Re: [asterisk-users] Had it with Dell Garbage - HP Question
On 00:05, Mon 31 Mar 08, Al Baker wrote: Could you elaborate a bit more on : For example, if I install zaptel from source, your support contract with them is void. Does this mean it is impossible to run Asterisk on Vendor Supported versions of RedHat or Suse ? Installing zaptel from source means you use a kernel module that is not tested/supported by RedHat/Suse. So if you call them for support they wont help you unless you unload this module and then reproduce the problem. Thanks Michiel van Baak wrote: On 02:34, Sat 29 Mar 08, Al Baker wrote: Helps a bunch !!! One follow up question - out of all of your possible choices for the OS how did you pick *Debian*. I 'm not saying is bad, I just know nothing about the particular disto. and and very curious what it brought to the table that made you pick over say *RedHat* - where you can *buy support *or *SUSE* - where you can *buy support*. My fear from hell is that I' get 50 or 60 of these boxes in, start having kernel panics, and have no damn body to help except the folks on mailing lists. Mind you these are often really smart people, very generously giving of their time, but not quite the say as a manned/paid support organization. I choose Debian because I was already using it. And because there are people out there that can help me. I dont want the support from suse or redhat because they wont help me when running anything that's not in their repositories. For example, if I install zaptel from source, your support contract with them is void. I also really like the Open and Free mindset of Debian. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards
Thank you for all your time om your most detailed response. It is extremely helpful. The vendor's web page is http://www.penguincomputing.com/index.php?option=com_contentid=170Itemid=209task=viewsysid=10007609 *PCI EXPANSION SLOTS* Number of Slots 5 Slot Speed PCI Express: two x8 slots, two x8 low profile slots; PCI-X: 64-bit/100MHz or if that doesn't display *PCI EXPANSION SLOTS* Number of Slots 5 Slot SpeedPCI Express: two x8 slots, two x8 low profile slots; PCI-X: 64-bit/100MHz or PCI EXPANSION SLOTS Number of Slots 5 Slot Speed PCI Express: two x8 slots, two x8 low profile slots; PCI-X: 64-bit/100MHz Nick Seraphin wrote: On Sat, 29 Mar 2008, Al Baker wrote: Detailed specs for the types of PCI slots on the system were posted each and every time I posted int the line Actually, your description wasn't 100% clear at all. _PCI Express_*: _two x8 slots*_, _two x8 low profile slots*_; *_PCI-X: 64-bit/100MHz_* 1) This description seems to IMPLY that there are 5 slots total. Do you know if this is in fact correct? It implies there are 2 PCI-E x8 slots, 2 PCI-E x8 low profile slots, and 1 PCI-X slot. I wouldn't rely on that however without talking to the vendor. 2) The first PCI Express: heading would normally imply that the slots listed afterwards are ALL PCI Express slots, however PCI-X is not PCI Express, so the vendor's description is confusing and misleading. 3) All these damn *'s you keep inserting, are those all done by you, or are some of them from the web page description? Most of the time when something has a * by it that means it's conditional on a footnote that appears at the bottom of the section or page. Are there footnotes we need to know about to clarify this? 4) Is this a rackmount server or a tower case? If rackmount, is it a 1u server or a 2u server or a 4u server? Just because the motherboard has 5 slots doesn't mean the case it is installed in will support 5 cards. A 1u case rarely supports more than 1 or 2 cards, and always requires a riser card. A 2u server rarely supports more than 2 cards unless they are low-profile. A 4u server might allow 5 cards, IF the case is designed with 5 slot openings in the back. 5) Are all the card slots open and available to you at time of shipping? Many options a customer orders with a server, such as a RAID controller or additional network ports will fill one or more of the available slots. You need to be sure all the slots you need are available to you when you get the server. As for types of cards. As others have already said, PCI-X is not PCI-Express and they are not interchangeable. A PCI-Express card slot can accomodate any PCI-Express card with the same number of lanes or less. So an x8 slot (8 lanes) will support an x1, x2, x4, or x8 card, but not an x16 card. I believe the Digium PCI Express cards are only x1 (one lane) so they should fit in any PCI Express slot, but you should check with Digium's web site to be 100% sure the card you are buying is a x1 card. Unless you specifically buy a low-pofile card, a normal PCI or PCI Express Card will NOT fit in a low-profile slot. So assuming Digium's cards are full height, you only have 3 possible options. The 2 PCI Express full height slots, and the 1 PCI-X slot, assuming all those slots are open and will be available with the case you're using. I would NOT base my purchasing decision on that vague description given by the vendor that you have listed in your messages. I would contact the vendor and clarify the total number of slots, types of slots, whether they are open or not, and whether the case will support them all. Many vendors will use a motherboard with 3-5 slots on the board, in a 1u rackmount case that only supports 1 physical card. One final word of warning... don't try to stick too many cards in one box without double checking with someone who can tell you if it will handle that capacity or not. There were a lot of problems with early Digium 4-port T1 cards where you couldn't use more than 1 or 2 cards at a time because of the interrupts. The newer cards, especially PCI Express, may not have that problem anymore... but I would double check before proceeding with more than 2 cards in one box. Can anyone out there clarify (for me as well) whether you can put, say, 4 or 5 4-port T1 cards in a single box now and have it work ok? Assuming enough RAM and a fast enough CPU of course. -- Nick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update
Re: [asterisk-users] Had it with Dell Garbage - HP Question
On Mon, Mar 31, 2008 at 01:04:53AM -0400, Al Baker wrote: There are people who will support your Debian / Centos / whatever boxes. If it is OK to ask on a non-commercial list, do you have a list of reliable O/S support folks. By this I mean companies with a support staff, as opposed to a really bright and talented guy who does it between classes in school. Historically our projects were on big HP iron with HP-UX support from HP Now that you mention HP: http://hp.com/go/debian -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clustering Meetme over multiple boxes?
Hi Matt, As you said, is this will work like this? 1. Student A will login in a conference room no 7789 2. Student B will login in a conference room no 7789 3. Student C will login in a conference room no 7789 4. Instructor for student A,B and C will login in a conference room no. 6689 5. When the instructor click a button the 7789 conference and 6689 conference will be merged in a listen mode Am I correct? If I am wrong please correct me. Thanks On Tue, Mar 4, 2008 at 11:22 PM, Matt Florell [EMAIL PROTECTED] wrote: Hello, I have actually done this both ways, with many small conferences and few large conferences. The best example of both is the voice_lab feature that is included with VICIDIAL(although not very well documented). What this feature does is it has students log into individual meetme rooms and then have an instructor dial into their own meetme room. When the students are all logged in and the instructor is ready the instructor clicks a button to initiate calls from all of the student meetme rooms to the instructor meetme room where they are in listen-only mode and the instructor speaks english phrases which the students then repeat in their own conference. The reason this is set up this way is to allow for supervisor monitoring of individual students as well as recording of each student individually as they hear and repeat the phrases. This application is in use in telemarketing schools in the Philippines to help students learn to better speak American English. I have tested this to 120 channels going into the instructor meetme room across 6 servers. Hope that helps, MATT--- On 3/4/08, Tony Mountifield [EMAIL PROTECTED] wrote: Hi Matt, thanks for your reply. In article [EMAIL PROTECTED] , Matt Florell [EMAIL PROTECTED] wrote: Hello, We have done this using IAX trunks between Asterisk servers to connect a PRI line on server A with a meetme room on server B. We have had hundreds of participants in meetme rooms across a dozen Asterisk servers using this method. Not knowing your setup I'm not sure if this would work easily for you, but this is a somewhat-easy, scalable method for expanding meetme capacity. Is it correct to understand that in your setup, a given conference only ever exists on a single server (presumably the one used by the first caller), and that calls arriving on a different server are proxied individually to whichever server is hosting the requested conference? I can see that this would be quite easy, and would work well for lots of smallish conferences, but might be a bit heavy for a system running a small number of huge conferences. In the latter case, I want to look at having a local conference on each box and bridging the conferences together. This is the bit that gets complicated for making conf-wide decisions :-) Cheers Tony MATT--- On 3/4/08, Tony Mountifield [EMAIL PROTECTED] wrote: Has anyone here done any work on clustering Meetme conferences over multiple Asterisk boxes? The scenario I am thinking of is where there are two or more boxes connected to a set of PRIs that all answer to the same PSTN number, and where it's not possible to know in advance on which box a call would arrive. So it would be possible to have some calls on one box and some on another, that should all be conferenced together, by somehow linking matching Meetme conferences on both/all boxes. Particular complications I can envisage are the handling of marked users (A, w and x options), call recording (r option), and MeetmeAdmin operations such as mute all and unmute all. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation
Re: [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3
Darrick Hartman (lists) wrote: snip / I didn't find it too much trouble in a Via C700N system. But I wouldn't use one of the mainstream distros for the OS. They chew up system resources just trying to accommodate any hardware. The solution is to roll-your-own. See this series of articles on my blog... http://www.theopensourcerer.com/tag/asterisk/ The C7 supports full i686 features. The C3 is an older chip that is fully i586 and partially i686 compatible. If you have a distribution that is compiled with i586 optimizations, you won't have problems. Darrick Yeah, hi Darrick. I sort of realised after my post what the issue was with the C3. Although my point about not using a regular distro still stands. If you roll your own, all the features of the host hardware can be used - perhaps more importantly, *only* those features - and your kernel and compiler appropriately optimised. Regular distros are great (I use Ubuntu on my desktop pc) but they do have to try and be all things to all men and suck up cycles and ram like the latest Dyson ;-) But for a low power 24/7 server that I won't be playing much with; a custom build is just fine. Consider that I have running concurrently on my little C7 with 1G of RAM (That I have *down-clocked* to 1Ghz): * Asterisk, * Samba, * Java/Tomcat: *Cosmo Calendar Server *ConcursiveSuiteCRM *Alfresco *OpenBravo * PostgreSQL, * MySQL, * Exim, * Apache, * Vtiger, SugarCRM, A few Joomla! instances, * Subversion Server * sshd, * ntpd, And some other stuff that I can't recall. I don't think that's too bad ;-) When I get some more free time, I'm planning to build Untangle too. Cheers Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] applicationmap in features.conf Asterisk 1.2 is ignoring DIAL tT options
Hi, I found out that GoTo in applicationmap is not working. OK, LOCAL is working. but I expected that applicationmap is using the DIAL option tT. But it doesnt, it works without tT Option, so also callee can use internal functions if callee knows the code. Any workaround avaiable? best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No voice in one direction, SIP, call manager
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, I have a problem with Asterisk 1.4.x and the call manager. When I originate a call by the call manager or by a dot-call file only the calling party can hear the called party, not vice versa. When I dial the same number directly from the SIP phone (Cisco 7960) everything is OK. The same configuration worked with Asterisk 1.2 last week before switching to 1.4. There is a gateway (Patton) to the telecom operator communicating with the Asterisk via SIP. I've checked the SIP channels with sip show channels and it's the same when the call is originated by the phone or the call manager. Is there something special to be set to make call manager originated calls working again? Dot-call used: # calling party Channel: SIP/CiscoPhone MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: sip Priority: 1 # called party Extension: +420phonenumber Call manager commands used: Action: login Username: call_manager Secret: call_password Events: off Action: originate Channel: SIP/CiscoPhone Context: sip Priority: 1 Timeout: 3 CallerID: Martin Edlman 38 Exten: +420phonenumber - -- Ragards, Martin Edlman Fortech, spol. s r.o, Ropkova 51, 57001 Litomyšl Public GPG key: http://edas.visaci.cz/#gpgkeys -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Fedora - http://enigmail.mozdev.org iD8DBQFH8KRoqmMakYm+VJ8RAh/gAKCsObn2hmsvuMqkrsnp9RJoYRBKNQCfSJzv rEkCQaLp6e0GOknasykg3K0= =zaws -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3
Apart from the tutorial itself, what I wanted to point out was that the way asterisk, zaptel and libpri are to be built is different for each project, and this is sub-optimal; and that by building Asterisk as required, you get a linkage error. l. On Sat, 29 Mar 2008 12:03:59 +0100, Alan Lord [EMAIL PROTECTED] wrote: Lenz wrote: Hello list, after spending the best part of an afternoon trying to build Asterisk on an old EPIA VIA C3, I thought that writing a tutorial would make life easier for future compilers: http://astrecipes.net/index.php?n=356 I had never compiled Asterisk for a different architecture, and I'm pretty disappointed at how complex it is - building Zaptel, Libpri and Asterisk requires discovering three different procedures, and even passing the required architecture to the autoconfig module was not enough for a clean build - libpthread and libresolv would not link, so you have to add them manually. Aybody got an idea of who should be notified of this immediate problem, apart for the time-wasteful general compilation procedure? Thanks l. Hi there, I didn't find it too much trouble in a Via C700N system. But I wouldn't use one of the mainstream distros for the OS. They chew up system resources just trying to accommodate any hardware. The solution is to roll-your-own. See this series of articles on my blog... http://www.theopensourcerer.com/tag/asterisk/ Cheers Al -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The most efficient way to know SIP phones IP addresses ?
Hi,
Sometimes, you need to send requests to SIP phones either from Linux command
line or from Asterisk dialplan.
Which is the most efficient way to know a SIP phone IP address ?
Today, I think I would use :
asterisk -rx sip show peer 692 | grep Addr-IP | awk '{print $3}'
I'm wondering if anything more concise and efficient exists ?
Regards
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Broken calls during conversation
Good morning, we face a problem with Atserisk 1.4.18.1 and Zaptel 1.4.9.2: calls are frequently ended during conversation or voicemail are not registring the entire messages given by callers. What we have -and seem strange- is: Module Size Used by ztdummy10312 0 zaptel200264 11 ztdummy ;- THIS crc_ccitt 6784 1 zaptel We have various phones brand (Snom, Polycom, Tiger) and they all have this behavior. The server is a Dell SC440 without any telephone card, Debian ETCH. ztdummy is used for meetme and voicemail application only. Another strange think in logs [Mar 31 10:15:19] VERBOSE[8370] logger.c: RTCP SR transmission error, rtcp halted Thanks for any hint. -- Daniel TOOTAi Networks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3
On Mon, Mar 31, 2008 at 09:11:00AM +0100, Alan Lord wrote: Darrick Hartman (lists) wrote: snip / I didn't find it too much trouble in a Via C700N system. But I wouldn't use one of the mainstream distros for the OS. They chew up system resources just trying to accommodate any hardware. The solution is to roll-your-own. See this series of articles on my blog... http://www.theopensourcerer.com/tag/asterisk/ The C7 supports full i686 features. The C3 is an older chip that is fully i586 and partially i686 compatible. If you have a distribution that is compiled with i586 optimizations, you won't have problems. Darrick Yeah, hi Darrick. I sort of realised after my post what the issue was with the C3. Although my point about not using a regular distro still stands. If you roll your own, all the features of the host hardware can be used - perhaps more importantly, *only* those features - and your kernel and compiler appropriately optimised. Regular distros are great (I use Ubuntu on my desktop pc) but they do have to try and be all things to all men and suck up cycles and ram like the latest Dyson ;-) But for a low power 24/7 server that I won't be playing much with; a custom build is just fine. You can easily take a standard distro and remove all the services you don't really need. Consider that I have running concurrently on my little C7 with 1G of RAM (That I have *down-clocked* to 1Ghz): One major point: one of the cool advantages of the VIA CPUs is that it can be run fanless. In your setup you couple it with a large HD, and hence your system has moving parts. * Asterisk, * Samba, * Java/Tomcat: *Cosmo Calendar Server *ConcursiveSuiteCRM *Alfresco *OpenBravo * PostgreSQL, * MySQL, * Exim, * Apache, * Vtiger, SugarCRM, A few Joomla! instances, * Subversion Server * sshd, * ntpd, Now, why would you run all of those things on the same system? Asterisk needs a responsive system. It will not play along well if you add heavy-duty file serving to the system, as the system will spend too much time serving files (in kernel space). There's a limit to what you can optimize away with real-time kernel features. Oh, and practically all of those can be installed as standard Debian packages, without a need for such a lengthy installation manual. I bet that in 1/2 a year after you install it, you'll end up with a system with quite a few known security holes. But you'll never bother fixing them. Xandros is one such vendor that never bothered following up on security fixes. Hence the eeepc was an easy target for exploiters. Need I say that I will not advise anyone to use software from Xandros? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The most efficient way to know SIP phones IP addresses ?
2008/3/31, Simon Elliston Ball [EMAIL PROTECTED]:
You could try:
asterisk -rx database get SIP/Registry 101 | cut -f 2 -d ':'
Which is not much shorter, but probably more efficient
That's fine !
Too bad one cannot input more specific database queries such as database
get SIP/Registry/Addr-IP 101.
Simon Elliston Ball
[EMAIL PROTECTED]
http://www.simonellistonball.com/
On 31 Mar 2008, at 10:02, Olivier wrote:
Hi,
Sometimes, you need to send requests to SIP phones either from Linux
command line or from Asterisk dialplan.
Which is the most efficient way to know a SIP phone IP address ?
Today, I think I would use :
asterisk -rx sip show peer 692 | grep Addr-IP | awk '{print $3}'
I'm wondering if anything more concise and efficient exists ?
Regards
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 --
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 --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The most efficient way to know SIP phones IP addresses ?
You could try:
asterisk -rx database get SIP/Registry 101 | cut -f 2 -d ':'
Which is not much shorter, but probably more efficient
Simon Elliston Ball
[EMAIL PROTECTED]
http://www.simonellistonball.com/
On 31 Mar 2008, at 10:02, Olivier wrote:
Hi,
Sometimes, you need to send requests to SIP phones either from Linux
command line or from Asterisk dialplan.
Which is the most efficient way to know a SIP phone IP address ?
Today, I think I would use :
asterisk -rx sip show peer 692 | grep Addr-IP | awk '{print $3}'
I'm wondering if anything more concise and efficient exists ?
Regards
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The most efficient way to know SIP phones IP addresses ?
The asterisk database system is really more of a hash table than a
full database, so it's unlikely to happen. It's actually berkeley db
underneath.
Of course you could always create your own table on calls by using
something like Set(DB(ips/692)=${SIPPEER(692|ip)}) in the dialplan,
but it's probably a lot easier to just use the registry database, just
depends on how often you're going to be doing the lookups.
simon
Simon Elliston Ball
[EMAIL PROTECTED]
http://www.simonellistonball.com/
On 31 Mar 2008, at 10:56, Olivier wrote:
2008/3/31, Simon Elliston Ball [EMAIL PROTECTED]: You
could try:
asterisk -rx database get SIP/Registry 101 | cut -f 2 -d ':'
Which is not much shorter, but probably more efficient
That's fine !
Too bad one cannot input more specific database queries such as
database get SIP/Registry/Addr-IP 101.
