[asterisk-users] Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network?

2012-02-17 Thread Frank Church
Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network? I have been having some troubles with a Linksys Sipura 2100 series, which suffers from NO AUDIO after a few calls.. Because it is on the same subnet as Asterisk it is configured with nat=no. When you think

Re: [asterisk-users] Asterisk V/s FreeSwitch

2012-02-07 Thread Frank Church
Freeswitch was engineered from scratch by some Asterisk developers who wanted to start afresh on a cleaner programming base. Asterisk is like Topsy, She just growed and had to maintain backward compatibility. The latest versions of Asterisk are reported to be much improved in that respect. On 7

[asterisk-users] Why doesn't this manager.conf code work on Asterisk 1.6.2 and 1.8.9?

2012-02-06 Thread Frank Church
Why doesn't this manager.conf code work on Asterisk 1.6.2 and 1.8.9? It works perfectly on Asterisk 1.4 In Asterisk 1.6 it appears to disconnect as soon as events occur and in 1.8.9 it can't be read at all. Apparently it has some syntax issues with 1.8.9. Is it possible to tell at a glance what

[asterisk-users] What packages are required to get cdr_adaptive_odbc to be compiled in Asterisk?

2012-02-06 Thread Frank Church
When I do a make menuselect cdr_adaptive_odbc is disabled. What packages are required to enable it? Even after executing apt-get install unixodbc libmyodbc odbc-postgresql tdsodbc unixodbc-bin it is still disabled. What am I missing? /voipfc --

Re: [asterisk-users] What packages are required to getcdr_adaptive_odbc to be compiled in Asterisk?

2012-02-06 Thread Frank Church
[mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Frank Church *Sent:* Monday, February 06, 2012 7:39 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] What packages are required to getcdr_adaptive_odbc to be compiled in Asterisk? When

[asterisk-users] Are there any ATAs that support IP6?

2012-02-06 Thread Frank Church
Are there any ATAs that support IPv6 in the wild, given that IP4 address are running out? /voipfc -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Does Asterisk permit multiple registrations to the same host?

2012-01-19 Thread Frank Church
: match_auth_username=yes Leandro 2012/1/19 Frank Church voi...@gmail.com: Does Asterisk permit multiple registrations to the same host? Each registration has a different username and password The purpose is for billing, handling incoming calls is not important, although it will be a bonus. I guess I

[asterisk-users] Does Asterisk permit multiple registrations to the same host?

2012-01-18 Thread Frank Church
Does Asterisk permit multiple registrations to the same host? Each registration has a different username and password The purpose is for billing, handling incoming calls is not important, although it will be a bonus. I guess I should also ask the converse, whether the receiving host can accept

[asterisk-users] What are the minimal permissions required to read the PeerStatus and Registry events?

2010-12-03 Thread Frank Church
I am logging events from the AMI and the PeerStatus and Registry events show that the privilege for them is System,All. Can a lower set of privileges be used? All looks pretty high to me. /Frank -- _ -- Bandwidth and

[asterisk-users] Should external hosts be able to register on Asterisk behind a firewall?

2010-08-07 Thread Frank Church
Of late I have noticed bruteforce attempts to register to a homebased Asterisk server which is behind a router firewall and I need to know whether the router on the firewall is the culprit here. It only stops after the router is restarted. The router is the HG521 from talktalk and the firewall

[asterisk-users] How does deny/permit work in sip.conf?

2010-08-06 Thread Frank Church
I have been seeing some attempts to register devices on my Asterisk and I want to reconfigure it so that devices will be registered only if they are from the correct address, ie 192.168.1.8/255.255.255.255. I thought using a config like deny=0.0.0.0/0.0.0.0 permit=192.168.1.8/255.255.255.255

Re: [asterisk-users] How does deny/permit work in sip.conf?

2010-08-06 Thread Frank Church
On 6 August 2010 16:21, Bruce Ferrell bferr...@baywinds.org wrote: On 08/06/2010 07:45 AM, Frank Church wrote: I have been seeing some attempts to register devices on my Asterisk and I want to reconfigure it so that devices will be registered only if they are from the correct address, ie

Re: [asterisk-users] How does deny/permit work in sip.conf?

2010-08-06 Thread Frank Church
On 7 August 2010 03:54, Bruce Ferrell bferr...@baywinds.org wrote: On 08/06/2010 07:30 PM, Bruce Ferrell wrote: On 08/06/2010 02:16 PM, Frank Church wrote: On 6 August 2010 16:21, Bruce Ferrell bferr...@baywinds.org wrote: On 08/06/2010 07:45 AM, Frank Church wrote: I have been seeing

[asterisk-users] MAC Address prefixes of Voip equipment

2010-07-12 Thread Frank Church
Is there a database of MAC address prefixes used the common VoIP devices. I see the Linksys Sipura devices state with 00:0E. Does the same apply to other Linksys VoIP equipment? Is there some way VoIP equipment allow themselves to be identified by requesting data from some ports? --

Re: [asterisk-users] MAC Address prefixes of Voip equipment

2010-07-12 Thread Frank Church
More googling got me this page - http://www.freepbx.org/v2/wiki/DevicesTakeTwo Very useful Thanks On 12 July 2010 16:41, Frank Church voi...@googlemail.com wrote: Is there a database of MAC address prefixes used the common VoIP devices. I see the Linksys Sipura devices state with 00:0E

[asterisk-users] Minimum modules required to run VoIP and CDR

2010-06-30 Thread Frank Church
What is the minimal module set required to run SIP with database CDR logging. I compiled Asterisk from source and I obviously compiled more stuff than I needed for VoIP and CDR logging to postgres. Sometimes there is a long gap between Asterisk starting and devices being able to register. sip

Re: [asterisk-users] Minimum modules required to run VoIP and CDR

2010-06-30 Thread Frank Church
The DNS setup itself is fine. The sip module just seems to take too much time to load. My modules.conf uses autoload=yes and it seems that many unwanted modules are loaded before sip itself starts. On 30 June 2010 13:52, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi!

[asterisk-users] Do AMI Events have timestamps?

2010-04-13 Thread Frank Church
I have been monitoring AMI events and realized that they don't have timestamps. Is that standard behaviour, or is there some way to get them to include timestamps? I am on 1.4. Is it available on 1.6? -- _ -- Bandwidth and

Re: [asterisk-users] [asterisk-biz] Asterisk system for church call center

2010-03-30 Thread Frank Church
On 30 March 2010 02:04, Mark Phillips g7...@g7ltt.com wrote: They say confession is good for the soul. Perhaps they are offering a phone in confessional service? Unfortunately the business of the church often flies in the face of the business of the Church. On 03/29/2010 07:48 PM, Alex

Re: [asterisk-users] Asterisk system for church call center

2010-03-30 Thread Frank Church
On 29 March 2010 21:46, Frank Church voi...@googlemail.com wrote: I have been asked by my church to recommend a VoIP system which can do the following. They do internet radio shows which are sometimes broadcast on radio. They are looking for a system which does the following for about 5

[asterisk-users] Asterisk system for church call center

2010-03-29 Thread Frank Church
thing for me. One of the distributions with SugarCRM comes to mind here. Sorry for cross-posting, but ready made and commercially supported systems are not ruled out, if they come within their budget. Regards Frank Church

[asterisk-users] Asterisk system for church call center

2010-03-29 Thread Frank Church
thing for me. One of the distributions with SugarCRM comes to mind here. Sorry for cross-posting, but ready made and commercially supported systems are not ruled out, if they come within their budget. Regards Frank Church

[asterisk-users] Is there a way for a peer to clear its registration from a server?

2010-03-11 Thread Frank Church
Is there a way for a client to tell a server where it is registered to remove the registration? /voipfc -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Is there a way for a peer to clear its registration from a server?

2010-03-11 Thread Frank Church
Thanks. Is there command is used for that? I have checked the help show and there is no command like sip register or sip unregister in the list. Is it available on version 1.4? On 11 March 2010 13:08, Kevin P. Fleming kpflem...@digium.com wrote: Frank Church wrote: Is there a way

Re: [asterisk-users] billsec is set to duration if call is not answered

2010-02-12 Thread Frank Church
and billsec is set to duration, or duration - 1. Is that behaviour that has been observed before? On 8 February 2010 17:00, Frank Church voi...@googlemail.com wrote: The behaviour of my Asterisk appears to have changed suddenly without any apparent cause. The version is use is 1.4.27.1 When a call

[asterisk-users] billsec is set to duration if call is not answered

2010-02-08 Thread Frank Church
The behaviour of my Asterisk appears to have changed suddenly without any apparent cause. The version is use is 1.4.27.1 When a call is not answered billsec is set to duration, and calls are charged. I can't see any change I could have made to cause this problem. Is it something already known in

[asterisk-users] Do the Linksys Sipura series have a known problem with Asterisk?

2010-02-04 Thread Frank Church
I have a Linksys Sipura SPA2102 connected to Asterisk 1.4.27 and sometimes it doesn't connect at all. I keep getting a busy signal when I try to dial. It appears to happen most often when both lines are registered. The 2 lines on Linksys lines also use different ports. Does that mean than it is

Re: [asterisk-users] Smallest possible Asterisk VM

2010-02-02 Thread Frank Church
I have developed a minimal call shop billing system that includes an Asterisk VM and I want it to be as small as to reduce the installation size. 100Mb is good On 2 February 2010 05:41, Frank Church voi...@googlemail.com wrote: How small can an Asterisk system be, in terms of disk space

[asterisk-users] Smallest possible Asterisk VM

2010-02-01 Thread Frank Church
How small can an Asterisk system be, in terms of disk space utilized? I am looking for just asterisk, with mysql, postgresql, or sqlite, with PHP and Python. After finishing the build and removing the tools, how small can the whole system be? 100Mb, 200Mb? Can packages be used to build the

[asterisk-users] Problems getting Asterisk to detect call in SUSE9.3, with FritzCard

2007-11-26 Thread Frank Church
I have installed an Asterisk 1.4 on Suse93 using a FritzCard. Some calls are logged to the ISDN log, but Asterisk is not detecting incoming calls. I wonder whether some other device or process is preventing Asterisk from gaining access to the isdn line? Is there some way to ensure that only

Re: [asterisk-users] Problems getting Asterisk to detect call in SUSE9.3, with FritzCard

2007-11-26 Thread Frank Church
On 26/11/2007, Per Jessen [EMAIL PROTECTED] wrote: Frank Church wrote: I have installed an Asterisk 1.4 on Suse93 using a FritzCard. Some calls are logged to the ISDN log, but Asterisk is not detecting incoming calls. I wonder whether some other device or process is preventing

Re: [asterisk-users] Problems getting Asterisk to detect call in SUSE9.3, with FritzCard

2007-11-26 Thread Frank Church
On 26/11/2007, Frank Church [EMAIL PROTECTED] wrote: I have installed an Asterisk 1.4 on Suse93 using a FritzCard. Some calls are logged to the ISDN log, but Asterisk is not detecting incoming calls. I wonder whether some other device or process is preventing Asterisk from gaining access

[asterisk-users] Are the ATAs which can allow multiple extensions from one network connection?

2007-11-04 Thread Frank Church
Are there ATAs that allow different phone numbers from one network connection? Such as supporting multiple IP addresses so that each RJ11 has a different extension or some other way? ___ --Bandwidth and Colocation Provided by

[asterisk-users] How to combine a Fritz ISDN card with analogue handsets

2007-10-27 Thread Frank Church
I want to use a Fritz AVM ISDN card to create a switch which is connected to 4 analogue extensions. I believe I need a 4 port FXS module for that, are there any cheap but reliable options out there? Are there some guides that go through the whole process? /voipfc

Re: [asterisk-users] Re: Where is the PlayDTMF command?

2006-10-15 Thread Frank Church
../../ast_trunk_manager_PlayDTMF.patch Regards On 10/13/06, Frank Church [EMAIL PROTECTED] wrote: When I try to apply this patch - ast_trunk_manager_PlayDTMF.patch - I receive the error below missing header for unified diff at line 3 of patch can't find file to patch at input line 3 Perhaps you used

Re: [asterisk-users] Calls being disconnected across VPN

2006-10-15 Thread Frank Church
Check the rpttimeout setting in sip.conf for the necessary extensions. As you are not making normal calls to outside parties, that setting may not applicable in your case. On 10/13/06, Jason Adams [EMAIL PROTECTED] wrote: Hey All, Sometimes we are running into issues with calls randomly

Re: [asterisk-users] Re: Where is the PlayDTMF command?

2006-10-13 Thread Frank Church
to this was: -- |--- app_senddtmf.c~2006-05-04 15:27:41.0 -0500 |+++ app_senddtmf.c 2006-05-04 15:29:21.0 -0500 -- File to patch: Is there something else I am missing On 10/12/06, Frank Church [EMAIL PROTECTED] wrote: Hi Moises, I have looked on that page

Re: [asterisk-users] Re: Where is the PlayDTMF command?

2006-10-12 Thread Frank Church
are never added to release branches, so you need to patch 1.2.12.1 adapting the trunk patch. Dont worry, is an easy patch. Regards On 10/11/06, Frank Church [EMAIL PROTECTED] wrote: Hi Moises, Ignore my last reply about the presence of the DTMF in 1.4. Do you have the source for patching

Re: [asterisk-users] Re: Where is the PlayDTMF command?

2006-10-11 Thread Frank Church
Hi Moises, does the you mentioned earlier at http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch include the DTMF event, or is it for PlayDTMF and SendDTMF? Looking through the actions on bug6082 it is hard to tell whether the DTMF event patch is still in there when I last compiled that branch

Re: [asterisk-users] Re: Where is the PlayDTMF command?

2006-10-11 Thread Frank Church
Hi Moises, Ignore my last reply about the presence of the DTMF in 1.4. Do you have the source for patching the DTMF event? There is no link to it on the bug6082 page, and I am not quite sure how it can be obtained from SVN. Regards Richard On 10/12/06, Frank Church [EMAIL PROTECTED] wrote

Re: [asterisk-users] Where is the PlayDTMF command?

2006-10-04 Thread Frank Church
Moises, do you know if the DTMF event in bug 6082 made it into version 1.4? When I last tried to compile that branch it needed the latest version of make 3.81, which trunk did not, and caused me to wonder if it had been committed to trunk. The DTMF detection events in trunk did not also

Re: [asterisk-users] Where is the PlayDTMF command?

2006-10-04 Thread Frank Church
On 10/4/06, Moises Silva [EMAIL PROTECTED] wrote: I could be wrong here, but I think that you're looking for SendDTMF and not PlayDTMF. getting it confuddled with PlayTones? He is not confused. PlayDTMF is a manager command, not an dial plan application, but included in the same module that

Re: [asterisk-users] How to send DTMF down a channel

2006-09-16 Thread Frank Church
] console = notice,warning,error,verbose,debug Regards On 9/15/06, Frank Church [EMAIL PROTECTED] wrote: The program in question is an adaptation an AGI calling card program. It is adapted for callback by setting by channelling the callback call into the context used for the normal inbound leg

[asterisk-users] How to send DTMF down a channel

2006-09-14 Thread Frank Church
the channel and the dtmf numbers as parameters and send the DTMF signals? Frank Church ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

Re: [asterisk-users] How to send DTMF down a channel

2006-09-14 Thread Frank Church
ideas of what the problem might be? On 9/14/06, Moises Silva [EMAIL PROTECTED] wrote: http://www.voip-info.org/wiki-Asterisk+cmd+SendDTMF Regards On 9/14/06, Frank Church [EMAIL PROTECTED] wrote: How can DTMF be sent down a channel? I am thinking of method where say a channel id can

Re: [asterisk-users] How to send DTMF down a channel

2006-09-14 Thread Frank Church
and it . On 9/15/06, Moises Silva [EMAIL PROTECTED] wrote: Frank. PlayDTMF and SendDTMF is the same as pressing keys at the phone. Im not understanding well, can you please explain a practical scenario of how do you expect it to work, and how actually works? :) Thanks Regards On 9/14/06, Frank Church

[asterisk-users] uConnect Voip device

2006-09-07 Thread Frank Church
Does this device allow connection to other phones besides Skype, like Xten Xlite? http://www.voipvoice.com/UConnect-2.html. Compatibility with standard voip is not mentioned on their website? ___ --Bandwidth and Colocation provided by Easynews.com --

[asterisk-users] Re: uConnect Voip device

2006-09-07 Thread Frank Church
Bump? On 9/7/06, Frank Church [EMAIL PROTECTED] wrote: Does this device allow connection to other phones besides Skype, like Xten Xlite? http://www.voipvoice.com/UConnect-2.html. Compatibility with standard voip is not mentioned on their website

Re: [asterisk-users] Re: uConnect Voip device

2006-09-07 Thread Frank Church
people who might have tried it before. On 9/7/06, Frank Church [EMAIL PROTECTED] wrote: Bump? On 9/7/06, Frank Church [EMAIL PROTECTED] wrote: Does this device allow connection to other phones besides Skype, like Xten Xlite? http://www.voipvoice.com/UConnect-2.html . Compatibility

[asterisk-users] phpagi syntax and SendDTMF

2006-09-01 Thread Frank Church
When SendDTMF is used on a channel, which party is the DTMF being sent to, the callee or the caller? What is the syntax for using SendDTMF in an AGI command? F Church ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing