[asterisk-users] cisco MC3810 weirdness with asterisk
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 is a definate plus. Thanks, --Tammy ** Disclaimer: This e-mail may contain trade secrets or privileged, undisclosed or otherwise confidential information. If you have received this e-mail in error, you are hereby notified that any review, copying or distribution of it is strictly prohibited. Please inform us immediately and destroy the original transmittal. Thank you for your cooperation. ** ___ -- 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] Rhino analog cards
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 06/10/2009 08:16 AM, Jeff LaCoursiere wrote: Had a fairly horrible lightning storm night before last, and four of eight ports in a 1.4.20 machine stopped answering. In the CLI: budsw*CLI zap show channels Chan Extension Context Language MOH Interpret pseudodefault en default 1from-zaptel en default 2from-zaptel en default 3from-zaptel en default 4from-zaptel en default 5from-zaptel en default 6from-zaptel en default 7from-zaptel en default 8from-zaptel en default budsw*CLI but in dmesg: r4fxo: module license 'unspecified' taints kernel. rcbfx 1: Rhino PCI BAR0 febff000 IOMem mapped at f8b12000 rcbfx 1: Waiting for response from card . rcbfx 1: Firmware Version 1.f rcbfx 1: Firmware File Version is 1.f rcbfx 1: Hardware version 11 rcbfx 1: G168 07 04 DSP Loader file size = 170 App file size = 48414 rcbfx 1: G168 DSP Ping DSP Version 106 rcbfx 1: G168 DSP Active and Servicing 4 Channels - f rcbfx 1: Starting DMA rcbfx 1: Spotted a Rhino: Rhino RCB4FXO (4 channels) rcbfx 2: Rhino PCI BAR0 febfe000 IOMem mapped at f8b14000 rcbfx 2: Waiting for response from card . rcbfx 2: Firmware Version 1.f rcbfx 2: Firmware File Version is 1.f rcbfx 2: Hardware version 11 rcbfx 2: G168 DSP App Loader Failed 4 rcbfx 2: Unable to intialize G168 DSP rcbfx 2: Starting DMA rcbfx 2: Spotted a Rhino: Rhino RCB4FXO (4 channels) So it seems the second card is fried? A reboot seems to result in the same messages. Trying to arrange for a power cycle - the site is remote. j ___ -- 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] Called party name with Cisco-2,811 gateway
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 names passed both ways. After upgrading the Cisco to the latest release (12.4.24T) it began honoring the remote-part-ID field sent by Asterisk and sends the called name to the Nortel. However, I still do not get the called name from the Nortel to Asterisk. Has anyone managed to make this working? There are a lot of choices when you set up your interconnectivity between your Cisco gear and the T1/PRI channels. If you post your dialpeer for this I'll take a look, although strictly speaking this isn't really an asterisk question. Here is the extract from the configuratio of the gateway: voice service voip clid network-provided qsig decode allow-connections sip to sip fax protocol t38 ls-redundancy 4 hs-redundancy 2 fallback pass-through g711alaw controller E1 0/0/0 pri-group timeslots 1-31 ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-qsig isdn overlap-receiving isdn not-end-to-end 64 isdn incoming-voice voice isdn supp-service name calling profile Network-Extension operation-value-tag local isdn negotiate-bchan isdn outgoing ie facility isdn outgoing ie extended-facility no isdn outgoing ie notify-indicator isdn outgoing display-ie isdn outgoing ie caller-number isdn outgoing ie called-number no cdp enable ! dial-peer voice 2 pots destination-pattern 0T progress_ind setup enable 3 no digit-strip direct-inward-dial port 0/0/0:15 Thanks, __Yehavi: ___ -- 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] OT - Aastra phones provisioning
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 /srv/tftp, a new Aastra can find its config files. When config files are stored in /srv/tftp/aastra, a new Aastra can't find its config files. I tried to using DHCP root-path option to tell Aastra phones to search the right subdirectory, but it doesn't seem to work. Any advice on this ? 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
Re: [asterisk-users] OT - Aastra phones provisioning
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 /srv/tftp. When config files are stored in /srv/tftp, a new Aastra can find its config files. When config files are stored in /srv/tftp/aastra, a new Aastra can't find its config files. I tried to using DHCP root-path option to tell Aastra phones to search the right subdirectory, but it doesn't seem to work. Any advice on this ? Regards How about /srv/tftp/aastraphones ? ___ -- 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] cant use h,1 at cancel!
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 dont execute the h,1, Know you why? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- 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] Query about tdm410 cards
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 the +12V connection? You only need it if you have analogue phones plugged in and not exchange lines.. I know - this is obvious and you probably do have analogue phones plugged in, but I'm just checking!!! 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] In Dahdi: what we use instead of /sbin/ztcfg -vv
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 dahdi-tools) And no: there's really no need to use -vv in a non-interactive run. It mearely serves to generate noise. If there's an error, you'll get an error message. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cant use h,1 at cancel!
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 not actually hung up? Generally the hangup context is used to 'clean up' or provide info about the call. If it didn't happen its a bit irrelevant. S ___ -- 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] Query about tdm410 cards
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 motherboard is too small. I know this mighs sound odd, but do you really need the +12V connection? You only need it if you have analogue phones plugged in and not exchange lines.. I have 2 fxs + 1fxo so I know - this is obvious and you probably do have analogue phones plugged in, but I'm just checking!!! we all miss the obvious at some time 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 -- If we are going to save a generation of young people, our children must know they will face bad consequences for criminal behavior. Sadly, too many youths are not getting that message. Our juvenile justice system must say to our children: We love you, but we are going to hold you accountable for your actions. - George W. Bush 01/01/2000 2000 Bush campaign literature signature.asc Description: Digital 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] cant use h,1 at cancel!
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 : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Steve Howes Envoyé : jeudi 11 juin 2009 10:59 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] cant use h,1 at cancel! 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 not actually hung up? Generally the hangup context is used to 'clean up' or provide info about the call. If it didn't happen its a bit irrelevant. S ___ -- 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] OT - Aastra phones provisioning
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 root directory. For instance, TFTP root directory is /srv/tftp. When config files are stored in /srv/tftp, a new Aastra can find its config files. When config files are stored in /srv/tftp/aastra, a new Aastra can't find its config files. I tried to using DHCP root-path option to tell Aastra phones to search the right subdirectory, but it doesn't seem to work. Any advice on this ? Regards How about /srv/tftp/aastraphones ? Do you mean using /srv/tftp/aastraphones should work while /srv/tftp/aastra wouldn't ? Anyway, I tried this and this doesn't change anything : Jun 11 11:37:33 osiris2 in.tftpd[2476]: connect from 192.168.100.119 (192.168.100.119) Jun 11 11:37:33 osiris2 tftpd[2477]: tftpd: trying to get file: security.tuz Jun 11 11:37:33 osiris2 tftpd[2477]: tftpd: serving file from /srv/tftp Jun 11 11:37:33 osiris2 in.tftpd[2478]: connect from 192.168.100.119 (192.168.100.119) Jun 11 11:37:33 osiris2 tftpd[2479]: tftpd: trying to get file: aastra.cfg Jun 11 11:37:33 osiris2 tftpd[2479]: tftpd: serving file from /srv/tftp Jun 11 11:37:35 osiris2 in.tftpd[2480]: connect from 192.168.100.119 (192.168.100.119) Jun 11 11:37:35 osiris2 tftpd[2481]: tftpd: trying to get file: security.tuz In /etc/dhcp3/dhcpd.conf, I included : option root-path /srv/tftp/aastraphones; If I'm not mistaken, it seems : - DHCP should include in its DHCPOFFER, a TFTP path set to /srv/tftp/aastraphones (using Wireshark, I can't see any root-path related value in DCHPOFFER from DHCP server nor in any consecutive DHCP or TFTP request but I can't say if this is normal or not) - from then, TFTP request should include something telling files should be searched in aastraphones subdirectory. ___ -- 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] Asterisk 1.4.26-rc2 Now Available
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 candidate. The most significant issues resolved include: * Treat an empty FORWARD_CONTEXT channel variable the same way we treat a non-existant one (issue #15056). * If using a deprecated musiconhold.conf format, reloading the module would cause the class to disappear (issue #14759). * Creation of generic call forward API, ast_call_forward() which is used to resolve a potentional SIP spiral detection problem (issue #13630). * Resolve a regression introduced after changes to module load order that were necessary to close another issue. The regression caused issues with the usage of #exec, affecting FreePBX users (issue #15189). Additionally, updates that went into Asterisk 1.4.25.1 to resolve a chan_iax2 issue have been merged into this release candidate (AST-2009-001). The original security advisory can be found here: http://downloads.asterisk.org/pub/security/AST-2009-001.html For a full list of changes in this release candidate, please see the ChangeLog: http://svn.asterisk.org/svn/asterisk/tags/1.4.26-rc2/ChangeLog Issues found in this release candidate can be reported at http://issues.asterisk.org Thank you for your continued support of Asterisk! ___ -- 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] cant use h,1 at cancel!
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 ${DIALSTATUS}-variable (which will be 'CANCEL')
and execute a command based on this.
GoToIf($[${DIALSTATUS}=ANSWER]?here:there)
Jonas.
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?
___
-- 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] Hello and Music on Hold question
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 and I've done this by deleting the old sound file that was in the music on hold directory as defined by the /etc/asterisk/musiconhold.conf and put a new one in there but when going on hold I can still hear the old music. Is there a cache somewhere that I need to clear? I have tried moh reload from the console but that just reloads the conf file. Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- 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 MC3810 weirdness with asterisk
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 need: 16 M of flash 64 M of memory A late SIP IOS installed in flash Obviously you will need an AVM card with proper analog modules. John Novack Tammy A. Wisdom wrote: 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 is a definate plus. Thanks, --Tammy ** Disclaimer: This e-mail may contain trade secrets or privileged, undisclosed or otherwise confidential information. If you have received this e-mail in error, you are hereby notified that any review, copying or distribution of it is strictly prohibited. Please inform us immediately and destroy the original transmittal. Thank you for your cooperation. ** ___ -- 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 -- Dog is my co-pilot ___ -- 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] cant use h,1 at cancel!
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 h,1,… Know you why? Because the call was cancelled and not actually hung up? Generally the hangup context is used to 'clean up' or provide info about the call. If it didn't happen its a bit irrelevant. I think you mean that it wasn't ANSWERED, and therefore, it cannot be hung up. -- Tilghman ___ -- 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] cant use h,1 at cancel!
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), it don’t execute the h,1,… Know you why? Because the call was cancelled and not actually hung up? Generally the hangup context is used to 'clean up' or provide info about the call. If it didn't happen its a bit irrelevant. I think you mean that it wasn't ANSWERED, and therefore, it cannot be hung up. Yes, the simple answer is to use Answer() and then play a ring while connecting the caller to the callee. ___ -- 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] cant use h,1 at cancel!
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 the caller hangup at ringing (cancel), it don’t execute the h,1,… Know you why? Because the call was cancelled and not actually hung up? Generally the hangup context is used to 'clean up' or provide info about the call. If it didn't happen its a bit irrelevant. I think you mean that it wasn't ANSWERED, and therefore, it cannot be hung up. Yes, the simple answer is to use Answer() and then play a ring while connecting the caller to the callee. Wouldn't that solution show all calls on the cdr as having a disposition of ANSWERED so rendering the cdr a bit useless? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- 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] OT - Aastra phones provisioning
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 are stored in /srv/tftp, a new Aastra can find its config
files.
When config files are stored in /srv/tftp/aastra, a new Aastra can't find
its config files.
I tried to using DHCP root-path option to tell Aastra phones to search the
right subdirectory, but it doesn't seem to work.
https://svn.amooma.com/gemeinschaft/trunk/usr/share/doc/gemeinschaft/misc/dhcpd-3-example.conf
:
class Aastra {
match if substring(hardware, 1, 3) = 00:08:5D;
option tftp-server-name
http://192.168.1.130/gemeinschaft/prov/aastra/;;
# Aastra does not support any :port in the URL, not even :80
# for firmware app versions 2.1.2
}
That should work analogously for TFTP.
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
--
___
-- 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 crash on DAHDI error: No more room in scheduler
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, measuring accuracy... 8192 samples in 8191.855 system clock sample intervals (99.998%) 8192 samples in 8191.584 system clock sample intervals (99.995%) 8192 samples in 8191.896 system clock sample intervals (99.999%) 8192 samples in 8191.832 system clock sample intervals (99.998%) 8192 samples in 8191.880 system clock sample intervals (99.999%) 8192 samples in 8191.865 system clock sample intervals (99.998%) 8192 samples in 8191.832 system clock sample intervals (99.998%) 8192 samples in 8191.848 system clock sample intervals (99.998%) 8192 samples in 8191.888 system clock sample intervals (99.999%) 8192 samples in 8191.856 system clock sample intervals (99.998%) 8192 samples in 8191.864 system clock sample intervals (99.998%) 8192 samples in 8191.888 system clock sample intervals (99.999%) 8192 samples in 8191.824 system clock sample intervals (99.998%) 8192 samples in 8191.865 system clock sample intervals (99.998%) 8192 samples in 8191.880 system clock sample intervals (99.999%) 8192 samples in 8191.816 system clock sample intervals (99.998%) 8192 samples in 8191.873 system clock sample intervals (99.998%) 8192 samples in 8191.897 system clock sample intervals (99.999%) 8192 samples in 8191.815 system clock sample intervals (99.998%) 8192 samples in 8191.856 system clock sample intervals (99.998%) 8192 samples in 8191.896 system clock sample intervals (99.999%) 8192 samples in 8191.848 system clock sample intervals (99.998%) 8192 samples in 8191.865 system clock sample intervals (99.998%) 8192 samples in 8191.872 system clock sample intervals (99.998%) 8192 samples in 8191.808 system clock sample intervals (99.998%) 8192 samples in 8191.840 system clock sample intervals (99.998%) 8192 samples in 8191.880 system clock sample intervals (99.999%) 8192 samples in 8191.816 system clock sample intervals (99.998%) --- Results after 28 passes --- Best: 99.999 -- Worst: 99.995 -- Average: 99.998143, Difference: 99.998143 #dahdi_scan [1] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 1 name=TE4/0/1 manufacturer=Digium devicetype=Wildcard TE420 (4th Gen) location=Board ID Switch 0 basechan=1 totchans=24 irq=169 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [2] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 2 name=TE4/0/2 manufacturer=Digium devicetype=Wildcard TE420 (4th Gen) location=Board ID Switch 0 basechan=25 totchans=24 irq=169 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [3] active=yes alarms=RED description=T4XXP (PCI) Card 0 Span 3 name=TE4/0/3 manufacturer=Digium devicetype=Wildcard TE420 (4th Gen) location=Board ID Switch 0 basechan=49 totchans=24 irq=169 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [4] active=yes alarms=RED description=T4XXP (PCI) Card 0 Span 4 name=TE4/0/4 manufacturer=Digium devicetype=Wildcard TE420 (4th Gen) location=Board ID Switch 0 basechan=73 totchans=24 irq=169 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF #cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 30618380 0 0 0IO-APIC-edge timer 1: 8 0 0 0IO-APIC-edge i8042 6: 6 0 0 0IO-APIC-edge floppy 8: 1 0 0 0IO-APIC-edge rtc 9: 0 0 0 0 IO-APIC-level acpi 12: 4 0 0 0IO-APIC-edge i8042 14: 1844 0 249693 0IO-APIC-edge ide0 50: 33 0 0 225302 PCI-MSI eth0 58: 31 0 91091 0 PCI-MSI eth1 169: 167784 0 12618802 1778 IO-APIC-level wct4xxp 233: 6264 0 624060 55148 IO-APIC-level ata_piix NMI:877880 1548 1480 LOC: 30618862 30618790 30618704 30618618 ERR: 0 MIS: 0 #cat /sys/module/dahdi/version 2.1.0.4 #cat /sys/module/dahdi/srcversion B180C6A9D92F9D67F3CEB2D #cat /sys/module/dahdi/parameters/ debugdeftaps #cat /sys/module/dahdi/parameters/debug 0 #cat /sys/module/dahdi/parameters/deftaps 64 #cat /sys/module/dahdi/refcnt 196 #cat /sys/module/dahdi/sections/ __kcrctab __ksymtab __ksymtab_strings __param__versions #cat /sys/module/dahdi/sections/__versions 0x8844a460 Log entries at
