[asterisk-users] how to know RTP por of a SIP client in the dialplan
Hi all, How can I get the RTP port one SIP client is using for sending/receiving RTP flow? Can I obtain it in from SIP_HEADER of something like that in the dialplan? Thank you! Isabel Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about queues.conf
thank you for your response after module unload app_queue.so module load app_queue.so i can do this operation, for the internal extension now i have another issue related to the same queues i have 2 providers for the first provider exten = 800,1,AgentLogin() exten = 52046,1,Answer() exten = 52046,2,Queue(hotline) there is no issue i can log in the queue 800 and recived the calls when i call this number 52046 for the first provider now the issue is with the secend provider the DID for the secen number from 500 to 600 exten = 800,1,AgentLogin() exten = 560,1,Answer() exten = 560,2,Queue(hotline) When i call this number 560 from my mobile i listen the music in hold for 2 second and after the call hang-up, i have noticed the same issue when i use meetme (for the first provider no issue but for the second there is no meetme) the log : -- Accepting call from '522343535' to '560' on channel 1/14, span 1 -- Executing [560@default:1] Answer(Zap/14-1, ) in new stack -- Executing [560@default:2] Queue(Zap/14-1, hotline) in new stack -- Started music on hold, class 'default', on Zap/14-1 -- Channel 1/14, span 1 got hangup, cause -1 -- Stopped music on hold on Zap/14-1 == Spawn extension (default, 560, 2) exited non-zero on 'Zap/14-1' -- Hungup 'Zap/14-1' please advice 2011/10/20 Warren Selby wcse...@selbytech.com On Thu, Oct 20, 2011 at 11:01 AM, salaheddine elharit salah.elharit...@gmail.com wrote: [Oct 20 15:34:00] WARNING[19179]: pbx.c:1832 pbx_extension_helper: No application 'Queue' for extension (agents, 666, 2) This line here indicates to me that you don't have app_queue.so loaded. Try, from the asterisk cli, the following: module unload app_queue.so module load app_queue.so And report back any error messages that may pop up. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP ports used by Asterisk in dialplan
You can use tcpdump portrange 1-2 udp 2011/10/20 Andrew Higgs andrew.m.hi...@gmail.com Hi Isabel, Could you not just filter out after the fact using something like Wireshark? Regards On Thu, Oct 20, 2011 at 3:28 PM, ISABEL ORDAS ARNAL i...@tid.es wrote: Dear all, ** ** Do you know if there is a way to know the 2 RTP ports that Asterisk is using for audio flow in a call in the dialplan? I would like to launch a Linux shell command “tcpdump” to capture audio flow in those 2 RTP ports before call starts and stop capturing at the end of the call. ** ** Regards, Isabel -- Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk dialplan macro output
Hi all,
Is there a way to read in the dialplan a macro output parameter?
For instance, in the following macro I would like to know the pid of the Linux
process for killing it when hanging up.
[macro-capture]
exten = s,1,NoOp(Caller IP = ${ARG1})
exten = s,n,NoOp(Filename = ${ARG2})
exten = s,n,Set(pid=${SHELL(bash -c /usr/local/CallMonitoring/launch-tshark.sh
${ARG1} ${ARG2})})
Thank you!
