[asterisk-users] Asterisk 12 questions
Hi. I'm attempting to compile Asterisk 12, but we want to use chan_sip instead of pjsip. I am missing something. I assumed that chan_sip was going to be added by default. Apparently not. I saw it in the menuconfig. Dumb question, but double xx beside It..does that mean not avail/not going to be installed? Can someone point me in the proper direction? A specific area where I can learn to build Asterisk 12 with chan_sip? I'm needing to use the chan_sip for the time being until I can learn the new SIP stack. Thanks - much appreciated! Glen Sent from my Android - if my spelling, diction or grammar is poor, please don't think that I am an idiot! -- _ -- 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 12 questions
On Thu, Jan 30, 2014 at 12:48 PM, Glen Millard glenmill...@gmail.comwrote: Hi. I'm attempting to compile Asterisk 12, but we want to use chan_sip instead of pjsip. Hi Glenn, I am missing something. I assumed that chan_sip was going to be added by default. Apparently not. I saw it in the menuconfig. Dumb question, but double xx beside It..does that mean not avail/not going to be installed? Yes, if theres x's it means it can't be installed - due to lack of a dependency, rather than it being an option which has been enabled or disabled. Can someone point me in the proper direction? A specific area where I can learn to build Asterisk 12 with chan_sip? I'm needing to use the chan_sip for the time being until I can learn the new SIP stack. Have you tried running ./contrib/scripts/install_prereq ? this should install everything you need if you're compiling from source. http://svn.asterisk.org/svn/asterisk/branches/12/contrib/scripts/install_prereq I don't know what you need to be able to install chan_sip as I've never had to have a look, but your first port of call should be the install_prereq script. Thanks - much appreciated! Glen Dan Sent from my Android - if my spelling, diction or grammar is poor, please don't think that I am an idiot! -- _ -- 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 ISDN
hello there. i do have debian 6 in alix 2d13 that runs asterisk 11.7. i do need to interface an ISDN line to asterisk and i bought an external isdn card usb. it is a dlink du 128ta. i have seen that i need misdn and lcr. in a clean install of debian 6.0.8 i installed through git the misdn and misdn user. unofortunatelly i am stuck with the lcr. i have installed misdn, misdnuser and LCR through git from misdn.eu. i am not familiar with LCR, but with dadhdi. i do not know how to place calls from asterisk through LCR to dlink usb. -- _ -- 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] Parking in Asterisk 12.0.0
Hi I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0. In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked. The syntax for ParkAndAnnounce I used was this (I don't want any announcement to be played): exten = s,n,ParkAndAnnounce(,3600,SIP/100) In the new Asterisk-version, the ParkAndAnnounce application gets called, but the call isn't parked. The only error I can see in the messages file is a DEBUG entry saying that the channel failed to join Bridge, like this: [Jan 30 21:00:01] DEBUG[7118][C-]: bridge_channel.c:1994 bridge_channel_internal_join: Bridge 9f437397-4864-4351-bf29-b37e6ccacf12: 0x16e3768(SIP/vpn-sbc-0001) failed to join Bridge Anyone else that has tried to convert old parking functionality into Asterisk 12.0.0 ? features.conf: parkswitch = *#,callee/caller,Macro(parkswitch) extensions.conf: [default] include = parkedcalls [macro-parkswitch] exten = s,1,ParkAndAnnounce(,,PARKED,SIP/100) messages: [Jan 30 21:00:00] DEBUG[7114][C-]: res_rtp_asterisk.c:2847 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at x.x.x.x:9530 [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4050 __ast_read: DTMF begin '*' received on SIP/at-tcty-ssw- [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4061 __ast_read: DTMF begin passthrough '*' on SIP/at-tcty-ssw- [Jan 30 21:00:00] DEBUG[7114][C-]: res_rtp_asterisk.c:2165 ast_rtp_update_source: Setting the marker bit due to a source update [Jan 30 21:00:00] DEBUG[7114][C-]: res_rtp_asterisk.c:2847 create_dtmf_frame: Creating END DTMF Frame: 42 (*), at x.x.x.x:9530 [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:3964 __ast_read: DTMF end '*' received on SIP/at-tcty-ssw-, duration 240 ms [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4005 __ast_read: DTMF end accepted with begin '*' on SIP/at-tcty-ssw- [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4034 __ast_read: DTMF end passthrough '*' on SIP/at-tcty-ssw- [Jan 30 21:00:00] DEBUG[7114][C-]: bridge_channel.c:1174 bridge_channel_feature: DTMF feature string on 0x7f6b8c10f998(SIP/at-tcty-ssw-) is now '*' [Jan 30 21:00:00] DEBUG[7114][C-]: res_rtp_asterisk.c:2847 create_dtmf_frame: Creating BEGIN DTMF Frame: 35 (#), at x.x.x.x:9530 [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4050 __ast_read: DTMF begin '#' received on SIP/at-tcty-ssw- [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4054 __ast_read: DTMF begin ignored '#' on SIP/at-tcty-ssw- [Jan 30 21:00:01] DEBUG[7114][C-]: res_rtp_asterisk.c:2847 create_dtmf_frame: Creating END DTMF Frame: 35 (#), at x.x.x.x:9530 [Jan 30 21:00:01] DTMF[7114][C-]: channel.c:3964 __ast_read: DTMF end '#' received on SIP/at-tcty-ssw-, duration 230 ms [Jan 30 21:00:01] DTMF[7114][C-]: channel.c:4034 __ast_read: DTMF end passthrough '#' on SIP/at-tcty-ssw- [Jan 30 21:00:01] DEBUG[7114][C-]: bridge_channel.c:1174 bridge_channel_feature: DTMF feature string on 0x7f6b8c10f998(SIP/at-tcty-ssw-) is now '*#' [Jan 30 21:00:01] DEBUG[7114][C-]: bridge_channel.c:1185 bridge_channel_feature: DTMF feature hook 0x7f6b8c1d9480 matched DTMF string '*#' on 0x7f6b8c10f998(SIP/ssw-) [Jan 30 21:00:01] DEBUG[7114][C-]: res_rtp_asterisk.c:2165 ast_rtp_update_source: Setting the marker bit due to a source update [Jan 30 21:00:01] DEBUG[7118][C-]: res_rtp_asterisk.c:2165 ast_rtp_update_source: Setting the marker bit due to a source update [Jan 30 21:00:01] DEBUG[7118][C-]: app.c:305 ast_app_exec_macro: SIP/vpn-sbc-0001 Original location: default,,1 [Jan 30 21:00:01] DEBUG[7118][C-]: pbx.c:4875 pbx_extension_helper: Launching 'ParkAndAnnounce' -- Executing [s@macro-parkswitch:1] ParkAndAnnounce(SIP/vpn-sbc-0001, ,,PARKED,SIP/100) in new stack [Jan 30 21:00:01] DEBUG[7118][C-]: bridge.c:486 find_best_technology: Bridge technology softmix does not have any capabilities we want. [Jan 30 21:00:01] DEBUG[7118][C-]: bridge.c:486 find_best_technology: Bridge technology simple_bridge does not have any capabilities we want. [Jan 30 21:00:01] DEBUG[7118][C-]: bridge.c:486 find_best_technology: Bridge technology native_rtp does not have any capabilities we want. [Jan 30 21:00:01] DEBUG[7118][C-]: bridge.c:505 find_best_technology: Chose bridge technology holding_bridge [Jan 30 21:00:01] DEBUG[7118][C-]: bridge.c:771 bridge_base_init: Bridge 9f437397-4864-4351-bf29-b37e6ccacf12: calling holding_bridge technology constructor [Jan 30 21:00:01] DEBUG[7118][C-]: bridge.c:779 bridge_base_init: Bridge
Re: [asterisk-users] Parking in Asterisk 12.0.0
I have converted the normal Park application and I can only alert you about the syntax change. I suspect also in the ParkAndAnnounce command, the parameters are ordered completely different. Leandro 2014-01-30 Anders Larsson aster...@adev.se: Hi I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0. In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked. The syntax for ParkAndAnnounce I used was this (I don't want any announcement to be played): exten = s,n,ParkAndAnnounce(,3600,SIP/100) In the new Asterisk-version, the ParkAndAnnounce application gets called, but the call isn't parked. The only error I can see in the messages file is a DEBUG entry saying that the channel failed to join Bridge, like this: [Jan 30 21:00:01] DEBUG[7118][C-]: bridge_channel.c:1994 bridge_channel_internal_join: Bridge 9f437397-4864-4351-bf29-b37e6ccacf12: 0x16e3768(SIP/vpn-sbc-0001) failed to join Bridge Anyone else that has tried to convert old parking functionality into Asterisk 12.0.0 ? features.conf: parkswitch = *#,callee/caller,Macro(parkswitch) extensions.conf: [default] include = parkedcalls [macro-parkswitch] exten = s,1,ParkAndAnnounce(,,PARKED,SIP/100) messages: [Jan 30 21:00:00] DEBUG[7114][C-]: res_rtp_asterisk.c:2847 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at x.x.x.x:9530 [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4050 __ast_read: DTMF begin '*' received on SIP/at-tcty-ssw- [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4061 __ast_read: DTMF begin passthrough '*' on SIP/at-tcty-ssw- [Jan 30 21:00:00] DEBUG[7114][C-]: res_rtp_asterisk.c:2165 ast_rtp_update_source: Setting the marker bit due to a source update [Jan 30 21:00:00] DEBUG[7114][C-]: res_rtp_asterisk.c:2847 create_dtmf_frame: Creating END DTMF Frame: 42 (*), at x.x.x.x:9530 [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:3964 __ast_read: DTMF end '*' received on SIP/at-tcty-ssw-, duration 240 ms [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4005 __ast_read: DTMF end accepted with begin '*' on SIP/at-tcty-ssw- [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4034 __ast_read: DTMF end passthrough '*' on SIP/at-tcty-ssw- [Jan 30 21:00:00] DEBUG[7114][C-]: bridge_channel.c:1174 bridge_channel_feature: DTMF feature string on 0x7f6b8c10f998(SIP/at-tcty-ssw-) is now '*' [Jan 30 21:00:00] DEBUG[7114][C-]: res_rtp_asterisk.c:2847 create_dtmf_frame: Creating BEGIN DTMF Frame: 35 (#), at x.x.x.x:9530 [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4050 __ast_read: DTMF begin '#' received on SIP/at-tcty-ssw- [Jan 30 21:00:00] DTMF[7114][C-]: channel.c:4054 __ast_read: DTMF begin ignored '#' on SIP/at-tcty-ssw- [Jan 30 21:00:01] DEBUG[7114][C-]: res_rtp_asterisk.c:2847 create_dtmf_frame: Creating END DTMF Frame: 35 (#), at x.x.x.x:9530 [Jan 30 21:00:01] DTMF[7114][C-]: channel.c:3964 __ast_read: DTMF end '#' received on SIP/at-tcty-ssw-, duration 230 ms [Jan 30 21:00:01] DTMF[7114][C-]: channel.c:4034 __ast_read: DTMF end passthrough '#' on SIP/at-tcty-ssw- [Jan 30 21:00:01] DEBUG[7114][C-]: bridge_channel.c:1174 bridge_channel_feature: DTMF feature string on 0x7f6b8c10f998(SIP/at-tcty-ssw-) is now '*#' [Jan 30 21:00:01] DEBUG[7114][C-]: bridge_channel.c:1185 bridge_channel_feature: DTMF feature hook 0x7f6b8c1d9480 matched DTMF string '*#' on 0x7f6b8c10f998(SIP/ssw-) [Jan 30 21:00:01] DEBUG[7114][C-]: res_rtp_asterisk.c:2165 ast_rtp_update_source: Setting the marker bit due to a source update [Jan 30 21:00:01] DEBUG[7118][C-]: res_rtp_asterisk.c:2165 ast_rtp_update_source: Setting the marker bit due to a source update [Jan 30 21:00:01] DEBUG[7118][C-]: app.c:305 ast_app_exec_macro: SIP/vpn-sbc-0001 Original location: default,,1 [Jan 30 21:00:01] DEBUG[7118][C-]: pbx.c:4875 pbx_extension_helper: Launching 'ParkAndAnnounce' -- Executing [s@macro-parkswitch:1] ParkAndAnnounce(SIP/vpn-sbc-0001, ,,PARKED,SIP/100) in new stack [Jan 30 21:00:01] DEBUG[7118][C-]: bridge.c:486 find_best_technology: Bridge technology softmix does not have any capabilities we want. [Jan 30 21:00:01] DEBUG[7118][C-]: bridge.c:486 find_best_technology: Bridge technology simple_bridge does not have any capabilities we want. [Jan 30 21:00:01] DEBUG[7118][C-]: bridge.c:486 find_best_technology: Bridge technology native_rtp does not have any capabilities we want. [Jan 30 21:00:01] DEBUG[7118][C-]: bridge.c:505 find_best_technology: Chose bridge technology
Re: [asterisk-users] Parking in Asterisk 12.0.0
On Thu, Jan 30, 2014 at 2:58 PM, Leandro Dardini ldard...@gmail.com wrote: I have converted the normal Park application and I can only alert you about the syntax change. I suspect also in the ParkAndAnnounce command, the parameters are ordered completely different. Leandro Please go ahead an open an issue for this - issues.asterisk.org. The problem here is that you are attempting to enter into a Parking bridge while you are still technically in a bridge. The DTMF features that account for the 'normal' mechanism of doing this - the one touch parking feature - recognize that you are in a bridge and do a safe transfer from the existing bridge to the parking bridge. By jumping out to a macro/gosub and directly going in through the ParkAndAnnounce application, you are bypassing that logic. The code in bridge_channel_internal_join is preventing you from going into the parking bridge as it knows that you have not yet safely left the bridge you are in. We'll take a look and see if there's a way to allow this to happen again. For now, you should use the one touch parking feature. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] callerid overwrite