Re: [asterisk-users] Asterisk - SIP - TCP and Exchange 2007 Unified Messaging
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 full control over your phone system but, I was thinking along the lines of using exchange really only as a messaging system - ie voice mail, email reader. From what I can make out MS are even going along the lines of doing speech to text with 2010 version (I think it has text to speech already). I would have to agree that the PBX side of things is held still by Asterisk and I don't see my view on that changing yet, but, I would imagine MS would dig their heels in rather than changing exchange. The Asterisk community, being more open minded to change, could easily(?) make this work. Thanks Wayne. ___ -- 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 - SIP - TCP and Exchange 2007 Unified Messaging
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 later. It's probably not perfect yet, but we'd be happy to hear how it works for you, and that will help the Asterisk developers make it better. As far as other things related to the vague notion of unified communications, there's the code that Terry Wilson just added on being able to read Exchange calendars (iCal/CalDAV are supported as well) from the Asterisk dialplan, there's plenty of Jabber work being done on the IM side, and Asterisk can already store voicemail in an IMAP mail server. (I've long since let go of my Windows skills, but I'm assuming that modern versions of Exchange still let you communicate via IMAP, right?) In short, there are a lot of exciting things happening in the world of Asterisk with regards to unified communications. -- Jared Smith Training Manager Digium, Inc. ___ -- 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] Automatic Calling Feature?
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, albeit a bit pricey. I've been wondering if Asterisk could do this for me? I don't really want to have to write scripts, but it would be great if it's already a feature. I don't have an Asterisk PBX running yet, but when I do it will probably have multiple T1 PRI lines, making it possible to dial all these numbers (100+) in a reasonable amount of time. Anyone know of a way to do this? -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- 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] Automatic Calling Feature?
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] On Behalf Of Christopher Stamper Sent: Thursday, June 11, 2009 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Automatic Calling Feature? 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, albeit a bit pricey. I've been wondering if Asterisk could do this for me? I don't really want to have to write scripts, but it would be great if it's already a feature. I don't have an Asterisk PBX running yet, but when I do it will probably have multiple T1 PRI lines, making it possible to dial all these numbers (100+) in a reasonable amount of time. Anyone know of a way to do this? -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- 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] Automatic Calling Feature?
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 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, albeit a bit pricey. I've been wondering if Asterisk could do this for me? I don't really want to have to write scripts, but it would be great if it's already a feature. I don't have an Asterisk PBX running yet, but when I do it will probably have multiple T1 PRI lines, making it possible to dial all these numbers (100+) in a reasonable amount of time. Anyone know of a way to do this? -- Christopher Stamper Email: christopherstam...@gmail.com mailto:christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- 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] Query about tdm410 cards
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 pins over and insulate it with a bit of hot melt or heatshrink. Probably as good as anything you'll buy. Ira ___ -- 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 hacked connection?
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 times we do not often call. Looking at Asterisk logs it shows SIP session from the internet connected in and making calls with account IDs we do not recognise - definitely none of ours. Very few calls have been made this way, trivial cost, but it is slightly worrying. Anyone any ideas on how this could be happening? Thank Paul ___ -- 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] SIP hacked connection?
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 telecoms provider picked up a few suspicious calls to places we do not normally call at times we do not often call. Looking at Asterisk logs it shows SIP session from the internet connected in and making calls with account IDs we do not recognise - definitely none of ours. Very few calls have been made this way, trivial cost, but it is slightly worrying. Anyone any ideas on how this could be happening? Thank Paul Posting some redacted logs would be very helpful. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] SIP hacked connection?
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 number they are asking for. if i changed it to context=internal which does have dialing rules, then people would be able to dial out through my asterisk box. Kyle On Thu, Jun 11, 2009 at 12:30 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 telecoms provider picked up a few suspicious calls to places we do not normally call at times we do not often call. Looking at Asterisk logs it shows SIP session from the internet connected in and making calls with account IDs we do not recognise - definitely none of ours. Very few calls have been made this way, trivial cost, but it is slightly worrying. Anyone any ideas on how this could be happening? Thank Paul ___ -- 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] Asterisk to CCM
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
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy
Ezell
Sent: Wednesday, June 10, 2009 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to CCM
As you can see below I am striping off the 8 before it ever goes
to CCM in the extensions.conf file.
exten =
_8XXX,1,Dial(${CISCOTRUNK}/${EXTEN:1...@172.16.200.10:1720)
I have the H323 gateway in CCM configured to use the same
Calling Search Space as my phone extensions.
Jimmy Ezell
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Austin
Sent: Tuesday, June 09, 2009 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Asterisk to CCM
Make sure you are stripping the 8 on inbound calls to
that H323 gateway
under CCM and that it has a valid search space to find
your extensions...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy
Ezell
Sent: Tuesday, June 09, 2009 3:13 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [asterisk-users] Asterisk to CCM
Hit another problem in my tutorial in converting over
from Cisco CallManager to Asterisk.
I have been following the instructions at :
http://voip-info.capatres.com/wiki/view/Asterisk+Cisco+CallManager+Integ
ration.html on intergrating Asterisk and Cisco CallManager.
I can make calls from CCM to Asterisk phones - and yes
that felt good to get that working.
My problem is that it does not work from the other
direction. I cannot make calls from CCM phones to Asterisk Phones.
I want to be able to dial 8 and the extension of the ccm
phone.
I am using CCM 3.3.(5) so I do not have the option to
use a SIP turnk because it is not supported. I am also using h323
instead of ooh323. Not sure if that might make a difference.
In Asterisk console I get:
-- Executing [8...@internal:1] Dial(SIP/207-08bd64c8,
H323/callman02/2...@172.16.200.10:1720) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called callman02/2...@172.16.200.10:1720
== Everyone is busy/congested at this time (1:0/0/1)
This is the contents of my h323.conf file:
=
[general]
port = 1720
bindaddr = 172.17.100.2
disallow=all
allow=gsm ; Always allow GSM, it's cool :)
allow=ulaw ; see doc/rtp-packetization for
framing options
allow=alaw
dtmfmode=rfc2833
gatekeeper = DISABLE
context=default
[callman02]
type=friend
context=default
ip=172.16.200.10
port=1720
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=rfc2833
nat=no
canreinvite=yes
qualify=yes
extensions.conf file
==
[globals]
CISCOTRUNK=H323/callman02
[cisco]
exten =
_8XXX,1,Dial(${CISCOTRUNK}/${EXTEN:1...@172.16.200.10:1720)
exten = _8XXX,n,Congestion()
exten = _8XXX,n,Hangup()
Jimmy Ezell
Converting CCM to Asterisk
http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html
___
-- 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] Query about tdm410 cards
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, cut off the wire end of the plug, put it back in bend the pins over and insulate it with a bit of hot melt or heatshrink. Probably as good as anything you'll buy. I have soldered to the back of the board, the molex pins go all the way through the pcb Ira ___ -- 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 -- I'm not available for comment.. signature.asc Description: Digital 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] Asterisk to CCM
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 have luck loading the SIP firmware on the Cisco phones and having the Cisco phones register directly as SIP devices with asterisk. That works great for us. We also have great success running SIP between the Cisco routers (3845) and asterisk. There's also a chan_skinny, aka SCCP support in asterisk, but I've never used it myself. I've never tried to integrate these things with Call Manager. First of all, I don't have a license for CCM, and second I already have asterisk setup the way I want it. ___ -- 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] Polycom Digitmap
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 two digitmaps, but I don't want to reintroduce the lockup issue I was having with the 600 digitmap on the 650. I've included my understanding of each definition below as well. I understand that the T at the end of certain definitions set a timeout, which is defined right after the digitmap. The Default timeout has a series of six 3's seperated by a |. The 600 definition has a single 3. What is the proper way to define these timeouts as well? It also seems some of the definitions in the 600 group are redundant. Can you think of any reason as to why these shouldn't be combined? Default: [2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT 600: [2-9]11|[*]xx|891xxx|[1-7]xxx|8[0-46-8]xx|8500|851xxx|9,1[2-9]x|9,xxxT 600 [2-9]11Information/Emergency (411 911 etc) [*]xx Service Numbers (*69) 891xxx 891 then 3 digits [1-7]xxx 4 Digit Extension (1801, 1101, 5674) 8[0-46-8]xx8 then 0-4 or 6-8 then 2 digits 8500 8500 851xxx 851 then 3 digits 9,1[2-9]x 9, then 1, then [2-9] then 9 digits 9,xxxT 9, then 7 digit number Default [2-9]11 Information/Emergency (411 911 etc) 0Operator 011xxx.T 011 then three digits, than anything [0-1][2-9]x 0-1 then 2-9 then 9 digits [2-9]x 2-9 then 9 digits [2-9]xxxT2-9 then 3 digits -- Justin Phelps www.onitato.com 850.866.6864 ___ -- 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] [SPAM] RE: SIP hacked connection?
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 him about calls being made from his company doing credit card soliciting. CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Redstone Sent: Thursday, June 11, 2009 3:30 PM To: Asterisk User Subject: [asterisk-users] SIP hacked connection? 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 times we do not often call. Looking at Asterisk logs it shows SIP session from the internet connected in and making calls with account IDs we do not recognise - definitely none of ours. Very few calls have been made this way, trivial cost, but it is slightly worrying. Anyone any ideas on how this could be happening? Thank Paul ___ -- 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] Writing for asterisk
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. ___ -- 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] Writing for asterisk
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 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. ___ -- 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 -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Current possible values for DIALSTATUS?
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! ___ -- 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] Writing for asterisk
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 Diazcarlos.ruizd...@gmail.com wrote: 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. ___ -- 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 -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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 ___ -- 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] Problems with ReceiveFAX (asterisk 1.6.0.3 and t38)
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. so ringcentral is trying for 3 times if the sending fax failed for the first time. what i observed is that for the first two attempts it failing with the UNEXPECTED MESSAGE RECEIVED and the last time it was successfull . and the above scenario is not always replicatable and some times its failing completely(3rd attempt also fails) i have the tcpdump's .cap files so if anybody want to look at them too i can send.i tried to send along with this mail but the mail was rejected may be because of exceeding the attachment size. Any help is appreciable Thanks and regards Srinivas Antarvedi ___ -- 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] FXO and fax-on-demand
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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