Isabel
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[asterisk-users] Video Softphone
Hi, Any video softphone that will send the video codec in the first INVITE, in eyebem and some other phones like Ekiga first we are getting audio and then there is a button SEND VIDEO, if we click that the re-invite is going with video codec, whereas i need to send the video at first invite itself, is there any softphone for that. I tested with Android phone with a application called IMS droid with Asterisk where in the first INVITE I am able to send the video. Can any one suggest any softphone which can send video codec in first INVITE. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No Voice path during NCS call with Asterisk 10.0.0
Hi, I tried to establish NCS call b/w 2 endpoints of same PacketCable MTA using asterisk-10.0.0. I observed that MDCX sent to aaln/1 contains its own SDP. Some I observed with aaln/2. So voice path is not established b/w aaln/1 and aaln/2. My Configurations: mgcp.cong: [mta84.globaledgesoft.com] host= mta84.globaledgesoft.com wcardep = aaln/* callwaiting = 1 ;canreinvite = 1 dtmfmode= rfc2833 ;amaflags= BILLING ncs = yes ; Use NCS 1.0 signalling ;pktcgatealloc = yes ; Allocate DQOS gate on CMTS ;hangupongateremove = yes ; Hangup the channel if the CMTS close the gate callerid= 3341 ;accountcode = test-362265 line= aaln/1 callerid= 3342 ;accountcode = test-362266 line= aaln/2 extension.conf: exten = 3341,1,Dial(MGCP/aaln/1...@mta84.globaledgesoft.com) exten = 3342,1,Dial(MGCP/aaln/2...@mta84.globaledgesoft.com) can anybody help me to resolve this issue. Regards Vikas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to know RTP por of a SIP client in the dialplan
Do you need to know to get it in dialplan? If I not, from shell (not Asterisk CLI) I usually use: netstata -a | grep asterisk By default Asterisk settings it should be something between 10k-20k -Bruce On Fri, Oct 21, 2011 at 3:46 AM, ISABEL ORDAS ARNAL i...@tid.es wrote: Hi all, How can I get the RTP port one SIP client is using for sending/receiving RTP flow? Can I obtain it in from SIP_HEADER of something like that in the dialplan? Thank you! ** ** Isabel -- Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skype Messaging with Asterisk 10?
Hello, is every development on chan_skype out of the question after Skypcrosoft pulled the plug, or can we hope for an Asterisk 10 Version that supports the new, shiny messaging-api in asterisk 10? Andreas... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype Messaging with Asterisk 10?
On 11-10-21 11:45 AM, Andreas Anderson wrote: Hello, is every development on chan_skype out of the question after Skypcrosoft pulled the plug, or can we hope for an Asterisk 10 Version that supports the new, shiny messaging-api in asterisk 10? Andreas... Nope, nobody submitted any patches for it. So anything now would have to be submitted into trunk, which would make Asterisk 11 the next version to support it. Again, assuming somebody submits a patch. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can we use MySQL native connector for ARA?
On 10/20/2011 05:59 AM, JR Richardson wrote: Hello Everyone, The documentation suggests using unixodbc for asterisk realtime. Is there any way we can just use native database clients such as libmysqlclient from MySQL? The native clients tend to be more up-to-date. Thanks in Advance, Nick. I've used the MySQL addon for years with great success, initiated the project to support read/write to separate databases for asterisk clustering. You will get more functionality for complex queries using the odbc connector, but for basic ARA applications, the MySQL addon works fine and I've never had a problem with stability. I also use the cdr_mysql as well. I wrote a couple of papers on asterisk_clustering_with_mysql_replication. They are a bit dated but still relevant. I'll send over if you like. Good luck. JR Hey JR! I've used the native MySQL connector too with great results. Maybe you could post a link to the paper(s) on voip-info.org for posterity. I know I'd love to seem them too Bruce Ferrell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about queues.conf
On Fri, Oct 21, 2011 at 4:50 AM, salaheddine elharit salah.elharit...@gmail.com wrote: -- Accepting call from '522343535' to '560' on channel 1/14, span 1 -- Executing [560@default:1] Answer(Zap/14-1, ) in new stack -- Executing [560@default:2] Queue(Zap/14-1, hotline) in new stack -- Started music on hold, class 'default', on Zap/14-1 -- Channel 1/14, span 1 got hangup, cause -1 -- Stopped music on hold on Zap/14-1 == Spawn extension (default, 560, 2) exited non-zero on 'Zap/14-1' -- Hungup 'Zap/14-1' please advice Do any calls from this provider work? In other words, if you changed it from Queue() to Dial() your sip extension (or whatever means you have of answering the call), does it work then or does it also hang up after two seconds? What version of Asterisk and Zaptel are you using? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dialplan macro output
On Fri, Oct 21, 2011 at 6:54 AM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Hi all,
** **
Is there a way to read in the dialplan a macro output parameter?
For instance, in the following macro I would like to know the pid of the
Linux process for killing it when hanging up.
I think what you're looking for is a GoSub that ends with a Return(value).
You then can pull up the value in ${GOSUB_RETVAL}. But I may be
misunderstanding what you're wanting to do.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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Re: [asterisk-users] Skype Messaging with Asterisk 10?