Simon Elliston Ball
[EMAIL PROTECTED]
http://www.simonellistonball.com/
On 31 Mar 2008, at 10:02, Olivier wrote:
Hi,
Sometimes, you need to send requests to SIP phones either from Linux
command line or from Asterisk dialplan.
Which is the most efficient way to know a SIP phone IP address ?
Today, I think I would use :
asterisk -rx sip show peer 692 | grep Addr-IP | awk '{print
$3}'
I'm wondering if anything more concise and efficient exists ?
Regards
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
--
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 --
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 --
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 --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3
Tzafrir Cohen wrote: snip / You can easily take a standard distro and remove all the services you don't really need. Yes, but you can't easily change the way the apps are built or setup, e.g. compiler optimisations, use of initrd when not necessary, kernel bloat just to accommodate any host. Consider that I have running concurrently on my little C7 with 1G of RAM (That I have *down-clocked* to 1Ghz): One major point: one of the cool advantages of the VIA CPUs is that it can be run fanless. In your setup you couple it with a large HD, and hence your system has moving parts. No. Fanless is useful, but it is power consumption I am more interested in. A typical AMD/Intel desktop processor will now chew upwards of 100W. That's without the mobo and external components. Also, can you find 300Gb of solid state storage for about £30. ;-) * Asterisk, * Samba, * Java/Tomcat: *Cosmo Calendar Server *ConcursiveSuiteCRM *Alfresco *OpenBravo * PostgreSQL, * MySQL, * Exim, * Apache, * Vtiger, SugarCRM, A few Joomla! instances, * Subversion Server * sshd, * ntpd, Now, why would you run all of those things on the same system? Because it is for home use where there is low, but relatively constant load (my wife and I both have home offices). Some of the apps are for testing/evaluation so do not get used heavily and will not last very long. I just wanted to show what is possible with a sub £100 7Watt piece of hardware. Asterisk needs a responsive system. It will not play along well if you add heavy-duty file serving to the system, as the system will spend too much time serving files (in kernel space). I have not experienced *any* performance issues - so far. And uptime is permanent - until I reboot as I've installed a new kernel or something. Oh, and practically all of those can be installed as standard Debian packages, without a need for such a lengthy installation manual. Yes, they can. But I might not like where and how Debian (for example) decides how and where they install and setup those apps. They also do not use the most up-to-date versions and you are in their hands about when and how to upgrade. I bet that in 1/2 a year after you install it, you'll end up with a system with quite a few known security holes. But you'll never bother fixing them. How much ;-) Seriously, if I find or notice for a major bug/hole it is trivial to update. I keep all my installation procedures noted (or scripted) so it is pretty easy just to a CMMI with a new version. I wouldn't recommend this route for everyone. But being a control freak I know what and where *everything* is on my server... I don't have that level of control when using a mainstream distro. Sudo apt-get install is nice, but you are totally ignorant about what's going on under the hood... Hey that sounds just like Windows! lol. Cheers Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7971
Matthew Gibson wrote: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+md5secret then in your sip.conf [ext] ... ;secret=123 md5secret=MD5SECRET Hey Martin, thanks for your response... Still no dice: Quick questions... Where are the following coming from? Is this something you placed, something generated, if so by what, CCM, the phone itself. authenticationURLhttp://YOUR.PBX.IP.HERE/cisco/authenticate.php/authenticationURL directoryURLhttp://YOUR.PBX.IP.HERE/cisco/directory.php/directoryURL informationURLhttp://YOUR.PBX.IP.HERE/cisco/help.php/informationURL servicesURLhttp://YOUR.PBX.IP.HERE/cisco/services.php/servicesURL Second... loadInformationSIP70.8-3-3S/loadInformation I don't have SIP70.8-3-3s I have term71.default.loads which includes all images listed inside the file: # cat term71.default.loads # This file contains a list of archive image files that will be requested by the # RELEASE load version 8-3-3ES2 # jar70sip.8-3-3ES2.sbn cnu70.8-3-3ES2.sbn apps70.8-3-3ES2.sbn dsp70.8-3-3ES2.sbn cvm70sip.8-3-3ES2.sbn I tried posting both term71.default and cvm70sip.8-3-3ES2 loadInformationterm71.default/loadInformation loadInformationcvm70sip.8-3-3ES2/loadInformation For NAT, when I have it set to true on SEP.xml, phone registers and this is what happens in the course of 5 seconds: natReceivedProcessingtrue/natReceivedProcessing natEnabledtrue/natEnabled -- Registered SIP '9' at 64.xxx.xxx.xx port 49344 expires 3600 -- Saved useragent Cisco-CP7971G-GE/8.3.0 for peer 9 [Mar 31 07:17:02] NOTICE[2743]: chan_sip.c:15322 sip_poke_noanswer: Peer '9' is now UNREACHABLE! Last qualify: 0 On sip show peer: (truncated) ToHost : 64.xxx.xxx.xx Addr-IP : 64.xxx.xxx.xx Port 49344 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 123 SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (g729:20,ulaw:20) Auto-Framing: No Status : UNREACHABLE Useragent: Cisco-CP7971G-GE/8.3.0 Reg. Contact : sip:[EMAIL PROTECTED]:5060;transport=udp So I set contact to match: astterm*CLI -- Registered SIP '9' at 192.168.1.145 port 5060 expires 3600 -- Saved useragent Cisco-CP7971G-GE/8.3.0 for peer 9 [Mar 31 07:28:12] NOTICE[2743]: chan_sip.c:15322 sip_poke_noanswer: Peer '9' is now UNREACHABLE! Last qualify: 0 Now it matches but the same disconnect occurs: sip show peer truncated ToHost : 64.xxx.xxx.xx Addr-IP : 192.168.1.145 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 9 SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (g729:20,ulaw:20) Auto-Framing: No Status : UNREACHABLE Useragent: Cisco-CP7971G-GE/8.3.0 Reg. Contact : sip:[EMAIL PROTECTED]:5060;transport=udp About to kick this 7971 ;) Nope, no firewall, clean connection, and no NAT is being used period. Most appreciated response if any. I'm definitely scratching my head on this one. 7970's I have working fine, never had a problem getting those to work. I'm wondering if its the sip firmware version I'm using at this point. J. Oquendo SGFA #579 (FW+VPN v4.1) SGFE #574 (FW+VPN v4.1) wget -qO - www.infiltrated.net/sig|perl http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x3AC173DB smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ENUMLOOKUP question.
Hi All,
I am trying to establish a call between two users [EMAIL PROTECTED] and
[EMAIL PROTECTED] using ENUMLOOKUP. The following is my configuration.
In the DNS for domain1 I have the following entry.
5.4.3.2.1.domain1.com. IN NAPTR 100 10 u sip+E2U
!^(.*)$!sip:[EMAIL PROTECTED].
My extensions.conf for the extension 12345 looks like this:
exten = 12345,1,Set(foo=${ENUMLOOKUP(+${EXTEN}domain1.com)})
exten = 12345,n,NoOp(Enum lookup = ${foo})
exten = 12345,n,Dial(SIP/${foo})
exten = 12345,n,Hangup()
When I analyze the network trace I am able to see that Asterisk does a
NAPTR query and does get an answer for the query.
But on the console the following is displayed.
-- Executing [EMAIL PROTECTED]:1] Set(SIP/ua2-08bbef98, foo=) in new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(SIP/ua2-08bbef98, Enum
lookup = ) in new stack
-- Executing [EMAIL PROTECTED]:3] Dial(SIP/ua2-08bbef98, SIP/)
in new stack
It seems that Asterisk is unable to parse the response it has received
from the NAPTR query.
In some cases, I get the following logs at the console.
[Mar 31 16:58:48] WARNING[18211]: enum.c:246 parse_naptr: NAPTR Regex
match failed.
[Mar 31 16:58:48] WARNING[18211]: enum.c:362 enum_callback: Failed to
parse naptr :(
[Mar 31 16:58:48] WARNING[18211]: dns.c:226 dns_parse_answer: Failed to
parse result
[Mar 31 16:58:48] WARNING[18211]: dns.c:267 ast_search_dns: DNS Parse
error for 5.4.3.2.1.domain1.com
Am I doing something wrong here. I would appreciate if someone could
help me out and point me in the right direction.
Thanks Regards,
Aadil
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UK FXO hangup detection with a twist
Hi, The twist? We actually have far-end hangup detection working fine, and that seems to be where the problem lies for most people. Our problem seems to be with requesting a hangup from our end reliably. If we originate the call, we can hang it up. This suggests to me that the Sangoma A200D is sending the correct hangup signaling. This way round, it is 100% reliable. If we accept a call originated elsewhere, then we cannot hang it up. Only the call originator seems to be able to do that. The upshot is that if asterisk hangs-up a line, and then tries to re-use it for an outbound call before the remote has disconnected, we are simply re-connected to the original caller, and start to play DTMF at them! Has anyone experienced this before? Anyone found a solution? Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] transfer call
Hi there, i'm beginner in Asterisk.. I have situation: when a caller ,some SIP account on local network, dial outside number trough some Zap channel, asterisk have to automaticly transfer call to another local SIP channelIn other words, he have to change caller.. For example: I have SIP accounts 100 and 101.. WHen i call some outside number , PBX must to automatic switch the caller...like from some softphone , when i press transfer... Is it posible trough asterisk , extensions.conf etc... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK FXO hangup detection with a twist
If we accept a call originated elsewhere, then we cannot hang it up. Only the call originator seems to be able to do that. The upshot is that if asterisk hangs-up a line, and then tries to re-use it for an outbound call before the remote has disconnected, we are simply re-connected to the original caller, and start to play DTMF at them! Has anyone experienced this before? Anyone found a solution? Yes I have seen that with many analogue lines in the UK. This behaviour is somewhat 'by design' - it occurs even if you just use 2 plain old telephones (and no asterisk). I always forced it on-hook for (I think it was) 15 seconds before attempting a new call on the same line, but I always had spare lines to dial out on, so no real need to dig deeper into this ;-) Conrad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jingle with Asterisk + PSTN
Hi Ali, On Fri, Mar 28, 2008 at 5:31 PM, Ali Jawad [EMAIL PROTECTED] wrote: Hi All I am developing a client that uses libjingle to do xmpp stuff with ejabberd. I can also make audio calls between those clients. What I am trying to archive now is to send calls to pstn using jingle. I was told in the jingle-dev community that asterisk can do that. Asterisk speaks Jingle indeed. If you're using a libjingle based client, you'll have to set up a GoogleTalk connection to Asterisk through the chan_gtalk channel driver. Those good pointers will help : http://www.voip-info.org/wiki/view/Asterisk+Google+Talk http://taug.ca/node/43 Note : the chan_jingle channel driver implements the Jingle (not GoogleTalk related), so it won't work with libjingle even though the names sound close. Is there any way to send jingle audio calls to asterisk and will it understand them ? If yes..can I forward those calls to PSTN ? PSTN, SIP, H323, SCCP, ... or any protocol implemented in Asterisk! Philippe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: DND answered as on the phone instead of not available
SIP response 486 Busy Here is returned unless a divert contact is set up in the phone config. I did a search through the SIP 3.0 Admin Guide and didn't see any way of returning a different SIP response. http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/SoundPointIP_SoundStationIP_AdminGuide_SIP3_0_Eng_Rev_A.pdf Andreas van dem Helge wrote: What is your extensions.conf setup? that has alot to do with it (I strongly suggest you use macros.) What SIP NNN code does the phone return when DND? On Mon, Mar 17, 2008 at 2:00 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: I am using Polycom IP600 phone. If I call a phone which has DND (do not disturb) enabled, the message to the caller will be The person on extension ... is on the phone, please leave a message Is there a way to pick the person ... not available message instead? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to customize voicemail greeting
Dear friends, I am trying to configure Asterisk so that it play differnt set of voicemail greets for differnt extensions. I put my customized .wav files under the extension, but it still does not work. Asterisk still plays the default voice file. debian:/var/spool/asterisk/voicemail/default/2000# ls -al total 116 drwxr-xr-x 6 root root 4096 2008-04-01 06:03 . drwxr-xr-x 7 root root 4096 2008-03-29 01:40 .. -rw-r--r-- 1 root root 9438 2008-04-01 06:00 busy.wav -rw-r--r-- 1 root root 1815 2008-04-01 06:00 greet.wav drwxr-xr-x 2 root root 4096 2008-04-01 06:03 INBOX drwxr-xr-x 2 root root 4096 2008-03-26 09:09 Old drwxr-xr-x 2 root root 4096 2008-03-20 07:20 temp drwxr-xr-x 2 root root 4096 2008-04-01 06:03 tmp -rwxr-xr-x 1 root root 66924 2008-03-22 23:25 unavail.tmp.wav -rw-r--r-- 1 root root 2772 2008-04-01 06:02 unavail.wav Is there anything I missed out? By the way, can i use .gsm for my customized voice files? Thanks, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tests in VMWare (was: Re: asterisk-users Digest, Vol 44, Issue 104)
On Mon, 2008-03-31 at 03:04 -0500, [EMAIL PROTECTED] wrote: Date: Mon, 31 Mar 2008 07:55:08 +0300 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] Tests in VMWare To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Sun, Mar 30, 2008 at 08:50:10PM -0400, Ein Bielaczyc wrote: I'm just wondering if any one else has tried to successfully install Asterisk on Ubuntu inside VM. What version of Ubuntu? What version of Asterisk? They're not allowed to tell you: NOTICE: This E-mail (including attachments) is covered by the Electronic Communications Privacy Act, 18 U.S.C.2510-2521, is confidential and may be legally privileged. If you are not the intended recipient, you are hereby notified that any retention, dissemination, distribution or copying of this communication is strictly prohibited. Please reply to the sender that you have received the message in error, then delete it. Hell, I wasn't even allowed to tell you that they're not allowed to tell you. -- Tzafrir Cohen -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK FXO hangup detection with a twist
On 31/03/2008, Steve Davies [EMAIL PROTECTED] wrote: Hi, The twist? We actually have far-end hangup detection working fine, and that seems to be where the problem lies for most people. Our problem seems to be with requesting a hangup from our end reliably. If we originate the call, we can hang it up. This suggests to me that the Sangoma A200D is sending the correct hangup signaling. This way round, it is 100% reliable. If we accept a call originated elsewhere, then we cannot hang it up. Only the call originator seems to be able to do that. The upshot is that if asterisk hangs-up a line, and then tries to re-use it for an outbound call before the remote has disconnected, we are simply re-connected to the original caller, and start to play DTMF at them! Has anyone experienced this before? Anyone found a solution? People regularly use this feature to answer a call, then decide they need to run upstairs to speak etc, so they put the receiver down, go upstairs (or wherever) and pick up the handset to speak. It dates back many years and I should think is designed in to the system in the UK. Not sure on modern exchanges how long it would take for the line to clear. Mike Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jingle with Asterisk + PSTN
So should I register directly on the asterisk server or should I send the voice calls through ejabberd to asterisk ? On Mon, Mar 31, 2008 at 4:55 PM, Philippe Sultan [EMAIL PROTECTED] wrote: Hi Ali, On Fri, Mar 28, 2008 at 5:31 PM, Ali Jawad [EMAIL PROTECTED] wrote: Hi All I am developing a client that uses libjingle to do xmpp stuff with ejabberd. I can also make audio calls between those clients. What I am trying to archive now is to send calls to pstn using jingle. I was told in the jingle-dev community that asterisk can do that. Asterisk speaks Jingle indeed. If you're using a libjingle based client, you'll have to set up a GoogleTalk connection to Asterisk through the chan_gtalk channel driver. Those good pointers will help : http://www.voip-info.org/wiki/view/Asterisk+Google+Talk http://taug.ca/node/43 Note : the chan_jingle channel driver implements the Jingle (not GoogleTalk related), so it won't work with libjingle even though the names sound close. Is there any way to send jingle audio calls to asterisk and will it understand them ? If yes..can I forward those calls to PSTN ? PSTN, SIP, H323, SCCP, ... or any protocol implemented in Asterisk! Philippe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with VoiceMailMain
Dear all, I noticed a very strange problem. When I tried using VoiceMailMain to record my unavailable message, the file does not get created even though I can find the corresponding mssage from asterisk: -- SIP/2001-b6307d78 Playing 'beep' (language 'en') -- x=0, open writing: /var/spool/asterisk/voicemail/default/2000/unavail.tmp format: wav, 0x82828c8 -- User ended message by pressing # -- SIP/2001-b6307d78 Playing 'auth-thankyou' (language 'en') -- SIP/2001-b6307d78 Playing 'vm-review' (language 'en') -- Saving message as is -- SIP/2001-b6307d78 Playing 'vm-msgsaved' (language 'en') -- SIP/2001-b6307d78 Playing 'vm-options' (language 'en') Also, if I copies the 'unavil.wav' inside the /2000/ directory myself, it gets deleted somehow. How come this is happening? What I want to do is to be able to copies some standard voicemail for different extension, bu so far, it does not seem to work for me yet. Can anyone give me some suggestion? Thank you very much for your kind help. Thanks, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 44, Issue 104
All too common and largely undocumented. I had this same problem. Installing ztdummy changes Asterisk to use it for timing of playback, apparently. Removing ztdummy fixed the problem. To get it all to work, I had to upgrade to to at least kernel 2.6.23.11 (previous versions are either missing options are just broken.) Which previous versions have you tried? I'll also note that the OP needs to get Zaptel working under Xen, which is probably a different issue than your own. I've tried 2.6.8, 2.6.18-5, 2.6.19, 2.6.21.3 and perhaps more. These are the only ones I recall. I tuned in late and didn't see they wanted Xen support, but I figure others may find it helpful via google. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI-python script
Any new about this? Thanks On Thu, Mar 27, 2008 at 11:29 AM, equis software [EMAIL PROTECTED] wrote: I was trying to trap SIGHUP, but could be another signal because it didn't work. I'm doing this class MyScript(): def logsignal(self,signum, frame): self.putCDR() def run(self): signal.signal(signal.SIGHUP, self.logsignal) def putCDR(): put my cdr in my db. I was tryin trap other signals to test this and work well def run(self): signal.signal(signal.SIGALRM, self.logsignal) signal.alarm(3) Thanks a lot! On Wed, Mar 26, 2008 at 4:54 PM, Steve Edwards [EMAIL PROTECTED] wrote: On Wed, 26 Mar 2008, equis software wrote: Hi! I have some IVRs made in python. If the caller hangup before the end of the script I can´t register in my database the cdr. From your description, I'm not sure exactly what you are asking, but 1 of these should solve your problem. 1) Trap SIGHUP. 2) Use the h extension. 3) Use deadagi(). Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK FXO hangup detection with a twist
On 31/03/2008, Mike Dent [EMAIL PROTECTED] wrote: On 31/03/2008, Steve Davies [EMAIL PROTECTED] wrote: The twist? We actually have far-end hangup detection working fine, and that seems to be where the problem lies for most people. Our problem seems to be with requesting a hangup from our end reliably. If we originate the call, we can hang it up. This suggests to me that the Sangoma A200D is sending the correct hangup signaling. This way round, it is 100% reliable. If we accept a call originated elsewhere, then we cannot hang it up. Only the call originator seems to be able to do that. The upshot is that if asterisk hangs-up a line, and then tries to re-use it for an outbound call before the remote has disconnected, we are simply re-connected to the original caller, and start to play DTMF at them! Has anyone experienced this before? Anyone found a solution? People regularly use this feature to answer a call, then decide they need to run upstairs to speak etc, so they put the receiver down, go upstairs (or wherever) and pick up the handset to speak. It dates back many years and I should think is designed in to the system in the UK. Not sure on modern exchanges how long it would take for the line to clear. I do the same myself, but for PABX use, that feature must be fatal! The line clearing time is long... I waited a couple of minutes at least. Is it possible to turn it off (call BT and ask for a certain feature to be enabled/disabled) or to shorten the line-clearing time to zero? Or perhaps Asterisk is able to detect the line state or the dialtone and act correctly to avoid re-using an open channel. In fact, the obvious way to do this might be if Asterisk could set the channel state to hanging up and wait for the far end signal (loop disconnect) that the line has actually been disconnected. This is a bit of Zaptel that I've never looked at, so I have no clue if what I am suggesting is even slightly possible. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jingle with Asterisk + PSTN
On Mon, Mar 31, 2008 at 4:51 PM, Ali Jawad [EMAIL PROTECTED] wrote: So should I register directly on the asterisk server or should I send the voice calls through ejabberd to asterisk ? You can't register an XMPP client on Asterisk, because it's not an XMPP server. The required steps to establish a Jingle voice call to Asterisk are : - register Asterisk on an XMPP server (ex. talk.google.com, but it can be any XMPP server) ; - register your XMPP (Jingle capable) client on any XMPP server ; - make sure both these XMPP clients are buddies ; - place your call directly to Asterisk. In order to have your Jingle call relayed to the PSTN, you must edit your Asterisk dialplan accordingly. Philippe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK FXO hangup detection with a twist
On Mon, 2008-03-31 at 16:25 +0100, Steve Davies wrote: On 31/03/2008, Mike Dent [EMAIL PROTECTED] wrote: On 31/03/2008, Steve Davies [EMAIL PROTECTED] wrote: The twist? We actually have far-end hangup detection working fine, and that seems to be where the problem lies for most people. Our problem seems to be with requesting a hangup from our end reliably. If we originate the call, we can hang it up. This suggests to me that the Sangoma A200D is sending the correct hangup signaling. This way round, it is 100% reliable. If we accept a call originated elsewhere, then we cannot hang it up. Only the call originator seems to be able to do that. The upshot is that if asterisk hangs-up a line, and then tries to re-use it for an outbound call before the remote has disconnected, we are simply re-connected to the original caller, and start to play DTMF at them! Has anyone experienced this before? Anyone found a solution? People regularly use this feature to answer a call, then decide they need to run upstairs to speak etc, so they put the receiver down, go upstairs (or wherever) and pick up the handset to speak. It dates back many years and I should think is designed in to the system in the UK. Not sure on modern exchanges how long it would take for the line to clear. I do the same myself, but for PABX use, that feature must be fatal! The line clearing time is long... I waited a couple of minutes at least. Is it possible to turn it off (call BT and ask for a certain feature to be enabled/disabled) or to shorten the line-clearing time to zero? Or perhaps Asterisk is able to detect the line state or the dialtone and act correctly to avoid re-using an open channel. In fact, the obvious way to do this might be if Asterisk could set the channel state to hanging up and wait for the far end signal (loop disconnect) that the line has actually been disconnected. This is a bit of Zaptel that I've never looked at, so I have no clue if what I am suggesting is even slightly possible. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You should ask for ground start signaling. This will resolve your issues. Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Simple Question
Does AMD (answering machine detect) need ztdummy or some other timer to function properly? -- Drew Miller Iowa Democratic Party Information Technology Director Office: (515) 974-1682 Cell: (515) 451-4509 AIM: ItsDrewMiller MSN: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK FXO hangup detection with a twist
On 31/03/2008, David Boyd [EMAIL PROTECTED] wrote: You should ask for ground start signaling. This will resolve your issues. Could you point me at some reference material for how this differs from KS, and what compatibility issues this might cause with other equipment? Has anyone tried this in the UK? Would BT even understand the request for ground-start signalling? I wonder if it is even possible with telco's other than BT in the UK... I can just imagine calling Virgin Media and asking for the line to be set to ground-start signalling... :) Any feedback welcomed. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 44, Issue 104
On Mon, Mar 31, 2008 at 11:08:09AM -0400, Norman Franke wrote: All too common and largely undocumented. I had this same problem. Installing ztdummy changes Asterisk to use it for timing of playback, apparently. Removing ztdummy fixed the problem. To get it all to work, I had to upgrade to to at least kernel 2.6.23.11 (previous versions are either missing options are just broken.) Which previous versions have you tried? I'll also note that the OP needs to get Zaptel working under Xen, which is probably a different issue than your own. I've tried 2.6.8, 2.6.18-5, 2.6.19, 2.6.21.3 and perhaps more. These are the only ones I recall. It is supposed to work for kernels = 2.6.22 . http://zaptel.tzafrir.org.il/#_kernel_configuration (That's the README file from the Zaptel tarball). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Question
No It does not require. Regards, Sanjay. - Original Message - From: Drew Miller [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, March 31, 2008 9:17:19 PM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Simple Question Does AMD (answering machine detect) need ztdummy or some other timer to function properly? -- Drew Miller Iowa Democratic Party Information Technology Director Office: (515) 974-1682 Cell: (515) 451-4509 AIM: ItsDrewMiller MSN: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP proxy screwing up peer addresses.
Hello, I am trying to test-call my own asterisk server to see if I can receive SIP calls properly. I use a softphone to call the SIP address, and because twinkle doesn't support SRV records, I go via a proxy. When the call comes in, asterisk says: handle_request_invite: Sending fake auth rejection for user martin f. krafft sip:[EMAIL PROTECTED];tag=fipzt and SIP debugging then prints: OPTIONS sip:sip05.insphone.ch SIP/2.0 Via: SIP/2.0/UDP 84.75.148.xxx:5060;branch=z9hG4bK71785803;rport From: asterisk sip:[EMAIL PROTECTED];tag=as05fc20f4 I am not calling as username asterisk, but I think this is the proxy substituting its name for mine. Why? Is it broken? Am I misunderstanding something? How can I fix/prevent his? Also, the IP is that of my asterisk server, the one which receives the call. It goes on: To: sip:sip05.insphone.ch I made the call to [EMAIL PROTECTED], not the unqualified sip05.insphone.ch (which is the proxy hostname). Contact: sip:[EMAIL PROTECTED] Again this is not the contact address. I see this often, that with SIP, the local part of a peer address is just changes, and I think it's similar to email header rewriting. However, header rewriting is rare and somewhat frowned upon, so why is it so commonplace with SIP? -- martin | http://madduck.net/ | http://two.sentenc.es/ die zeit für kleine politik ist vorbei. schon das nächste jahrhundert bringt den kampf um die erdherrschaft. - friedrich nietzsche spamtraps: [EMAIL PROTECTED] digital_signature_gpg.asc Description: Digital signature (see http://martin-krafft.net/gpg/) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3
On Mon, 31 Mar 2008, Alan Lord wrote: Also, can you find 300Gb of solid state storage for about £30. ;-) Where?? Gordon___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK FXO hangup detection with a twist
On Mon, 31 Mar 2008, Steve Davies wrote: On 31/03/2008, David Boyd [EMAIL PROTECTED] wrote: You should ask for ground start signaling. This will resolve your issues. Could you point me at some reference material for how this differs from KS, and what compatibility issues this might cause with other equipment? Has anyone tried this in the UK? Would BT even understand the request for ground-start signalling? I wonder if it is even possible with telco's other than BT in the UK... I can just imagine calling Virgin Media and asking for the line to be set to ground-start signalling... :) Any feedback welcomed. AIUI: You request BT to set the Disconnect Clear Time on the circuit to whatever your PBX requires - 800mS is a usual figure... Taken from another list, or even archives of this one, or somewhere else A google of this reveals: http://www.voipuser.org/forum_topic_7470.html and others... Good luck! Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsterPas ObjectPascal Based FastAGI Server goes Open Source
