posting is because I believe it's a stupid personal preference rule that someone made it. I don't care if one is pre-dinosaur age or not. In addition, if all one has to say is Search google or oh my god, my eyes hurt, due to your WHATEVER WAY you post, I see nothing but whining. What is the list for then? If you are smart to answer or have your news up on the air you don't have to bother to answer. I personally never respond to a post if I am going to say use Google or unless I am sure I know what I am talking about. So, in order to be courteous to each other, please move on if you don't want to respond. I can't believe there are people who are setting behind their desks all day waiting to moderate the Asterisk user list while there is no moderation on to be done on this list. I suggest play a game of pacman rather than smart alek responses. At the end I also want to give credit for many smart people out here who without any prejudice do respond and do understand what they are talking about. But there are also the occasional whiners.meh who cares... Thanks for bringing this up. On Fri, Jan 14, 2011 at 8:30 PM, Andrew Latham lath...@gmail.com wrote: Seconded. Although I've succumbed to bottom posting on occasion when following the convention of the ongoing thread. On 01/14/2011 07:42 PM, Don Kelly wrote: Bruce et al… I’m posting a new thread with the “Top Posting” subject so I won’t draw complaints about “hijacking” the 4-port thread. snip When I post (which is rarely, as I have little to offer the list), I top post and explain that it’s my preference and I don’t know how to do it effectively otherwise. This gives everyone fair warning to delete my posts before reading them. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax As mentioned in the past, trimming your post is the best first step on mailing lists. Many of the top post vs bottom post comments happen on the 5+ post on a thread when the size of the email becomes an issue. I have blindly replied in the past and was unable to understand my own email. Take a moment and trim out the messy bits. Use a (snip) or snip to note huge missing areas. As you will note in Don's post there is a history to the argument. Also note Don's multiple signatures which I think he will review after he sees it in action. :) Above all, be polite... ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- -- _ -- 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] Bruce B
LOL what a looser. Are you a fat admin behind a desk who is going to loose his job due to recession and is pissed off? Here is your first response to one of my first posts: I was going to respond with some very insightful and helpful information but I'm not a PRI Guru. Sorry, maybe next time. One stupid useless line because you had an issue with the word Guru. P.S. I have no time to go through your collected list but I should say Good Job. I have a position open for you for data entry. You seem to perform data entry and retrieval jobs very well. I am not being sarcastic! -Bruce On Fri, Jan 14, 2011 at 8:31 PM, Tim Nelson tnel...@fudnet.net wrote: You've been officially added to my kill file [1]. The lists are here to get suggestions and assistance with various issues [2]. They are *NOT* your one stop shop for everyone doing your homework [3][4][5][6][7][8][9]. You make it abundantly clear that you're making no effort whatsoever to find answers to the questions you post. And, rather than listen to answers given, or even suggestions about your list etiquette, you instead choose to ignore those suggestions and ask more questions [10]. AND, to make matters worse, this isn't the only list you actively abuse [11][12][13]. Also, since you're unable to seek information on your own, I've taken the liberty of keeping references to all of the above points for you. If I were a mod, I'd drop you from the list. But alas, pushing your useless drivel to /dev/null will have to suffice [14]. I'll just sit here listening to a very relevant song [15] while I get back to the regularly scheduled programming. --Tim [1] http://en.wikipedia.org/wiki/Kill_file [2] http://en.wikipedia.org/wiki/Mailing_list [3] http://lists.digium.com/pipermail/asterisk-users/2011-January/257684.html [4] http://lists.digium.com/pipermail/asterisk-users/2011-January/257685.html [5] http://lists.digium.com/pipermail/asterisk-users/2011-January/257762.html [6] http://lists.digium.com/pipermail/asterisk-users/2011-January/257832.html [7] http://lists.digium.com/pipermail/asterisk-users/2011-January/257888.html [8] http://lists.digium.com/pipermail/asterisk-users/2011-January/257962.html [9] http://lists.digium.com/pipermail/asterisk-users/2011-January/257992.html [10] http://lists.digium.com/pipermail/asterisk-users/2011-January/257991.html [11] http://www.mail-archive.com/support@pfsense.com/msg21300.html [12] http://www.mail-archive.com/support@pfsense.com/msg21307.html [13] http://www.mail-archive.com/support@pfsense.com/msg21119.html [14] http://en.wikipedia.org/wiki//dev/null [15] http://en.wikipedia.org/wiki/Don't_Go_Away_Mad_(Just_Go_Away) http://en.wikipedia.org/wiki/Don%27t_Go_Away_Mad_%28Just_Go_Away%29 -- _ -- 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] Do I need a sip proxy?