Hi Paul, is every development on chan_skype out of the question after Skypcrosoft pulled the plug, or can we hope for an Asterisk 10 Version that supports the new, shiny messaging-api in asterisk 10? Nope, nobody submitted any patches for it. So anything now would have to be submitted into trunk, which would make Asterisk 11 the next version to support it. E, please correct me if i'm wrong, but the out-of-call-messaging-api is in asterisk 10 and currently supports sip and xmpp...? But i'm not asking for an extension of asterisk in any way, but of chan_skype that was sold 'til July... Again, assuming somebody submits a patch. Since when can someone submit a patch for chan_skype?? Did i miss an announcement that it has been opensourced? I'm under the impression that digium is the only party who *can* extend chan_skype... Kind regards, Andreas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to know RTP por of a SIP client in the dialplan
On Fri, Oct 21, 2011 at 2:46 AM, ISABEL ORDAS ARNAL i...@tid.es wrote: Hi all, How can I get the RTP port one SIP client is using for sending/receiving RTP flow? Can I obtain it in from SIP_HEADER of something like that in the dialplan? Thank you! ** ** I don't think you can pull this information from a dialplan native application, but you could probably write an AGI that pulls this information for you. The AGI Environment data includes things like the current channel in use, which should be able to start you off in the right direction. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about queues.conf
yes if i chang it from queues or meetme to dial there is no issue it'works withou issue i call the same numbers 560 and i can reponse this call in sip 1000 without issue exten = 560,1,Dial(SIP/1000, 30) the asterisk version is Asterisk 1.4-r110474M zaptel-1.4.12.1 i want to know also why for the first provider we put all the number in extensions .conf but for the second provider we put just the last 3 numbers thanks for your help 2011/10/21 Warren Selby wcse...@selbytech.com On Fri, Oct 21, 2011 at 4:50 AM, salaheddine elharit salah.elharit...@gmail.com wrote: -- Accepting call from '522343535' to '560' on channel 1/14, span 1 -- Executing [560@default:1] Answer(Zap/14-1, ) in new stack -- Executing [560@default:2] Queue(Zap/14-1, hotline) in new stack -- Started music on hold, class 'default', on Zap/14-1 -- Channel 1/14, span 1 got hangup, cause -1 -- Stopped music on hold on Zap/14-1 == Spawn extension (default, 560, 2) exited non-zero on 'Zap/14-1' -- Hungup 'Zap/14-1' please advice Do any calls from this provider work? In other words, if you changed it from Queue() to Dial() your sip extension (or whatever means you have of answering the call), does it work then or does it also hang up after two seconds? What version of Asterisk and Zaptel are you using? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype Messaging with Asterisk 10?
Since when can someone submit a patch for chan_skype?? Did i miss an announcement that it has been opensourced? I'm under the impression that digium is the only party who *can* extend chan_skype... Paul was a little confused and thought something would have to be added to Asterisk. But, with that said, the source to chan_skype.c is available in the download and could be modified (just the library for interacting with the underlying Skype library is binary-only). Only chan_skype.c would need to be modified for messaging API support. Basically, just modify new_chat_message() to allocate a new ast_msg, fill in the appropriate values with the ast_msg_set_set_() functions, and then call ast_msg_queue() on it. Terry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about queues.conf
On Fri, Oct 21, 2011 at 12:24 PM, salaheddine elharit salah.elharit...@gmail.com wrote: yes if i chang it from queues or meetme to dial there is no issue it'works withou issue Please do the call again, this time please show us the output also with a sip debug and a zap debug. the asterisk version is Asterisk 1.4-r110474M zaptel-1.4.12.1 These are both very old versions. The current release of asterisk is currently five generations newer than what you're using, and Zaptel isn't even used anymore, the tool was renamed to DAHDI. It may make more sense to update to the latest version of at least the 1.4 branch of asterisk (currently 1.4.42 I think?) and make the switch to DAHDI. This will require some effort on your part, so don't do this without planning on a production box. i want to know also why for the first provider we put all the number in extensions .conf but for the second provider we put just the last 3 numbers I don't know why you only need 3 numbers for your second provider, perhaps that's all that they are sending you? You will probably need to ask the provider why they are not sending you the full number like you're expecting. There may be more reasons hidden in your extensions.conf, if you want to share it maybe someone here can go over it and spot anything that sticks out? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype Messaging with Asterisk 10?
