Re: [asterisk-users] rtcachefriends qualify sip reload
From my experience prune does not take effect without reload. And after reload ALL your phones are unreachable for 2 minutes! Imagine you have several thousands devices unreachable for 2 minutes. How much calls will fail during that time? Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Tuesday, March 02, 2010 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] rtcachefriends qualify sip reload On Tue, 2010-03-02 at 15:59 +0100, jonas kellens wrote: On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote: If you are changing RealTime config in your DB you need to do a sip prune realtime either directly from asterisk cli or using AMI. You really do not need to do a SIP reload when changing the config of one sip extension. I notice that after a sip prune realtime all I also loose all of my realtime sip peers. Same result actually as with sip reload. I close the softphone of gerrie2 (becomes unspecified) asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime gerrie005/gerrie005192.168.1.106D N 5060 OK (4 ms) Cached RT gerrie002/gerrie002(Unspecified)D N 0 UNKNOWNCached RT gerrie001/gerrie001192.168.1.105D N 5060 OK (11 ms) Cached RT I prune the realtime peers to no longer have gerrie002 in cache : asterisk*CLI sip prune realtime all 3 peers pruned. 2 users pruned. [Mar 2 15:42:19] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 91 The realtime peers are all gone : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime Internal call fails : [Mar 2 15:46:38] WARNING[558]: app_dial.c:1272 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Mar 2 15:46:38] == Everyone is busy/congested at this time (1:0/0/1) [Mar 2 15:46:38] == Auto fallthrough, channel 'SIP/gerrie001-09f631e0' status is 'CHANUNAVAIL' I re-register 2 softphones (gerrie001 gerrie005) : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime gerrie002/gerrie002(Unspecified)D N 0 UNREACHABLE Cached RT gerrie001/gerrie001192.168.1.105D N 5060 OK (11 ms) Cached RT gerrie005/gerrie005192.168.1.106D N 5060 OK (7 ms) Cached RT The SIP-peer 'gerrie002' is still in the cache ! Don't know where this is coming from ?? I prune again : asterisk*CLI sip prune realtime all 3 peers pruned. 1 users pruned. [Mar 2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 [Mar 2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 [Mar 2 15:52:01] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 And again no more peers until I re-register : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime This realtime thing isn't really working out here... What exactly do I need to do to clear the cache and thus the old SIP-peers so they can no longer be used ?? Do not prune all peers, only the peer you wish to reload or eliminate! Do sip prune realtime peer peername. That way you do not lose all the other registrations. I really do not see this as a problem as the phones will usually re register quickly or if the user dials any number. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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] realtime call peers status
Hi The link you put in your email was the starting point that I used myself. It should give you a good grounding of where to start and how to proceed. Ish lore wrote: Hi, thanks a lot for the reply, yes I would like to put data in a web interface (maybe php made better if already done :) ). I'm reading something about dymanic realtime: could be ok for my needs? Or is better spent my time on this docs : http://www.voip-info.org/wiki/view/Asterisk+manager+API ? 2010/3/2 Ishfaq Malik i...@pack-net.co.uk: lore wrote: Hi all, I need to check in realtime the calls that my asterisk is menaging: 1) SIP peers status and with who are talking. 2) IAX peers status and with who are talking 3) elapsed talking time Some one could show me the way to realize that? Any help are really appreciated Thanks a lot in advance From asterisk cli core show channels core show channel channel name If you need to put it into a pretty front end you can use the AMI Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] rtcachefriends qualify sip reload
Hi We run production servers for various customers all using realtime with web interfaces so they can change their own config whenever they want. Prune works fine for us and we never do sip reloads (1.4.17) Ish Mindaugas Kezys wrote: From my experience prune does not take effect without reload. And after reload ALL your phones are unreachable for 2 minutes! Imagine you have several thousands devices unreachable for 2 minutes. How much calls will fail during that time? Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Tuesday, March 02, 2010 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] rtcachefriends qualify sip reload On Tue, 2010-03-02 at 15:59 +0100, jonas kellens wrote: On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote: If you are changing RealTime config in your DB you need to do a sip prune realtime either directly from asterisk cli or using AMI. You really do not need to do a SIP reload when changing the config of one sip extension. I notice that after a sip prune realtime all I also loose all of my realtime sip peers. Same result actually as with sip reload. I close the softphone of gerrie2 (becomes unspecified) asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime gerrie005/gerrie005192.168.1.106D N 5060 OK (4 ms) Cached RT gerrie002/gerrie002(Unspecified)D N 0 UNKNOWNCached RT gerrie001/gerrie001192.168.1.105D N 5060 OK (11 ms) Cached RT I prune the realtime peers to no longer have gerrie002 in cache : asterisk*CLI sip prune realtime all 3 peers pruned. 2 users pruned. [Mar 2 15:42:19] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 91 The realtime peers are all gone : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime Internal call fails : [Mar 2 15:46:38] WARNING[558]: app_dial.c:1272 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Mar 2 15:46:38] == Everyone is busy/congested at this time (1:0/0/1) [Mar 2 15:46:38] == Auto fallthrough, channel 'SIP/gerrie001-09f631e0' status is 'CHANUNAVAIL' I re-register 2 softphones (gerrie001 gerrie005) : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime gerrie002/gerrie002(Unspecified)D N 0 UNREACHABLE Cached RT gerrie001/gerrie001192.168.1.105D N 5060 OK (11 ms) Cached RT gerrie005/gerrie005192.168.1.106D N 5060 OK (7 ms) Cached RT The SIP-peer 'gerrie002' is still in the cache ! Don't know where this is coming from ?? I prune again : asterisk*CLI sip prune realtime all 3 peers pruned. 1 users pruned. [Mar 2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 [Mar 2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 [Mar 2 15:52:01] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 And again no more peers until I re-register : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status Realtime This realtime thing isn't really working out here... What exactly do I need to do to clear the cache and thus the old SIP-peers so they can no longer be used ?? Do not prune all peers, only the peer you wish to reload or eliminate! Do sip prune realtime peer peername. That way you do not lose all the other registrations. I really do not see this as a problem as the phones will usually re register quickly or if the user dials any number. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] dahdi and oslec
On Wed, 2010-03-03 at 11:31 +0530, Chandrakant Solanki wrote: Hi All, I have followed below steps to enable echo cancellation. # cd /usr/src # wget http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.28.tar.bz2 # tar xjf linux-2.6.28.tar.bz2 # tar zxvf dahdi-linux-2.1.0.4.tar.gz # ln -s /usr/src/dahdi-linux-2.1.0.4 /usr/src/dahdi # mkdir /usr/src/dahdi/drivers/staging # cp -fR /usr/src/linux-2.6.28/drivers/staging/echo /usr/src/dahdi/drivers/staging # sed -i s|#obj-m += dahdi_echocan_oslec.o|obj-m += dahdi_echocan_oslec.o| /usr/src/dahdi/drivers/dahdi/Kbuild # sed -i s|#obj-m += ../staging/echo/|obj-m += ../staging/echo/| /usr/src/dahdi/drivers/dahdi/Kbuild # echo 'obj-m += echo.o' /usr/src/dahdi/drivers/staging/echo/Kbuild # cd /usr/src/dahdi # make # make install # cd /usr/src # tar zxvf dahdi-tools-2.1.0.2.tar.gz # cd /usr/src/dahdi-tools-2.1.0.2 # ./configure # make # make install # wget http://www.rowetel.com/ucasterisk/downloads/oslec-0.2.tar.gz # tar xvzf oslec-0.2.tar.gz # cd oslec-0.2 # make # insmod kernel/oslec.ko when i restart /etc/init.d/dahdi service it gives me following error in /var/log/message Mar 3 11:06:37 server1 kernel: echo: exports duplicate symbol oslec_hpf_tx (owned by oslec) Mar 3 11:06:37 server1 modprobe: WARNING: Error inserting echo (/lib/modules/2.6.18-92.1.22.el5/staging/echo/echo.ko): Invalid module format Mar 3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol oslec_create Mar 3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol oslec_update Mar 3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol oslec_free Mar 3 11:06:37 server1 modprobe: FATAL: Error inserting dahdi_echocan_oslec (/lib/modules/2.6.18-92.1.22.el5/dahdi/dahdi_echocan_oslec.ko): Unknown symbol in module, or unknown parameter (see dmesg) # cat /etc/dahdi/system.conf loadzone= in defaultzone = in span=1,1,7,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 echocanceller=oslec,1-15,17-31 Is there anything missing or i am going wrong.. Help me out. Thanks in advance... -- Regards, Chandrakant Solanki -- _ -- 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 hehe ;) You are already built dahdi with oslec. You will not load manually this module. Try! Build dahdi, modprobe your module (my module is wcfxo) modprobe wcfxo: (dmesg) wcfxo :00:09.0: PCI INT A - GSI 17 (level, low) - IRQ 17 wcfxo: DAA mode is 'FCC' cat /etc/dahdi/system.conf fxsks = 1 echocanceller =oslec,1-240 loadzone = ru defaultzone = ru dahdi_cfg -vv DAHDI Tools Version - 2.2.0 * Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to oslec Hope it helps.. -- Best regards, Vince Mallow xmpp: w...@jabber.slan.ru web: http://gentoo-way.blogspot.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
[asterisk-users] Getting verbose or debug tracing in Asterisk
Hi, For some reason I can't get Asterisk to produce debug or verbose tracing output. I connect to asterisk using 'asterisk -r' Then issue the command: Core set debug 10 And Core set verbose 10 And it confirms that the correct level has been set. I then attempt a connection from an x-lite client. However, no debug or verbose output appears in the console or in any of the log files in /var/log/asterisk. I'm using Asterisk 1.6.2.2. Anybody know what I'm doing wrong? Thanks, Tim - Tim Culhane, Critical Path Ireland, 42-47 Lower Mount Street, Dublin 2. Direct line: 353-1-2415107 phone: 353-1-2415000 tim.culh...@criticalpath.net http://www.criticalpath.net Critical Path a global leader in digital communications -- _ -- 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] how can I release trunks after transferring 2 calls connected on trunks between the same machines.
Hello, I made 3 questions because they are linked and actually dealing with the same need of releasing trunks after transferring 2 calls connected on trunks between the same machines. 1) I have a machine with Asterisk 1.4 connected with a SIP trunk to a PBX. A (on the PBX) calls B (a SIP phone on Asterisk). B answers and puts A on hold. Then B calls C (on the PABX) and does an attended transfer. How should I configure Asterisk to send a SIP REFER message to the PBX at the transfer ? I need it to release the SIP trunks and let the transferees talk through the PBX without involving Asterisk any longer in their call. 2) If I connect a first Asterisk via SIP trunk to a second Asterisk 1.6 and connect this second Asterisk to a PBX via QSIG. A (on the PBX) calls B (a SIP phone on the first Asterisk). B answers and puts A on hold. Then B calls C (on the PABX) and does an attended transfer. Provided the first Asterisk can send a REFER message to the second Asterisk, how should I configure the second Asterisk to make it send the message Facility: Call Transfer Complete to the PBX via QSIG at this transfer ? I need it to allow the PBX to reply with the message Path Replacement Purpose, have the QSIG trunks released and let the transferees talk through the PBX without involving the 2 Asterisk any longer in their call. Which hw/QSIG board shall I use on the second Asterisk to get this behavior ? Any PRI board ? 3) This is basically the same question of 2) but with a IAX trunk. If I connect a first Asterisk via IAX trunk to a second Asterisk 1.6 and connect this second Asterisk to a PBX via QSIG. A (on the PBX) calls B (a SIP phone on the first Asterisk). B answers and puts A on hold. Then B calls C (on the PABX) and does an attended transfer. At the transfer, the first Asterisk correctly sends TXREQ to the second Asterisk. How should I configure the second Asterisk to make it send the message Facility: Call Transfer Complete to the PBX via QSIG at this transfer ? I need it to allow the PBX to reply with the message Path Replacement Purpose, have the QSIG trunks released and let the transferees talk through the PBX without involving the 2 Asterisk any longer in their call. Thank you and Regards, Raoul Trevisi -- _ -- 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] Echo cancellation on DAHDI
- DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu: Hi, Carlos I checked dmesg on my server and i found following message what is meaning for this ? i cant understand VPM400: Not Present VPM450: echo cancellation for 128 channels VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) regards Dhaval That means you have a VPM450 echo cancelling module attached to your digital board. All you need to do in order to activate echo cancelling is setting echocancel=yes on your chan_dahdi.conf. After that you can check if the echo canceller is really enabled by triggering a dahdi show channel X on the Asterisk CLI. X should be a channel that's currently on a call. Here's an example: stara*CLI dahdi show channel 64 Channel: 64 File Descriptor: 77 Span: 3LI Extension: Dialing: no Context: pabx Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: ISDN PRI Radio: 0I Owner: DAHDI/64-1 Real: DAHDI/64-1 Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently ON PRI Flags: Call PRI Logical Span: Implicit Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook Of course, the interesting line to you is the Echo Cancellation one. -- _ -- 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] cli_originate malfunction after upgrade from 1.6.2.0 to 1.6.2.1-5
On Tue, Mar 02, 2010 at 12:44:06PM +0100, Andreas Brodmann wrote: Hi all, We encountered a strange phenomenon when trying to upgrade from 1.6.2.0 to any newer releases: We use the following cli command to feed a wave/mp3 file into an existing conference on an other serve: /opt/asterisk/sbin/asterisk -r -x channel originate Local/confgongad...@xy_features extension confgongp...@xy_features The corresponding extensions.conf part looks like that: -- [XY_Features] exten = ConfGongAdmin,1,NoCDR() exten = ConfGongAdmin,n,Set(TIMEOUT(absolute)=10) exten = ConfGongAdmin,n,Dial(SIP/12...@server) exten = ConfGongPlay,1,Answer() exten = ConfGongPlay,n,Set(TIMEOUT(absolute)=10) exten = ConfGongPlay,n,Wait(2) exten = ConfGongPlay,n,Playback(/etc/asterisk/sounds/gong) --- Until asterisk-1.6.2.0 this worked fine. With later releases including 1.6.2.5 asterisk does a call to confgongad...@xy_features but once that stands does not continue with a call to ConfGongPlay. IIRC this issue is fixed in latest SVN, and also in 1.2.6.3-rc2 (1.2.6.5 is based on 1.2.6.2). -- 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 -- 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] dialplan reload: not working with large dialplans
On Tue, Mar 02, 2010 at 03:19:36PM +0100, Andreas Brodmann wrote: Hi Tzafrir, yes, I will have to 'anonymize' the dialplan, Can you reproduce it with any other large dialplan? is this list the right place though? That, or a bug report in http://issues.asterisk.org/ . -- 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 -- 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] Getting verbose or debug tracing in Asterisk
In logger.conf do you have verbose and debug on the console line? If not add
them and do logger reload.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 3, 2010, at 2:13 AM, Tim Culhane wrote:
Hi,
For some reason I can't get Asterisk to produce debug or verbose tracing
output.
I connect to asterisk using 'asterisk -r'
Then issue the command:
Core set debug 10
And
Core set verbose 10
And it confirms that the correct level has been set.
I then attempt a connection from an x-lite client.
However, no debug or verbose output appears in the console or in any of
the log files in /var/log/asterisk.
I'm using Asterisk 1.6.2.2.
Anybody know what I'm doing wrong?
Thanks,
Tim
-
Tim Culhane,
Critical Path Ireland,
42-47 Lower Mount Street,
Dublin 2.
Direct line: 353-1-2415107
phone: 353-1-2415000
tim.culh...@criticalpath.net
http://www.criticalpath.net
Critical Path
a global leader in digital communications
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Re: [asterisk-users] dahdi and oslec
You might have to load the canceller with a modprobe (modprobe mg2 for example) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of wins mallow Sent: Wednesday, March 03, 2010 1:28 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] dahdi and oslec On Wed, 2010-03-03 at 11:31 +0530, Chandrakant Solanki wrote: Hi All, I have followed below steps to enable echo cancellation. # cd /usr/src # wget http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.28.tar.bz2 # tar xjf linux-2.6.28.tar.bz2 # tar zxvf dahdi-linux-2.1.0.4.tar.gz # ln -s /usr/src/dahdi-linux-2.1.0.4 /usr/src/dahdi # mkdir /usr/src/dahdi/drivers/staging # cp -fR /usr/src/linux-2.6.28/drivers/staging/echo /usr/src/dahdi/drivers/staging # sed -i s|#obj-m += dahdi_echocan_oslec.o|obj-m += dahdi_echocan_oslec.o| /usr/src/dahdi/drivers/dahdi/Kbuild # sed -i s|#obj-m += ../staging/echo/|obj-m += ../staging/echo/| /usr/src/dahdi/drivers/dahdi/Kbuild # echo 'obj-m += echo.o' /usr/src/dahdi/drivers/staging/echo/Kbuild # cd /usr/src/dahdi # make # make install # cd /usr/src # tar zxvf dahdi-tools-2.1.0.2.tar.gz # cd /usr/src/dahdi-tools-2.1.0.2 # ./configure # make # make install # wget http://www.rowetel.com/ucasterisk/downloads/oslec-0.2.tar.gz # tar xvzf oslec-0.2.tar.gz # cd oslec-0.2 # make # insmod kernel/oslec.ko when i restart /etc/init.d/dahdi service it gives me following error in /var/log/message Mar 3 11:06:37 server1 kernel: echo: exports duplicate symbol oslec_hpf_tx (owned by oslec) Mar 3 11:06:37 server1 modprobe: WARNING: Error inserting echo (/lib/modules/2.6.18-92.1.22.el5/staging/echo/echo.ko): Invalid module format Mar 3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol oslec_create Mar 3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol oslec_update Mar 3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol oslec_free Mar 3 11:06:37 server1 modprobe: FATAL: Error inserting dahdi_echocan_oslec (/lib/modules/2.6.18-92.1.22.el5/dahdi/dahdi_echocan_oslec.ko): Unknown symbol in module, or unknown parameter (see dmesg) # cat /etc/dahdi/system.conf loadzone= in defaultzone = in span=1,1,7,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 echocanceller=oslec,1-15,17-31 Is there anything missing or i am going wrong.. Help me out. Thanks in advance... -- Regards, Chandrakant Solanki -- _ -- 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 hehe ;) You are already built dahdi with oslec. You will not load manually this module. Try! Build dahdi, modprobe your module (my module is wcfxo) modprobe wcfxo: (dmesg) wcfxo :00:09.0: PCI INT A - GSI 17 (level, low) - IRQ 17 wcfxo: DAA mode is 'FCC' cat /etc/dahdi/system.conf fxsks = 1 echocanceller =oslec,1-240 loadzone = ru defaultzone = ru dahdi_cfg -vv DAHDI Tools Version - 2.2.0 * Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to oslec Hope it helps.. -- Best regards, Vince Mallow xmpp: w...@jabber.slan.ru web: http://gentoo-way.blogspot.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] Getting verbose or debug tracing in Asterisk
The problem is on your x-lite end. If you were speaking to Asterisk (even incorrectly), it would at least indicate a bad connection. IMO, it is better to use numbers for extensions as opposed to user1, but that is irrelevant. Make sure the x-lite client has the IP of your asterisk box. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Culhane Sent: Wednesday, March 03, 2010 4:13 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting verbose or debug tracing in Asterisk Hi, For some reason I can't get Asterisk to produce debug or verbose tracing output. I connect to asterisk using 'asterisk -r' Then issue the command: Core set debug 10 And Core set verbose 10 And it confirms that the correct level has been set. I then attempt a connection from an x-lite client. However, no debug or verbose output appears in the console or in any of the log files in /var/log/asterisk. I'm using Asterisk 1.6.2.2. Anybody know what I'm doing wrong? Thanks, Tim - Tim Culhane, Critical Path Ireland, 42-47 Lower Mount Street, Dublin 2. Direct line: 353-1-2415107 phone: 353-1-2415000 tim.culh...@criticalpath.net http://www.criticalpath.net Critical Path a global leader in digital communications -- _ -- 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] CALLERID(num) not working
I am having a problem setting the caller ID that shows when I make an outbound call over my PRI line. If I make a call from a SIP phone registered with the Asterisk box the PRI is connected to the correct ID shows on my cell phone. If I make a call from an IAX trunk connected asterisk box calling the same number as call one and setting the caller ID to the same number as call one the caller ID shown on my cell phone is the default caller ID for the PRI line. The system the PRI line is connected to is running Asterisk 1.6.0.13. Here are the two calls as shown on the CLI of the asterisk box with the PRI line. The first works and the second does not. [2010-03-02 13:32:09.520] == Using SIP RTP TOS bits 184 [2010-03-02 13:32:09.520] == Using SIP RTP CoS mark 5 [2010-03-02 13:32:09.520] == Using SIP VRTP TOS bits 136 [2010-03-02 13:32:09.520] == Using SIP VRTP CoS mark 6 [2010-03-02 13:32:09.617] -- Executing [9111...@context:1] Set(SIP/username-114ffe50, MyChan=SIP) in new stack [2010-03-02 13:32:09.617] -- Executing [9111...@context:2] GotoIf(SIP/username-114ffe50, 0?ISLOCAL) in new stack [2010-03-02 13:32:09.618] -- Executing [9111...@context:3] GotoIf(SIP/username-114ffe50, 0?DODIAL) in new stack [2010-03-02 13:32:09.618] -- Executing [9111...@context:4] Macro(SIP/username-114ffe50, outgoing,) in new stack [2010-03-02 13:32:09.618] -- Executing [...@macro-outgoing:1] GotoIf(SIP/username-114ffe50, 1?NEEDUSER:HAVEUSER) in new stack [2010-03-02 13:32:09.618] -- Goto (macro-outgoing,s,2) [2010-03-02 13:32:09.618] -- Executing [...@macro-outgoing:2] Macro(SIP/username-114ffe50, getmyUserID) in new stack [2010-03-02 13:32:09.619] -- Executing [...@macro-getmyuserid:1] GotoIf(SIP/username-114ffe50, 1?FROMCHAN) in new stack [2010-03-02 13:32:09.619] -- Goto (macro-getmyUserID,s,4) [2010-03-02 13:32:09.619] -- Executing [...@macro-getmyuserid:4] Set(SIP/username-114ffe50, MyChan=SIP/username) in new stack [2010-03-02 13:32:09.619] -- Executing [...@macro-getmyuserid:5] Set(SIP/username-114ffe50, __UserID=username) in new stack [2010-03-02 13:32:09.619] -- Executing [...@macro-getmyuserid:6] GotoIf(SIP/username-114ffe50, 0?NOUSER) in new stack [2010-03-02 13:32:09.620] -- Executing [...@macro-getmyuserid:7] Goto(SIP/username-114ffe50, done) in new stack [2010-03-02 13:32:09.620] -- Goto (macro-getmyUserID,s,12) [2010-03-02 13:32:09.620] -- Executing [...@macro-getmyuserid:12] Verbose(SIP/username-114ffe50, 2,getmyUserID set ID to username) in new stack [2010-03-02 13:32:09.620] == getmyUserID set ID to username [2010-03-02 13:32:09.621] -- Executing [...@macro-outgoing:3] Set(SIP/username-114ffe50, DB(users/username/LDNumber)=911) in new stack [2010-03-02 13:32:09.624] -- Executing [...@macro-outgoing:4] Set(SIP/username-114ffe50, DB(users/username/LDContext)=context) in new stack [2010-03-02 13:32:09.624] -- Executing [...@macro-outgoing:5] Set(SIP/username-114ffe50, RCStatus=0) in new stack [2010-03-02 13:32:09.624] -- Executing [...@macro-outgoing:6] Verbose(SIP/username-114ffe50, 2,Record Call Status: 0) in new stack [2010-03-02 13:32:09.624] == Record Call Status: 0 [2010-03-02 13:32:09.624] -- Executing [...@macro-outgoing:7] GotoIf(SIP/username-114ffe50, 0?Record:NoRecord) in new stack [2010-03-02 13:32:09.625] -- Goto (macro-outgoing,s,12) [2010-03-02 13:32:09.625] -- Executing [...@macro-outgoing:12] Verbose(SIP/username-114ffe50, 2,Not going to record the call) in new stack [2010-03-02 13:32:09.625] == Not going to record the call [2010-03-02 13:32:09.625] -- Executing [...@macro-outgoing:13] NoOp(SIP/username-114ffe50, ) in new stack [2010-03-02 13:32:09.625] -- Executing [9111...@context:5] Goto(SIP/username-114ffe50, DODIAL) in new stack [2010-03-02 13:32:09.625] -- Goto (context,911,7) [2010-03-02 13:32:09.625] -- Executing [9111...@context:7] Set(SIP/username-114ffe50, CALLERID(num)=22) in new stack [2010-03-02 13:32:09.625] -- Executing [9111...@context:8] Dial(SIP/username-114ffe50, Dahdi/G1/11,40,g) in new stack [2010-03-02 13:32:09.626] -- Requested transfer capability: 0x00 - SPEECH [2010-03-02 13:32:09.626] -- Called G1/11 [2010-03-02 13:32:09.778] -- DAHDI/23-1 is proceeding passing it to SIP/username-114ffe50 [2010-03-02 13:32:12.426] -- DAHDI/23-1 is making progress passing it to SIP/username-114ffe50 [2010-03-02 13:32:20.285] -- DAHDI/23-1 answered SIP/username-114ffe50 [2010-03-02 13:32:21.976] -- Channel 0/23, span 1 got hangup request, cause 16 [2010-03-02 13:32:21.990] -- Hungup 'DAHDI/23-1' [2010-03-02 13:32:21.991] -- Executing [9111...@context:9] Verbose(SIP/username-114ffe50, 2,Dahdi call just got status ANSWER) in new stack [2010-03-02 13:32:21.991] == Dahdi call just got status ANSWER [2010-03-02 13:32:21.991] --
Re: [asterisk-users] dahdi and oslec
On Wed, Mar 03, 2010 at 08:19:12AM -0600, Danny Nicholas wrote: You might have to load the canceller with a modprobe (modprobe mg2 for example) It's 'dahdi_echocan_mg2' . And dahdi should modprobe it for you when you run dahdi_cfg . -- 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 -- 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 and cellphone/GSM voicemailbox
On Tue, 2010-03-02 at 14:42 -0500, Fred Posner wrote: On Mar 2, 2010, at 2:37 PM, jonas kellens wrote: Does Asterisk know when it hits a voicemailbox ? When calling to a cell-phone or GSM, after some rings and no pickup you arrive at a voicemailbox. If Asterisk does not know it's a voicemailbox that has answered the call, the voicemailbox will contain 60minutes of 'silence'. This is very expensive 'silence'. How to avoid this ? Jonas You can avoid this is several ways... one of the ways I like best is to dial with a macro that then requires the recipient to press 1 or some dtmf confirmation to accept the call. Very good at avoiding voicemail, cell phone service messages, etc. Have you a link to documentation on how to implement this ? I wonder what to do with, and how to connect the caller with the callee, until this callee presses '1'... Jonas. -- _ -- 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] Is this a bug?
Hi List, I'm working on making one of my applications multi-lingual and find that I have this problem. The SayDigits and SayNumber functions in 1.4.26.2 recognize but don't process the CHANNEL(language) function. Here's a snippet to verify. exten = 317,1,Answer exten = 317,n,playback(tt-monkeysintro) exten = 317,n,Set(CHANNEL(language)=es) exten = 317,n,Wait(2) exten = 317,n,SayDigits(123) exten = 317,n,SayNumber(1) exten = 317,n,playback(vm-goodbye) exten = 317,n,Set(CHANNEL(language)=en) exten = 317,n,Wait(2) exten = 317,n,SayDigits(123) exten = 317,n,SayNumber(1) exten = 317,n,playback(vm-goodbye) exten = 317,n,hangup I can work around it, but would like to use the built-in functions. Regards, Danny Nicholas -- _ -- 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] Getting verbose or debug tracing in Asterisk
Hi, Not sure what I changed, but when I open x-lite now, I get the following verbose output on the CLI: [Mar 3 15:31:22] NOTICE[2273]: chan_sip.c:21331 handle_request_subscribe: Received SIP subscribe for peer without mailbox: user2 Does this indicate a successful registration of user2? Is the 'without mailbox' important? Thanks, Tim -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 03 March 2010 14:17 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Getting verbose or debug tracing in Asterisk The problem is on your x-lite end. If you were speaking to Asterisk (even incorrectly), it would at least indicate a bad connection. IMO, it is better to use numbers for extensions as opposed to user1, but that is irrelevant. Make sure the x-lite client has the IP of your asterisk box. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Culhane Sent: Wednesday, March 03, 2010 4:13 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting verbose or debug tracing in Asterisk Hi, For some reason I can't get Asterisk to produce debug or verbose tracing output. I connect to asterisk using 'asterisk -r' Then issue the command: Core set debug 10 And Core set verbose 10 And it confirms that the correct level has been set. I then attempt a connection from an x-lite client. However, no debug or verbose output appears in the console or in any of the log files in /var/log/asterisk. I'm using Asterisk 1.6.2.2. Anybody know what I'm doing wrong? Thanks, Tim - Tim Culhane, Critical Path Ireland, 42-47 Lower Mount Street, Dublin 2. Direct line: 353-1-2415107 phone: 353-1-2415000 tim.culh...@criticalpath.net http://www.criticalpath.net Critical Path a global leader in digital communications -- _ -- 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
Re: [asterisk-users] Getting verbose or debug tracing in Asterisk
W/O mailbox should just be a warning. Sip show peers will tell you if the registration was actually successful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Culhane Sent: Wednesday, March 03, 2010 9:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Getting verbose or debug tracing in Asterisk Hi, Not sure what I changed, but when I open x-lite now, I get the following verbose output on the CLI: [Mar 3 15:31:22] NOTICE[2273]: chan_sip.c:21331 handle_request_subscribe: Received SIP subscribe for peer without mailbox: user2 Does this indicate a successful registration of user2? Is the 'without mailbox' important? Thanks, Tim -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 03 March 2010 14:17 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Getting verbose or debug tracing in Asterisk The problem is on your x-lite end. If you were speaking to Asterisk (even incorrectly), it would at least indicate a bad connection. IMO, it is better to use numbers for extensions as opposed to user1, but that is irrelevant. Make sure the x-lite client has the IP of your asterisk box. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Culhane Sent: Wednesday, March 03, 2010 4:13 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting verbose or debug tracing in Asterisk Hi, For some reason I can't get Asterisk to produce debug or verbose tracing output. I connect to asterisk using 'asterisk -r' Then issue the command: Core set debug 10 And Core set verbose 10 And it confirms that the correct level has been set. I then attempt a connection from an x-lite client. However, no debug or verbose output appears in the console or in any of the log files in /var/log/asterisk. I'm using Asterisk 1.6.2.2. Anybody know what I'm doing wrong? Thanks, Tim - Tim Culhane, Critical Path Ireland, 42-47 Lower Mount Street, Dublin 2. Direct line: 353-1-2415107 phone: 353-1-2415000 tim.culh...@criticalpath.net http://www.criticalpath.net Critical Path a global leader in digital communications -- _ -- 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 -- _ -- 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] Getting verbose or debug tracing in Asterisk
Here is my output of 'sip show peers' user1/user110.41.3.12 D N 10434Unmonitored user2/user210.41.3.12 D N 65293Unmonitored user3/user3(Unspecified)D N 5060 Unmonitored user4/user4(Unspecified)D N 5060 Unmonitored 4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 4 online, 0 offline] So, does this mean the registration worked? What is the difference between monitored and unmonitored? Tim -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 03 March 2010 15:42 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Getting verbose or debug tracing in Asterisk W/O mailbox should just be a warning. Sip show peers will tell you if the registration was actually successful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Culhane Sent: Wednesday, March 03, 2010 9:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Getting verbose or debug tracing in Asterisk Hi, Not sure what I changed, but when I open x-lite now, I get the following verbose output on the CLI: [Mar 3 15:31:22] NOTICE[2273]: chan_sip.c:21331 handle_request_subscribe: Received SIP subscribe for peer without mailbox: user2 Does this indicate a successful registration of user2? Is the 'without mailbox' important? Thanks, Tim -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 03 March 2010 14:17 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Getting verbose or debug tracing in Asterisk The problem is on your x-lite end. If you were speaking to Asterisk (even incorrectly), it would at least indicate a bad connection. IMO, it is better to use numbers for extensions as opposed to user1, but that is irrelevant. Make sure the x-lite client has the IP of your asterisk box. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Culhane Sent: Wednesday, March 03, 2010 4:13 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting verbose or debug tracing in Asterisk Hi, For some reason I can't get Asterisk to produce debug or verbose tracing output. I connect to asterisk using 'asterisk -r' Then issue the command: Core set debug 10 And Core set verbose 10 And it confirms that the correct level has been set. I then attempt a connection from an x-lite client. However, no debug or verbose output appears in the console or in any of the log files in /var/log/asterisk. I'm using Asterisk 1.6.2.2. Anybody know what I'm doing wrong? Thanks, Tim - Tim Culhane, Critical Path Ireland, 42-47 Lower Mount Street, Dublin 2. Direct line: 353-1-2415107 phone: 353-1-2415000 tim.culh...@criticalpath.net http://www.criticalpath.net Critical Path a global leader in digital communications -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Deadlock while using MGCP on Asterisk
Hello guys, Finally I have done the upgrade. Theres no more deadlock now ! Thanks. Something still goes wrong and I dont find anything on that : Most of users connected on Asterisk/MGCP cannot place calls because a hang up ringback tone triggered while typing the phone number on the phone. What I can see with Wireshark is : CPE NTFY ; Asterisk OK ; Asterisk CRCX With SDP ; Asterisk RQNT ; CPE OK with SDP ; CPE OK for 4/5 times whereas with a good call this happened 1 time. And in Warning/full : WARNING[11145] chan_mgcp.c: Maximum retries exceeded for transaction 33708 on [030303030303] [Mar 2 19:30:05] NOTICE[11145] chan_mgcp.c: Removing message from 026244104989 transaction 49481 [Mar 2 19:30:07] NOTICE[11145] chan_mgcp.c: Got response back on [026244104989] for transaction 49470 we aren't sending? [Mar 2 19:30:07] NOTICE[11145] chan_mgcp.c: Got response back on [026244104989] for transaction 49471 we aren't sending? I dont understand these outputs. Can you help me to clarify ? Regards, Adrien .L De : Adrien Lemoine [mailto:alemo...@legos.fr] Envoyé : jeudi 25 février 2010 18:57 À : 'Miguel Molina'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc : 'mmichel...@digium.com' Objet : RE: [asterisk-users] Deadlock while using MGCP on Asterisk Thank you guys for your feedback. I consider the upgrading to 1.4.29.1. Does it can definitively prevent me from this kind of freeze ? Regards, Adrien .L De : Miguel Molina [mailto:mmol...@millenium.com.co] Envoyé : jeudi 25 février 2010 18:21 À : alemo...@legos.fr; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Deadlock while using MGCP on Asterisk Adrien Lemoine escribió: Hello all, Im running Asterisk 1.2.35 with chan_mgcp activated. The process host around 2,4K users. Along the day Ive got some debug reports like : Feb 24 22:25:42 DEBUG[28546] channel.c: Avoiding deadlock for 'MGCP/aaln/1...@028421223635-1' Feb 24 22:29:04 DEBUG[28670] channel.c: Avoiding initial deadlock for 'MGCP/aaln/1...@028421223635-1' Then, at random time (around 10~16 hours after a restart), Asterisk comes into deadlocks : Feb 25 16:28:22 WARNING[8149] channel.c: Avoided deadlock for '0xb713cb60', 9 retries! Feb 25 16:29:07 WARNING[8180] channel.c: Avoided initial deadlock for '0xb713cb60', 9 retries! Feb 25 16:40:21 WARNING[8629] channel.c: Avoided initial deadlock for '0xb713cb60', 9 retries! Avoided seems to correlate that Asterisk is in deadlock status. I put in attached a gdb output during the deadlock if it can helps. How can I correct these errors and avoid the crash not the deadlock J Regards, Adrien .L That kind of Avoided deadlock... messages, typical for early 1.2 systems have gone on recent versions on 1.4.X and higher. Did you consider upgrading? Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- 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] Getting verbose or debug tracing in Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tim Culhane wrote: Here is my output of 'sip show peers' user1/user110.41.3.12 D N 10434Unmonitored user2/user210.41.3.12 D N 65293Unmonitored user3/user3(Unspecified)D N 5060 Unmonitored user4/user4(Unspecified)D N 5060 Unmonitored 4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 4 online, 0 offline] So, does this mean the registration worked? What is the difference between monitored and unmonitored? Tim user1 user2 have registered. user3 user4 have not Unmonitored means that you have not specified qualify=yes in the peer configuration. Barry - -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLjoj2CFu3bIiwtTARArj4AKCh99NSCRHISUuNv/G72zGERoj8fwCfXpIv nuD43cWZ3m9k8TxFDhx/vdo= =Zptj -END PGP SIGNATURE- -- _ -- 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] forward problem!
Hello all, Here my architecture : Proxy1-asterisk1-proxy2-phone1 If a call arrived from proxy1 to phone1 AND phone1 always forward to proxy, asterisk1 say: -- Now forwarding SIP/phone1-001d to 'Local/969990...@proxy2' (thanks to SIP/proxy2-001e) Why it use Local ? I just need to use as a normal call, not a local Thank you Francois -- _ -- 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] Getting verbose or debug tracing in Asterisk
Ok thanks everybody. I seem to have verbose output to the CLI working with the addition of 'verbose' to the console line in logger.conf. Also, I now seem to have a couple of users registered via x-lite ... Though I don't really know why x-lite - asterisk connection suddenly decided to work. Tim -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry L. Kline Sent: 03 March 2010 16:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Getting verbose or debug tracing in Asterisk -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tim Culhane wrote: Here is my output of 'sip show peers' user1/user110.41.3.12 D N 10434 Unmonitored user2/user210.41.3.12 D N 65293 Unmonitored user3/user3(Unspecified)D N 5060 Unmonitored user4/user4(Unspecified)D N 5060 Unmonitored 4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 4 online, 0 offline] So, does this mean the registration worked? What is the difference between monitored and unmonitored? Tim user1 user2 have registered. user3 user4 have not Unmonitored means that you have not specified qualify=yes in the peer configuration. Barry - -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLjoj2CFu3bIiwtTARArj4AKCh99NSCRHISUuNv/G72zGERoj8fwCfXpIv nuD43cWZ3m9k8TxFDhx/vdo= =Zptj -END PGP SIGNATURE- -- _ -- 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] Echo cancellation on DAHDI
On Wed, 2010-03-03 at 10:14 +0530, DHAVAL INDRODIYA wrote: Hi, Carlos I checked dmesg on my server and i found following message what is meaning for this ? i cant understand VPM400: Not Present VPM450: echo cancellation for 128 channels VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) Well, that means that your card does have the echo cancellation module installed and it is active. Please post your DAHDI configuration to make sure your channels are properly configured. You should not have echo on any channel but remember that the E1 is not the only source of echo for calls. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] 911, channel full
Hi,
I am trying to implement 911 funtionality in my PBX. A call should
drop if all lines are busy. Here is my context nineoneone from
extensions.conf
[nineoneone]
exten = s,1,Set(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,Set(EMERGENCY=1,g)
exten = s,n,Set(SET_EMERG_FLAG=1)
exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY}=1]?inprogress)
exten = s,n,SoftHangup(${EMERGENCY_TRUNK}-1)
exten = s,n,Wait(12)
exten = s,n,Goto(checkavail)
exten = s,s+2(inprogress),Congestion
exten = s,checkavail+101(notavail),Goto(trunkbusy)
exten = h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3)
exten = h,3,Set(EMERGENCY=0,g)
If all lines connecting to PSTN are busy. I get busy tone upon dialing
911 and following message is generated by CLI.
app_dial.c:1547 dial_exec_full: Unable to create channel of type
'DAHDI' (cause 34 - Circuit/channel congestion)
I would appreciate if somebody help me solve this issue.
Regards
Shahnawaz
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[asterisk-users] Best practise for ISDN Video Conferencing..
Hi All, I'm about to setup an Asterisk install to take over an old legacy PBX system. At present, the legacy system has modules in it which provides 4 * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on site, these use the ISDN30 (uk) that the normal voice calls go over. Is it possible to emulate this in asterisk? I've seen zapras but I'm not sure if that's right. Is there a better way to do Video conferencing over ISDN in asterisk that will work with the Tandberg unit? Thanks, Mark -- _ -- 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 SIP, SIPAddHeader() and Cisco GED-125
Greetings: I'm in the situation where I'm trying to splash information picked off by an asterisk IVR into a Cisco call center environment. I'm under the impression that the ONLY way to do this is to setup socket connections with the Cisco voice processor, or CVP, and send packets corresponding to GED-125. Cisco has a detailed 100+-page document detailing the internals of what these packets need to look like. But wouldn't it be nice if instead, you could use SIPAddHeader() with X tags and have Cisco pick off the out-of-band values from SIP packets? Wouldn't it be even nicer if there was a middleware that spoke GED-125 out of one side, and spoke SIP X headers on the other side? I will soon be able to tell you about the bowels of this interaction, but before I go down this road, does anybody want to speak up with lessons learned from doing this themselves? I'm assuming I'm going to end up creating a library in Perl to help me do this (that is, the out-of-band conversation with the CVP). -- _ -- 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] 911, channel full
On 3 Mar 2010, at 17:21, mir shahnawaz wrote:
[nineoneone]
exten = s,1,Set(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,Set(EMERGENCY=1,g)
exten = s,n,Set(SET_EMERG_FLAG=1)
exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY}=1]?inprogress)
exten = s,n,SoftHangup(${EMERGENCY_TRUNK}-1)
exten = s,n,Wait(12)
exten = s,n,Goto(checkavail)
exten = s,s+2(inprogress),Congestion
exten = s,checkavail+101(notavail),Goto(trunkbusy)
exten = h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3)
exten = h,3,Set(EMERGENCY=0,g)
If all lines connecting to PSTN are busy. I get busy tone upon dialing
911 and following message is generated by CLI.
app_dial.c:1547 dial_exec_full: Unable to create channel of type
'DAHDI' (cause 34 - Circuit/channel congestion)
Can you tell us the other lines too? i.e. the bit where it attempts to
actually do the hangup..
S
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Re: [asterisk-users] Best practise for ISDN Video Conferencing..
- Mark Adams m...@campbell-lange.net escreveu: Hi All, I'm about to setup an Asterisk install to take over an old legacy PBX system. At present, the legacy system has modules in it which provides 4 * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on site, these use the ISDN30 (uk) that the normal voice calls go over. Is it possible to emulate this in asterisk? I've seen zapras but I'm not sure if that's right. Is there a better way to do Video conferencing over ISDN in asterisk that will work with the Tandberg unit? Thanks, Mark I don't think Asterisk can do video over ISDN. It would be great if anyone can prove me wrong thought. -- _ -- 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] Uverse, Asterisk and SIP
Warren Selby wrote: You need to set your firewall public ip to dhcp in order for Uverse dmz to work. Thanks, --Warren Selby On Mar 2, 2010, at 8:53 PM, sean darcy seandar...@gmail.com wrote: Fred Posner wrote: On Mar 2, 2010, at 6:27 PM, sean darcy wrote: I've just got Uverse installed. I had dsl, but ATT insisted I couldn't keep my old dsl, but had to switch to Uverse internet - vdsl. My setup: linux box as router : 10.10.11.252 asterisk box: 10.10.11.180 10.10.11.252 is multihomed and connected to the Uverse Residential Gateway. I've set it up as DMZplus, and it shows the public ip address as eth1. I can ssh into the linux box from outside. sip worked fine with dsl. I used teliax, junction and direct sip to the asterisk box in the office. I can ssh from 10.10.11.180 to the office. But not now. The asterisk box sends out sip messages, but nothing comes in. In the office asterisk box, I don't see the sip messages come in. Is anybody using sip behind a Uverse RG? Care to share the magic? sean Sean, I had att u-verse up until a week ago and loved it. Ran Asterisk behind it with great success. (I only left u-verse because of a physical move). Anyway, by default the u-verse router simply will block upd like noone's business. Make sure you have a firewall and then tell the u- verse router to open everything to that firewall (and proceed like you did on dsl). If you change the mac of your firewall, you'll need to reauth it again. ---fred http://qxork.com Well, I think I did that by setting the linux box to DMZplus: View Firewall Summary-View Firewall Details Current Settings: Custom Device AllowedApps AppType Protocol PortNumber(s) PublicIP 76.xxx.yyy.zzzAll-(all)(all)76.xxx.yyy.zzz and Edit Advanced Firewall Settings unchecked all the Security Settings, and unchecked all the Attack Detection. Anything else? sean Well at least my RG doesn't let you use DMZplus _unless_ you've chosen dhcp. So I did. And the RG shows the router as DMZplus. And I can ssh into my router from the internet. Anybody else got this working? sean -- _ -- 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] Uverse, Asterisk and SIP
On Wed, Mar 3, 2010 at 12:03 PM, sean darcy seandar...@gmail.com wrote: Well at least my RG doesn't let you use DMZplus _unless_ you've chosen dhcp. So I did. And the RG shows the router as DMZplus. And I can ssh into my router from the internet. Anybody else got this working? sean I know when I first got Uverse, I had to call their second level tech support and get some ports opened that were by default closed off on their end (not in the RG, but higher upstream) - you may want to contact them and see if they're blocking SIP? I run my asterisk server from a datacenter and I only have a phone at home behind my RG, so I can't speak to your specific situation. -- Thanks, --Warren Selby 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] Uverse, Asterisk and SIP
On Mar 3, 2010, at 1:03 PM, sean darcy wrote: Well at least my RG doesn't let you use DMZplus _unless_ you've chosen dhcp. So I did. And the RG shows the router as DMZplus. And I can ssh into my router from the internet. Anybody else got this working? sean What are the issues? First, do you have a public IP or private IP from the DHCP server. If it's private, then it's not set up correctly. If it's public, make sure you've updated your sip.conf with the public ip as an external address. ---fred http://qxork.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] 911, channel full
Thanks for your reply. This all I have, am I missing something? Please
help in this regard. Here is full output from CLI
-- Executing [...@default:1] Goto(SIP/501-0137,
nineoneone,s,1) in new stack
-- Goto (nineoneone,s,1)
-- Executing [...@nineoneone:1] Set(SIP/501-0137,
SET_EMERG_FLAG=0) in new stack
-- Executing [...@nineoneone:2] ChanIsAvail(SIP/501-0137,
DAHDI/g0) in new stack
-- Executing [...@nineoneone:3] Set(SIP/501-0137,
EMERGENCY=1,g) in new stack
-- Executing [...@nineoneone:4] Set(SIP/501-0137,
SET_EMERG_FLAG=1) in new stack
-- Executing [...@nineoneone:5] Dial(SIP/501-0137,
DAHDI/g0/91234567) in new stack
[Mar 3 11:26:06] WARNING[28572]: app_dial.c:1547 dial_exec_full:
Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel
congestion)
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/501-0137' status is 'CONGESTION'
Regards
Shahnawaz
On Wed, Mar 3, 2010 at 10:54 AM, Steve Howes steve-li...@geekinter.net wrote:
On 3 Mar 2010, at 17:21, mir shahnawaz wrote:
[nineoneone]
exten = s,1,Set(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,Set(EMERGENCY=1,g)
exten = s,n,Set(SET_EMERG_FLAG=1)
exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY}=1]?inprogress)
exten = s,n,SoftHangup(${EMERGENCY_TRUNK}-1)
exten = s,n,Wait(12)
exten = s,n,Goto(checkavail)
exten = s,s+2(inprogress),Congestion
exten = s,checkavail+101(notavail),Goto(trunkbusy)
exten = h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3)
exten = h,3,Set(EMERGENCY=0,g)
If all lines connecting to PSTN are busy. I get busy tone upon dialing
911 and following message is generated by CLI.
app_dial.c:1547 dial_exec_full: Unable to create channel of type
'DAHDI' (cause 34 - Circuit/channel congestion)
Can you tell us the other lines too? i.e. the bit where it attempts to
actually do the hangup..
S
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[asterisk-users] Looking for a configuration guru to collaborate with
I work with various fixes in the Asterisk source tree... cross-compilation to new platforms, adding new features (channels, resources, etc), and adding new configuration samples that do useful things: https://issues.asterisk.org/view.php?id=16090 https://issues.asterisk.org/view.php?id=15858 https://issues.asterisk.org/view.php?id=15857 https://issues.asterisk.org/view.php?id=12293 https://issues.asterisk.org/view.php?id=11969 https://issues.asterisk.org/view.php?id=11487 and I'd like to do more: * simplify SLA (shared line appearance) through macros, etc. * add E.164 support * add Freenum/ISN support * add diversion support (P-Asserted-Identity, etc) to SIP * add find-me/follow-me via dialplan rules * add telephone auto-configuration hooks etc. if anyone is especially good at troubleshooting dialplan and sip (i.e. extensions.conf and sip.conf) issues, that would be really useful. If you can help me polish some examples of how to do these, I'll file documentation enhancement fixes in mantis (the Asterisk bug tracker) and get the changes integrated into the source tree, as I did above. Thanks! -Philip -- _ -- 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] Free 'Locked up' Channels
Hi All, Asterisk 1.4.25.1 .22 .29 - pretty much every Asterisk install we have out there exhibits this. Just wondering how to free a channel that will stay eternally busy ala: carl*CLI core show channels Channel Location State Application(Data) SIP/101-Dotnet-09bb2 *...@from-inside-dotne Down(None) 1 active channel 0 active calls This channel is not active. But Asterisk will never free it. Unfortunately it affects SIP subscriptions so people think this extension is always busy. Restart when convenient is no use because Asterisk will always think this channel is in use. I can force a restart but I would prefer if there was a way to free this channel from the CLI. TIA. Brian -- _ -- 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] Best practise for ISDN Video Conferencing..
Hi, thanks for your response. I'm not sure if I explained correctly. I need asterisk to provide an ISDN data function, whilst also routing voice calls over the same PRI. Is this possible? Regards, Mark On 3 Mar 2010, at 17:58, Vinícius Fontes vinic...@canall.com.br wrote: - Mark Adams m...@campbell-lange.net escreveu: Hi All, I'm about to setup an Asterisk install to take over an old legacy PBX system. At present, the legacy system has modules in it which provides 4 * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on site, these use the ISDN30 (uk) that the normal voice calls go over. Is it possible to emulate this in asterisk? I've seen zapras but I'm not sure if that's right. Is there a better way to do Video conferencing over ISDN in asterisk that will work with the Tandberg unit? Thanks, Mark I don't think Asterisk can do video over ISDN. It would be great if anyone can prove me wrong thought. -- _ -- 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] Identify scripts connecting to the asterisk manager
Is there any easy way to identify which script or service is connecting to the Asterisk manager? Somewhere on my system a script or service is trying to connect with a bad user name or password. I get the following error: connect attempt from '127.0.0.1' unable to authenticate I thought maybe I could do a tcpdump on port 5038 and try to fish out the bad username or password but I wasn't able to see any passwords or usernames in plain text. Any way I could maybe change the logging in Asterisk to show me the username that is not able to authenticate? - Jason -- _ -- 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] forward problem!
On Wed, Mar 3, 2010 at 8:30 AM, BERGANZ Francois franc...@acropolistelecom.net wrote: Hello all, Here my architecture : Proxy1—asterisk1—proxy2—phone1 If a call arrived from proxy1 to phone1 AND phone1 always forward to proxy, asterisk1 say: -- Now forwarding SIP/phone1-001d to 'Local/969990...@proxy2' (thanks to SIP/proxy2-001e) Why it use Local ? I just need to use as a normal call, not a local Thank you Francois -- _ -- 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 What version of Asterisk are you running? I have a very different setup than you but it looks like the following bug might apply to your case too. https://issues.asterisk.org/view.php?id=16865 Dave -- _ -- 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] CallerID and distinctive ring detection
Using distinctive ring detection with bell202 cid, is there any way to tell DAHDI to sometimes expect the cid after the 2nd ring, other times after the 1st? I just added RingMaster service (2nd DID w/ distinctive ring) to a TDM800P FXO line. No problem setting dringcontext for the 2nd DID. The 1st DID works normally, but I get no CallerID on the 2nd because the call is picked up before the FSK spill is sent. In both cases, the spill is sent about 2.8 secs after the start of the 1st ring, and 0.7 secs after the (1st or 2nd) ring ends. But after the default cadence, DAHDI waits for the spill. After the dring cadence, the pickup is almost immediate (about 0.5 sec). Anybody have any suggestions? distinctiveringaftercid doesn't help. -- Barry -- _ -- 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] Best practise for ISDN Video Conferencing..
Search bugs.asterisk.org and enter 'digital' in the search field. It probably will is my answer. I currently am not using it, so YMMV. Alec Davis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Adams Sent: Thursday, 4 March 2010 10:39 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best practise for ISDN Video Conferencing.. Hi, thanks for your response. I'm not sure if I explained correctly. I need asterisk to provide an ISDN data function, whilst also routing voice calls over the same PRI. Is this possible? Regards, Mark On 3 Mar 2010, at 17:58, Vinícius Fontes vinic...@canall.com.br wrote: - Mark Adams m...@campbell-lange.net escreveu: Hi All, I'm about to setup an Asterisk install to take over an old legacy PBX system. At present, the legacy system has modules in it which provides 4 * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on site, these use the ISDN30 (uk) that the normal voice calls go over. Is it possible to emulate this in asterisk? I've seen zapras but I'm not sure if that's right. Is there a better way to do Video conferencing over ISDN in asterisk that will work with the Tandberg unit? Thanks, Mark I don't think Asterisk can do video over ISDN. It would be great if anyone can prove me wrong thought. -- _ -- 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] how to create a dummy call
Hi all, What i'm going to do is that enable caller sing while playing a background music (likes karaoke). My approach is using Monitor and Meetme apps.Caller make a call to asterisk, asterisk join caller in to a voice conference and create a dummy caller which will play music, then Monitor app record both music and singer's voice. But i dont know how to create a dummy caller or throw a dummy call in order to do above task. Any idea or comment is appreciated. Thanks Quyps -- _ -- 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 create a dummy call
Hi all, It maybe not clear that what i'm going to do. What i want to do is that enable user to call to a number then a background music will be played and he/she sing to mobilephone, the voice will be recorded and synchronized with the music. Any idea? There is an approach which using Monitor and Meetme application, it however need to throw an extra call to playing music, and this call should be thrown automatically by Asterisk. Again, any idea? Please help, thanks Quyps On Thu, 2010-03-04 at 10:37 +0700, Pham Quy wrote: Hi all, What i'm going to do is that enable caller sing while playing a background music (likes karaoke). My approach is using Monitor and Meetme apps.Caller make a call to asterisk, asterisk join caller in to a voice conference and create a dummy caller which will play music, then Monitor app record both music and singer's voice. But i dont know how to create a dummy caller or throw a dummy call in order to do above task. Any idea or comment is appreciated. Thanks Quyps -- _ -- 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 create a dummy call
This may help you: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out On Wed, Mar 3, 2010 at 11:20 PM, Pham Quy qu...@vega.com.vn wrote: Hi all, It maybe not clear that what i'm going to do. What i want to do is that enable user to call to a number then a background music will be played and he/she sing to mobilephone, the voice will be recorded and synchronized with the music. Any idea? There is an approach which using Monitor and Meetme application, it however need to throw an extra call to playing music, and this call should be thrown automatically by Asterisk. Again, any idea? Please help, thanks Quyps On Thu, 2010-03-04 at 10:37 +0700, Pham Quy wrote: Hi all, What i'm going to do is that enable caller sing while playing a background music (likes karaoke). My approach is using Monitor and Meetme apps.Caller make a call to asterisk, asterisk join caller in to a voice conference and create a dummy caller which will play music, then Monitor app record both music and singer's voice. But i dont know how to create a dummy caller or throw a dummy call in order to do above task. Any idea or comment is appreciated. Thanks Quyps -- _ -- 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 -- Pascal B. http://www.kameleonlabs.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
[asterisk-users] [asterisk-user] SIP / Echo Cancellation
Hello I have successfully compiled OSLEC for echo cancellation for DAHDI channel. Is there any way to do echo cancellation for SIP Channel. Is any, please suggest me.?? Thanks in advance.. -- Regards, Chandrakant Solanki -- _ -- 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 Audio on pstn call
Hello, I'm facing problem where as whenever there are incoming call from pstn, there will be no audio coming in. User at the other end also could not hear my voice. This happens few days back. Im using asterisk 1.6.1.2 with dahdi tool 2.2.0. I thought it was time to upgrade, so upgraded to dahdi 2.2.1 and asterisk 1.6.2.5. However, it does not help at all. My current config as follows :- X100P clone card /etc/dahdi/system.conf # Span 1: WCFXO/0 Generic Clone Board 1 (MASTER) fxsks=1 echocanceller=mg2,1 /etc/asterisk/dahdi-channels.conf ; Span 1: WCFXO/0 Generic Clone Board 1 (MASTER) ;;; line=1 WCFXO/0/0 FXSKS (SWEC: MG2) signalling=fxs_ls callerid=asreceived group=0 context=from-pstn channel = 1 callerid= group= context=default /etc/asterisk/chan_dahdi.conf [trunkgroups] [channels] language = my ; usecallerid = yes callwaiting = yes usecallingpres = yes callwaitingcallerid = yes threewaycalling = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes mailbox = 5000 echocancel = yes echocancelwhenbridged = yes rxgain = 2.0 txgain = 3.0 group = 1 callgroup = 1 pickupgroup = 1 faxdetect = both signalling = fxs_ls callerid = asreceived group = 0 channel = 1 callerid = group = context = default #include dahdi-channels.conf my call plan will execute voicemail when there;s incoming call from pstn. result as shwon here -- Executing [...@from-pstn:1] Set(DAHDI/1-1, CallTime=20100304 13:45:30) in new stack -- Executing [...@from-pstn:2] Set(DAHDI/1-1, CallerIDString= 01935x) in new stack -- Executing [...@from-pstn:3] System(DAHDI/1-1, /bin/echo 20100304 13:45:30 01935x [] - to pstn /var/log/asterisk/call_log) in new stack -- Executing [...@from-pstn:4] Answer(DAHDI/1-1, ) in new stack -- Executing [...@from-pstn:5] VoiceMail(DAHDI/1-1, 5000,u) in new stack -- Stopped music on hold on DAHDI/1-1 -- Playing 'vm-theperson.gsm' (language 'my') -- Playing 'digits/5.gsm' (language 'my') -- Playing 'digits/0.gsm' (language 'my') -- Playing 'digits/0.gsm' (language 'my') -- Playing 'digits/0.gsm' (language 'my') -- Playing 'vm-isunavail.gsm' (language 'my') -- Playing 'vm-intro.gsm' (language 'my') -- Playing 'beep.gsm' (language 'my') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/5000/tmp/ksOvKw format: wav, 0x91bfb68 -- Recording automatically stopped after a silence of 10 seconds -- Playing 'auth-thankyou.gsm' (language 'my') -- Executing [...@from-pstn:8] Hangup(DAHDI/1-1, ) in new stack how ever, starting from line 5 onwards, theres no audio at all. anybody can help ? thank you. -- _ -- 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] No Audio on pstn call
additional info on the system Linux home 2.6.30.3-SLACKWARE #1 Sun Feb 7 09:09:33 MYT 2010 i686 Intel(R) Pentium(R) 4 CPU 2.00GHz GenuineIntel GNU/Linux Asterisk 1.6.2.5, Copyright (C) 1999 - 2009 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.5 currently running on home (pid = 11838) Verbosity is at least 7 home*CLI module show like dahdi Module Description Use Count codec_dahdiGeneric DAHDI Transcoder Codec Translato 0 app_dahdibarge.so Barge in on DAHDI channel application0 chan_dahdi.so DAHDI Telephony Driver 0 app_dahdiscan.so Scan DAHDI channels application 0 app_dahdiras.soDAHDI ISDN Remote Access Server 0 res_timing_dahdi.soDAHDI Timing Interface 0 on the other hand, calls made internally are ok. On Thu, Mar 4, 2010 at 2:43 PM, Siti Zalifah Md Yatim ctzali...@gmail.com wrote: Hello, I'm facing problem where as whenever there are incoming call from pstn, there will be no audio coming in. User at the other end also could not hear my voice. This happens few days back. Im using asterisk 1.6.1.2 with dahdi tool 2.2.0. I thought it was time to upgrade, so upgraded to dahdi 2.2.1 and asterisk 1.6.2.5. However, it does not help at all. My current config as follows :- X100P clone card /etc/dahdi/system.conf # Span 1: WCFXO/0 Generic Clone Board 1 (MASTER) fxsks=1 echocanceller=mg2,1 /etc/asterisk/dahdi-channels.conf ; Span 1: WCFXO/0 Generic Clone Board 1 (MASTER) ;;; line=1 WCFXO/0/0 FXSKS (SWEC: MG2) signalling=fxs_ls callerid=asreceived group=0 context=from-pstn channel = 1 callerid= group= context=default /etc/asterisk/chan_dahdi.conf [trunkgroups] [channels] language = my ; usecallerid = yes callwaiting = yes usecallingpres = yes callwaitingcallerid = yes threewaycalling = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes mailbox = 5000 echocancel = yes echocancelwhenbridged = yes rxgain = 2.0 txgain = 3.0 group = 1 callgroup = 1 pickupgroup = 1 faxdetect = both signalling = fxs_ls callerid = asreceived group = 0 channel = 1 callerid = group = context = default #include dahdi-channels.conf my call plan will execute voicemail when there;s incoming call from pstn. result as shwon here -- Executing [...@from-pstn:1] Set(DAHDI/1-1, CallTime=20100304 13:45:30) in new stack -- Executing [...@from-pstn:2] Set(DAHDI/1-1, CallerIDString= 01935x) in new stack -- Executing [...@from-pstn:3] System(DAHDI/1-1, /bin/echo 20100304 13:45:30 01935x [] - to pstn /var/log/asterisk/call_log) in new stack -- Executing [...@from-pstn:4] Answer(DAHDI/1-1, ) in new stack -- Executing [...@from-pstn:5] VoiceMail(DAHDI/1-1, 5000,u) in new stack -- Stopped music on hold on DAHDI/1-1 -- Playing 'vm-theperson.gsm' (language 'my') -- Playing 'digits/5.gsm' (language 'my') -- Playing 'digits/0.gsm' (language 'my') -- Playing 'digits/0.gsm' (language 'my') -- Playing 'digits/0.gsm' (language 'my') -- Playing 'vm-isunavail.gsm' (language 'my') -- Playing 'vm-intro.gsm' (language 'my') -- Playing 'beep.gsm' (language 'my') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/5000/tmp/ksOvKw format: wav, 0x91bfb68 -- Recording automatically stopped after a silence of 10 seconds -- Playing 'auth-thankyou.gsm' (language 'my') -- Executing [...@from-pstn:8] Hangup(DAHDI/1-1, ) in new stack how ever, starting from line 5 onwards, theres no audio at all. anybody can help ? thank you. -- _ -- 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 create a dummy call
On Wednesday 03 March 2010 22:20:40 Pham Quy wrote: It maybe not clear that what i'm going to do. What i want to do is that enable user to call to a number then a background music will be played and he/she sing to mobilephone, the voice will be recorded and synchronized with the music. Any idea? There is an approach which using Monitor and Meetme application, it however need to throw an extra call to playing music, and this call should be thrown automatically by Asterisk. You really don't need to generate any call at all. Just Answer, Monitor, and Playback the sound file. Monitor will take care of mixing the sound file and the user's voice. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.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