look like the issue continues, I am unable to overwrite callerid from
sip.conf in extensions.conf,
In sip.conf under
[general]
trustrpid = no should i change it to yes?
Thanks
On Tue, Jan 28, 2014 at 1:06 PM, motty cruz motty.c...@gmail.com wrote:
Thank you for your reply, I updated extensions.conf file to reflect your
suggestion, I will monitor Asterisk for any more issues,
Thanks,
On Tue, Jan 28, 2014 at 11:23 AM, Andres and...@telesip.net wrote:
On 1/28/14, 1:55 PM, motty cruz wrote:
Hi all,
I'm having issues with overwrite caller id, when I call someone my caller
id should be mycompanyinc but instead my id shows up as my extension
number 101.
this is what i have in sip.conf
[101]
type=friend
context=sipphones
call-limit=99
callerid=iuser 101
disallow=all
allow=ulaw
allow=alaw
username=101
secret=Passwd
dtmfmode=rfc2833
host=dynamic
mailbox=101@default
nat=yes
canreinvite=no
this is what i have in extensions.conf
[outbound]
exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)
This is how we have it and it works fine on Asterisk 1.8:
Set(CALLERID(number)=insert your number here)
exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)
any ideas? as this happens random,
--
Technical Supporthttp://www.cellroute.net
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[asterisk-users] DAHDI-Linux and DAHDI-Tools 2.9.0 Now Available
The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.0 DAHDI-Tools-v2.9.0 dahdi-linux-complete-2.9.0+2.9.0 This release is 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 - Introduces support for Digium's new TE131 and TE132 products. - Updates firmware for existing TE133 and TE134 products. - New documentation and support tool improvements for configurable span/channel numbering - Currently, span/channel ordering is determined by module load order - Work arounds are used to specify channel assignment order by blacklisting all modules and then loading them in a specific order to preserve channel assignments. - We have been driving towards moving span/chan assignments out of kernel space and into user space. - This is a much more robust solution which allows for: - hotplugging, surprise device removal and installation while maintaining channel ordering - parallel module loading (much faster booting on dense systems) - discrete control over span and channel ordering via configuration files - sticky channel assignments which can be tied to specific hardware ids or pci slots - This new system is enabled by setting the module parameter of dahdi auto_assign_spans=0 - More info here: http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/278656/match=auto_assigned_spans Shortlog of dahdi-linux changes since v2.8.0.1: Oron Peled (3): xpp: deprecate dahdi_autoreg xpp: continue xpp.dahdi_autoreg deprecation sysfs: new device attribute: registration_time Russ Meyerriecks (6): wcte13xp: wcaxx: Fix broken devicetype attributes wcte13xp: Update firmware to 0x780017 wcte13xp: Add support for te131 and te132 products Revert dahdi: Change auto_assign_spans default from 1 to 0. wcte13xp: wcaxx: wcte43x: Remove VPM_SUPPORT compile option. wcte13xp: wcxb: Add delayed reset firmware feature Shaun Ruffell (10): wctdm24xxp: Reset module specific type information on probe. dahdi: Move clearing of DAHDI_ALARM_NOTOPEN to __dahdi_assign_span(). dahdi: Change auto_assign_spans default from 1 to 0. wcaxx, wcte13xp, wcte43x: Honor max_latency module parameter. wcte13xp: Export max_latency module parameter. wcte43x, wcte13xp: Use MSI interrupts if possible. dahdi: Do not access invalid memory if invalid local span number is passed to spantype attribute. wcte43x: Trivial drop of unnecessary local variables. wct4xxp: Trivial drop of unnecessary local variables. wcte43x, wcte13xp, wcaxx: Bump irqmisses counter when there are DMA underruns. Tzafrir Cohen (4): README: xpp.dahdi_autoreg is deprecated README: the new registration_time device attribute README: The sysfs class now includes no channels sysfs: registration_time: use ktime_get_ts Shortlog of dahdi-tools changes since v2.8.0: Oron Peled (6): Makefile: do install all man-pages hotplug modularization: move sources to a subdir hotplug modularization: split logic to scriptlets new dahdi_waitfor_span_assignments tool dahdi_span_types: allow defaults + overrides Change span-type.conf generation policy Russ Meyerriecks (2): wcte13xp: Teach tools about te131 te132 products dahdi.init: Don't exit on lack of /etc/dahdi/system.conf Shaun Ruffell (8): dahdi_cfg: Wait for all spans to be assigned. dahdi_span_config: Do not run auto span configuration if spans are auto assigned. dahdi_handle_device, dahdi_span_config: Check for auto_assign_spans only when ACTION is add. dahdi_genconf: Add 'modules', 'spantypes', and 'assignedspans' to list of available generators. dahdi_span_types: Show location of configuration file in help message. dahdi_handle_device: Auto assign only the device being added. dahdi_cfg: Add semaphore to prevent parallel execution. dahdi_cfg: Allow dynamic spans to handle udev based span assignment. Tzafrir Cohen (16): dahdi.rules: Replace SYSFS with ATTRS dahdi.rules: use += for RUN .gitignore: more generated files README: indentation level for config samples README: document initialization README: Update the install targets span_types/assignments: no * in device list dahdi_genconf: don't generate spantypes by default dahdi_span_assignments.8: s/register/assign/ dahdi_span_types: hush warning of missing attribute programmable bash completion for some commands dahdi_perl: fix regression with an AB with no modules bash_completion: fix dahdi_genconf hyphen/minus fixes in man pages hotplug: document asterisk scriptlet README: udev hooks run scripts from directories The diffstat from the dahdi-linux
[asterisk-users] how to get full channel name - AMI cuts off
Using Dahdi/PRI, I end up with channel names like 'DAHDI/i8/9995551212-4d6B', but when I do a 'core show channels' it cuts off those names to only 'DAHDI/i8/9995551212-'. This is the same for AMI. Is there a way to get the full channel name within AMI? I'm using asterisk 11.7.0 Thanks, -Justin -- _ -- 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 get full channel name - AMI cuts off [solved]
After posting this, I ran across 'core channel show concise', which gives the data in a more machine friendly format. -Justin From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Thursday, January 30, 2014 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] how to get full channel name - AMI cuts off Using Dahdi/PRI, I end up with channel names like 'DAHDI/i8/9995551212-4d6B', but when I do a 'core show channels' it cuts off those names to only 'DAHDI/i8/9995551212-'. This is the same for AMI. Is there a way to get the full channel name within AMI? I'm using asterisk 11.7.0 Thanks, -Justin -- _ -- 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 12 questions
Looks like I figured It out! A couple of things seemed to be getting in the way: 1. Old leftovers - I had a previous version of Asterisk kicking about. I used the package manager and removed It. 2. Openssl-dev libraries. Learned that it's a dependency of chan_sip. I am embarrassed to say that I did not know! Now, I've a working version of Asterisk 12 with chan_sip . Now - to decipher the AMI... Glen Sent from my Android - if my spelling, diction or grammar is poor, please don't think that I am an idiot! On Jan 30, 2014 7:53 AM, Daniel Jenkins dan.jenkin...@gmail.com wrote: On Thu, Jan 30, 2014 at 12:48 PM, Glen Millard glenmill...@gmail.comwrote: Hi. I'm attempting to compile Asterisk 12, but we want to use chan_sip instead of pjsip. Hi Glenn, I am missing something. I assumed that chan_sip was going to be added by default. Apparently not. I saw it in the menuconfig. Dumb question, but double xx beside It..does that mean not avail/not going to be installed? Yes, if theres x's it means it can't be installed - due to lack of a dependency, rather than it being an option which has been enabled or disabled. Can someone point me in the proper direction? A specific area where I can learn to build Asterisk 12 with chan_sip? I'm needing to use the chan_sip for the time being until I can learn the new SIP stack. Have you tried running ./contrib/scripts/install_prereq ? this should install everything you need if you're compiling from source. http://svn.asterisk.org/svn/asterisk/branches/12/contrib/scripts/install_prereq I don't know what you need to be able to install chan_sip as I've never had to have a look, but your first port of call should be the install_prereq script. Thanks - much appreciated! Glen Dan Sent from my Android - if my spelling, diction or grammar is poor, please don't think that I am an idiot! -- _ -- 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] how to get full channel name - AMI cuts off [solved]
On Thu, Jan 30, 2014 at 5:48 PM, Justin Killen jkil...@allamericanasphalt.com wrote: After posting this, I ran across 'core channel show concise', which gives the data in a more machine friendly format. That may work over AMI, but in general, it isn't recommended. The command class authorization, EVENT_CLASS_COMMAND, is relatively powerful and shouldn't be exposed to a general AMI action without a lot of forethought. What's more, CLI commands are generally viewed as being appropriate for end users, and not programs controlling Asterisk. While 'core show channels concise' is unlikely to change in future versions, it certainly isn't versioned in the same fashion as AMI events/actions. If you need to get a dump of all active channels in the system over AMI, I'd recommend the CoreShowChannels AMI action [1]. It will send the information back for each channel as an event, and doesn't require the same level of permission as the corresponding CLI command. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_CoreShowChannels Matt -- _ -- 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] callerid overwrite
Howdy,
Before changing any configuration I would highly recommend reading through
the entry in the sample file. Trust remote party ID may be set to 'no' for
a very good reason on your PBX, please take care to understand why it
should be changed before doing so.
Before digging into that though, what does the CLI tell you if you do a
NoOp() after having Set() the Caller ID function [1]?
[1] Something like;
exten = _9NXX,1,Set(CALLERID(name)=mycompanyinc)
same = n,NoOp(The caller ID has been set to ${CALLERID(name)})
same = n,Dial(SIP/att/${EXTEN:1},80)
Hope this helps.
Justin Hester
Digium, Inc. · Technical Trainer
445 Jan Davis Drive NW · Huntsville, AL 35806 · USA
ph: +1 256 428 6238
Check us out at: http://digium.com · http://asterisk.org
On Thu, Jan 30, 2014 at 5:29 PM, motty cruz motty.c...@gmail.com wrote:
look like the issue continues, I am unable to overwrite callerid from
sip.conf in extensions.conf,
In sip.conf under
[general]
trustrpid = no should i change it to yes?
Thanks
On Tue, Jan 28, 2014 at 1:06 PM, motty cruz motty.c...@gmail.com wrote:
Thank you for your reply, I updated extensions.conf file to reflect your
suggestion, I will monitor Asterisk for any more issues,
Thanks,
On Tue, Jan 28, 2014 at 11:23 AM, Andres and...@telesip.net wrote:
On 1/28/14, 1:55 PM, motty cruz wrote:
Hi all,
I'm having issues with overwrite caller id, when I call someone my
caller id should be mycompanyinc but instead my id shows up as my
extension number 101.
this is what i have in sip.conf
[101]
type=friend
context=sipphones
call-limit=99
callerid=iuser 101
disallow=all
allow=ulaw
allow=alaw
username=101
secret=Passwd
dtmfmode=rfc2833
host=dynamic
mailbox=101@default
nat=yes
canreinvite=no
this is what i have in extensions.conf
[outbound]
exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)
This is how we have it and it works fine on Asterisk 1.8:
Set(CALLERID(number)=insert your number here)
exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)
any ideas? as this happens random,
--
Technical Supporthttp://www.cellroute.net
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