On 11-10-21 01:25 PM, Terry Wilson wrote: Since when can someone submit a patch for chan_skype?? Did i miss an announcement that it has been opensourced? I'm under the impression that digium is the only party who *can* extend chan_skype... Paul was a little confused and thought something would have to be added to Asterisk. But, with that said, the source to chan_skype.c is available in the download and could be modified (just the library for interacting with the underlying Skype library is binary-only). Only chan_skype.c would need to be modified for messaging API support. Basically, just modify new_chat_message() to allocate a new ast_msg, fill in the appropriate values with the ast_msg_set_set_() functions, and then call ast_msg_queue() on it. Terry What Terry said. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any help with these error messages???
Moved to Asterisk 1.8.7, most of the watnings/errors are gone. I have a new error though: [Oct 21 13:40:40] ERROR[15709] ais/clm.c: Could not initialize cluster membership service: Try Again And I get a warning that no music on hold classes are configured. Never mind. The chan_dahdi warnings about the Pseudo channel was already fixed in -r331955 of the v1.8 SVN branch and is in v1.8.7. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI-Linux 2.5.0.2 and DAHDI-Tools 2.5.0.2 Released
The Asterisk Development Team is pleased to announce the release of DAHDI-Linux 2.5.0.2 and DAHDI-Tools 2.5.0.2. 2.5.0.2 is a bug fix release. It is recommended that current users of v2.5 to upgrade. DAHDI-Linux 2.5.0.2, DAHDI-Tools 2.5.0.2, and DAHDI-Linux-Complete 2.5.0.1+2.5.0.1 are available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete Issues closed by this release: DAHLIN-257: wcb4xxp shows hardware EC in /proc/dahdi/* for B410P although vpmsupport=0 DAHLIN-260: wctdm24xxp/base.c fails to compile in 2.6.16 kernel The DAHDI-Linux short log from v2.5.0.1 is: Shaun Ruffell (8): dahdi: Decrease the initial coretimer delay to 4ms from 1 second. wctdm24xxp, wcte12xp: Advertise VPMOCT032 presence in dahdi_span.devicetype. dahdi: Check for master in DAHDI_STARTUP / resolves MeetMe regression. wctdm24xxp: Set dahdi_span.devicetype string in one place. wctc4xxp: Allow G723 SID frames to pass to the hardware decoder. wct4xxp: Fix condition where hardware echo canceler erroneously mutes DTMF. wcb4xxp: Do not show LASVEGAS2 as echocan name if vpmsupport is set to 0 dahdi: Move WARN_ON_ONCE from wctc4xxp driver to include/dahdi/kernel.h Tzafrir Cohen (1): xpp: fxs: bugfix for 2fxs+6fxo cards The DAHDI-Tools short log from v2.5.0.1 is: Shaun Ruffell (2): dahdi_genconf: Assume spans with unknown term types are software selectable. dahdi_genconf: Use 'dahdi_scan' to determine span_type for B410P cards. And the diff stat for this release from v2.5.0.1 is: drivers/dahdi/dahdi-base.c |6 +- drivers/dahdi/wcb4xxp/base.c|2 +- drivers/dahdi/wct4xxp/base.c|2 -- drivers/dahdi/wctc4xxp/base.c | 21 - drivers/dahdi/wctdm24xxp/base.c | 31 +++ drivers/dahdi/wcte12xp/base.c |8 +++- drivers/dahdi/xpp/card_fxs.c| 39 +++ include/dahdi/kernel.h | 13 + 8 files changed, 84 insertions(+), 38 deletions(-) and for DAHDI-Tools: xpp/perl_modules/Dahdi/Span.pm | 37 +++-- 1 files changed, 35 insertions(+), 2 deletions(-) For a full list of changes in these releases, please see the ChangeLog at http://svn.asterisk.org/svn/dahdi/linux/tags/2.5.0.2/ChangeLog and http://svn.asterisk.org/svn/dahdi/tools/tags/2.5.0.2/ChangeLog . Issues found in this release can be reported in the DAHDI-Linux [1] and DAHDI-Tools [2] projects at https://issues.asterisk.org/jira [1] https://issues.asterisk.org/jira/browse/DAHLIN [2] https://issues.asterisk.org/jira/browse/DAHTOOL Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