Thanks for the info. I did get it working without any SIP Proxy. There is a bug in pfSense v1.2.3 where certain configs are not removed and some inconsistencies exist in the xml config file. Once I cleaned that and when I limited my Asterisk servers to use different port ranges for UDP traffic now everything is working great. On Tue, Jan 18, 2011 at 7:26 AM, Pan B. Christensen p...@ibidium.no wrote: Hello Bruce, Sorry for the delay. I don't really have time to follow this list much. In your original setup, you did use a sort of SIP Proxy (the central Asterisk feeding the others) depending on your definition. A SIP Proxy would probably solve your issue, but as I stated in my previous mail, you should not need one. Fixing (or exchanging) Pfsense should also solve your issue and then you'll have one less device that can bring your system down. Fixing Pfsense will probably require you to troubleshoot the issue some more to see exactly what happens, so you know what you need to fix. Compare the SIP traffic between your Asterisks and Pfsense to the traffic between Pfsense and your provider. Capture the traffic in .pcap format with ngrep, tcpdump, wireshark or other packet dumping tools, then analyze it in wireshark. To capture traffic outside Pfsense, you'll probably need to mirror a switch port, install a hub or ask your provider to send you a dump. This will require some understanding of the SIP message format and TCP/IP, but it should not be very complicated. I'm quite sure Pfsense changes the contents of the SIP message itself in ways it should not do possibly in addition to changing the IP packets in ways it should not do. It may also possibly block incoming traffic it should not block. If you decide to use a SIP proxy, then going back to your original design (using Asterisk as a proxy) would probably be the easiest for you. Of the alternatives you've listed, I only have experience with Kamailio. In simple setups, its default configuration will not need to be altered much to get it working. Its logic is VERY different to Asterisk, though. I know that Kamailio would be a very good choice for this role. I believe the alternatives would be as well. With kind regards, Pan B. Christensen Senior technician Ibidium AS http://www.ibidium.no/ - Original Message - *From:* Bruce B bruceb...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Tuesday, January 11, 2011 4:37 PM *Subject:* Re: [asterisk-users] Do I need a sip proxy? Thanks a lot for the great input Pan. I think you are right on point with this one. I have STATIC PORT enabled in my outbound WAN. I am not sure if it was set for SIP or OpenVPN use but it is there for a reason. So, I try to mingle a bit with Siproxd package. I am a bit fuzzy on it though. If I have the Siproxd enabled, does it act as a one single server that connects multiple times to my provider or providers and then I connect to the Siproxd in return? Or, I can still register from Asterisk directly with the provider(s) and Siproxd will take care of the SIP packets to be handled nicely? If it's the latter then it sounds fine to use otherwise it would not only be complicated but also a downtime to Siproxd mean downtime to all Asterisk servers. ***In addition I have setup Siproxd according to pfsense guide online but once I save the configurations and return to it there are no configs left. I know this question is for pfsense forum but maybe someone else experienced this? ***And to return to my original question, do I need a SIP proxy and which one would be suit my needs? I still like to get an input on my previous e-mail. I have to stay with pfsense for now as it has proven to be a good router in all other aspect. Thanks, On Tue, Jan 11, 2011 at 7:38 AM, Pan B. Christensen p...@ibidium.nowrote: Hello Bruce, Your understanding of NAT is correct, and your setup should work. I’m not familiar with Pfsense, but I suspected that your problem was due to a SIP ALG. Pfsense seems to have a SIP ALG and other special handling of VoIP traffic. Hence, you are not using plain NAT. Pfsense is probably rewriting the SIP packets in addition to the IP packets. Try reconfiguring Pfsense or swapping it for something else. A good way to troubleshoot your scenario is to compare the traffic in your end to the traffic on your providers end (or on either side of pfsense). Pay attention to the source and destination IP and ports in addition to the contents of the SIP messages. http://doc.pfsense.org/index.php/VoIP_Configuration http://en.wikipedia.org/wiki/Application-level_gateway With kind regards, Pan *From:* Bruce B bruceb...@gmail.com *Sent:* Tuesday, January 11, 2011 8:58 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Subject:* [asterisk-users] Do I need a sip proxy? Hi Everyone, I
Re: [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?
Yes, it does. Bell provides the same as well and it works with Asterisk. -Bruce On Fri, Jan 21, 2011 at 7:11 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Hi list, For a client I am setting up a system which will use T1 PRI from Primus, who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only used switchtypes euroISDN and National. Although the documentation says Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If anybody has experience with Primus, it'll be more helpful. Thanks Zeeshan A Zakaria -- www.visionvoip.com www.ilovetovoip.com www.pbxforall.com -- _ -- 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
[asterisk-users] SIP channel status - Why is it different when calling an internal extension rather than an outside line over SIP?
Hi Everyone, I want to call first party using a .callfile and a second party using a context and then bridge the two calls. I MUST make sure that first party picks up first and then the second party should be dialed. Trying the following using an internal extension works nicely and the playback file is play after the extension picks up. But using the same method for calling an outside phone number (using a good quality SIP provider) does not wait for the channel to come up and starts the Playback line right away. What is the fault behind this and what is workaround? This works: *originate sip/101 extension s@dial_wait* [dial_wait] exten = s,1,Answer exten = s,n,Playback(Please_wait_as_dial_the_second_party) exten = s,n,NoOp(Calling second party) exten = s,n,Dial(SIP/sip_provider/1214555) This doesn't wait for channel to come up and jumps to Playback (s,2) without even the first party yet picking up: *originate SIP/sip_provider/1214888 extension s@dial_wait* * * *Thanks,* -- _ -- 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] Can a duration limit be specified in spool call file?
Hi Everyone, I don't see any parameter for limiting duration of a call in the .call file for Asterisk spool outgoing directory. I'd rather not use a MeetMe to drop the call in a conference room and to then limit the call duration as that complicates things unnecessarily. I am wondering if there is anything else I can do or if the Channel parameter take call duration like the DIAL parameter? Thanks, -- _ -- 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] Any voice changer applications for Asterisk?
Hello, Are there any other other voice changer applications to Asterisk other than the one from Lobstertech? (http://lobstertech.com/voice_changer.html) Specifically interested in open-source but can have a look at economical commercial alternatives as well. Thanks -- _ -- 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] Any voice changer applications for Asterisk?
AAhem. https://wiki.asterisk.org/wiki/display/AST/Function_PITCH_SHIFT Granted, it's in 1.8, but it's in the documentation ;-) Cheers Thanks for the pointer. Unfortunately, I am using 1.6 for all my servers now. But I would like to know if anyone tested the new pitch changer and can give me some feedback. With Lobstertech voice changer I am experiencing feedback on the receiver side along with delay. I guess delay is introduced by the engine that lowers the pitch. I am wondering if the Ast 1.8 version of pitch change does it in a different way that doesn't introduce any delay and makes the sound to be more real. I have also noticed sound degradition with Lobstertech voice changer but again that is probably due to the way the pitch is changed. 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
Re: [asterisk-users] Can a duration limit be specified in spool call file?
