Re: [asterisk-users] Dahdi and Junghanns QuadBRI
On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote: 2009/12/4 Olivier oza-4...@myamail.com Trying with a Junghanns PCI OctoBRI, I've got : # dahdi_hardware pci::08:00.0 qozap- 1397:16b8 Generic OctoBRI ISDN card My initial thought was that wcb4xxp driver could not support PCIe cards, at the moment. Should wcb4xxp driver support Junghanns PCI OctoBRI ? In trunk it does. It has been commited there quite a while ago but I guess you should use trunk. Or use rpm packages from Elastix. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Starting and installing Dahdi (2.2.0)?
Hello Unless I overlooked it, the Asterisk Reference Information Version 1.6.1.6 at www.asterisk.org/docs doesn't include instruction on how to start Dahdi when used to drive a TDP PCI card (OpenVox A400P with a single FXO module www.openvox.cn/products/show.php?itemid=20lang=2). I'd like to know... 1. How to start Dahdi manually. Is this the right way? modprobe wctfxo modprobe wctdm modprobe zaptel 2. How to add a startup script in CentOS through chkconfig If this is covered in an up-to-date document on the Net, please tell me where it can be found. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting and installing Dahdi (2.2.0)?
It looks like make config takes care of installing an init script, so I can just run /etc/init.d/dahdi start to load the required modules. I get the following error, however: --- # /etc/init.d/dahdi start Loading DAHDI hardware modules: wcfxo: [ OK ] Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) [FAILED] --- FWIW, the PCI card seems to be correctly detected: --- # lspci -v 03:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b100:0003 Flags: bus master, medium devsel, latency 64, IRQ 12 I/O ports at a000 [size=256] Memory at e200 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 --- Here's /etc/dahdi/system.conf: --- loadzone=fr defaultzone=fr fxsks=1 --- Any idea what is wrong? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting and installing Dahdi (2.2.0)?
... but ls -l /dev/dahdi/ doesn't return channel #1 :-/ # ls -l /dev/dahdi/ total 0 crw-rw 1 root root 196, 254 Dec 8 13:38 channel crw-rw 1 root root 196, 0 Dec 8 13:38 ctl crw-rw 1 root root 196, 255 Dec 8 13:38 pseudo crw-rw 1 root root 196, 253 Dec 8 13:38 timer # lsmod dahdi_dummy 8484 0 wcfxo 16032 0 dahdi 192392 2 dahdi_dummy,wcfxo # dahdi_cfg -vvv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) Could it be some incompatibily between this TDM card and the motherboard/PCI bus? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 with IAX
Of course, as long as your endpoints support it. Read more about it and purchase G.729 channel licenses for Asterisk from Digium: http://www.digium.com/en/products/g729codec.php Once you have the codec properly installed, enable it for your peer in your iax.conf file allow=g729. Restart asterisk and go to it. Also, Google is your friend. Search: g729 iax for lots of information and examples. On Tue, Dec 8, 2009 at 1:03 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: dear All, can I use G729 with IAX trunk or IAX calls regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting and installing Dahdi (2.2.0)?
I got it figured out: Modules must be listed in /etc/dahdi/modules: wcfxo wctdm dahdi /etc/init.d/dahdi start dahdi_cfg -vvv HTH, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and Junghanns QuadBRI
2009/12/8 Tzafrir Cohen tzafrir.co...@xorcom.com On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote: 2009/12/4 Olivier oza-4...@myamail.com Trying with a Junghanns PCI OctoBRI, I've got : # dahdi_hardware pci::08:00.0 qozap- 1397:16b8 Generic OctoBRI ISDN card My initial thought was that wcb4xxp driver could not support PCIe cards, at the moment. Should wcb4xxp driver support Junghanns PCI OctoBRI ? In trunk it does. Do you mean Dahdi Tools trunk version or shall I also use Dahdi-Linux trunk version (I used Dahdi-Tools revision 6822 and Dahdi-Linux 2.2.0.2) ? It has been commited there quite a while ago but I guess you should use trunk. Or use rpm packages from Elastix. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Voicemail
Hi List! I am running 'Asterisk 1.4.22 built by root' I have an issue with voicemails. In the var/spool/asterisk/voicemail/default/ext/inbox the msgnnn.txt files are flagged as readable, this causes asterisk to just skip over the voicemails when listening. drwx-w 2 asterisk 4096 2009-12-03 15:17 . drwx-w 7 asterisk 4096 2009-12-02 10:55 .. -r--rw-rw- 1 asterisk 267 2009-12-03 15:17 msg.txt -rwx-w 1 asterisk 44204 2009-12-03 15:17 msg.wav I know there was something that had to be changed in voicemail.c and asterisk recompiled some time ago but I can't recall it. This system has been installed about 3 months but has only gone into production this week. If anyone could help would be appreciated. Regards and TIA. Brian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Voicemail
Hi List, Apologies if this appears twice.. Apple mail seemed to post a follow up last time that isn't appearing.. I am running 'Asterisk 1.4.22 built by root' I have an issue with voicemails. In the var/spool/asterisk/voicemail/default/ext/inbox the msgnnn.txt files are flagged as readable, this causes asterisk to just skip over the voicemails when listening. drwx-w 2 asterisk 4096 2009-12-03 15:17 . drwx-w 7 asterisk 4096 2009-12-02 10:55 .. -r--rw-rw- 1 asterisk 267 2009-12-03 15:17 msg.txt -rwx-w 1 asterisk 44204 2009-12-03 15:17 msg.wav I know there was something that had to be changed in voicemail.c and asterisk recompiled some time ago but I can't recall it. This system has been installed about 3 months but has only gone into production this week. If anyone could help would be appreciated. Regards and TIA. Brian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo issue
Hi - I am having echo issues on our Asterisk box using a PRI circuit. I was using the software echo cancellation and that helped a bit but didn't solve it completely. So I went and bought a Digium echo cancellation module for the TE121 card. That made it even worst, getting more echo on external calls and between internal extension to extension. The echo doesn't happen all the time, but enough to get complaints from our users. Completely fed up with the issue, I removed the module from the card. Can someone guide me on how to fix/tune/address the echo issues. You can likely eliminate most echo on a PRI by setting txgain and rxgain. Are you using dahdi or zaptel? If Dahdi, what do your system.conf and chan_dahdi.conf look like? If zaptel, what do your zaptel.conf and zapata.conf look like? When you say you have echo on calls that are internal extension to internal extension, are the endpoints using dahdi/zaptel or some voip technology (sip, iax, mgcp, skinny, etc)? If voip, any echo is acoustically generated by the endpoints themselves. On voip calls I've often had this happen when the endpoints are using headsets, or have gain levels set very high. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and Junghanns QuadBRI
On Tue, Dec 08, 2009 at 03:47:52PM +0100, Olivier wrote: 2009/12/8 Tzafrir Cohen tzafrir.co...@xorcom.com On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote: 2009/12/4 Olivier oza-4...@myamail.com Trying with a Junghanns PCI OctoBRI, I've got : # dahdi_hardware pci::08:00.0 qozap- 1397:16b8 Generic OctoBRI ISDN card My initial thought was that wcb4xxp driver could not support PCIe cards, at the moment. Should wcb4xxp driver support Junghanns PCI OctoBRI ? In trunk it does. Do you mean Dahdi Tools trunk version or shall I also use Dahdi-Linux trunk version (I used Dahdi-Tools revision 6822 and Dahdi-Linux 2.2.0.2) ? Both. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail issues
Hi List, Apologies if this appears more than once.. Apple mail seemed to post a follow up last time that isn't appearing so I've moved to webmail to send.. I am running 'Asterisk 1.4.22 built by root' I have an issue with voicemails. In the var/spool/asterisk/voicemail/default/ext/inbox the msgnnn.txt files are flagged as readable, this causes asterisk to just skip over the voicemails when listening. drwx-w 2 asterisk 4096 2009-12-03 15:17 . drwx-w 7 asterisk 4096 2009-12-02 10:55 .. -r--rw-rw- 1 asterisk 267 2009-12-03 15:17 msg.txt -rwx-w 1 asterisk 44204 2009-12-03 15:17 msg.wav I know there was something that had to be changed in voicemail.c and asterisk recompiled some time ago but I can't recall it. This system has been installed about 3 months but has only gone into production this week. If anyone could help would be appreciated. Regards and TIA. Brian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting and installing Dahdi (2.2.0)?
On Tue, Dec 08, 2009 at 02:15:40PM +0100, Vincent wrote: Hello Unless I overlooked it, the Asterisk Reference Information Version 1.6.1.6 at www.asterisk.org/docs doesn't include instruction on how to start Dahdi when used to drive a TDP PCI card (OpenVox A400P with a single FXO module www.openvox.cn/products/show.php?itemid=20lang=2). I'd like to know... 1. How to start Dahdi manually. Is this the right way? modprobe wctfxo Not needed. modprobe wctdm modprobe zaptel zaptel? You use dahdi, not zaptel. And zaptel will be pulled by wctdm anyway. Hint: dahdi_hardware . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting and installing Dahdi (2.2.0)?
On Tue, Dec 08, 2009 at 03:37:26PM +0100, Vincent wrote: I got it figured out: Modules must be listed in /etc/dahdi/modules: wcfxo wctdm dahdi You actually only need 'wctdm' . And in fact, you could have generated that file with: dahdi_genconf modules /etc/init.d/dahdi start dahdi_cfg -vvv '/etc/init.d/dahdi start' also runs '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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] might have found and issue
I call into a box running asterisk 1.4.27.1 - this works. on that box I run the CLI and enter the command core show channels concise initially I see the ALSA/default.. and all that which is correct. I continue to speak and continue to do the core show channels concise. I continue to see the ALSA/default ... etc... After some time I am still connected as I hear audio but when I do the command core show channels concise nothing is printed. Seems to happen after 40-45 seconds. However - when I hangup I do see all the normal Hangup on Console stuff. Why after 40-45 seconds is the core show channels concise not printing anything? Thanks Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail issues
We got the last two. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Directory application: First DTMF digit is missed if pressed during using your touch tone keypad... announcement
If you're an asterisk 1.6 user, and use the 'Directory application', have you noticed that the first keypress is always missed if you press it during the part of the announement where alison says using you touch tone keypad If this includes you, have a look at mantis bug https://issues.asterisk.org/view.php?id=16409 If you are keen, please apply the patch and report back to either the list or add a comment to the reported bug.. Alec Davis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk throws error using the alsa module
Hello, I can't get the sound over alsa to work with Asterisk. My current version is 1.4.21.2~dfsg-3 running on debian stable. All settings are the default ones with exception of: /etc/asterisk/modules.conf: load = chan_alsa.so noload = chan_oss.so /etc/asterisk/alsa.conf: input_device=default output_device=default asterisk is started up and doesn't complain about alsa in the logfiles. anyway, if i connect via asterisk -r and type 'dial s' i get: puppy*CLI core set verbose 20 Verbosity was 0 and is now 20 puppy*CLI console dial s -- Executing [...@local:1] Wait(ALSA/default, 1) in new stack -- Executing [...@local:2] Answer(ALSA/default, ) in new stack Console call has been answered -- Executing [...@local:3] Set(ALSA/default, TIMEOUT(digit)=5) in new stack -- Digit timeout set to 5 -- Executing [...@local:4] Set(ALSA/default, TIMEOUT(response)=10) in new stack -- Response timeout set to 10 -- Executing [...@local:5] BackGround(ALSA/default, demo-congrats) in new stack -- ALSA/default Playing 'demo-congrats' (language 'en') [Dec 8 18:24:48] ERROR[10571]: chan_alsa.c:693 alsa_read: Read error: Resource temporarily unavailable I couldn't find any hints using google... does anyone has an idea? Thanks, vitaminx -- mail: vitam...@callistix.net www : http://www.callistix.net irc : #callistix @ chat.freenode.net:8001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk throws error using the alsa module
On Tue, Dec 08, 2009 at 06:25:46PM +0100, vitaminx wrote: Hello, I can't get the sound over alsa to work with Asterisk. My current version is 1.4.21.2~dfsg-3 running on debian stable. All settings are the default ones with exception of: /etc/asterisk/modules.conf: load = chan_alsa.so noload = chan_oss.so /etc/asterisk/alsa.conf: input_device=default output_device=default asterisk is started up and doesn't complain about alsa in the logfiles. anyway, if i connect via asterisk -r and type 'dial s' i get: puppy*CLI core set verbose 20 Verbosity was 0 and is now 20 puppy*CLI console dial s -- Executing [...@local:1] Wait(ALSA/default, 1) in new stack -- Executing [...@local:2] Answer(ALSA/default, ) in new stack Console call has been answered -- Executing [...@local:3] Set(ALSA/default, TIMEOUT(digit)=5) in new stack -- Digit timeout set to 5 -- Executing [...@local:4] Set(ALSA/default, TIMEOUT(response)=10) in new stack -- Response timeout set to 10 -- Executing [...@local:5] BackGround(ALSA/default, demo-congrats) in new stack -- ALSA/default Playing 'demo-congrats' (language 'en') [Dec 8 18:24:48] ERROR[10571]: chan_alsa.c:693 alsa_read: Read error: Resource temporarily unavailable The sound device is being used by something else? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E1 Channel Numbering - Your Comments.
All This is a small issue that I stumbled onto that has to do with the channel numbering on an E1 connection into an Asterisk Zaptel/DAHDI system. As most of us already know an E1 has 32 channels of which 30(1-15 17-31) are B-channels and 1 (16) is a D-Channel. The 32nd channel is not presented in Asterisk Zaptel/DAHDI. There are other configurations but this is the most common. *** Everything currently works and the community is humming along with the current implementation. *** Is it or would it be worth the time to note the 32nd channel of the E1? I noticed this issue on a 4 port card with 4 E1s. I quickly expected the last B-channel at 127 where it was actually at 124 (my first time with E1s). As telcom lingo goes I have seen this 32nd channel referred to as time slot 0 and as framing channel and expect there to be other names as well. As an example, the current method: bchan=1-15 dchan=16 bchan=17-31 ... bchan=32-46 dchan=47 bchan=48-62 ... bchan=63-77 dchan=78 bchan=79-93 ... bchan=94-108 dchan=109 bchan=110-124 and an example of my first thoughts: bchan=1-15 dchan=16 bchan=17-31 uchan=32 ... bchan=33-47 dchan=48 bchan=49-63 uchan=64 ... bchan=65-79 dchan=80 bchan=81-95 uchan=96 ... bchan=97-111 dchan=112 bchan=113-127 uchan=128 I quickly grabbed the uchan to mean Unused or Unallocated. I am going to assume that there are some better ideas and names for this. One thought was Framing or fchan which does not exactly describe the channel. I did notice that depending on your locale the channel had several names. What are your ideas? ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and Junghanns QuadBRI
2009/12/8 Tzafrir Cohen tzafrir.co...@xorcom.com On Tue, Dec 08, 2009 at 03:47:52PM +0100, Olivier wrote: 2009/12/8 Tzafrir Cohen tzafrir.co...@xorcom.com On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote: 2009/12/4 Olivier oza-4...@myamail.com Trying with a Junghanns PCI OctoBRI, I've got : # dahdi_hardware pci::08:00.0 qozap- 1397:16b8 Generic OctoBRI ISDN card My initial thought was that wcb4xxp driver could not support PCIe cards, at the moment. Should wcb4xxp driver support Junghanns PCI OctoBRI ? In trunk it does. Do you mean Dahdi Tools trunk version or shall I also use Dahdi-Linux trunk version (I used Dahdi-Tools revision 6822 and Dahdi-Linux 2.2.0.2) ? Both. Strange : I'm still getting : # dahdi_hardware pci::08:00.0 qozap- 1397:16b8 Generic OctoBRI ISDN card # dahdi_genconf modules cat /etc/dahdi/modules # Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules) on Tue Dec 8 18:25:30 2009 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. qozap I think I installed latest trunk versions for both Dahdi Linux and Dahdi Tools : cd /usr/src/dahdi-linux svn checkout http://svn.asterisk.org/svn/dahdi/linux/trunk . make make install ... Any hint ? I was about to drop dahdi_genconf for a try but th problem seems deeper as make config output includes: List of detected DAHDI devices: pci::08:00.0 qozap- 1397:16b8 Generic OctoBRI ISDN card -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Channel Numbering - Your Comments.
Andrew Latham wrote: and an example of my first thoughts: bchan=1-15 dchan=16 bchan=17-31 uchan=32 Well, you've missed an important point: the DAHDI drivers for E1 cards would have to be modified to make this 32nd channel in each span actually exist, before any configuration in chan_dahdi.conf would be relevant. If that was done, there wouldn't actually be any changes required in chan_dahdi.conf at all; if you wished to, you could put in a comment for each 32nd channel to indicate that you are intentionally skipping it, but there is no need to make chan_dahdi actually aware of that channel at all. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Channel Numbering - Your Comments.
On Tue, Dec 8, 2009 at 2:58 PM, Kevin P. Fleming kpflem...@digium.com wrote: Andrew Latham wrote: and an example of my first thoughts: bchan=1-15 dchan=16 bchan=17-31 uchan=32 Well, you've missed an important point: the DAHDI drivers for E1 cards would have to be modified to make this 32nd channel in each span actually exist, before any configuration in chan_dahdi.conf would be relevant. I did think of that but my fingers totally forgot to put it in the email. ;) I shall assume it is either really hard or really easy to change this. If that was done, there wouldn't actually be any changes required in chan_dahdi.conf at all; if you wished to, you could put in a comment for each 32nd channel to indicate that you are intentionally skipping it, but there is no need to make chan_dahdi actually aware of that channel at all. This is where my query lives... What if... Imagine 2+ E1s sharing the first E1's D-channel for timing and some manufacturer thought about selling some hardware that would allow the use of 32 channels on the next E1 and so on. So something like dchan=16 bchan=1-15,17-31,32-63 could happen. How would the current DAHDI act if B-Channels existed at channel 32. Does it just skip them _because_ it is an E1 or does it see the channel and skip it. My apologies, its a holiday here and I am catching up with my email. This topic was in my email. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk throws error using the alsa module
[Dec 8 18:24:48] ERROR[10571]: chan_alsa.c:693 alsa_read: Read error: Resource temporarily unavailable I agree, this looks like some form of conflict for the sound device. The first thing I'd suggest doing, is trying to reproduce the error with a command-line tool, with asterisk out of the loop entirely. You'd use a command such as aplay -D default /path/to/demo-congrats.wav See if it plays back properly. A resource temporarily unavailable error from ALSA would tend to suggest one of two sorts of conflicts: [1] A low-level (e.g. IRQ) conflict for the sound device itself. This could occur as a result of motherboard misconfiguration... for example, if the sound card/chip was configured to use IRQ 2 or 3, and there was also a serial port in use which made use of this interrupt. Check (e.g.) /proc/interrupts to see if you can find such a conflict. [2] A higher-level conflict for use of the sound card, e.g. between two different (and incompatible) ALSA accesses, or between a native ALSA access and a user of ALSA's OSS driver- or library-level API emulation. One not-uncommon culprit is having an X Window desktop up and running. Some of the newer desktop packages have their own sound-management architecture (e.g. ESD, the Enlightenment Sound Daemon, or the JACK audio toolkit, or PulseAudio). These management systems often open the underlying sound device (in a non-shared mode) and then provide their own APIs for arbitrating access, mixing multiple outputs together, etc., and a separate native ALSA access from Asterisk will often be unable to share access to the card. When doing native ALSA access, it's often possible to share access to the sound card (playing back two or more sounds simultaneously). Some sound cards have this capability in hardware. Many do not... and for those that do not, you can resolve the conflict by telling all of the playback apps to use the dmix plugin. This is a software mixer... it opens the underlying sound-card PCM output in an exclusive- access mode, and then accepts connections from any number of ALSA clients and mixes the audio together before sending it to the sound card. The trick about dmix is that *all* of the clients have to agree to use it. If the first client to open the sound card doesn't use dmix (but opens the default hardware device directly), then any further clients (dmix or otherwise) will be locked out. Similarly, if dmix is in use. any attempt by an ALSA client to access the hardware directly will probably be rejected (unless the hardware itself can do the mixing for it). On older versions of ALSA, it's necessary to specify the dmix device manually e.g. aplay -D dmix /foo/bar/baz.wav On some more recent versions of ALSA, using the default device will give you the hardware device directly *if* the hardware can handling mixing... and will give you the dmix device otherwise. Any sound client which is manually configured to access the hardware directly (via e.g. hw:) or the direct rate-and-format-conversion plugin (e.g. plughw) will not be going through the dmix plugin. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma A101DE with Dell PE 2850
Hi friends, I am about to install an asterisk server using a Sangoma A101DE over a Dell PE 2850 Server but I have doubts about PCI requirements. First I see at sangoma page that A101DE is PCI-Express (I think x1 for the size of the connector) And the specs for the PE 2850 is For PCI-X one 3.3-V, 64-bit, 100-MHz or three 3.3-V, 64 bit, 133MHz for PCI Express one x4 lane width one x8 lane width I can connect the card to any of the slots?, or only to PCI-Express Slots? (is compatible the card with x4 and x8 PCI-Express slots?) Thanks in advance. Atte. Ricardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Channel Numbering - Your Comments.
Andrew Latham wrote: This is where my query lives... What if... Imagine 2+ E1s sharing the first E1's D-channel for timing and some manufacturer thought about selling some hardware that would allow the use of 32 channels on the next E1 and so on. So something like dchan=16 bchan=1-15,17-31,32-63 could happen. The D-channels are not used for timing, the timing channels are. As far as I am aware (but my knowledge base is limited) the 32nd channel on an E1 cannot be used for data transfer under any circumstances. Please be sure you are not confusing the 32nd channel (used for timing) with the 16th channel (used as a D-channel for ISDN PRI usage). Asterisk already supports using all 31 available channels on an E1 as B-channels if the D-channel signaling is coming from a different span. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automon = *1 one touch recording
On Mon, 2009-12-07 at 19:39 -0700, Joseph wrote: After pressing *1 console is not showing anything indicating that the call is being recorded: I find that I often have to adjust the featuredigittimeout setting in features.conf, as users tend to take their time between the * and 1 keys when turning on automon. -- Jared Smith Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A101DE with Dell PE 2850
2009/12/8 Ricardo Melendez rmelen...@utep.com.mx: First I see at sangoma page that A101DE is PCI-Express (I think x1 for the size of the connector) Yes, it is PCIe x1. There is an A101D wich is PCI(-X). for PCI Express one x4 lane width one x8 lane width I can connect the card to any of the slots?, or only to PCI-Express Slots? (is compatible the card with x4 and x8 PCI-Express slots?) Yes, the A101DE runs in PCIe x4 or x8 and the A101D will run in PCI or PCI-X Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Channel Numbering - Your Comments.
On Tue, 2009-12-08 at 14:47 -0300, Andrew Latham wrote: As most of us already know an E1 has 32 channels of which 30(1-15 17-31) are B-channels and 1 (16) is a D-Channel. The 32nd channel is not presented in Asterisk Zaptel/DAHDI. There are other configurations but this is the most common. As an aside, I've seen different documentation in various places that shows this sync channel as being channel zero (coming before the first bearer channel), not the 32nd channel. I'm not familiar enough with E1s myself to be able to say definitively that this is the case, but thought I'd throw this out there for discussion (and hopefully more enlightenment). -- Jared Smith Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] network config
Slightly OT? A client has two offices in the Virgin Islands that MUST maintain data connectivity, and there are no available leased line options to run a P2P link between them. To date, broadband Internet connections at both offices have been used as the link, with a VPN tunnel, and phones in one location use the tunnel (Sonicwall) to talk with asterisk at the other location. Although this functions well, it only takes an (unfortunately frequent) hiccup to lose calls and/or severely impact quality. The client has decided to get a second Internet connection at both sites, and use the Sonicwall or any other possible firewall to manage the tunnel over both links, such that the phones won't know what link is being traversed, or (hopefully) that a link has gone down. So the first question is - has anyone attempted anything similar and made it work? Do you lose an in progress call when the tunnel switches from one link to the other? And finally - is there a device that will manage the tunnel such that a high water mark of latency will also cause the tunnel to switch to the other link, rather than actual packet loss? Thanks for any tips, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 Channel Numbering - Your Comments.
On Tue, Dec 8, 2009 at 4:01 PM, Kevin P. Fleming kpflem...@digium.com wrote: Andrew Latham wrote: This is where my query lives... What if... Imagine 2+ E1s sharing the first E1's D-channel for timing and some manufacturer thought about selling some hardware that would allow the use of 32 channels on the next E1 and so on. So something like dchan=16 bchan=1-15,17-31,32-63 could happen. The D-channels are not used for timing, the timing channels are. As far as I am aware (but my knowledge base is limited) the 32nd channel on an E1 cannot be used for data transfer under any circumstances. Please be sure you are not confusing the 32nd channel (used for timing) with the 16th channel (used as a D-channel for ISDN PRI usage). Asterisk already supports using all 31 available channels on an E1 as B-channels if the D-channel signaling is coming from a different span. Yes we are on the same page here. After some further reading I could not find a single instance (other than data) which could use the 32nd channel. I see why the DAHDI (and Zaptel) software is the way it is. The mistake I made was regarding a 32 Channel MUX which was 32 channels _of_ E1s... So the 32nd channel in an E1 should never (knocks on wood) be used by voice circuits. Is the numbering alone worth changing DAHDI? Probably not. I can even imagine users wondering why they can't do anything with this new magical 32nd channel. Sorry for the waste of time. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and Junghanns QuadBRI
On Tue, Dec 08, 2009 at 06:51:12PM +0100, Olivier wrote:
2009/12/8 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Dec 08, 2009 at 03:47:52PM +0100, Olivier wrote:
2009/12/8 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote:
2009/12/4 Olivier oza-4...@myamail.com
Trying with a Junghanns PCI OctoBRI, I've got :
# dahdi_hardware
pci::08:00.0 qozap- 1397:16b8 Generic OctoBRI ISDN card
My initial thought was that wcb4xxp driver could not support PCIe
cards,
at
the moment.
Should wcb4xxp driver support Junghanns PCI OctoBRI ?
In trunk it does.
Do you mean Dahdi Tools trunk version or shall I also use Dahdi-Linux
trunk
version
(I used Dahdi-Tools revision 6822 and Dahdi-Linux 2.2.0.2) ?
Both.
Strange : I'm still getting :
# dahdi_hardware
pci::08:00.0 qozap- 1397:16b8 Generic OctoBRI ISDN card
# dahdi_genconf modules cat /etc/dahdi/modules
# Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules) on
Tue Dec 8 18:25:30 2009
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
qozap
I think I installed latest trunk versions for both Dahdi Linux and Dahdi
Tools :
cd /usr/src/dahdi-linux
svn checkout http://svn.asterisk.org/svn/dahdi/linux/trunk .
make
make install
...
Any hint ?
I was about to drop dahdi_genconf for a try but th problem seems deeper as
make config output includes:
List of detected DAHDI devices:
pci::08:00.0 qozap- 1397:16b8 Generic OctoBRI ISDN card
What's the output of:
lspci -v -nn -s 08:00.0
(or if I got the syntax wrong: the full entry for this specific PCI
device, and specifically, the sub-vendor and sub-product IDs)
http://svn.asterisk.org/svn/dahdi/linux/trunk/drivers/dahdi/wcb4xxp/base.c
has:
static struct pci_device_id b4xx_ids[] __devinitdata =
...
{ 0x1397, 0x16b8, 0x1397, 0xb552, 0, 0, (unsigned long)hfc8s },
The equivalent in
http://svn.asterisk.org/svn/dahdi/tools/xpp/perl_modules/Dahdi/Hadware/PCI.pm
is:
# Lookup algorithm:
# First match 'vendor:product/subvendor:subproduct' key
# Else match 'vendor:product/subvendor' key
# Else match 'vendor:product' key
# Else not a dahdi hardware.
my %pci_ids = (
...
'1397:08b4/1397:b556' = { DRIVER = 'wcb4xxp', DESCRIPTION =
'Junghanns DuoBRI ISDN card' },
...
'1397:16b8' = { DRIVER = 'qozap', DESCRIPTION = 'Generic
OctoBRI ISDN card' },
--
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
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Re: [asterisk-users] network config
snip A client has two offices in the Virgin Islands that MUST maintain data connectivity, and there are no available leased line options to run a P2P link between them. snip Is there line of sight? I've been wanting to do a long-shot wifi link and my company would give it a shot if you want :). snip Do you lose an in progress call when the tunnel switches from one link to the other? /snip Any 'fail-over' router with links from separate providers that don't route the same subnets (cable/dsl) will have to change its default route when it 'fails-over'. As such, the VPN tunnel will be disconnected and reconnected. I'm sure you could make it brief, but yes, calls will likely be completely dropped. snip And finally - is there a device that will manage the tunnel such that a high water mark of latency will also cause the tunnel to switch to the other link, rather than actual packet loss? /snip See above. Fail-over routers have to wait some criteria are met in order to fail over (ping latency, ping loss, etc). This means that the connection you're using as the 'default' WILL go 'down' BEFORE it switches to the other one, regardless of the criteria used. Another plan would be to set up two routers at the site with two separate VPN tunnels across the two different links, both tunnels being always on. You could then use a SIP proxy or iptables magic to choose which tunnel was the best at any given time. I would go for the wifi. Maybe because I want to do a long-shot link. Also because I want to go to the virgin islands :). Good luck! -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automon = *1 one touch recording
2009/12/8 Joseph syscon...@gmail.com: After pressing *1 console is not showing anything indicating that the call is being recorded: -- Executing [...@office-closed:1] Playback(SIP/479-1270-680060b0, transfer) in new stack -- SIP/479-1270-680060b0 Playing 'transfer' (language 'en') -- Executing [...@office-closed:2] Dial(SIP/479-1270-680060b0, SIP/11IAX2/iaxy-322|30|rwW) in new stack -- Called 11 -- Called iaxy-322 -- Call accepted by 10.0.0.108 (format ulaw) -- Format for call is ulaw -- SIP/11-007855f0 is ringing -- IAX2/iaxy-322-6005 is ringing -- SIP/11-007855f0 answered SIP/479-1270-680060b0 -- Hungup 'IAX2/iaxy-322-6005' == Spawn extension (office-closed, 11, 2) exited non-zero on 'SIP/479-1270-680060b0' Did you make sure that your telephone actually sends the DTMF tones (the right way)? It seems that asterisk does not recognise incoming DTM or your verbosity level is not high enough. Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network config
Hi David, On Tue, 8 Dec 2009, David Gibbons wrote: snip A client has two offices in the Virgin Islands that MUST maintain data connectivity, and there are no available leased line options to run a P2P link between them. snip Is there line of sight? I've been wanting to do a long-shot wifi link and my company would give it a shot if you want :). Sadly no, because cruise ships park (dock?) directly in front of one the locations, which is directly between them. Worse high intensity radar blasts seem to give any kind of wireless signal we have attempted lots of trouble. If it weren't for the ships, this would work well I think, but as its happens the ships are the source of the client's revenue! snip And finally - is there a device that will manage the tunnel such that a high water mark of latency will also cause the tunnel to switch to the other link, rather than actual packet loss? See above. Fail-over routers have to wait some criteria are met in order to fail over (ping latency, ping loss, etc). This means that the connection you're using as the 'default' WILL go 'down' BEFORE it switches to the other one, regardless of the criteria used. Hmm, an excellent point. I suppose some amount of tweaking might cause the switch to happen before asterisk or the endpoint decides that the call is lost? Are these SIP timers that we might play with? Some amount of silent interruption might be tolerated during a switch, but a lost call is hard to accept. Another plan would be to set up two routers at the site with two separate VPN tunnels across the two different links, both tunnels being always on. You could then use a SIP proxy or iptables magic to choose which tunnel was the best at any given time. Hmm, another good thought. Now its getting complicated :) I would go for the wifi. Maybe because I want to do a long-shot link. Also because I want to go to the virgin islands :). Heh. Come on down! Water is fine... j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realm authentication
Actually yhe best one who answered me before is xavimes, but did not understand well his explaination, so I am still searching and need a help. The realm is like a domain and it is used for authentication, this kind of authentication is used when we are going to register from a wireless phone (like nokia mobile and so on), it gives more security. Well, the client software on the mobile is SIP, but how to setup at Asterisk the username and password under the realm domain, and how to setup the domain name, I am not able to know until now and I am looking for help. Any help? I wrote below what xavimes wrote for me: Regards Bilal From: Xavier Mesquida xavi...@yahoo.com Subject: Re: [asterisk-users] The SIP in the Mobile Phones are not able toregister on asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 141856.82049...@web30803.mail.mud.yahoo.com Content-Type: text/plain; charset=iso-8859-1 Have you set the realm in the sip settings in the mobile? Default one is asterisk . It's important too, defining Registration to Always on, because if not, it doesn't enable the wifi connection. Finally, don't enable compression and security You want something different than sip.conf? AFAIK it's all in there. And what you name realm should possibly be a context in asterisk language. Or did I get you wrong? Eckhard bilal ghayyad wrote: Hello List; Anyone can advise how realm authentication method is working? I mean, where to create the SIP username and password and where to create the realm that will be used for the authentication method for registration? Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo issue
The echo between our extensions (using Polycom 550 handsets) disappears once I removed the Digium echo module. We are still experiencing some echo on land line calls, using dahdi to connect to our PRI circuit. What kind of settings do you recommend for the txgain and rxgain? Do I make the gain changes in chan_dahdi.conf? Thank you! This is my system.conf: === # Span 1: WCT1/0 Wildcard TE121 Card 0 (MASTER) span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 echocanceller=mg2,1-23 # Span 2: WCTDM/0 Wildcard AEX410 Board 1 fxoks=25 echocanceller=mg2,25 fxoks=26 echocanceller=mg2,26 fxoks=27 echocanceller=mg2,27 # channel 28, WCTDM/0/3, no module. # Global data loadzone= us defaultzone= us This is my chan_dahdi.conf [trunkgroups] [channels] context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no faxdetect=incoming echotraining=800 rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 ;Uncomment these lines if you have problems with the disconection of your analog lines ;busydetect=yes ;busycount=3 immediate=no #include dahdi-channels.conf #include chan_dahdi_additional.conf From: Noah Miller noahisaacmil...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tue, December 8, 2009 7:37:28 AM Subject: Re: [asterisk-users] Echo issue Hi - I am having echo issues on our Asterisk box using a PRI circuit. I was using the software echo cancellation and that helped a bit but didn't solve it completely. So I went and bought a Digium echo cancellation module for the TE121 card. That made it even worst, getting more echo on external calls and between internal extension to extension. The echo doesn't happen all the time, but enough to get complaints from our users. Completely fed up with the issue, I removed the module from the card. Can someone guide me on how to fix/tune/address the echo issues. You can likely eliminate most echo on a PRI by setting txgain and rxgain. Are you using dahdi or zaptel? If Dahdi, what do your system.conf and chan_dahdi.conf look like? If zaptel, what do your zaptel.conf and zapata.conf look like? When you say you have echo on calls that are internal extension to internal extension, are the endpoints using dahdi/zaptel or some voip technology (sip, iax, mgcp, skinny, etc)? If voip, any echo is acoustically generated by the endpoints themselves. On voip calls I've often had this happen when the endpoints are using headsets, or have gain levels set very high. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo issue
On Tue, Dec 8, 2009 at 7:37 AM, Noah Miller noahisaacmil...@gmail.com wrote: Hi - I am having echo issues on our Asterisk box using a PRI circuit. I was using the software echo cancellation and that helped a bit but didn't solve it completely. So I went and bought a Digium echo cancellation module for the TE121 card. That made it even worst, getting more echo on external calls and between internal extension to extension. The echo doesn't happen all the time, but enough to get complaints from our users. Completely fed up with the issue, I removed the module from the card. Can someone guide me on how to fix/tune/address the echo issues. You can likely eliminate most echo on a PRI by setting txgain and rxgain. Are you using dahdi or zaptel? If Dahdi, what do your system.conf and chan_dahdi.conf look like? If zaptel, what do your zaptel.conf and zapata.conf look like? When you say you have echo on calls that are internal extension to internal extension, are the endpoints using dahdi/zaptel or some voip technology (sip, iax, mgcp, skinny, etc)? If voip, any echo is acoustically generated by the endpoints themselves. On voip calls I've often had this happen when the endpoints are using headsets, or have gain levels set very high. - Noah We found ourselves in a similar situation during our rollout and solved it with a quad-span Ditech echo-cancellation appliance (http://www.ditechnetworks.com/products/quad-2_echo-canceller.html). It's a couple of grand, but after months of playing with software EC, the hardware modules and every zaptel setting we could find, this appliance removed echo like flipping a switch. The metrics we later obtained from it clearly showed that we simply had tail on a long loop to an old CO switch that exceeded the maximum 128ms that either software EC or the hardware module could handle. The side benefits are that we get all sorts of metrics from the appliance, and we also get adaptive gain, which solved another problem we had with trying to find gain settings that suited both softly-spoken and strident users. The support from Ditech was excellent and we haven't looked back. CP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] meetme.conf adminpin - what does it do?
I can't seem to locate any documentation on what this does. I tested it out with a simple static conference room: exten = conference,1,MeetMe(,1aMqw) and a static room defined in meetme.conf: conf = 123456,22,1 Users can get in with either of the pins, but I don't see that it does anything - I can't access the admin menu, nor does it set the user as marked to open up the conference in a open-on-marked-enter conference. Ideally it would be great if this could be used as a hook to grant them as the marked user so a conference bridge could be opened up / taken off hold when they enter. hose ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme.conf adminpin - what does it do?
What you say...Hose (hose+aster...@bluemaggottowel.com): I can't seem to locate any documentation on what this does. I tested it out with a simple static conference room: exten = conference,1,MeetMe(,1aMqw) and a static room defined in meetme.conf: conf = 123456,22,1 Users can get in with either of the pins, but I don't see that it does anything - I can't access the admin menu, nor does it set the user as marked to open up the conference in a open-on-marked-enter conference. Ideally it would be great if this could be used as a hook to grant them as the marked user so a conference bridge could be opened up / taken off hold when they enter. Argh, nevermind, figured out the issue - it DOES work in letting an admin start the conference, but the dialplan was also waiting for a marked user (w flag). Removing that fixed it... hose ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Easy way to see what dahdi channels are being used
Hi, I have just recently been using DAHDI, and I wanted to know how to monitor capacity. Let's say I have two DS1 (23 channels) coming in, one for Florida (let's say) and one for New York. How can I get a reading of how many channels of each T1 port is being used at any given moment? Ideally, have two values, one for each T1. dahdi show channels doesn't show outgoing calls. Is there another command I am not aware of? Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Easy way to see what dahdi channels are being used
Core show channels shows all calls. you will get two entries for most calls, 1 for the dahdi channel and one for the sip phone using it. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Tuesday, December 08, 2009 3:45 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Easy way to see what dahdi channels are being used Hi, I have just recently been using DAHDI, and I wanted to know how to monitor capacity. Let's say I have two DS1 (23 channels) coming in, one for Florida (let's say) and one for New York. How can I get a reading of how many channels of each T1 port is being used at any given moment? Ideally, have two values, one for each T1. dahdi show channels doesn't show outgoing calls. Is there another command I am not aware of? Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Easy way to see what dahdi channels are being used
From the CLI: asterisk -rx 'core show channels' | grep DAHDI | sort -n Channels with a value of 1-23 are on your primary DS1, channels with a value of 25-47 are on your second DS1. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Mike l...@virtutel.ca wrote: Hi, I have just recently been using DAHDI, and I wanted to know how to monitor capacity. Let's say I have two DS1 (23 channels) coming in, one for Florida (let's say) and one for New York. How can I get a reading of how many channels of each T1 port is being used at any given moment? Ideally, have two values, one for each T1. dahdi show channels doesn't show outgoing calls. Is there another command I am not aware of? Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Limits
Thanks Jared, That solution was perfect! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith Sent: 07 December 2009 14:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Limits On Sun, 2009-12-06 at 08:49 -0500, Dan Journo wrote: I’m trying to figure out how to limit the number of concurrent calls a client can make. I prefer to use the GROUP() and GROUP_COUNT() dialplan functions to enforce arbitrary call limits in Asterisk -- Jared Smith Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Easy way to see what dahdi channels are being used
Thanks Tim and Danny. It seems a more direct way should be there, but that`ll work. Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Tuesday, December 08, 2009 16:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Easy way to see what dahdi channels are being used From the CLI: asterisk -rx 'core show channels' | grep DAHDI | sort -n Channels with a value of 1-23 are on your primary DS1, channels with a value of 25-47 are on your second DS1. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Mike l...@virtutel.ca wrote: Hi, I have just recently been using DAHDI, and I wanted to know how to monitor capacity. Let's say I have two DS1 (23 channels) coming in, one for Florida (let's say) and one for New York. How can I get a reading of how many channels of each T1 port is being used at any given moment? Ideally, have two values, one for each T1. dahdi show channels doesn't show outgoing calls. Is there another command I am not aware of? Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi restart kills server
I`ve just experience a dead server, because I ran /etc/init.d/dahdi restart . I had to reboot the server. Should I worry about something not being right in my install, or is there a known problem with doing this while Asterisk is running? I expected DAHDI channels to die, but not the whole server! Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Easy way to see what dahdi channels are being used
On Tue, 2009-12-08 at 19:04 -0500, Mike wrote: Thanks Tim and Danny. It seems a more direct way should be there, but that`ll work. A more direct way would be to use SNMP in Asterisk and keep statistics with Cacti. That way you will have an historical view of usage by hour, day, week and year. Even if you do not use cacti to create graphs you can still do an snmpwalk to query how many lines of what type are in use at any moment. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sequential dialing preferences
At 10:38 AM on 06 Dec 2009, Thomas Perron wrote:
I am trying to use a simple tool in the Dial plan so that if the first
number does not connect the logic will go to the second and/or third.
Basically, I want the call to ring and connect to the first number
Then, if it is not answered I want another number to try to get
connected Then, if second number does not answer I want the third to
be tried i only list the scenario for the first two numbers
Here is what I have now which works fine for the one and only
number...
exten = s,n,Dial(SIP/callwithus/12135551212,120,A(ginger3)) ;
Service line
so, will this work ... ..
exten =
s,n,Dial(SIP/callwithus/12135551212[SIP/callwithus/12145551212],120,A(ginger3))
;
Service line
Please send comments to make this work.
It'll work without the square brackets. The square brackets that are
shown in core show application Dial aren't meant to be put in
literally. They just signify that the stuff inside them is optional.
However, using Dial that way won't do what you're looking for. Instead,
it'll ring both (or all) devices at once, and the first one to answer
will get the call. The others will just be disconnected. If you want
it to ring the second number only after the first one didn't work,
you'll have to do that in your dialplan by checking ${DIALSTATUS} after
Dial.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
signature.asc
Description: PGP signature
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Re: [asterisk-users] Easy way to see what dahdi channels are being used
That`s my plan exactly, but for that I need some value to poll, and I was looking for the most efficient way to know that 12 out of 23 channels are being used. Seems that I need to massage the data more than I wanted, instead of using a dahdi show port 3 command. That`s what I meant by it being indirect. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Tuesday, December 08, 2009 19:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Easy way to see what dahdi channels are being used On Tue, 2009-12-08 at 19:04 -0500, Mike wrote: Thanks Tim and Danny. It seems a more direct way should be there, but that`ll work. A more direct way would be to use SNMP in Asterisk and keep statistics with Cacti. That way you will have an historical view of usage by hour, day, week and year. Even if you do not use cacti to create graphs you can still do an snmpwalk to query how many lines of what type are in use at any moment. -- Telecomunicaciones Abiertas de M xico S.A. de C.V. Carlos Ch vez Prats Director de Tecnolog a +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi restart kills server
you have to stop asterisk before restarting dahdi service On Dec 8, 2009, at 7:06 PM, Mike wrote: I`ve just experience a dead server, because I ran /etc/init.d/dahdi restart . I had to reboot the server. Should I worry about something not being right in my install, or is there a known problem with doing this while Asterisk is running? I expected DAHDI channels to die, but not the whole server! Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jamie A. Stapleton CBSi - Connecting your problems with solutions. Telephone: (804) 412-1601 Facsimile: (804) 412-1611 VideoConf: callto:jstapleton.computer-business.com Meet me on LinkedInhttp://www.linkedin.com/in/jstapleton Have I exceeded your expectations? Please share your experience with our Founder, Fred W. Brumbaughmailto:fbrumba...@computer-business.com LEGAL DISCLAIMER The information transmitted is intended solely for the individual or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of or taking action in reliance upon this information by persons or entities other than the intended recipient is prohibited. If you have received this email in error please contact the sender and delete the material from any computer. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Interesting problem with IP's
Have a trunk 1.4 asterisk, running on centos on the lan at work. A long story, but we had the entire work network on a public address range (90.1.0.x), going to a firewall, then out to the net. At home (192.168.1.x network) I have a router that connects to the firewall via a vpn tunnel. All was great. My cisco 7960 (192.168.1.100) was able to register with the asterisk server on 90.1.0.76 - and there was no audio problems whatsoever. I also must stress that I had nat=no and no nat-specific flags set in asterisk. However,the day came where the techs decided that we should be on a private internal network, and moved all of the devices onto a 10.0.x.x internal network. Needless to say, it wasn't an easy task. Now, although my vpn is connected to the new network, and I can access all of the machine as I used to be able to, I now only have 1-way audio on my phone !! (I can hear, and it gets progressively worse,the other party cannot hear me) Why would this have changed ? Do I need to do nat stuff now ? and why ? Interesting. Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi restart kills server
On Tue, Dec 08, 2009 at 07:06:52PM -0500, Mike wrote: I`ve just experience a dead server, because I ran /etc/init.d/dahdi restart . I had to reboot the server. What version of DAHDI (tools, linux)? What DAHDI hardware (if any) do you have? What do you have on /etc/dahdi/modules ? Should I worry about something not being right in my install, or is there a known problem with doing this while Asterisk is running? The module dahdi cannot be unloaded when Asterisk is running. Thus this restart is pointless. It should not crash the system, IIRC. Why did you restart dahdi? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automon = *1 one touch recording
On 12/08/09 11:11, Jared Smith wrote: On Mon, 2009-12-07 at 19:39 -0700, Joseph wrote: After pressing *1 console is not showing anything indicating that the call is being recorded: I find that I often have to adjust the featuredigittimeout setting in features.conf, as users tend to take their time between the * and 1 keys when turning on automon. -- Jared Smith Digium, Inc. Well, ;transferdigittimeout = 3 (default is 3 seconds) but this does not work or does not take any effect, this feature worked perfectly in Asterisk 1.2 I just tried it, I set: transferdigittimeout = 4 it doesn't work. I'm using cordless phone and I'm 100% sure that it take me less then 1.5 seconds to press *1 with one finger. However, when I tried pressing *1 using two fingers it worked. So, it seems to me transferdigittimeout setting doesn't work or doesn't take any effect. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and Junghanns QuadBRI
What's the output of: lspci -v -nn -s 08:00.0 # lspci -v -nn -s 08:00.0 08:00.0 ISDN controller [0204]: Cologne Chip Designs GmbH ISDN network Controller [HFC-8S] [1397:16b8] (rev 01) Subsystem: Cologne Chip Designs GmbH Device [1397:b552] Flags: medium devsel, IRQ 10 I/O ports at ec00 [size=8] Memory at febff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Kernel modules: hfcmulti, wcb4xxp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recording from billsec
I want to rebuild my mixmonitor file.But this time I just want the recording is from the time when the client answer the call,not from the beginning. Anybody can help? Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automon = *1 one touch recording
Joseph wrote: On 12/08/09 11:11, Jared Smith wrote: On Mon, 2009-12-07 at 19:39 -0700, Joseph wrote: After pressing *1 console is not showing anything indicating that the call is being recorded: I find that I often have to adjust the featuredigittimeout setting in features.conf, as users tend to take their time between the * and 1 keys when turning on automon. -- Jared Smith Digium, Inc. Well, ;transferdigittimeout = 3 (default is 3 seconds) but this does not work or does not take any effect, this feature worked perfectly in Asterisk 1.2 I just tried it, I set: transferdigittimeout = 4 it doesn't work. I'm using cordless phone and I'm 100% sure that it take me less then 1.5 seconds to press *1 with one finger. However, when I tried pressing *1 using two fingers it worked. So, it seems to me transferdigittimeout setting doesn't work or doesn't take any effect. Hmmm... That would possibly also explain why I always succeed in doing *2 xfers, and my wife always fails... I always have 2 fingers on those buttons, and she is the single-finger-typing-kind'o'gal... Weird though that unattended (##) xfers DO work for her as well... --FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automon = *1 one touch recording
There is another setting which I can't find at the moment which controls this -- its normally set to 500ms. Francesco Peeters france...@fampeeters.com wrote: Joseph wrote: On 12/08/09 11:11, Jared Smith wrote: On Mon, 2009-12-07 at 19:39 -0700, Joseph wrote: After pressing *1 console is not showing anything indicating that the call is being recorded: I find that I often have to adjust the featuredigittimeout setting in features.conf, as users tend to take their time between the * and 1 keys when turning on automon. -- Jared Smith Digium, Inc. Well, ;transferdigittimeout = 3 (default is 3 seconds) but this does not work or does not take any effect, this feature worked perfectly in Asterisk 1.2 I just tried it, I set: transferdigittimeout = 4 it doesn't work. I'm using cordless phone and I'm 100% sure that it take me less then 1.5 seconds to press *1 with one finger. However, when I tried pressing *1 using two fingers it worked. So, it seems to me transferdigittimeout setting doesn't work or doesn't take any effect. Hmmm... That would possibly also explain why I always succeed in doing *2 xfers, and my wife always fails... I always have 2 fingers on those buttons, and she is the single-finger-typing-kind'o'gal... Weird though that unattended (##) xfers DO work for her as well... --FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_voicemail. Help me to find typo source ...
Hi, In /var/log/asterisk/full (Asterisk 1.6.2-rc6), I can see : [Dec 8 15:02:17] VERBOSE[10283] config.c: == Parsing '/var/spool/asterisk/voicemail/default/103/INBOX/msg.txt': [Dec 8 15:02:17] VERBOSE[10283] config.c: == Found [Dec 8 15:02:17] VERBOSE[10283] file.c: -- SIP/103-0784 Playing 'vm-message.gsm' (language 'fr') [Dec 8 15:02:18] WARNING[10283] file.c: File vm-recieved does not exist in any format [Dec 8 15:02:18] WARNING[10283] file.c: Unable to open vm-recieved (format 0x8 (alaw)): No such file or directory [Dec 8 15:02:18] WARNING[10283] say.c: Unable to play message vm-recieved Strangely, I can't find any line in *.c files containing such vm-recieved string. If I add ln -s vm-received vm-recieved in appropriate directory, those error messages disappear. Though I've found a work around, I would be very happy to pin point root cause but I'm a bit lost in the amount of files in Asterisk code. Any hint ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