Announcement: We are pleased to announce that we have released AsterPas FastAGI ObjectPascal Script Server for Asterisk PBX under license. What is AsterPas? AsterPas is a FastAGI server which allows real-time scripting of Asterisk PBX call flow using ObjectPascal based scripting. AsterPas includes many built objects available from scripts such as Cepstral TTS Engine class, database access class for FirebirdSQL, MySQL 4.1-5.0 and SQLite3 databases, Call File generation and more. Because AsterPas is a TCP socket server, it can be installed on the local Asterisk PBX or on a different computer to offload processing. AsterPas is written in 100% ObjectPascal using the Lazarus IDE for the FreePascal Compiler. Yes, ObjectPascal, It's not your mom and dad's pascal ;) AsterPas has been compiled and tested on: CentOS 4/Linux Windows 2000/XP Windows Server 2000 More information on AsterPas can be found on its web page at: http://www.datatrakpos.com/pos/datatalk/asterpas.aspx Source Code can be downloaded via svn or viewed from here: http://leebo.dreamhosters.com/asterpas/ Note: AsterPas relies on several 3rd party libraries: - Synapse (http://synapse.ararat.cz) (Open Source) - Pascal Data Object (http://pdo.sourceforge.net/) (Open Source) - sqlite3ds (included with lazarus/Freepascal) (Open Source) - TPasAGI (included with AsterPas sources, written by me ;) ) (Open Source) - RemObjects Pascal Script (http://www.remobjects.com) (Free with Source) -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards
Dear list, this is getting ridiculous - I would read the specs and compare From the link below it says: * Two Available PCI Express x8 Slots * Two Available PCI Express x8 Low Profile Slots * One Available 64-bit/100MHz PCI-X slot The list has already answered what goes in what slot so I won't repeat that. It is a 1U unit and the above taken directly from the specs at : http://www.penguincomputing.com/index.php?option=com_contenttask=viewid=386Itemid=598 Oh, will repeat some... Digium's TE420 PCI Express card provides termination of up to 120 channels of voice or data across four E1, T1, or J1 interfaces in a PCIe x1 form factor. Selectable on a per-port or per-card basis, the TE420 allows E1 and T1 circuits to be mixed with full channel synchronization. Supporting PCIe x1, the TE420 may be used in any available PCIe 1.0 compliant slot - 1x, x4, x8, and x16 without consider So, you can fit 2 of those in the: Two Available PCI Express x8 Slots Forget about the 2 low profile slots when it comes to Digiumand AFAIK everyone else too... For the PCI-X (if penguin computing can confirm that it is at least half length) slot you may use for example OpenVox D410P, that will work. When it comes to the Digium cards I can't say but Digium sales should be to tell you which one (5.5V or 3.3V or none of them). I am sure their cards do work, as would several other manufacturers cards (don't have the experience with them yet). E.g. PIKA says: Slot Requirements: Standard x86 PCI Half-size Slot (Compatible with PCI-X slot) for their PIKA for Asterisk T1/E1 PCI card. So that should work. I would say - kind of forget about the 2 half height PCI Express slots as AFAIK no one is providing half-height E1/T1 cards. //Peter - short and always getting flamed Al Baker wrote: Thank you for all your time om your most detailed response. It is extremely helpful. The vendor's web page is http://www.penguincomputing.com/index.php?option=com_contentid=170Itemid=209task=viewsysid=10007609 *PCI EXPANSION SLOTS* Number of Slots 5 Slot Speed PCI Express: two x8 slots, two x8 low profile slots; PCI-X: 64-bit/100MHz or if that doesn't display *PCI EXPANSION SLOTS* Number of Slots 5 Slot Speed PCI Express: two x8 slots, two x8 low profile slots; PCI-X: 64-bit/100MHz or PCI EXPANSION SLOTS Number of Slots 5 Slot Speed PCI Express: two x8 slots, two x8 low profile slots; PCI-X: 64-bit/100MHz Nick Seraphin wrote: On Sat, 29 Mar 2008, Al Baker wrote: Detailed specs for the types of PCI slots on the system were posted each and every time I posted int the line Actually, your description wasn't 100% clear at all. _PCI Express_*: _two x8 slots*_, _two x8 low profile slots*_; *_PCI-X: 64-bit/100MHz_* 1) This description seems to IMPLY that there are 5 slots total. Do you know if this is in fact correct? It implies there are 2 PCI-E x8 slots, 2 PCI-E x8 low profile slots, and 1 PCI-X slot. I wouldn't rely on that however without talking to the vendor. 2) The first PCI Express: heading would normally imply that the slots listed afterwards are ALL PCI Express slots, however PCI-X is not PCI Express, so the vendor's description is confusing and misleading. 3) All these damn *'s you keep inserting, are those all done by you, or are some of them from the web page description? Most of the time when something has a * by it that means it's conditional on a footnote that appears at the bottom of the section or page. Are there footnotes we need to know about to clarify this? 4) Is this a rackmount server or a tower case? If rackmount, is it a 1u server or a 2u server or a 4u server? Just because the motherboard has 5 slots doesn't mean the case it is installed in will support 5 cards. A 1u case rarely supports more than 1 or 2 cards, and always requires a riser card. A 2u server rarely supports more than 2 cards unless they are low-profile. A 4u server might allow 5 cards, IF the case is designed with 5 slot openings in the back. 5) Are all the card slots open and available to you at time of shipping? Many options a customer orders with a server, such as a RAID controller or additional network ports will fill one or more of the available slots. You need to be sure all the slots you need are available to you when you get the server. As for types of cards. As others have already said, PCI-X is not PCI-Express and they are not interchangeable. A PCI-Express card slot can accomodate any PCI-Express card with the same number of lanes or less. So an x8 slot (8 lanes) will support an x1, x2, x4, or x8 card, but not an x16 card. I believe the Digium PCI Express cards are only x1 (one lane) so they should fit in any PCI Express slot, but you should check with Digium's web site to be 100% sure the card you are buying is a x1 card. Unless you specifically buy a low-pofile card, a
Re: [asterisk-users] AsterPas ObjectPascal Based FastAGI Server goes Open Source
Lee Jenkins wrote: Announcement: We are pleased to announce that we have released AsterPas FastAGI ObjectPascal Script Server for Asterisk PBX under license. Oops. That should be LGPL license ;) -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI cards, Digium vs. Sangoma
On Sun, Aug 26, 2007 at 9:49 AM, Doug Lytle [EMAIL PROTECTED] wrote: I have, on many occasions, had kernel panics when trying to shut down wanrouter. I don't have this 'fear' with Digium cards. I never have had those issues if you don't execute zaptel init.d script, because it tries to unmod all zaptel dependant modules including wanrouter which need to be unmoded with wanrouter script. (A matter of order in the unload process). Perhaps this tip helps you avoiding that fear. This makes an auto-reboot after a kernel panic occurs. /etc/sysctl.conf: kernel.panic = 1 OR echo 1 /proc/sys/kernel/panic OR pass panic=1 as a kernel parameter in your grub.conf/lilo.conf -- Octavio H. Ruiz Cervera ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to give user a prompt before connecting the call
Hello, Is it possible to request for the premission from the called party through a prompt before routing the call? For instance, before actually connecting two parties through the use of DIAL command in the dialplan, I want to let Asterisk to automatically ask for the called party to decide whether he/she would like to be connected. ( ex. Press 1 to connect and 2 to hangup). Can this function be done? If so, how to do it? Thank you . Pete Dao ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI cards, Digium vs. Sangoma
Octavio Ruiz wrote: On Sun, Aug 26, 2007 at 9:49 AM, Doug Lytle [EMAIL PROTECTED] wrote: /etc/sysctl.conf: kernel.panic = 1 OR echo 1 /proc/sys/kernel/panic OR pass panic=1 as a kernel parameter in your grub.conf/lilo.conf Now that is nice to know, thanks! 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Control of RTP open ports
Dear all, I have an Asterisk 1.4.13 with SIP protocol and several voip clients (Twinkle, X-Lite and SJPhone). Every call among voip clients pass through the Asterisk server, so there isn't any voip packet client-to-client. Can Asterisk control the RTP open ports the voip clients use ??? Or the RTP open ports depend on the voip clients ??? Special thanks Alejandro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR Timestamps (cdr-custom)
We have just implemented cdr-custom. Works fine minus the timestamps that appear in the CDR. The system's timezone is GMT. In the configuration usegmtime=yes is set. However, all of the CDRs in the Custom CDR comes as GMT-5. Another oddity is that the standard cdr/Master.csv is using GMT. Please advise. Thanks, kw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI cards, Digium vs. Sangoma
On Mon, Mar 31, 2008 at 11:54:14AM -0600, Octavio Ruiz wrote: On Sun, Aug 26, 2007 at 9:49 AM, Doug Lytle [EMAIL PROTECTED] wrote: I have, on many occasions, had kernel panics when trying to shut down wanrouter. I don't have this 'fear' with Digium cards. I never have had those issues if you don't execute zaptel init.d script, because it tries to unmod all zaptel dependant modules including wanrouter which need to be unmoded with wanrouter script. (A matter of order in the unload process). Perhaps this tip helps you avoiding that fear. This makes an auto-reboot after a kernel panic occurs. Removing all the modules on shutdown is not really needed. Many people seem to prefer it for some strange reason. In addition, what you describe here means a specific init.d script's stop action needs to run after that of another. There are a number of ways to order that. In most distributions it is done by explicit ordering. In some it is done by dependencies. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to give user a prompt before connecting thecall
Yes it is. I'm remote at the moment so I can't send you the code but google for mobile remote receiver and you'll find what you are looking for. Lots of people do it so they don't have calls to cell phones picked up by voicemail. Cheers dean -Original Message- From: Pete Kay [EMAIL PROTECTED] Sent: Monday, March 31, 2008 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] How to give user a prompt before connecting thecall Hello, Is it possible to request for the premission from the called party through a prompt before routing the call? For instance, before actually connecting two parties through the use of DIAL command in the dialplan, I want to let Asterisk to automatically ask for the called party to decide whether he/she would like to be connected. ( ex. Press 1 to connect and 2 to hangup). Can this function be done? If so, how to do it? Thank you . Pete Dao ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3
Gordon Henderson wrote: On Mon, 31 Mar 2008, Alan Lord wrote: Also, can you find 300Gb of solid state storage for about £30. ;-) Where?? Gordon Sorry my bad. It was a question... Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI cards, Digium vs. Sangoma
On Mon, Mar 31, 2008 at 12:58 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: There are a number of ways to order that. In most distributions it is done by explicit ordering. In some it is done by dependencies. Gentoo is one of them, where if you run directly from CLI /etc/init.d/zaptel stop and wanrouter script have the zaptel dependency first wanrouter is going to be stopped as a direct dependency of zaptel. http://www.gentoo.org/doc/en/handbook/handbook-x86.xml?part=2chap=4 Have you any other in mind? -- Octavio H. Ruiz Cervera ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gentilini, Paul is out of the office.
I will be out of the office starting Mon 03/31/2008 and will not return until Tue 04/01/2008. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI cards, Digium vs. Sangoma
Tzafrir Cohen wrote: On Mon, Mar 31, 2008 at 11:54:14AM -0600, Octavio Ruiz wrote: On Sun, Aug 26, 2007 at 9:49 AM, Doug Lytle [EMAIL PROTECTED] wrote: I have, on many occasions, had kernel panics when trying to shut down wanrouter. I don't have this 'fear' with Digium cards. I never have had those issues if you don't execute zaptel init.d script, because it tries to unmod all zaptel dependant modules including wanrouter which need to be unmoded with wanrouter script. (A matter of order in the unload process). Perhaps this tip helps you avoiding that fear. This makes an auto-reboot after a kernel panic occurs. Removing all the modules on shutdown is not really needed. Many people seem to prefer it for some strange reason. In addition, what you describe here means a specific init.d script's stop action needs to run after that of another. There are a number of ways to order that. In most distributions it is done by explicit ordering. In some it is done by dependencies. Right, this Kernel panic issue was a surprise to me as well. I even complained to Sangoma. In order to avoid it at customer sites, we edit the /etc/init.d/zaptel script and in the stop section we put in 'service wanrouter stop' before anything gets unloaded. Andres http://www.neuroredes.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7965 SIP Firmware
I have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S). Does anyone have a valid XMLDefault.cnf.xml they could share? I have tried the version at voip-infoinfo.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIPview_comment_id=14768#Troubleshootingfor the 7941/7961 but unfortunately /var/log/messages shows in.tftp stops sending after XMLDefault.cnf.xml (see below), so i'm assuming the 7965 doesn't like my XMLDefault.cnf.xml. Mar 31 21:34:47 fsvr dhcpd: DHCPACK on 192.168.10.174 to 00:1f:6c:61:1f:72 via eth0 Mar 31 21:34:47 fsvr in.tftpd[3577]: RRQ from 192.168.10.174 filename CTLSEP001F6C611F72.tlv Mar 31 21:34:47 fsvr in.tftpd[3577]: sending NAK (1, File not found) to 192.168.10.174 Mar 31 21:34:47 fsvr in.tftpd[3578]: RRQ from 192.168.10.174 filename SEP001F6C611F72.cnf.xml Mar 31 21:34:47 fsvr in.tftpd[3578]: sending NAK (1, File not found) to 192.168.10.174 Mar 31 21:34:47 fsvr in.tftpd[3579]: RRQ from 192.168.10.174 filename XMLDefault.cnf.xml Thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Tutorial: Asterisk on EPIA VIA C3
On Mon, 31 Mar 2008, Alan Lord wrote: Gordon Henderson wrote: On Mon, 31 Mar 2008, Alan Lord wrote: Also, can you find 300Gb of solid state storage for about £30. ;-) Where?? Gordon Sorry my bad. It was a question... Ah, Doh... I mis-read it all.. Curse my dyslexia! However I've moved my own workstations and dev. servers over to running Debian in a 4GB device with /home mounted on a similar spec. server as yours (1GHz Via C3 - same platform as my PBXs) which also boots off flash with a pair of low-power WD SATA drives in it ... Seems to work just fine! Gordon___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7965 SIP Firmware
Razza wrote: I have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S). Does anyone have a valid XMLDefault.cnf.xml they could share? I have tried the version at voip-info info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIPview_comment_id=14768#Troubleshooting for the 7941/7961 but unfortunately /var/log/messages shows in.tftp stops sending after XMLDefault.cnf.xml (see below), so i'm assuming the 7965 doesn't like my XMLDefault.cnf.xml. YMMV Change to reflect your firmware (e.g. P003-07-4-xx) .. Line 20 and 21 are one line... Default callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort mgcpPorts listen2427/listen keepAlive2428/keepAlive /mgcpPorts /ports processNodeName/processNodeName /callManager /member /members /callManagerGroup loadInformation8 model=IP Phone 7940P003-07-4-00/loadInformation8 loadInformation7 model=IP Phone 7960P003-07-4-00/loadInformation7 loadInformation6 model=IP Phone 7970term71.default/loadInformation6 authenticationURL/authenticationURL directoryURL/directoryURL idleURL/idleURL informationURL/informationURL messagesURL/messagesURL servicesURL/servicesURL /Default -- J. Oquendo SGFA #579 (FW+VPN v4.1) SGFE #574 (FW+VPN v4.1) wget -qO - www.infiltrated.net/sig|perl http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x3AC173DB smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need some input for Quad T1 and channel banks.
I'm looking to install a system with 80 FXS analog phones. At this time the only cost effective solution is using a 4 port T1 card and addit 600 channel bank. Has anyone tried this solution? any good documents beside http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check as far as i know, addit 600 T1 interface is not PRI (please correct me if i'm wrong) its CAS robbed bit, will that work with new Digium quad T1 like TE410P ?( I prefer to use Digium if possible) The system is connected to the Telco through SIP trunk so all we have in terms of analog is local loop, Do we need to have echo cancel in this scenario ? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7965 SIP Firmware
On 31/03/2008, J. Oquendo [EMAIL PROTECTED] wrote: YMMV Change to reflect your firmware (e.g. P003-07-4-xx) 8 SNIP 8 I removed the following lines: loadInformation8 model=IP Phone 7940P003-07-4-00/loadInformation8 loadInformation7 model=IP Phone 7960P003-07-4-00/loadInformation7 And tried both of these: loadInformation6 model=IP Phone 7965term65.default/loadInformation6 and loadInformation6 model=IP Phone 7965SIP45.8-3-4SR1S/loadInformation6 But again I get no further than /var/log/messages showing in.tftp stops sending after XMLDefault.cnf.xml Any suggestions? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7965 SIP Firmware
On Mon, 2008-03-31 at 23:07 +0100, Razza wrote: On 31/03/2008, J. Oquendo [EMAIL PROTECTED] wrote: YMMV Change to reflect your firmware (e.g. P003-07-4-xx) 8 SNIP 8 I removed the following lines: loadInformation8 model=IP Phone 7940P003-07-4-00/loadInformation8 loadInformation7 model=IP Phone 7960P003-07-4-00/loadInformation7 And tried both of these: loadInformation6 model=IP Phone 7965term65.default/loadInformation6 and loadInformation6 model=IP Phone 7965SIP45.8-3-4SR1S/loadInformation6 But again I get no further than /var/log/messages showing in.tftp stops sending after XMLDefault.cnf.xml Any suggestions? For a 7965, you might try loadinformation to be 335.. I have had to match up CCM tk.prod values to match on newer phones in the past to be what cisco uses in their internal database before I could get them to work. Although, leaving those lines out completely will work as well assuming they already have the SIP firmware loaded.. -Greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some input for Quad T1 and channel banks.
Al lists wrote: I'm looking to install a system with 80 FXS analog phones. Each channel bank can handle 48 analog channels, 2 PRIs per box. as far as i know, addit 600 T1 interface is not PRI (please correct me if i'm wrong) its CAS robbed bit, will that work with new Digium quad T1 like TE410P ?( I prefer to use Digium if possible) This is incorrect. I've got ours setup at a PRI. esf, b8zs. The system is connected to the Telco through SIP trunk so all we have in terms of analog is local loop, Do we need to have echo cancel in this scenario ? I'd say anything with analog will need EC. If you can afford it on a quad card, I would. 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some input for Quad T1 and channel banks.
Doug Lytle wrote on Monday, March 31, 2008 5:40 PM Al lists wrote: I'm looking to install a system with 80 FXS analog phones. Each channel bank can handle 48 analog channels, 2 PRIs per box. as far as i know, addit 600 T1 interface is not PRI (please correct me if i'm wrong) its CAS robbed bit, will that work with new Digium quad T1 like TE410P ?( I prefer to use Digium if possible) This is incorrect. I've got ours setup at a PRI. esf, b8zs. This does not sound right. If it is 2 PRIs then it should be 46 channels (or 47 channels if sharing a D channel). A quick google search indicates that these channel banks can deal with PRI in a drop and insert mode only, not for termination. (I use Adtran channel banks which are not PRI so I may be confused here). Don Pobanz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unicall + incomplete DNIS on international calls
Hello everybody, i'm from Mexico, at the time i´m working on a production server with asterisk 1.2.25 + spandsp-0.0.4 + libmfcr2-0.0.3+libsupertone-0.0.2+libunicall-0.0.3 and zaptel-1.2.22. I installed this version of astunicall that i downloaded from http://www.moythreads.com/astunicall/ Everything works fine, i'm able to make outgoing calls and recive incoming calls with all ANI and DNIS digits, except for International incoming call. My phone provider(Telmex) gives me 10 digits of ANI and 4 digits of DNIS, that i´ve configured on my unicall.conf. My main issue becomes when i recive an internationall incoming call, there is no ANI, appears with only one digit instead four, and that digit it's always a number 1( i attach unicall log). I already talked with my phone provider about this issue, and, as they told me, all DNIS and ANI of international incoming calls are just bypassed by them directly to my server. They mentioned something about timers that may avoid my server to recive all values (DNIS and ANI), but i'm not quite sure about this. On my file unicall.conf i added some timers that moises commented on his forum. Any clue what would be the reason of my issue ? Here are my files: -- unicall.conf -- [channels] language=en context=from-pstn usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 relaxdtmf=no rxgain=0 txgain=0 group=1 callgroup=0 pickupgroup=0 immediate=no callerid=asreceived amaflags=default musiconhold=default protocolclass=mfcr2 protocolvariant=mx,10,4,7,t1=15000,t2=24000,t3=15000,max-seize-wait-ack=2000 channel=1-10 loglevel=255 -- zapata.conf --- [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no --- zaptel.conf --- loadzone=us defaultzone=us #Sangoma A101 port 1 [slot:0 bus:10 span:1] wanpipe1 span=1,1,0,cas,hdb3 cas=1-10:1101 dchan=16 - DEBUG UNICALL --- Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 - 0001 [1/IDLE/Idle /Idle ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 Detected Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 Creating a new call with CRN 32769 Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 1101 - [2/DETECTED/Seize ack /Seize ack] Mar 31 13:10:35 NOTICE[14902] chan_unicall.c: Unicall/8 event Detected Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 - 1 on [2/DETECTED/Seize ack /Seize ack] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 6 on - [2/DETECTED/Group C /Category req ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 - 1 off [2/DETECTED/Group C /Category req ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 6 off - [2/DETECTED/Group C /Category req ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 - 2 on [2/DETECTED/Group C /Category req ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 1 on - [2/DETECTED/Group C /ANI request ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 - 2 off [2/DETECTED/Group C /ANI request ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 1 off - [2/DETECTED/Group C /ANI request ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 - F on [2/DETECTED/Group C /ANI request ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 5 on - [2/DETECTED/Group A /DNIS request ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 - F off [2/DETECTED/Group A /DNIS request ] Mar 31 13:10:35 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 5 off - [2/DETECTED/Group A /DNIS request ] Mar 31 13:10:50 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 3 on - [2/DETECTED/Group B /Go to grp II ] Mar 31 13:10:50 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 3 off - [2/DETECTED/Group B /Go to grp II ] Mar 31 13:10:50 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 - 2 on [2/DETECTED/Group B /Go to grp II ] Mar 31 13:10:50 NOTICE[14902] chan_unicall.c: Unicall/8 event Offered Mar 31 13:10:50 NOTICE[14902] chan_unicall.c: CRN 32769 - Offered on channel 0 (ANI: , DNIS: 1, Cat: 1) -- The values of ANI and DNIS are incorrect Mar 31 13:10:50 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 Call control(5) Mar 31 13:10:50 DEBUG[14902] chan_unicall.c: MFC/R2 UniCall/8 Accept call Mar 31 13:10:50 DEBUG[14902] chan_unicall.c:
Re: [asterisk-users] Need some input for Quad T1 and channel banks.
On Mon, Mar 31, 2008 at 6:01 PM, Al lists [EMAIL PROTECTED] wrote: I'm looking to install a system with 80 FXS analog phones. At this time the only cost effective solution is using a 4 port T1 card and addit 600 channel bank. Has anyone tried this solution? any good documents beside http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check as far as i know, addit 600 T1 interface is not PRI (please correct me if i'm wrong) its CAS robbed bit, will that work with new Digium quad T1 like TE410P ?( I prefer to use Digium if possible) The system is connected to the Telco through SIP trunk so all we have in terms of analog is local loop, Do we need to have echo cancel in this scenario ? Thanks! I am not sure of your budget but I would go with a SIP to FXS gateway such as the Quintum Tenor AX. If you consider the amount you would be paying for the T1 card and the Adits, it may not be that big of a difference and certainly more simple. Disclaimer, I have not done a price comparison or done the channel bank solution on that scale (only 24 ports for me) but the SIP-FXS is so much better in my opinion. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to give user a prompt before connecting thecall
It can be done via the 'visit a macro' part of the dial command... If anyone would like, i can post a code sample. PaulH On Mon, 2008-03-31 at 15:33 -0400, Dean Collins wrote: Yes it is. I'm remote at the moment so I can't send you the code but google for mobile remote receiver and you'll find what you are looking for. Lots of people do it so they don't have calls to cell phones picked up by voicemail. Cheers dean -Original Message- From: Pete Kay [EMAIL PROTECTED] Sent: Monday, March 31, 2008 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] How to give user a prompt before connecting thecall Hello, Is it possible to request for the premission from the called party through a prompt before routing the call? For instance, before actually connecting two parties through the use of DIAL command in the dialplan, I want to let Asterisk to automatically ask for the called party to decide whether he/she would like to be connected. ( ex. Press 1 to connect and 2 to hangup). Can this function be done? If so, how to do it? Thank you . Pete Dao ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some input for Quad T1 and channel banks.
Don Pobanz wrote: Doug Lytle wrote on Monday, March 31, 2008 5:40 PM This does not sound right. If it is 2 PRIs then it should be 46 channels I may have the terminology incorrect. I don't have a D channel, so I guess this would be called a T1 then? 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some input for Quad T1 and channel banks.
Doug Lytle wrote: Don Pobanz wrote: Doug Lytle wrote on Monday, March 31, 2008 5:40 PM This does not sound right. If it is 2 PRIs then it should be 46 channels I may have the terminology incorrect. I don't have a D channel, so I guess this would be called a T1 then? Doug A channel bank does not do ISDN. You will be using what is called a channelized T1. You will probably set it up as 24 voice channels useing ESF B8ZS. When you use a channelized T1, each channel carries it's own signaling state and called number info is sent over the voice path(unless you have rotatory phones). Caller ID is also sent out via the voice channel. ISDN puts all the signaling on a single data stream called the D channel and you need to have two phone switches that talk to each other over the D channel. The signaling channel carries the calling and called number as well as the busy/idle state for each of the voice channels. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to give user a prompt before connecting thecall
Please do! From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Paul Hales [EMAIL PROTECTED] Sent: Monday, March 31, 2008 7:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to give user a prompt before connecting thecall It can be done via the 'visit a macro' part of the dial command... If anyone would like, i can post a code sample. PaulH On Mon, 2008-03-31 at 15:33 -0400, Dean Collins wrote: Yes it is. I'm remote at the moment so I can't send you the code but google for mobile remote receiver and you'll find what you are looking for. Lots of people do it so they don't have calls to cell phones picked up by voicemail. Cheers dean -Original Message- From: Pete Kay [EMAIL PROTECTED] Sent: Monday, March 31, 2008 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] How to give user a prompt before connecting thecall Hello, Is it possible to request for the premission from the called party through a prompt before routing the call? For instance, before actually connecting two parties through the use of DIAL command in the dialplan, I want to let Asterisk to automatically ask for the called party to decide whether he/she would like to be connected. ( ex. Press 1 to connect and 2 to hangup). Can this function be done? If so, how to do it? Thank you . Pete Dao ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some input for Quad T1 and channel banks.
Im guessing T1cas not PRI,just because its giving 24 fxs per T1. Steve, what are my options for SIP to fxs? thank you! On 3/31/08, Doug Lytle [EMAIL PROTECTED] wrote: Don Pobanz wrote: Doug Lytle wrote on Monday, March 31, 2008 5:40 PM This does not sound right. If it is 2 PRIs then it should be 46 channels I may have the terminology incorrect. I don't have a D channel, so I guess this would be called a T1 then? 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to give user a prompt before connecting thecall
Something like this:
Dialling:
exten = s,n(dial),Dial($ZAP/G1/${number},15,M(check)gm)
exten = s,n,Dbget(next/number)
exten = s,n,Goto(dial)
{macro-check}
exten = s,n,Playback(${heresacall})
exten = s,n,Read(response,options,1)
exten = s,n,Goto(${response},1)
exten = 1,1,Macroexit
exten = 2,1,Playback(thanksfortakingthecall)
This hasn't been tested. Give it a red hot go.
Another option is to set up a queue with external numbers as members,
and set the queue as need the memebrs to accept the calls. (not that I
can remember that option)
PaulH
On Mon, 2008-03-31 at 20:55 -0500, Jeremy Mann wrote:
Please do!
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Paul Hales [EMAIL
PROTECTED]
Sent: Monday, March 31, 2008 7:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to give user a prompt before connecting
thecall
It can be done via the 'visit a macro' part of the dial command...
If anyone would like, i can post a code sample.
PaulH
On Mon, 2008-03-31 at 15:33 -0400, Dean Collins wrote:
Yes it is.
I'm remote at the moment so I can't send you the code but google for mobile
remote receiver and you'll find what you are looking for.
Lots of people do it so they don't have calls to cell phones picked up by
voicemail.
Cheers
dean
-Original Message-
From: Pete Kay [EMAIL PROTECTED]
Sent: Monday, March 31, 2008 2:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] How to give user a prompt before connecting
thecall
Hello,
Is it possible to request for the premission from the called party through
a prompt before routing the call?
For instance, before actually connecting two parties through the use of DIAL
command in the dialplan, I want to let Asterisk to automatically
ask for the called party to decide whether he/she would like to be
connected. ( ex. Press 1 to connect and 2 to hangup).
Can this function be done? If so, how to do it?
Thank you .
Pete Dao
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and entity(ies)
to whom it is addressed. If you are the intended recipient, further
disclosures are prohibited without proper authorization. If you are not the
intended recipient, any disclosure, copying, printing, or use of this
information is strictly prohibited and possibly a violation of federal or
state law and regulations. If you have received this information in error,
please notify Texas Health Management Group immediately at 1-817-310-4999.
Texas Health Management Group, its subsidiaries, and affiliates hereby claim
all applicable privileges related to this information.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