On Fri, Jan 28, 2011 at 7:49 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote: On Friday 28 January 2011 18:27:15 Bruce B wrote: Hi Everyone, I don't see any parameter for limiting duration of a call in the .call file for Asterisk spool outgoing directory. I'd rather not use a MeetMe to drop the call in a conference room and to then limit the call duration as that complicates things unnecessarily. I am wondering if there is anything else I can do or if the Channel parameter take call duration like the DIAL parameter? No, but you can specify a Local channel as the channel in the call file and then set a TIMEOUT(absolute) for the call, before you Dial() the actual channel you want to use. Keep in mind that the actual channel could be specified by a Set variable in the callfile. -- Tilghman Can you you please explain the Local Channel concept. I am not sure what should be the Channel line: Channel: xxx/yyy/ Thanks for the input. -- _ -- 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] Any voice changer applications for Asterisk?
On Mon, Feb 7, 2011 at 8:39 AM, Steve Underwood ste...@coppice.org wrote: On 02/06/2011 05:05 PM, Sherwood McGowan wrote: AAhem. https://wiki.asterisk.org/wiki/display/AST/Function_PITCH_SHIFT Granted, it's in 1.8, but it's in the documentation ;-) Cheers That seems to do exactly what the Lobstertech code does. What do people use this for? The Lobstertech one was a fun toy, but seems to be of no practical use. Changing female to male, child to adult, etc. seems pretty useful, but these modules make no attempt to perform a meaningful voice change. They would need to control the formants independent of the pitch to produce anything like a plausible voice adjustment. Thanks for the clarification. I got to agree that it's not of practical use. I was hoping there is a way around the echo and long delay that is generated. I guess not yet. -- _ -- 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] Can a duration limit be specified in spool call file?
On Mon, Feb 7, 2011 at 12:40 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: oh and didn't you guys already have your little histrionics sessin about trimming the goddamned emails, mailing list etiquette about top posting versus bottom, etc../.. My complaint is not something as trivial as where one should reply in a mailing list email, or if one should trim emails every time he replies to the list... The reason why I have my occasional bitch session at a few emails is not that I want to be a policeman..it's twofold... 1. I don't let my son pull that shit when I know he can at least partially figure it out...He'll learn it 4-5 times slower if he's just handed answers to problems...Look at it this way, at least I only reply to emails playing mailing list policeman (gotta love how that was thrown in just before the gentlemen played his own version of enforcer of etiquette) occasionally...my kid hears it all the time 2. I can only take so many years of doing my best to NOT be one of those people who get a ticket from the list police, learning 98% of my trade by experimentation and research, until I start feeling like it's fairly rude to see the amount of please give me the information without me having to put in much work that this list has become...go read the archives...it's getting exponentially worse... Let me close by saying, Bruce, I did go overboard on your email. I apologize. You'll find that I don't do THAT often either, but I am a man, and I'm not playing the it's the internet so my balls are 250% bigger game. Novack, I'll close by saying, have a nice day...and I'll keep trimming occassionally, as I always have...what can I say, I'm not THAT concerned with space...I have plenty of screens...etc...etc... To the rest of you, I sincerely hope you didn't waste your time watching this childish set of insults and rants... *grabs a bucket...tosses water on the laptop*** flame off, here's hoping we can get back to work Good to know you are not short of space :-) I hope everyone else upgrades as well. I wouldn't have minded you saying, *This topic is fully covered in /Doc folder and in asterisk Wiki. Reference URL: .* in one or two lines that was relevant. To go on a lecture. I may still not agree with your points but apology accepted and no hurt feelings here either. I hope that Asterisk mailing list becomes a less hostile list day by day. Cheers (for real) -- _ -- 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] Callback through extensions.conf?
Asterisk runs as root but what about the bash script or the php file that creates the file? Maybe comment the mv command and check the file permissions by *ls -la call-filename.call* to be sure. *chown root.root call-filename* (if root is really the user running Asterisk) and then the mv command line should do the trick. If you are sure that permissions are not the problem and you have archive set to yes then you can browse the */var/spoo/asterisk/outgoing_done* folder to see if the call file is transferred there or not. The file should contain some info to help you and it's existence also means that somehow you are not seeing the call through your CLI as it's processed. However I doubt this is happening. -Bruce On Mon, Feb 7, 2011 at 11:46 AM, Gilles codecompl...@free.fr wrote: On Mon, 7 Feb 2011 07:57:07 -0800 (PST), Steve Edwards asterisk@sedwards.com wrote: sudo /usr/sbin/asterisk -d -d -d -n -v -v -v Oops. A '-c' should be in there :) Thanks Steve for the help. I launched * with asterisk -d -d -d -n -v -v -v -c, and ran module show to check that pbx_spool.so is loaded: = *CLI module show like pbx_spool.so Module Description Use Count pbx_spool.so Outgoing Spool Support 0 1 modules loaded = Next, I moved the following callfile to /var/spool/asterisk/outgoing: = #callfileSIP.call Channel: SIP/xlite Context: callback-dialtone-auth Extension: s Priority: 1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 = Asterisk runs as root, and owns this file as well. Unfortunately, nothing shows up in the console, the xlite extension isn't called, even after waiting for a few minutes. Could it be that pbx_spool.so isn't really loaded, or is Asterisk somehow configured to ignore callfiles? Thank you. -- _ -- 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] Callback through extensions.conf?
In my (1.4.X) experience, the file just stays in /var/spool/asterisk/outgoing and gets “little tags” added until you get the problem resolved or delete the file. That is absolutely true if the file is not processed. I guess he can again do a ls -la in that folder to check permissions for the file not processed. -Bruce -- _ -- 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] Can a duration limit be specified in spool call file?
Thanks Faisal. That is it. I was confused by the fact that there is also the Context, Extension, and Priority in the .call file that should be filled along with the Channle: local. I found out that the call file first calls the local channel context and once that is connected then it moves onto the second context that is defined in subsequent the variables. Indeed this was what was throwing me off. -Bruce On Tue, Feb 8, 2011 at 1:57 AM, fai...@vopium.com wrote: Hi, If you need full control on both legs of call you can redirect Leg-1 to your dialplan as Channel: Local/your-extension@your-context/nLocal/your-extension@your-context/n and from there you control the Leg-1 using dial-plan or AGI as you like while Leg is normally comes to dialplan and totally in controll. Regards, Faisal scussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Can a duration limit be specified in spool call file? Bruce, All in all, I don't think it's that hostile, it just goes through cycles...maybe a good number of us may indeed have estrogen issues and it's the moon, who knows ;-) LOL Cheers (and I always mean it, seriously :D ) Sherwood McGowan Yes, THAT Mick -- _ -- 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
[asterisk-users] IP ban list by country
Hi everyone, I know it's off topic from Asterisk directly but yet related. What sources do you use to limit SIP connecting customers to specific countries by IP (e.g. allowing USA and not China). It would help me a lot of you can note the sources you trust that are complete and up to date. Thanks -- _ -- 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 Call File using Local Channel not passing Variable back to Dialplan
Thanks for the input. Lack of proper documentation really causes issues with
things like this. I think it's noted somewhere on Voipinfo that SetVar was
to be used with version 1.0 and prior to 1.6. It turned out that the issue
was with s,1,Answer() for first leg of the call. Taking that out fixed the
issue and now the whole context is read. Previously I was mistakenly looking
at the second leg which had everything garbled.
On Tue, Feb 15, 2011 at 12:05 AM, Mike Diehl mdi...@diehlnet.com wrote:
The syntax that I use is:
SetVar: username=justincase
I'm using Asterisk 1.6.2. Hope that helps.
Mike.
Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
I am trying to pass a variable using the .call files but it turns out
blank.
Can someone please point out what might be wrong here:
*/tmp/spool-file.sh*
*--*
echo Channel: Local/s@callback_leg*1*/n
CallerID: \Call-back\ 123456
MaxRetries: 0
RetryTime: 10
WaitTime: 45
Context: callback_leg*2*
Extension: s
Priority: 1
Set: myNum=$1
Set: partyNum=$2
AlwaysDelete: Yes
Archive: Yes $call_file
*Context - callback_leg1*
*--*
[callback_leg1]
exten = s,1,Answer()
exten = s,1,NoOp(PartyNum=${partyNum})
exten = s,n,NoOp(MyNum=${myNum})
exten = s,n,Hangup()
Both the NoOp for $partyNum and $myNum come back blank. Does this have
anything to do with the use of Local Channel?
Thanks
--
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[asterisk-users] No ring tone on inbound call - but channel connects fine
Hi Everyone, I have a SIP turnk which works fine with both inbound and outbound calling. However, the only issue is that there is no Ring Tone if someone calls us. The phones used are Aastra and Polycom connected to the PBX via VPN (SIP). I do get an outbound ring tone, so it's not that there is any media loss between the phones and the PBX. But when the DID is called there is dead silence until the call is picked up. What is generally causing something like this? and where should I start looking? Much appreciate your experienced tips. Thanks -- _ -- 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] No ring tone on inbound call - but channelconnects fine
Thanks. Indeed ringing instead of MoH which was missing files fixed the issue. Thanks for the quick great tip. Simple things hide from us sometime. On Wed, Feb 16, 2011 at 3:48 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* Wednesday, February 16, 2011 2:33 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] No ring tone on inbound call - but channelconnects fine Hi Everyone, I have a SIP turnk which works fine with both inbound and outbound calling. However, the only issue is that there is no Ring Tone if someone calls us. The phones used are Aastra and Polycom connected to the PBX via VPN (SIP). I do get an outbound ring tone, so it's not that there is any media loss between the phones and the PBX. But when the DID is called there is dead silence until the call is picked up. What is generally causing something like this? and where should I start looking? Much appreciate your experienced tips. Thanks This sounds like a dialplan problem. My thought is that your SIP trunk should go to an “incoming” context that does something like this: In-house phones are 1000 and 1001 [incoming] Exten = s,1,answer Exten = s,n,Dial(SIP/1000SIP/1001,30,mKkTt) -- _ -- 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
[asterisk-users] PRI wanrouter status shows disconnected - system problem or Telco?
Hi everyone, I am reading through Sangoma Wiki right now. But someone may already and quickly notice this. I have a system that is down since the morning (maybe power intruptions). All seems fine except for wanrouter status shows disconnected. Following are the alarms raised. Should I call telco (they have long wait times) or should I just keep searching online for troubleshooting tips. *wanpipe1| AFT TE1 | N/A | Disconnected | * This is what shows. root@pbx:~ $ wanpipemon -i w1g1 -c Ta * w1g1: T1 Rx Alarms (Framer) * *ALOS: OFF | LOS: ON* *RED:ON | AIS: OFF* *LOF:ON | RAI: OFF* * * ** w1g1: T1 Rx Alarms (LIU) ** * * *Short Circuit: OFF* *Open Circuit: OFF* *Loss of Signal: ON* * w1g1: T1 Tx Alarms * AIS:ON | YEL: ON * w1g1: T1 Performance Monitoring Counters * Line Code Violation : 0 Bit Errors (CRC6/Ft/Fs) : 0 Out of Frame Errors : 0 Rx Level: -36db root@pbx:~ $ ifconfig w1g1 Link encap:Point-to-Point Protocol UP POINTOPOINT NOARP MTU:8 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:0 errors:0 dropped:0 overruns:0 carrier:4383 collisions:0 txqueuelen:100 RX bytes:0 (0.0 b) TX bytes:0 (0.0 b) Interrupt:177 Memory:f8c5-f8c51fff root@pbx:~ $ wanrouter hwprobe verbose - | Wanpipe Hardware Probe Info (verbose) | - 1 . AFT-A101-SH : SLOT=4 : BUS=4 : IRQ=9 : CPU=A : PORT=1 : HWEC=32 : V=36 +01:DS26521: PCIe: PLX2 Card Cnt: A101-2=1 root@pbx:~ $ wanrouter status Devices currently active: wanpipe1 Wanpipe Config: Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate | wanpipe1| N/A | A101/1D/A102/2D/4/4D/8| 177 | 4 | 1| N/A | 0 | Wanrouter Status: Device name | Protocol | Station | Status| *wanpipe1| AFT TE1 | N/A | Disconnected |* *pbx*CLI pri show span 1* Primary D-channel: 24 Status: Provisioned, In Alarm, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 Logical Channel Mapping: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 Thanks, -- _ -- 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] Any good tutorials for setting up Asterisk SNMP and Cacti for remote monitoring?
Hi Everyone, I have been searching the web and I don't know if SNMP is just that complex to setup or that not many people use SNMP to monitor Asterisk but the information is scattered all over. I have got to the point to configure SNMP with Asterisk and then it's all confusing from there on to actually see the graphs in Cacti. I would appreciate it if you can post your steps or point me to a good guide posted somewhere on the web. I have followed this but it's not complete: http://www.voipphreak.ca/2008/10/28/asterisk-snmp-with-cacti-howto-upgraded-for-asterisk-16-and-ubuntu/ ***Please don't post any smart-aleck comments like google it. Thanks, -- _ -- 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] Any good tutorials for setting up Asterisk SNMP and Cacti for remote monitoring?
Thanks. This comes really close. My asterisk currently has snmp setup properly and I can see it shows the output when I do snmpwalk command. I am stuck at Cacti end. Wondering what to do to setup the asterisk remote end. The tutorial you provided is for Nagios (which I tend to stay away due to it's install complications) Thanks again, On Mon, Mar 7, 2011 at 2:14 AM, Faisal Hanif fai...@vopium.com wrote: http://www.danielaliaman.com/blog/files/AsteriskSNMPtutorial.pdf *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* Sunday, March 06, 2011 10:59 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Any good tutorials for setting up Asterisk SNMP and Cacti for remote monitoring? Hi Everyone, I have been searching the web and I don't know if SNMP is just that complex to setup or that not many people use SNMP to monitor Asterisk but the information is scattered all over. I have got to the point to configure SNMP with Asterisk and then it's all confusing from there on to actually see the graphs in Cacti. I would appreciate it if you can post your steps or point me to a good guide posted somewhere on the web. I have followed this but it's not complete: http://www.voipphreak.ca/2008/10/28/asterisk-snmp-with-cacti-howto-upgraded-for-asterisk-16-and-ubuntu/ ***Please don't post any smart-aleck comments like google it. Thanks, -- _ -- 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
[asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?
Hi everyone, Installed asterisk from yum repository but I think H.323 is not supported as I tried commands like this and they don't work: - *h.323 debug*: Enable chan_h323 debug - *h.323 gk cycle*: Manually re-register with the Gatekeper - *h.323 hangup*: Manually try to hang up a call - *h.323 no debug*: Disable chan_h323 debug - *h.323 no trace*: Disable H.323 Stack Tracing - *h.323 show codecs*: Show enabled codecs - *h.323 show tokens*: Manually try to hang up a call - *h.323 trace*: Enable H.323 Stack Tracing Of course I can't go to source since I am using the repository. How can I install H.323. Is that OH323 I should look for? Thanks -- _ -- 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
