[asterisk-users] cisco MC3810 weirdness with asterisk

2009-06-11 Thread Tammy A. Wisdom
Has anyone here successfully gotten a cisco MC3810 talking with asterisk? I am getting the dreaded - Got SIP response 400 Bad Request - 'Malformed/Missing URL' back from xxx.xxx.xxx.xxx If you've gotten it to work you can feel free to email me off list. If your willing to share config's that also

Re: [asterisk-users] Rhino analog cards

2009-06-11 Thread Darrick Hartman
Jeff, Contact their tech support. You will need to send the card in for service, but they may be able to repair it. You should look into getting some sort of surge protection on the analog lines if you don't already have something. The surgegate stuff seems to work well. Darrick On

Re: [asterisk-users] Called party name with Cisco-2,811 gateway

2009-06-11 Thread Yehavi Bourvine
Sorry for the delay... 2009/6/7 David Backeberg dbackeb...@gmail.com On Sun, Jun 7, 2009 at 4:20 AM, Yehavi Bourvineyehavi.bourv...@gmail.com wrote: Hello, I am using a Cisco 2,811 gateway to connect Asterisk over PRI to our Nortel TX-1 PBX. Up to now I had only the calling party

[asterisk-users] OT - Aastra phones provisioning

2009-06-11 Thread Olivier
Hi, I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new Aastra SIP phones can be auto-provisioned when config files are stored in a specific TFTP subdirectory instead of TFTP root directory. For instance, TFTP root directory is /srv/tftp. When config files are stored in

Re: [asterisk-users] OT - Aastra phones provisioning

2009-06-11 Thread Grygoriy Dobrovolskyy
2009/6/11 Olivier oza-4...@myamail.com Hi, I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new Aastra SIP phones can be auto-provisioned when config files are stored in a specific TFTP subdirectory instead of TFTP root directory. For instance, TFTP root directory is

[asterisk-users] cant use h,1 at cancel!

2009-06-11 Thread BERGANZ François
Hello, In my dialplan, I do s,n,DIAL(…) If my called phone response and after hangup, asterisk execute the h,1,… But, if I the caller hangup at ringing (cancel), it don’t execute the h,1,… Know you why? Thank you Cordialement, BERGANZ François P Pensez à

Re: [asterisk-users] Query about tdm410 cards

2009-06-11 Thread Gordon Henderson
On Wed, 10 Jun 2009, Alex Samad wrote: Hi recently bought a soekris net5501 and a tdm410 to place in there. I am having some issues attaching 12V power to the card via the molex card - basically the box for the motherboard is too small. I know this mighs sound odd, but do you really need

Re: [asterisk-users] In Dahdi: what we use instead of /sbin/ztcfg -vv

2009-06-11 Thread Tzafrir Cohen
On Wed, Jun 10, 2009 at 05:02:35PM -0700, bilal ghayyad wrote: Hi All; In dahdi: what we use instead of ztcfg -vv (that is existed /sbin/ztcfg -vv). ? http://docs.tzafrir.org.il/dahdi-tools/#_dahdi_tools (This is taken verbatim from the file UPDATE.txt in the source tarball of

Re: [asterisk-users] cant use h,1 at cancel!

2009-06-11 Thread Steve Howes
On 11 Jun 2009, at 08:59, BERGANZ François wrote: In my dialplan, I do s,n,DIAL(…) If my called phone response and after hangup, asterisk execute the h,1,… But, if I the caller hangup at ringing (cancel), it don’t execute the h,1,… Know you why? Because the call was cancelled and

Re: [asterisk-users] Query about tdm410 cards

2009-06-11 Thread Alex Samad
On Thu, Jun 11, 2009 at 09:02:37AM +0100, Gordon Henderson wrote: On Wed, 10 Jun 2009, Alex Samad wrote: Hi recently bought a soekris net5501 and a tdm410 to place in there. I am having some issues attaching 12V power to the card via the molex card - basically the box for the

Re: [asterisk-users] cant use h,1 at cancel!

2009-06-11 Thread BERGANZ François
I found a bug in asterisk 1.6: http://lists.digium.com/pipermail/asterisk-dev/2009-April/037684.html in fact, the h,1 in the macro don’t work with cancel! Cordialement, BERGANZ François  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De :

Re: [asterisk-users] OT - Aastra phones provisioning

2009-06-11 Thread Olivier
2009/6/11 Grygoriy Dobrovolskyy megaho...@gmail.com 2009/6/11 Olivier oza-4...@myamail.com Hi, I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new Aastra SIP phones can be auto-provisioned when config files are stored in a specific TFTP subdirectory instead of TFTP

[asterisk-users] Asterisk 1.4.26-rc2 Now Available

2009-06-11 Thread Asterisk Development Team
The Asterisk Development Team is pleased to announce the second release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc2 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release fixes some issues reported by the community since the first release

Re: [asterisk-users] cant use h,1 at cancel!

2009-06-11 Thread jonas kellens
How about this : if you add the option 'g' in your Dial()-command, then when the caller hangs up Asterisk will continue to execute the commands hat follow. You could then read the ${DIALSTATUS}-variable (which will be 'CANCEL') and execute a command based on this.

[asterisk-users] Hello and Music on Hold question

2009-06-11 Thread Ishfaq Malik
Hello all, I have just joined this list and I'm currently working with asterisk 1.4.17 using RealTime. Not quite sure the level of queries you get but hopefully I'll be able to help with some input as well as questions of my own. Now to my first query. I'm changing the hold music on our system

Re: [asterisk-users] cisco MC3810 weirdness with asterisk

2009-06-11 Thread John Novack
Many CNET users have Cisco 3810's working with Asterisk 1.2, 1.4 and even 1.6 as well as Astlinux versions. Some are even using the T1 card to connect to a channel bank with number of calls limited to the 6 DSP card in the 3810. Contact users on the CNET VOIP list for details, but you will

Re: [asterisk-users] cant use h,1 at cancel!

2009-06-11 Thread Tilghman Lesher
On Thursday 11 June 2009 03:59:01 Steve Howes wrote: On 11 Jun 2009, at 08:59, BERGANZ François wrote: In my dialplan, I do s,n,DIAL(…) If my called phone response and after hangup, asterisk execute the h,1,… But, if I the caller hangup at ringing (cancel), it don’t execute the

Re: [asterisk-users] cant use h,1 at cancel!

2009-06-11 Thread Anthony
Tilghman Lesher wrote: On Thursday 11 June 2009 03:59:01 Steve Howes wrote: On 11 Jun 2009, at 08:59, BERGANZ François wrote: In my dialplan, I do s,n,DIAL(…) If my called phone response and after hangup, asterisk execute the h,1,… But, if I the caller hangup at ringing (cancel),

Re: [asterisk-users] cant use h,1 at cancel!

2009-06-11 Thread Ishfaq Malik
Anthony wrote: Tilghman Lesher wrote: On Thursday 11 June 2009 03:59:01 Steve Howes wrote: On 11 Jun 2009, at 08:59, BERGANZ François wrote: In my dialplan, I do s,n,DIAL(…) If my called phone response and after hangup, asterisk execute the h,1,… But, if I

Re: [asterisk-users] OT - Aastra phones provisioning

2009-06-11 Thread Philipp Kempgen
Olivier schrieb: I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new Aastra SIP phones can be auto-provisioned when config files are stored in a specific TFTP subdirectory instead of TFTP root directory. For instance, TFTP root directory is /srv/tftp. When config files

Re: [asterisk-users] asterisk crash on DAHDI error: No more room in scheduler

2009-06-11 Thread Allan Oepping
We had another dahdi problem this morning and got a dahdi_test -v. When calling into Span 2 it was just dead air. There was also an HDLC Abort at the start of the problem this time as well, but not the very first time it happened. Thanks. Allan #dahdi_test -v Opened pseudo dahdi interface,

Re: [asterisk-users] Asterisk - SIP - TCP and Exchange 2007 Unified Messaging

2009-06-11 Thread Wayne
David Backeberg wrote: I would ask the question the other way around. Are there any plans for Microsoft to release a unified messaging product that will comply with SIP over UDP? I do see your point in a potential (ok who are kidding - real) risk of a system crash with using MS having

Re: [asterisk-users] Asterisk - SIP - TCP and Exchange 2007 Unified Messaging

2009-06-11 Thread Jared Smith
On Wed, 2009-06-10 at 23:00 +0100, Wayne wrote: I was wondering what the current development plans / patches etc are to allow Asterisk to talk to Exchange 2007 Unified Messaging with respect to adding SIP over TCP support? There is experimental support for SIP over TCP in Asterisk 1.6.0 and

[asterisk-users] Automatic Calling Feature?

2009-06-11 Thread Christopher Stamper
Right now, my organization is using a commercial service (OneCallNow.com), that gives telephone notifications to all numbers in a predefined list. Example: -Admin records a voice message -Service calls each number in the list, and plays the message back to them It's a pretty handy service,

Re: [asterisk-users] Automatic Calling Feature?

2009-06-11 Thread Danny Nicholas
Nerdvittles.com has a nice example of this, when they are up. They used it for Phone trees for a school or something like that. Took less than 30 minutes to put in my dialplan and use. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]

Re: [asterisk-users] Automatic Calling Feature?

2009-06-11 Thread Duncan Turnbull
Not too hard to do, you can have a script generate a list of call files which automatically ring the callers in the list and play a message http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Cheers Duncan Christopher Stamper wrote: Right now, my organization is using a

Re: [asterisk-users] Query about tdm410 cards

2009-06-11 Thread Ira
At 02:01 PM 6/10/2009, you wrote: http://www.cyberguys.com/product-search/?keyword=molex doesn't look like it, really need a 90 degree plug and I am in OZ not usa so postage is going to kill me I'd buy a standard one, pull the pins, cut off the wire end of the plug, put it back in bend the

[asterisk-users] SIP hacked connection?

2009-06-11 Thread Paul Redstone
Hi Running 1.2.26 BRI stuffed. Calls made via PSTN via ISDN interface (Junghanns). SIP ports mapped through firewall as we often connect from outside, but all SIP accounts have good passwords. However our telecoms provider picked up a few suspicious calls to places we do not normally call at

Re: [asterisk-users] SIP hacked connection?

2009-06-11 Thread Steve Totaro
On Thu, Jun 11, 2009 at 3:30 PM, Paul Redstonepaul.redst...@solica.com wrote: Hi Running 1.2.26 BRI stuffed. Calls made via PSTN via ISDN interface (Junghanns). SIP ports mapped through firewall as we often connect from outside, but all SIP accounts have good passwords. However our

Re: [asterisk-users] SIP hacked connection?

2009-06-11 Thread Kyle Kienapfel
I can only suggest the most obvious cause without knowing how its configured, sorry. Take a look at the default context in sip.conf for me: [general] context=default my default context doesn't exist, so if a call comes in from an unknown user, asterisk complains about not matching whatever

Re: [asterisk-users] Asterisk to CCM

2009-06-11 Thread Jimmy Ezell
Still no luck getting this to work. I have been looking at the CallManager Logs but so far that is worse then useless. Anyone out there have any luck connecting Asterisk 1.4 and Cisco CallManager 3.3(5)? Jimmy Ezell http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html

Re: [asterisk-users] Query about tdm410 cards

2009-06-11 Thread Alex Samad
On Thu, Jun 11, 2009 at 11:14:47AM -0700, Ira wrote: At 02:01 PM 6/10/2009, you wrote: http://www.cyberguys.com/product-search/?keyword=molex doesn't look like it, really need a 90 degree plug and I am in OZ not usa so postage is going to kill me I'd buy a standard one, pull the pins,

Re: [asterisk-users] Asterisk to CCM

2009-06-11 Thread David Backeberg
On Thu, Jun 11, 2009 at 5:04 PM, Jimmy Ezelljez...@hmhca.com wrote: Still  no luck getting this to work.  I have been looking at the CallManager Logs but so far that is worse then useless.  Anyone out there have any luck connecting Asterisk 1.4 and Cisco CallManager 3.3(5)? Not exactly. We

[asterisk-users] Polycom Digitmap

2009-06-11 Thread Justin Phelps
I'm working on replacing a SoundPoint 600 with a 650. I need to merge these two sets of digitmaps in the polycom sip.cfg file, because the 650 locks up when I try to use the digitmap from the 600. I've included the default digitmap from a 3.1.3 RevB polycom release. I'd like to merge these

[asterisk-users] [SPAM] RE: SIP hacked connection?

2009-06-11 Thread C. Savinovich
Very few calls have been made this way, trivial cost, but it is slightly worrying. That's what I thought when they hacked into one of my systems, but it is not the cost of the calls, it is the purposed of the calls you should watch out for. The FBI contacted the owner of the PBX, and inquired

[asterisk-users] Writing for asterisk

2009-06-11 Thread Carlos Ruiz Diaz
Hello, Where can I found information about writing modules, applications and low level interactions for asterisk? At http://www.asterisk.org/developers I was unable to find tutorials for doing what I mentioned above. Thanks in advance. Carlos. ___ --

Re: [asterisk-users] Writing for asterisk

2009-06-11 Thread Moises Silva
this is possibly the best you can find: http://www.russellbryant.net/blog/2008/06/30/how-to-write-an-asterisk-module-part-3/ On Thu, Jun 11, 2009 at 7:50 PM, Carlos Ruiz Diazcarlos.ruizd...@gmail.com wrote: Hello, Where can I found information about writing modules, applications and low level

[asterisk-users] Current possible values for DIALSTATUS?

2009-06-11 Thread John Regal
Hi, As of v 1.6.1.1, can anyone tell me what the current possible values for DIALSTATUS could be? I found http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe it is outdated since there is no FAIL or FAILED in this list. Thanks!

Re: [asterisk-users] Writing for asterisk

2009-06-11 Thread Carlos Ruiz Diaz
Thank you! It is really useful, On Thu, Jun 11, 2009 at 9:13 PM, Moises Silva moises.si...@gmail.comwrote: this is possibly the best you can find: http://www.russellbryant.net/blog/2008/06/30/how-to-write-an-asterisk-module-part-3/ On Thu, Jun 11, 2009 at 7:50 PM, Carlos Ruiz

[asterisk-users] Problems with ReceiveFAX (asterisk 1.6.0.3 and t38)

2009-06-11 Thread srinivas Antarvedi
Hello users, have been facing problems with t38 passthrough using asterisk 1.6.0.3. observed also that in case of SendFAX we are not having major issues, its almost successfull. ReceiveFAX has problems most of the time. we have been using a ringcentral account for testing this receivefax.

[asterisk-users] FXO and fax-on-demand

2009-06-11 Thread AshikAli.m
Dear all , Is there any way for fax-on-demand on TDM FXO port using asterisk ? . I am seeking for guidence to meet this goal . with regards, ashik ali . m chen...@india . ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --