[asterisk-users] Purposely setting red alarm on PRI for testing purposes
Does anyone know if is possible to purposely set red alarm status on PRI circuit for testing purposes (other than unplugging it). I have looked for a console command which might allow this ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a variable for SIP response codes?
Once the call is hung up it is too late. I need to interpret the SIP
response codes prior to hangup so I can play an appropriate recorded voice
announcement.
On 4/9/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
Eric Bishop wrote:
> Hi all,
>
> I want to implement certain actions based on SIP response codes. Is
there a
> similar variable such as ${DIALSTATUS} that comes back with the relevant
> SIP
> response code for a call?
I believe there is SIPGetHeader, but Asterisk tries to translate
whatever code it gets from the specific technology (PRI, SIP, IAS2,
MGCP, SCCP, H323, etc) into an Asterisk HANGUPCAUSE which is mostly
Q.931 codes. HANGUPCAUSE will not tell you the SIP response code, but
it will tell you much more than DIALSTATUS will.
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[asterisk-users] Is there a variable for SIP response codes?
Hi all,
I want to implement certain actions based on SIP response codes. Is there a
similar variable such as ${DIALSTATUS} that comes back with the relevant SIP
response code for a call?
--- Thanks
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[asterisk-users] Any way to get rid of AEL created contexts?
"show dialplan" keeps showing contexts created by AEL. I tried deleting /etc/asterisk/extensions.ael but kept getting these messages in the Asterisk log: Feb 14 21:39:53 WARNING[6074] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Feb 14 21:39:53 WARNING[6074] pbx.c: Requested contexts didn't get merged Is there any way to delete or disable AEL? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 603 driving me nuts
I do need MWI notifcation, just not on this particulary trunk. Is there anyway to to turn off MWI on a particular trunk or can it only be done globally? On 2/22/07, Olle E Johansson <[EMAIL PROTECTED]> wrote: 22 feb 2007 kl. 08.24 skrev Davy Chan: > **>I have one Asterisk box registering to another via SIP and on > the registar > **>console I keep getting: > **> > **>-- Got SIP response 603 "Declined (no dialog)" back from > xxx.xxx.xxx.xx > **> > **>Anyone know how to turn off this "feature"? > > Look at: > > http://lists.digium.com/pipermail/asterisk-users/2007-February/ > 179168.html > > The message is popping up because Asterisk's new behavior to > SIP NOTIFY messages carrying Message Waiting Indication (MWI) info. > > See ya... Why enable MWI notification when you don't need it? /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 603 driving me nuts
Surely there must be a simpler way than patching the Asterisk code? After all this is Asterisk-to-Asterisk registration not some third party softswitch idiosyncrasy. Would setting up fake voicemail boxes help? On 2/22/07, Davy Chan <[EMAIL PROTECTED]> wrote: **>I have one Asterisk box registering to another via SIP and on the registar **>console I keep getting: **> **>-- Got SIP response 603 "Declined (no dialog)" back from xxx.xxx.xxx.xx **> **>Anyone know how to turn off this "feature"? Look at: http://lists.digium.com/pipermail/asterisk-users/2007-February/179168.html The message is popping up because Asterisk's new behavior to SIP NOTIFY messages carrying Message Waiting Indication (MWI) info. See ya... d.c. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP response 603 driving me nuts
I have one Asterisk box registering to another via SIP and on the registar console I keep getting: -- Got SIP response 603 "Declined (no dialog)" back from xxx.xxx.xxx.xx Anyone know how to turn off this "feature"? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Passing a variable from one Asterisk box to another
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten => _23XX,1,SetVar(Foo=1234)
exten => _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call dials into Box 2 the variable Foo does not get passed...
Does anyone have any clever ideas?
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[asterisk-users] Fwd: Can anyone help me out with Polycom 2.1 firmware please?
Any kind Polycom dealers out there? -- Forwarded message -- From: Eric Bishop <[EMAIL PROTECTED]> Date: Feb 14, 2007 8:10 PM Subject: Can anyone help me out with Polycom 2.1 firmware please? To: Asterisk Users Mailing List - Non-Commercial Discussion < asterisk-users@lists.digium.com> Would be greatly appreciated ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme - is this statement from the Wiki still true?
"The conference bridge runs Ulaw codec by default. If you let people connect with GSM or other codecs, Asterisk will use CPU power to convert audio between codecs" ... What about alaw channels is there any transcoding work being done there? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Native format prompts
Hi all, I am trying to implement native format (ulaw) voice prompts and music on hold. Different documentation has different file extensions. Does Asterisk recognise them all? So far I have .ulaw .ul .pcm . Which should I use so Asterisk recognises them as native uLaw files ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can anyone help me out with Polycom 2.1 firmware please?
Would be greatly appreciated ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP privacy headers
thanks for that. Do you know what P-Asserted-Identity needs to be set to to hide caller ID via privacy headers? On 2/5/07, Darryl Dunkin <[EMAIL PROTECTED]> wrote: Look here: http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Eric Bishop *Sent:* Sunday, February 04, 2007 15:43 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] SIP privacy headers Hi, Out ITSP has told us to user "SIP privacy headers" to hide outbound caller ID. Does anyone know how or if this can be done in Asterisk. I tried exten => s,3,SIPAddHeader(privacy=on) prior to executing Dial but no luck. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP privacy headers
Hi, Out ITSP has told us to user "SIP privacy headers" to hide outbound caller ID. Does anyone know how or if this can be done in Asterisk. I tried exten => s,3,SIPAddHeader(privacy=on) prior to executing Dial but no luck. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Polycom Firmware -- Was: [asterisk-users] Asterisk 1.4 & Polycom buddy status
I second that request On 1/25/07, Kenneth Padgett <[EMAIL PROTECTED]> wrote: > I ran into this problem with an early batch of IP650s. Polycom's firmware > version 2.0.3b made this issue go away. Speaking of Polycom firmware, anyone have an up to date source for the stuff? The site I ordered from took down their FTP site that had it. :( -Kenneth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone know what this warning is about? Nothing in list history about it either..
On inbound calls from my SIP provider I get multiple warnings as follows: WARNING[5351]: chan_sip.c:7086 check_via: 'MODH6QD' is not a valid host Everything else works but these warnings are a pain and I don't know what they are about Nothing on previos lists or Google explains... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which g729 module for HP DL 360 G3 (Xeon CPU's)?
I am running a HP DL360 G3 ans want to know the optimal g729 module for it. There don't seem to be any optimised for Xeon's. I am currently using i686, but is there a better one to match my Xeon CPU's? [EMAIL PROTECTED] ~]# cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 2 model name : Intel(R) Xeon(TM) CPU 2.80GHz stepping: 7 cpu MHz : 2799.656 cache size : 512 KB physical id : 3 siblings: 1 core id : 3 cpu cores : 1 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe cid xtpr bogomips: 5602.71 processor : 1 vendor_id : GenuineIntel cpu family : 15 model : 2 model name : Intel(R) Xeon(TM) CPU 2.80GHz stepping: 9 cpu MHz : 2799.656 cache size : 512 KB physical id : 0 siblings: 1 core id : 0 cpu cores : 1 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe cid xtpr bogomips: 5597.58 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] g726 voice prompts
Anyone know if it posible to make voice promps native g726 or g711 format? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click to dial apps always show from "asterisk"
I have seen the answer to this question previously, perhaps I am just not asking the question correctly. For manager-based apps that do not explicitly set a callerid is there anyway to overide the system default of "asterisk" On 11/28/06, Tim Panton <[EMAIL PROTECTED]> wrote: On 28 Nov 2006, at 03:01, Eric Bishop wrote: > I am trying to do it with FOP and Calling Circles. Both have closed > code. Anyway to do it from Asterisk? > You could use the 'Local' channel as the argument to the originate command and then set it in the dialplan. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Do extra CPU's help?
Do extra CPU's without hyperthreading help? On 11/28/06, Don <[EMAIL PROTECTED]> wrote: hyperthreading screws ours up...we actually run better with hyperthreading off... hyperthreading results seem to vary from different people you talk too. - Original Message ----- *From:* Eric Bishop <[EMAIL PROTECTED]> *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Sent:* Monday, November 27, 2006 10:54 PM *Subject:* [asterisk-users] Do extra CPU's help? Hi all, We have Xeon-based system with only 1 (hyperthreaded) CPU (in a HP DL360). We are seeing high load on multiple meetme session as well as g729 transcoding. My question is will putting an extra CPU help or does Asterisk just run on a single CPU. -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.17/553 - Release Date: 11/27/2006 4:00 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Do extra CPU's help?
Hi all, We have Xeon-based system with only 1 (hyperthreaded) CPU (in a HP DL360). We are seeing high load on multiple meetme session as well as g729 transcoding. My question is will putting an extra CPU help or does Asterisk just run on a single CPU. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click to dial apps always show from "asterisk"
I am trying to do it with FOP and Calling Circles. Both have closed code. Anyway to do it from Asterisk? On 11/27/06, mitcheloc <[EMAIL PROTECTED]> wrote: You can use the CallerID parameter of the Originate command to override the default caller id. It's listed on the wiki with examples: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate Cheers On 11/27/06, Eric Bishop <[EMAIL PROTECTED]> wrote: > We have calls that originate click-to-dial apps that use the manager > interface. As most of you know these apps first ring your handset so that > you pickup the handset and then place the outbound call once you have picked > up. > > When they first ring my handset (before me picking up the handset) the call > shows as being "from asterisk". Is there any way to change this "from" name > to something the average joe understands such as "PBX"? > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to kill a meet me room at midnight
You can't hanup channels with a call file you can only create them no? On 11/28/06, Noah Miller <[EMAIL PROTECTED]> wrote: > You could write an extension which executes meetme kick, for all the > channels, but I am not sure how to execute such a thing at a given > time. Create a call file, and schedule it to run with cron. The following page on the wiki shows something similar: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out You can adapt it to suit your needs. - Noah > on Saturday 11/25/2006 Eric Bishop([EMAIL PROTECTED]) wrote > > Not quite what I'm looking for. I ant to hang up all channels (zap or sip) > > in meetme room 5 > > > > On 11/23/06, Michiel van Baak <[EMAIL PROTECTED]> wrote: > > > > > > On 19:18, Thu 23 Nov 06, Eric Bishop wrote: > > > > Other than rebooting the server or restarting Asterisk from cron does > > > anyone > > > > know how to kill a meetme room at midnight. Or perhaps other creative > > > ways > > > > people deal with callers who don't hang up. > > > > > > You can use soft hangup > > > > > > -- > > > Michiel van Baak > > > [EMAIL PROTECTED] > > > http://michiel.vanbaak.eu > > > GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD > > > > > > "Why is it drug addicts and computer afficionados are both called users?" > > > > > > ___ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > Not quite what I'm looking for. I ant to hang up all channels (zap or sip) in meetme room 5On 11/23/06, Michiel van Baak < > > [EMAIL PROTECTED]> wrote:On 19:18, Thu 23 Nov 06, Eric Bishop wrote:> Other than rebooting the server or restarting Asterisk from cron does anyone > > > know how to kill a meetme room at midnight. Or perhaps other creative ways> people deal with callers who don't hang up.You can use soft hangup <chan>--Michiel van Baakmailto:[EMAIL PROTECTED]"> > > [EMAIL PROTECTED]http://michiel.vanbaak.eu";> http://michiel.vanbaak.euGnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD";> http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD > > "Why is it drug addicts and computer afficionados are both called users?"___--Bandwidth and Colocation provided by http://Easynews.com";> > > Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users";> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > --Bandwidth and Colocation provided by Easynews.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 > [EMAIL PROTECTED] > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Click to dial apps always show from "asterisk"
We have calls that originate click-to-dial apps that use the manager interface. As most of you know these apps first ring your handset so that you pickup the handset and then place the outbound call once you have picked up. When they first ring my handset (before me picking up the handset) the call shows as being "from asterisk". Is there any way to change this "from" name to something the average joe understands such as "PBX"? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to kill a meet me room at midnight
Not quite what I'm looking for. I ant to hang up all channels (zap or sip) in meetme room 5 On 11/23/06, Michiel van Baak <[EMAIL PROTECTED]> wrote: On 19:18, Thu 23 Nov 06, Eric Bishop wrote: > Other than rebooting the server or restarting Asterisk from cron does anyone > know how to kill a meetme room at midnight. Or perhaps other creative ways > people deal with callers who don't hang up. You can use soft hangup -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD "Why is it drug addicts and computer afficionados are both called users?" ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to kill a meet me room at midnight
Other than rebooting the server or restarting Asterisk from cron does anyone know how to kill a meetme room at midnight. Or perhaps other creative ways people deal with callers who don't hang up. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calls "from asterisk"
When we have calls that originate click-to-daial apps that use the manager interface they always originate "from asterisk" is there any way to change the "from" name? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hosted asterisk
Dean, I know Qtec definately do, however their offering is pretty much focused only on businesses and they offer their service only via their private IP network - not via the Internet. http://www.qtec.com.au -- Eric On 11/17/06, Dean Collins <[EMAIL PROTECTED]> wrote: I have a client who is looking for hosted asterisk in Australia, as far as I can tell ATP is the only company offering this. Does anyone else on this list know of someone? Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SIP 2.0.2 firmware
I second that request. On 11/4/06, Kevin Bockman <[EMAIL PROTECTED]> wrote: Hi,Would anyone be kind enough to send me the 2.0.2 SIP firmware? I askedVoipSupply for it on Wednesday, nagged them again on Thursday and theydid not even send the request yet. I was supposed to have it 'Friday morning' at the latest. I'm doing equipment upgrades this weekend sothis is the time to do it. I've been having random phone crashing using2.0.1.I also asked VoipSupply for firmware a month or so ago and they never sent it.It is not listed yet on the freedom file site.Thanks,Kevin___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FOP run control for CentOS/RHEL
Anyone have a sane rc script for FOP on CentOS/RHEL systems? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Buddy Watch Setup help request
Do you have anything special in your sip.conf for the Polycom phones?On 10/4/06, Scott Higginbotham <[EMAIL PROTECTED] > wrote:Here is an example of what I have:in extensions.conf:exten => 2111,hint,SIP/2111 exten => 2111,1,Dial(SIP/2111,60)my Polycom's all pull config's via TFTP. Due to the nature of our setup, Ihave individual configuration files for each phone. I have in my file the following entries for the users I wantto watch via presence:UserJoe 21112130 010and in my file for the phones I have:up.useDirectoryNames="1" feature.1.name="presence" feature.1.enabled="1"included in my .That should be all you need. Hope that helps. Scott HigginbothamSystems / Network Operations Manager215.259.2185 or 1.800.835.5710 ext 2185[EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]On Behalf Of Robert JenkinsSent: Tuesday, October 03, 2006 4:09 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [asterisk-users] Polycom Buddy Watch Setup help request(Subject changed from 'Re: [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?' as it was a bit off topic).>From: [EMAIL PROTECTED][mailto: [EMAIL PROTECTED]] On Behalf Of Eric Bishop>Sent: 03 October 2006 07:34>To: Asterisk Users Mailing List - Non-Commercial Discussion>Subject: Re: [asterisk-users] Polycom Buddy Watch Broken with 2.0.1Firmware?>>Does anyone have an end-to-end summary of how they have successfully set upthe buddy feature including all the relevant Asterisk and Polycom config>snippets. All I have been able to do so far is scrounge up bits and peices from the list and Wiki - nothing that covers the entire process... I think a>lot of people would benefit from that (myself included)...I second this!I'm about to set up some Polycom 601 + Sidecars and I'm also having difficulty finding anything covering the overall 'buddy' config.Examples would be greatly appreciated.Robert Jenkins___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?
Does anyone have an end-to-end summary of how they have successfully set up the buddy feature including all the relevant Asterisk and Polycom config snippets. All I have been able to do so far is scrounge up bits and peices from the list and Wiki - nothing that covers the entire process... I think a lot of people would benefit from that (myself included)... On 10/3/06, Paul Dugas <[EMAIL PROTECTED]> wrote: Install went fine. No troubles other than this and it'd be minor if oneof the reasons for the update wasn't to expand the number of buddiesallowed on the IP601+sidecards we're adding for the attendant. Ugh... Anyway, directory entries haven't changed:^M DoeJane10011 1The config entries you referred to are set in my global sip.cfg andapply to all of the units. Looks right to me. Did some sniffing and Asterisk is sending a NOTIFY like so:...xmlns:pp="urn:ietf:params:xml:ns:pidf:person" xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status"xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"entity="sip:[EMAIL PROTECTED] ">Ready sip:[EMAIL PROTECTED]open--- Extension Changed 1001 new state Idle for Notify User x1002 pbx*CLI>HmmmOn Mon, 2006-10-02 at 22:14 -0400, Scott Higginbotham wrote:> I did the same thing with the Polycom's - upgraded all mine from 1.6.x to> 2.0.1 but I had great success and no problem with the buddy watch / presence > feature --- if anything, it works a little better.>> Whats your -directory.xml configuration file look like? Did> you make any changes to the > line of:>> up.useDirectoryNames="1" feature.1.name="presence" feature.1.enabled="1">> In the config?>> Scott Higginbotham> Systems / Network Operations Manager > 215.259.2185 or 1.800.835.5710 ext 2185> [EMAIL PROTECTED]>> -Original Message-> From: [EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED]]On Behalf Of Paul Dugas> Sent: Monday, October 02, 2006 8:01 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion> Subject: [asterisk-users] Polycom Buddy Watch Broken with 2.0.1> Firmware?>>> I updated a batch of Polycom IP501 phones and an IP601 to the 2.0.1> firmware to get the new NAT keep-alive feature and the ability to watch> more than a handful of buddy contacts but it appears to have broken the> buddy-watch feature. Is anyone seeing this? Anybody know if it's a > Polycom problem or something on the Asterisk end?>> I'm running a recent (2 days ago) copy of the 1.2 trunk. In a rather> bone-headed move, I updated the firmware and Asterisk at the same time > so I'm unable to tell which is the culprit.>> Curious,>> Paul>> --> Paul Dugas, Computer EngineerDugas Enterprises, LLC> [EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Park> http://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA> --> This e-mail and any attachments are confidential. If you receive > this message in error or are not the intended recipient, you should> not retain, distribute, disclose or use any of this information and> you should destroy the e-mail and any attachments or copies. >> ___> --Bandwidth and Colocation provided by Easynews.com -->> asterisk-users mailing list> To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users--Paul Dugas, Computer EngineerDugas Enterprises, LLC [EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Parkhttp://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA--This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you shouldnot retain, distribute, disclose or use any of this information andyou should destroy the e-mail and any attachments or copies.___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How can I stop lost DNS from killing Asterisk?
Hi All,When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Leased line interconnect
So would both work with Asterisk as an interconnect? The configuration I want is:Asterisk/TE410 (Site A) <---> 30 channels in the service provider network <---> Asterisk/TE410 (Site B)With the 31x64kbps leased line the service provider gives the following specifications: Max bandwidth: 1984kbpsFraming: G.704Line encoding: HDB3Customer interface: Electrical G.703 With the E-1 the service provider gives the following specifications: Max bandwidth: 2048kbps Framing: Clear Channel Line encoding: HDB3 Customer interface: Electrical G.703 The question is will either one work with Asterisk? And what are pros and cons of each service for use in conjunction with each. Could I run a PRI protocol over either one since I will cintrol both ends? On 9/23/06, Jay R. Ashworth <[EMAIL PROTECTED]> wrote: On Sat, Sep 23, 2006 at 08:22:18AM +1000, Eric Bishop wrote:>We are looking to interconnect 2 Asterisk boxes at seperate sites via a TDM>leased line, rather than IP mainly for commercial reasons. Our network >provider is offering us either a 31x64kbps leased line or an E1. Am I just>ignorant or are these the same thing? An E1 has 30 B channels and 1 D>channel. That is also I guess what we would run over the 31x64kbps leased >line so what is the difference? Which should I choose and would work best>with Asterisk.My intuition is you're gonna spend a whole lot of extra money oninterface cards doing it that way. An E-1 is a 32-timeslot circuit; usually broken up as 30B+1D+sync.If you get it as unframed data, rather than a PRI, you can have all 31slots to yourself, since you don't need one to talk to the switch. You might want, if you can, to expand on your "commercial reasons". Ifyou're getting a leased line, you can *still* put the appropriatehalf-bridges on each end, and run IP over Ethernet over that line... which would probably be cheaper than the T-cards.Plus, I'm told that it's a Bad Idea to have multiple T-cards in onechassis... though I can't imagine why...Cheers,-- jra--Jay R. Ashworth [EMAIL PROTECTED]Designer Baylink RFC 2100Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274"That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Leased line interconnect
Hi all,We are looking to interconnect 2 Asterisk boxes at seperate sites via a TDM leased line, rather than IP mainly for commercial reasons. Our network provider is offering us either a 31x64kbps leased line or an E1. Am I just ignorant or are these the same thing? An E1 has 30 B channels and 1 D channel. That is also I guess what we would run over the 31x64kbps leased line so what is the difference? Which should I choose and would work best with Asterisk. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What I always get asked in SME * deployments
When ever we do a roll out of Asterisk in a small business environment replacing an old key system or legacy PBX the receptionist always asks us, "How do I know if someone is on a call before transferring them?". My typical answer is "why do you need to know, just do an attended transfer and if they can take the call they will, if they can't just tell the caller the person is busy". If the receptionist insists on "knowing" we give them FOP. Has anyone out there devised a better way to let a receptionist "know if someone is on a call"? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Binary/unreadable configuration files?
Anyone know if it possible to create binary/obfuscated/ human unreadable extensions.conf/sip.conf etc.? We would like to deploy a system in an environment where not giving out root is still not enough. We want to hide the contents of these normally plain text files. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone out there using Junghanns ISDNguard?
Do you need BRI stuff to use the ISDNguard? Also can you make the switch manually rather than relying on heartbeat auto failover?On 7/11/06, Tzafrir Cohen <[EMAIL PROTECTED] > wrote:On Tue, Jul 11, 2006 at 08:55:41PM +1000, Eric Bishop wrote:> If so can you comment on how well it has (or hasn't) worked for you? Trivia: bristuff includes an Asterisk application called 'Segfault'intended to help test the ISDNGuard.The code is pretty simple. And I bet that the feature of that app hasbeen a bug of other apps in the past ;-) --Tzafrir Cohen sip:[EMAIL PROTECTED]icq#16849755 iax:[EMAIL PROTECTED]+972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone out there using Junghanns ISDNguard?
If so can you comment on how well it has (or hasn't) worked for you? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Blended?
What us meant by "blended rate"? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any Polycom dealers willing help out?
Hi All, We are in search of the latest Polycom firmware SIP 1.6.6 as per http://www.polycom.com/resource_center/1,,pw-492,00.html Can someone help? We have legitimately obtained these phones but even our official distributor can't get their hands on updated firmware. The only thing we have found is http://www.freedomphones.net/polycom/files/?M=A which has only older versions. Are there any kind Polycom authorized dealers who can help me? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "sip show inuse" is useless!
I have tried it with 1.2.7.1 and 1.2.9.1. Same issue with both and only on the SIP trunk, not on endpoints.On 6/19/06, William Piper <[EMAIL PROTECTED]> wrote: What version of * are you using? I am running 1.2.7.1 with call-limit= and it works fine. bp On 6/19/06, Eric Bishop < [EMAIL PROTECTED]> wrote: Hi all,We have a SIP trunk with * and even when there are calls in progress "sip show inuse" always shows 0 calls in progress. I have outgoinglimit and incominglimit limit set and have also tried call-limit. "sip show inuse" works fine with SIP handsets though very frustrating. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] "sip show inuse" is useless!
Hi all, We have a SIP trunk with * and even when there are calls in progress "sip show inuse" always shows 0 calls in progress. I have outgoinglimit and incominglimit limit set and have also tried call-limit. "sip show inuse" works fine with SIP handsets though very frustrating. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] "Reserving" a conference room
Hi all, We have executives who use conference rooms. The typical scenario is that one of them will organise a conference a few hours in advance and email everyone the details, however is there anyway the they can "reserve" a conference room number? For example if they organise a conference in room 123 at 4pm and send out all the details then someone else in the meantime dynamically creates conference room 123. This is what we are trying to avoid. I know we can set up a static conference room and pin, but we would like to do it with dynamic conference rooms. thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 941 missing blind transfer soft button?
Hi all, I have previously (and briefly) use a Sipura 941 before. I could have sworn that it has a blind transfer soft key when on a call. Now running the latest firmware (4.1.12a) the only soft keys that come up while on a call are for attended transfer and 2 way conference. Can anyone tell me if I was just imagining it being there? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-941 called number distinctive ring with Personal Directory
Hi All, I know this can be acheived in the Asterisk dial plan however for non-technical reasons I need to be able to do it using the SPA-941 Personal Directory feature. An entry such as the following matches the CALLING number fine but I need the match the CALLED number. In all the specs of the SPA-941 it says it can do distinctive ring based on called and calling number, however I can only seem make a CALLING number match. n=Joe Bloggs;p=0123456789;r=1 Can anyone help? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] US telco lingo
Could someone explain to a non-US dummy the following phrases I have seen on the list. "I can provide you with tier 1 termination 6/6. I can blend or NPANXX breakout." "We provide US48 termination, blended rate for 1 MOU and above is .008 with 6/6." What is 6/6? What is US48? What is blended? What is MOU? What is NPANXX breakout? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail WAV to PDA Problems
I also have an 8700g. Have you managed to figure out how to play .wav voicemails?On 5/13/06, Kerry Garrison < [EMAIL PROTECTED]> wrote:Our system is running all of the latest code and freepbx and would send the attachment to my MDA just fine and I was able to play it without anyproblem. My problem was that the MDA is a worthless turd and a complete jokeas a phone. I took it back and switched to the backberry 8700g which has its own attachment problems.> -Original Message-> From: [EMAIL PROTECTED]> [mailto: [EMAIL PROTECTED]] On Behalf Of> Peder @ NetworkOblivion> Sent: Friday, May 12, 2006 9:02 AM> To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Voicemail WAV to PDA Problems>> Our asterisk server has been up and running for over a year> and it works great. I have emails going to my account as an> attachment and I can listen to them on the desktop and it > works fine. I just got a T-Mobile MDA that runs Windows> Pocket (or whatever they call it) and it can check email. If> I have it download the email, it gets the attachment, but it> can't seem to play it (it CAN play wav files). If I take the > email that was sent to my home account and then "forward it> to myself" and let the MDA pick it up, then it can play the> attachment. So clearly it isn't an issue playing WAV's, or> even WAV's from Asterisk, it's some email attachment issue > with the way Asterisk or Postfix sends the attachment.> Has anybody else run into this problem? If so, any help> would be appreciated.>> Peder>> ___ > --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users>___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone willing to share an Australian dialplan.xml file for Cisco phones?
___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting ptime attribute in SDP invite
Is it possible for Asterisk to set the ptime attribute on outbound calls in SDP invite? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any Polycom dealer willing to help?
Hi All, We are in search of the latest Polycom firmware SIP 1.6.5 and BootROM 3.1.3 as per http://www.polycom.com/resource_center/1,,pw-492,00.html Can someone help? We have legitimately obtained these phones but even our official distributor can't get their hands on updated firmware. The only thing we have found is http://www.freedomphones.net/polycom/files/?M=A which has only old versions. Are there any kind Polycom authorized dealers who can help me? -- Eric ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P Really Bad Echo
Is this with the TE411P? Also what do you mean by "pulled the zaptel trunk source"?On 2/17/06, Stagg Shelton < [EMAIL PROTECTED]> wrote:This is my last update to my issue. Finally my echo problem is resolved. On Monday morning 2/13/06 I pulled the the zaptel trunksource. That night after my customers core business hours we built thenew zaptel drivers, rebuilt libpri, asterisk, asterisk-addons. My echo disappeared almost entirely we made a few tweaks with the tx and rxgain settings. My echo problem disappeared completely with theadditional tweaks to txgain. Occasionally at the very beginning of alocal call echo will exist for a second or so, but then it goes away. In two operating days there has only been one notice from a user aboutexperiencing an echo. All the users were informed that they shouldnotify us of any echo experiences.Here are my final configurations zaptel trunk pulled 2/13/06 approx 10:00am est.Asterisk 1.2.4LibPri 1.2.2Asterisk-Addons 1.2.1Asterisk-Sounds 1.2.1/etc/zaptel.conf=span=1,1,0,esf,b8zsbchan=1-23 dchan=24#bchan=25-47#dchan=48#bchan=49-71#dchan=72#bchan=73-95#dchan=96fxoks=97-100loadzone = usdefaultzone=us/etc/asterisk/zapata.confcontext=from-pstn switchtype=nationalpridialplan=nationalsignalling=pri_cperesetinterval=neverfaxdetect=incomingusecallerid=yesechocancel=yesechotraining=800rxgain=4.5txgain=-13.5group=0channel=>1-23 Thank You for all of your pointers and support in this issue.Stagg Sheltonwww.oneringnetworks.com___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to create latency on purpose
Hi All, I have a Digium card in my Asterisk server configured as pri_net and I want to introduce latency on it in order to simulate PSTN conditions and test some echo canceller hardware. Is it possible to purposefully introduce latency and echo in a controlled fashion in order to do so? Thanks.. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P Really Bad Echo
Nope only bad feedback here. The software EC in Asterisk worked much better for me than did the VPM on the TE411P.On 2/13/06, Isaac Xiao (KVB Kunlun Pty Limited) <[EMAIL PROTECTED]> wrote: What version of Asterisk and Zaptel you were using? Did you try latest Asterisk 1.2.4 and Zaptel 1.2.3? Anyone has good feedback for TE411P? Isaac Xiao Stagg Shelton wrote: > It was Digium's opinion that perhaps the card had a VPM. We got a > replacement TE411P, I implemented it tonight and still the exact same > echo problem. At this point I feel like I can rule out failed hardware. > > I contacted Digium support and now they are telling me it's something > with my carrier, and I should call them. I called Bellsouth, and they > ran a full stress test on the circuit taking me offline for about 30 > minutes. > > The end result is that the circuit test passed with no errors. > Bellsouth says it's not in their network, Digium says its not their > card, and I have a te411p with VPM disabled in the wct4xx kernel > module because something doesn't work the way it should. My customer > is wanting to know about sangoma cards with the echo cancellation, and > at this point I'm nervous to recommend any hardware. I'm going to > look into the sangoma that you suggested. Are there any other kinds > of products that I could look into either Passive or Active.> > Thanks > > Stagg Shelton > www.oneringnetworks.com> ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any way to grep through fast moving console messages?
Or perhaps slow them down or pipe to a file? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P Really Bad Echo
Kevin, I have experienced the same issue. I get worse echo with the VPM installed than with software EC. Have had it at 2 different sites with 2 different TE411P's. - EricOn 2/6/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: Stagg Shelton wrote:> I just implemented a system using a TE411P hardware echo cancellation> card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as> I always have. To my surprise calls out to the PSTN had a terrible > echo. 1 - 2 second delay, and quite clear. The echo was so bad that I1 to 2 _seconds_? There is no echo canceler anywhere that will handlethat much echo delay.Did you actually remove the VPM, or just disable it? Please check whether you have this problem with the VPM installed but disabled,because it could be a bad VPM.Finally... I know it's Sunday night, but you should really pursue thiswith Digium Support tomorrow morning :-) ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.3 CentOS 4.x RPMS
Do you have step by step instructions on how you created these RPMs. I would like to create a few of my own but compiled for my own custom kernel and patchea and am not very familiar with RPM packagingOn 1/27/06, Andrew McRory <[EMAIL PROTECTED] > wrote:Available in the usual place. ftp://ftp.linuxsys.com/pub/releases/CentOS-4.0This release includes minor spec changes, spandsp 0.0.2pre23, a newSangoma wanpipe RPM for use with the LSE kernel rpm and an AMPinstallation document. Best Regards,--Andrew McRory - President/CTOLinux Systems Engineers, Inc. - http://www.linuxsys.comLocated in beautiful Tallahassee, FloridaOffice 850-224-5737 Office 850-575-7213Mobile 850-294-7567___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.2.2 RPMS for CentOS 4.x
will they work with CentOS 4.2?On 1/19/06, Andrew McRory <[EMAIL PROTECTED]> wrote: I have compiled a set of RPMS from svn and put them in the regular place.Link:ftp://ftp.linuxsys.com/pub/releases/CentOS-4.0/asterisk-1.2.2/ Best Regards,--Andrew McRory - President/CTOLinux Systems Engineers, Inc. - http://www.linuxsys.comLocated in beautiful Tallahassee, FloridaOffice 850-224-5737 Office 850-575-7213Mobile 850-294-7567___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phones unbeatable echo
Would you mind sharing with the list the tellabs hardware and how you got it up and running (ie pinouts etc)? On 1/15/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: Hello Dan,I was fighting with echo on a number of circumstances, and came to thefollowing conclusions.If you are on a distant loop, or analog lines with issues, those issuesneed to be addressed or you need a workaround. In a few cases, I converted to ISDN-BRI, which has been one of my bestdecisions, because I get excellent quality as well as high-speed callcompletion...In one case, I put in an ADIT 600 channel bank, and still had serious echo problems. I tried and tried, but found no simple solution bymessing with the zapata drivers. Installing a hardware tellabs echocanceller totally solved the echo issue. I have the zapata.conf echocancellation totally off, and the lines sound great. These are also lines that are odd, meaning about 15K feet from the CO, with periodicinstabilities during rain/snow.I went through the various tweaks, milliwatt tests, etc, but only thehardware could solve it (and in minutes after installation as opposed to the hours I spend working with software).Depending on the amount of channels you have, you may consider achannelbank with tellabs, or one of the new digium analog cards with ec,though I have not used the new digiums yet myself. They are expensive solutions, but the best solutions too.I wish there were 4 port card that had great EC, but there isn't. Iwait for the day that we have pci-express voip cards at our disposal,that would be something... Asterisk would take off entirely at that point, since the latencies that cause so many problems would be gone,and the capacities would be so much higher.Just in case I went over your head here, sip>sip should produce no echo.If it does there are other issues. If you are going analog>analog and hear no echo, I would have a look at the network itself.Regards,Greg-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dan ElderSent: Thursday, January 12, 2006 2:53 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP phones unbeatable echoHey all again, I'm wrestling with echo problems on our sip extensions.I've set these items in zapata.conf but tweaking these values doesn't seem to make much differenceechocancel=yesechocancelwhenbridged=yesechotraining=2500rxgain=8.0txgain=1.0are there other settings that can help me tame this beast? Beensearching but not turning up anything that'll work here. Thanks in advance.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No "native bridge" on outbound SIP channels
Yes the 7960 is also set only to use alaw. I was under the impression though that nat=yes did not effect this. And if it does why does it native bridge ok on inbound calls with the same nat=yes On 1/15/06, Jonathan Feally <[EMAIL PROTECTED]> wrote: I'm guessing that you have a similar entry in your sip.conf for the 7960?? The 7960 has a setting for preferred codec. It defaults to g711 U-Law. You might try changing this setting also as the 7960 doesn't know that you only want to speak A-Law. You will also want to make sure that the nat settings are disabled on both devices as they are on the same network. nat=never is a better choice than nat=no. You might also check your extensions.conf to verify that the calling from 1760 to 7960 is the same as from 7960 to 1760. You could also try moving both devices to using U-Law instead. -Jon Eric Bishop wrote: Hi all, I have a Cisco 1760 gateway and and Cisco 7960 VoIP phone running via Asterisk. Both are running g711A codecs and SIP. On inbound calls I get a native bridge, however on outbound calls I never get a native bridge. With other SIP gateways I do get a native bridge on the outbound call. My sip.conf is as follows: [cisco1760] type=friend context=incoming host=192.168.0.55 insecure=yes nat=no canreinvite=no dtmfmode=rfc2833 disallow=all allow=alaw I have also confirmed while on an outbound calls that both are using the exact same codecs. sip show channels shows pbx*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 192.168.0.55 123456789 4ea2e1314cd 00102/0 alaw No Tx: ACK 192.168.0.58 200 0013c427-f4 00101/00102 alaw No Rx: ACK 2 active SIP channels Anyone have an idea what's going on? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No "native bridge" on outbound SIP channels
Hi all, I have a Cisco 1760 gateway and and Cisco 7960 VoIP phone running via Asterisk. Both are running g711A codecs and SIP. On inbound calls I get a native bridge, however on outbound calls I never get a native bridge. With other SIP gateways I do get a native bridge on the outbound call. My sip.conf is as follows: [cisco1760] type=friend context=incoming host=192.168.0.55 insecure=yes nat=no canreinvite=no dtmfmode=rfc2833 disallow=all allow=alaw I have also confirmed while on an outbound calls that both are using the exact same codecs. sip show channels shows pbx*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 192.168.0.55 123456789 4ea2e1314cd 00102/0 alaw No Tx: ACK 192.168.0.58 200 0013c427-f4 00101/00102 alaw No Rx: ACK 2 active SIP channels Anyone have an idea what's going on? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo tail stats
Wouldn't that large spike be the primary sound rather than the echo?On 1/15/06, James Harper <[EMAIL PROTECTED] > wrote: Here's the graph of the echo coefficients I grabbed from a x100p card on my asterisk server. If my interpretation is correct, it shows that most of the echo comes in at about the 28th tap, and assuming a sample rate of 8000hz, that would be about 3.5ms. Will that tell you the sort of things you need to know? Thanks james From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Eric Bishop Sent: Sunday, 15 January 2006 11:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] echo tail stats Yes, plese do post it On 1/15/06, James Harper <[EMAIL PROTECTED]> wrote: I'm working on some code to be able to preload the echo cancellers (and obviously dump the data too, which I've already done). I should have a patch ready tonight or tomorrow. If you are interested I can attach a plot of the coefficients against time which might tell you the sort of thing you need to know if you could do it on your data. The image is only an 11k gif… would it be okay to post it to the list? James From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Eric Bishop Sent: Sunday, 15 January 2006 10:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] echo tail stats Does anyone know how to determine the echo tail size (in ms) of a particular call? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo tail stats
Yes, plese do post itOn 1/15/06, James Harper <[EMAIL PROTECTED]> wrote: I'm working on some code to be able to preload the echo cancellers (and obviously dump the data too, which I've already done). I should have a patch ready tonight or tomorrow. If you are interested I can attach a plot of the coefficients against time which might tell you the sort of thing you need to know if you could do it on your data. The image is only an 11k gif… would it be okay to post it to the list? James From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Eric Bishop Sent: Sunday, 15 January 2006 10:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] echo tail stats Does anyone know how to determine the echo tail size (in ms) of a particular call? ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] echo tail stats
Does anyone know how to determine the echo tail size (in ms) of a particular call? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Turning off 2100 Hz tone detection without editing zconfig.h and recompiling
Anyone know how to ignore the 2100 Hz tone detection without editing zconfig.h and recompiling? I am getting a lot of false "zaptel Disabled echo canceller because of tone (rx) on channel xx" The wiki mentiones that this can be disabled at run time. See http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation the last paragraph. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 small issues with Cisco 1760 gateway and Asterisk
Hi all, We have 1760 working perfectly here with Asterisk for in and outbound calls except for: 1) Outgoing calls sound like they have silence suppression on them (inbound calls are totally fine though). Have tried "no vad" and and different VICs. 2) On outgoing calls on the Cisco console I get %SIP-3-BADPAIR: Unexpected event 16 (SIPSPI_EV_CC_CALL_MEDIA_CHANGED) in state 11 (STATE_RECD_INVITE) substate 0 (SUBSTATE_NONE) . I have googled for this error and checked Cisco site to no avail. I am not sure if these two issues are related.. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Turning off hardware echo can on TE411P
Anyone know if Asterisk 1.2.1 supports turning off the hardware echo canceller WITHOUT recompiling the driver like I had to in 1.0.X? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Attack dialing
Anyone have eny elegant dial plan config for attack dialing? Basically I just want to automatically and continuously dial a busy until it is answered and then hand it over to a SIP hanset. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco FXO hangup detection
I am using a Cisco 1760V with FXO card in Australia to provide ports into Asterisk. I was wondering if anyone out there has a config for the cisco to detect the disconnect or hangup signal for Australian tones. If the calling party hangs up while leaving a voice mail for example, it takes around 15 seconds for the call to time out. I believe the Cisco can be configured to detect the hangup or disconnect tone, but l can't find any details in my searching. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MySQL CDR - MySQL starting too late
Well that didn't work. When I rebooted MySQL didn't start at allOn 11/21/05, JP Carballo <[EMAIL PROTECTED] > wrote:JP Carballo wrote:> Eric Bishop wrote:>>> I have: >>>> [EMAIL PROTECTED] ~]# chkconfig --list | grep mysql>> mysqld 0:off 1:off 2:off 3:on4:off 5:off 6:off>> [EMAIL PROTECTED] ~]# chkconfig --list | grep asterisk>> asterisk0:off 1:off 2:on3:on4:on5:on6:off>>>> What would you suggest I do?>>>>> > > Holy crap, this kind of replying is getting me dizzy! Up, down, what > next? Left and right?> Why can't we just agree to delete all previous text, anyway we all> have threaded readers...don't we?> >> chkconfig --level 3 mysqld off> chkconfig --level 2 mysqld on > chkconfig --level 2 asterisk off>I forgot to add that you should get this:([EMAIL PROTECTED]:asterisk)# chkconfig --list | grep "asterisk\|mysqld"mysqld 0:off1:off2:on3:off4:off5:off6:offasterisk 0:off1:off2:off3:on4:off5:off6:off--JP Carballohttp://www.netfone2x.comBringing the world closer.It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MySQL CDR - MySQL starting too late
I have: [EMAIL PROTECTED] ~]# chkconfig --list | grep mysql mysqld 0:off 1:off 2:off 3:on 4:off 5:off 6:off [EMAIL PROTECTED] ~]# chkconfig --list | grep asterisk asterisk 0:off 1:off 2:on 3:on 4:on 5:on 6:off What would you suggest I do? On 11/21/05, Matt Riddell <[EMAIL PROTECTED]> wrote: Eric Bishop wrote:> Hi All,>> I am running Asterisk (1.0.9.) on CentOS 4 with CDR recording being> output to MySQL. However whenever the system boots up after a reboot I> am needing to manually restart Asterisk because MySQL is after Asterisk > in the service startup sequence and I get>> ERROR[3367]: Failed to connect to mysql database cdr on localhost.>> Anyone know of a simple and elegant way to fix this?>> I'd prefer not to have to hack either MySQL or Asterisk init scripts If it's running using services, you could set MySQL to start on level 2 andAsterisk on level 3.chkconfig --list--Cheers,Matt Riddell___ http://www.sineapps.com/news.php (Daily Asterisk News - html)http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk MySQL CDR - MySQL starting too late
Hi All, I am running Asterisk (1.0.9.) on CentOS 4 with CDR recording being output to MySQL. However whenever the system boots up after a reboot I am needing to manually restart Asterisk because MySQL is after Asterisk in the service startup sequence and I get ERROR[3367]: Failed to connect to mysql database cdr on localhost. Anyone know of a simple and elegant way to fix this? I'd prefer not to have to hack either MySQL or Asterisk init scripts ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Can anyone explain reason for this echo
On 11/18/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: On Friday 18 November 2005 00:30, Eric Bishop wrote:> I purchased the following item:> http://www.oriontelecom.com/echo_canceller/1u_telnet/e1_1u_19inch_ec.html >> As you can see not a very highly spec'd product but does the job well.Perhaps not highly specc'd but with tail lengths of 64ms bidirectional or128ms unidirectional, it's already more capable than the software cancellers (16ms unidirectional with echocancel=128) and I believe that the VPM isn'tall that much better, but I'm not 100% sure now that I can't find the specson it.> I don't accept the fact that mine is a special case. In fact if anything it > should be better than most other scenarios as we are using Tier 1 hardware> (all HP), Digium Rev 2 firmware and our rack is about 10 metres from the> CO.None of that really matters -- it's the overall disance from your RJ48 to the far end's phone that determines the TDM delay, and delays in yourmotherboard's PCI implementation that cause echo. By far mostly the latter. Andrew, I really don't buy that. Everyone seems to blame everything else other Digium for faulty products. Would you be as soft on the vendor if it was Microsoft? The simple fact is they have made a fantastic PBX software product and along with some reasonably priced PC telephony hardware. However echo elimination is clearly lacking in all their products. If you simply take this fact into account when planning installations, you will save yourself a lot time and trouble. -A.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Can anyone explain reason for this echo
I purchased the following item: http://www.oriontelecom.com/echo_canceller/1u_telnet/e1_1u_19inch_ec.html As you can see not a very highly spec'd product but does the job well. I don't accept the fact that mine is a special case. In fact if anything it should be better than most other scenarios as we are using Tier 1 hardware (all HP), Digium Rev 2 firmware and our rack is about 10 metres from the CO. On 11/18/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: On Thursday 17 November 2005 21:01, Eric Bishop wrote:> I got sick of tweaking and playing with Digium's ridiuculous voodoo so I> just bought a dedicated E1 PRI echo canceller and bingo, problem solved.> Digium make some good IP PBX software and hardware but all their echo > cancellers, hardware and software are complete rubbish.There are many, many of us who disagree. The echo was not solveable on yourparticular installation. You could have a longer tail than the software echo can Asterisk has can handle, and longer than the Digium hardware echo can canmanage. I am interested in the echo can you settled upon, and what its specsare.Would you mind sharing this information? -A.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Can anyone explain reason for this echo
I got sick of tweaking and playing with Digium's ridiuculous voodoo so I just bought a dedicated E1 PRI echo canceller and bingo, problem solved. Digium make some good IP PBX software and hardware but all their echo cancellers, hardware and software are complete rubbish. On 11/18/05, Doug Meredith <[EMAIL PROTECTED]> wrote: Eric Bishop <[EMAIL PROTECTED]> wrote:>If I call our Asterisk box via Disa and then place a call to one of the>problem analogue numbers (native Zap bridge) I don't get any echo. So the >echo seems to occur only when using a SIP handset and making a call to an>analogue number.The echo is probably always there. You only notice it with the SIPphone because of the additional latency that this introduces. Doug--Doug Meredith ([EMAIL PROTECTED])SystemGuard - Oracle remote support877-974-8273 (87-SYSGUARD)506-854-7997 www.systemguard.com___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dedicated echo canceller hardware
Yes I am referring to TE411. I have not used TE406 which is the same product, just different slot type. On 11/15/05, George Pajari <[EMAIL PROTECTED]> wrote: Eric Bishop wrote:> I have recently seen the light and started using dedicated echo> cancellation hardware. It works great with our E1 PRI's, much better> than either of Digium's hardware or software echo cancellation products. What make/model are you using? And when you say "Digium's hardware ...echo can" are you referring to the 406/411 products?g.--George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dedicated echo canceller hardware
Hi All, I have recently seen the light and started using dedicated echo cancellation hardware. It works great with our E1 PRI's, much better than either of Digium's hardware or software echo cancellation products. I have had trouble however finding a simlar device for use with analogue lines and the Digium TDM400P. Does anyone have any recomendations? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS Server Failure wreaks havoc
I agree about Asterisk being terrible with DNS failure, but how can you avoid using DNS on *nix system?On 11/7/05, Eric ManxPower Wieling < [EMAIL PROTECTED]> wrote:Brian Capouch wrote:> I don't think this is a new issue--I've seen it talked about on the list > before. I don't know if I've ever seen anyone post a fix.>> My DNS server went out last night in a horrendous storm when an upstream> link went down. The madness is that the behavior of the whole server, > including the part that's handling my POTS lines, gets wigged out on a> DNS failure, making the whole system unusable. I have two questions;> being able to solve either would be wonderful:Asterisk is horrible at handleing DNS failures. Don't use DNS with Asterisk.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ABE - Are you happy with it?
Can any one who has gone from the open source version of Asterisk to ABE comment on their experiences? Specifically: - How does the quality compare to the open source stable versions? - How often do updates come out? - How far is it behind CVS HEAD in terms of features? - How good had Digium support been? - Overall was the switch worth the money? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk and reverse DNS
Nope, never really found a satisfactory solution to this.. On 11/3/05, Tom Rymes <[EMAIL PROTECTED]> wrote: Hi there. I noticed a post you made to asterisk-users backin June regarding problems you were having with Asteriskif your internet connection went down. I am having thesame problem here, and I was wondering if you found any good solution to this issue.Sorry to bother you,Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any experiences with Orion hardware echo cancellers?
I am looking to buy wither the 1U or desktop E1 echo canceller from Orion. Has anyone had any experiences either good or bad with these units? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can anyone explain reason for this echo
Our configuration is as follows: SIP phones -> TE410P -> PSTN When a SIP handset makes a call to other ISDN numbers - no problem. When a SIP handset make a call to analogue numbers - echo. I know for certain that the problem is at our end. Why? If I call our Asterisk box via Disa and then place a call to one of the problem analogue numbers (native Zap bridge) I don't get any echo. So the echo seems to occur only when using a SIP handset and making a call to an analogue number. Can anyone provide a logical explanation for this? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Re] Re: [Asterisk-Users] Echo canceller on TE406 & Asterisk
I would also be very interested in what hardware you used. On 10/29/05, Robert Augustyn <[EMAIL PROTECTED]> wrote: Darren,Can you elaborate on what echocan did you use and how?Thanks.robert> -Original Message-> From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED]] On Behalf Of> Darren Wright> Sent: Friday, October 28, 2005 7:35 AM> To: [EMAIL PROTECTED]; Asterisk Users Mailing List -> Non-Commercial Discussion> Subject: RE: [Re] Re: [Asterisk-Users] Echo canceller on> TE406 & Asterisk>> I have given up totally on Digium based echo cancel, hardware > or software. The KB1 is the best so far, but still> unacceptable. I installed a hardware echocan FACING the T1> card in the asterisk box, and> all is perfect. No complaints from any of my clients since > taking that> leap.>> -Darren>>> ___> --Bandwidth and Colocation sponsored by Easynews.com -->> Asterisk-Users mailing list> Asterisk-Users@lists.digium.com> http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users>___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo Canceller question- is there a viable solution?
I am running CVS HEAD. How can I tell which software echo canceller I am using?On 10/28/05, Matthew Fredrickson < [EMAIL PROTECTED]> wrote:On Oct 27, 2005, at 12:38 AM, [EMAIL PROTECTED] wrote:> My question is, what is the direction in relation to analog boards and> such?Right now, it looks like the current fad of the asterisk group ishardware echocancelation. However, there is work that is occurring on the software echocans to improve them. In fact, I just committed basically an update toKB1(which was until now the latest and greatest version of MEC2) that issupposedto provide somewhat significant improvements. >> Quite a few people tend to have difficulties with echo, and although> the> WIKI has some very helpful advice, from a business standpoint I would> think that it would be an important step to come up with a final > solution to the problem.>> Many companies who make the higher end equipment seem to have tackled> the issue on their hardware.>> Do we know if digium is spending time on solving the issue? For > example, having a tool to run on a digium analog or t1 board to analyze> the line statistics and come up with the proper gain settings could be> extremely helpful.>> Such a tool would require a firm knowledge of the causes and solutions > to echo however, but I would assume that digium should have a grasp on> this.>> It just seems difficult to suggest to companies to use an asterisk> based> solution (if they do not use pri) when there is the possibility that an > installation will have issues with echo.>> At this point, it feels more like a trial experience to eliminate echo> in various environments.Unfortunately, that's the way it is right now. Getting to the point where you haveenough knowledge to be able to work on these things is not aninsignificant task.It seems like we're slowly getting there, and now that we have somemore intereston improving the software echo cans we might be a little be closer to getting to thepoint where it "just works".>> I have used local tone from the CO to help narrow things down, but a> tool that would loop dial a line and do an analysis could reduce the > implementation time from days to hours.Well, there isn't anything that does the "whole job" right now.There's a bunchof pieces that go together, and if you have the necessary knowledge of how toput the pieces together, you can get pretty close to it "just working". It's not thatbad though, one can also see it as job security as well :-)Matthew Fredrickson___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where does Asterisk put it's files
Does anyone have a full list of places Asterisk puts all config files and binaries. I need this to be able to fully rollback if I have a failed upgrade of Asterisk/Zaptel/LibPRI. So far I have: /etc/zaptel.conf /etc/asterisk/ /usr/sbin/safe_asterisk /usr/sbin/asterisk /usr/lib/asterisk/modules/ /usr/include/asterisk/ /lib/modules/`uname -r`/misc /usr/lib/ /usr/include/ Anything I have missed? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo canceller on TE406 & Asterisk
I replaced a TE410P (1st Gen) with a TE411P (2nd gen with hardware echo canceller) and the echo actually got much worse! Very disappointing! On 10/28/05, Boris Bakchiev <[EMAIL PROTECTED]> wrote: Hi,I have TE406P (2nd gen card with echo cancellation on-board).We still notice quite often echo on our PBX that is connected to one ofthe spans on TE406P (with calls routers to PRI provider on another span).I've tried to experiment with the echo cancellation on asterisk.I enabled echo cancellation in Zapata.conf to see if I can improve thesituation and users started reporting "warping bubble" (description I got from one of the users) sound on calls from PABX->Asterisk->PRI (andother way).I was expecting that asterisk would disable its echo cancellation onceit find on-board module.The strange thing I noticed that after system reboot things are now better.Although I cannot say for sure because the system was ever rebooted 2times.Can anyone shed some light on this? Has anyone had similar problems?Or point me into right direction for troubleshooting? RegardsBoris___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Taking the plung to CVS HEAD
We are running 1.0.9 STABLE on all of our machines. I am about try and upgrade one machine to CVS HEAD as all this echo cancellation improvements sound enticing. Can anyone recommend a) A procedure to cleanly upgrade from STABLE to HEAD b) A procedure to ensure I can back out and go back to 1.0.9 easily I have looked on the wiki but couldn't find much about this. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium G.729 codec modules updated
On a dual processor Xeon (EM64T) would you reccomend turning hypertreading on or off? I tend go for it off dual processor machines just in case 2 processes end up on the one physical processor rather than 2 processes on 2 different physical processors. What do you think?On 10/9/05, Kevin P. Fleming <[EMAIL PROTECTED] > wrote:Eric Bishop wrote:> Have you founy any real life performance benefit of x86_64 (particularly > EM64T on Xeon) as apposed to plan old x86?Yes. On my Athlon-64 2200+ system on my desk, the 'opteron' versionencodes a 6722 block sample file in 478ms; the 'i686' version does it in514ms. The 'i386' version is somewhere near 600ms, since it has no fancy instruction scheduling.Interestingly, _all_ of the 32-bit x86 optimized versions run just fineon that machine, meaning that GCC did not opt to use any instructionsthat are specific to a processor model/family... the performance improvements come only from scheduling the instruction flow.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium G.729 codec modules updated
Have you founy any real life performance benefit of x86_64 (particularly EM64T on Xeon) as apposed to plan old x86? On 10/9/05, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: Dinesh Nair wrote:> and they still go for US$10 a pop ?Patent indemnification licenses are completely separate from the codecbinary you choose to use. There is no price difference for CPU type, OS platform or anything else.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7960 show queue status
Hi all, We have a small call centre here running with Asterisk 1.0.9. All the agents use Cisco 7960's with SIP 7.5 firmware. Is there any way we can show queue status on the those nice big LCD's. Especially we would like to display whether the agent is currently logged in or not. Is this possible? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Performance tuning on dual Xeon EM64T and x86_64 Linux
Hi all, Just a couple of quick questions. I have a HP DL360 G4 (dual Xeon EM64T 3.0Ghz processors). I am using a TE411P in the system. 1. Should I run the a x86_64 Linux (CentOS) or just go with the plain old x86 version? Is there any benefit (or things to be aware of) on x86_64 vs x86? 2. This being a dual processor system, should I turn on or off hyper thrreading? Thanks in advance! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HP DL360 G4 EM64T and hyperthreading options
Hi all, Just a couple of quick questions. I have a HP DL360 G4 (dual 3.0Ghz processors). The processors are EM64T. I am using a TE411P in the system. 1. Should I run the a x86_64 Linux (CentOS) or just go with the plain old 32 bit version? 2. This being a dual processor system, should I turn on or off hyper thrreading? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P zapata.conf, monitoring echo cancellation and echo tail size
Can someone from Digium comment on this? On 9/9/05, Cory Andrews <[EMAIL PROTECTED]> wrote: I don't think you can switch the echo tail size, I could be mistaken, but I think the more channels you are utilizing the smaller amount of MS you have allocated to each channel. Cory AndrewsPartner / PurchasingVOIPSupply.com++454 Sonwil DriveBuffalo, NY 14225++v - 800.398.VOIP Ext 22f - 716.630.1548e - [EMAIL PROTECTED] -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Eric BishopSent: Thursday, September 08, 2005 5:47 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] TE411P zapata.conf,monitoring echo cancellation and echo tail sizeHi all,1. Just bought a new TE411P and about to install it replacing the existing TE410P. I am assuming I need to set echocancel=no and echocancelwhenbridged=no now that it will be done in hardware, correct?2. Is there any way to monitor hardware echo cancellation to ensure it is working (apart from being on a call)?3. The Digium webite says "By supporting 16ms with 128 channels or 64ms on 32 channels this card will perform in the most difficult of environments.". How do I switch the echo tail size between 16ms and 64ms?Thanks. ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE411P zapata.conf, monitoring echo cancellation and echo tail size
Hi all, 1. Just bought a new TE411P and about to install it replacing the existing TE410P. I am assuming I need to set echocancel=no and echocancelwhenbridged=no now that it will be done in hardware, correct? 2. Is there any way to monitor hardware echo cancellation to ensure it is working (apart from being on a call)? 3. The Digium webite says "By supporting 16ms with 128 channels or 64ms on 32 channels this card will perform in the most difficult of environments.". How do I switch the echo tail size between 16ms and 64ms? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial command nor progressing on Zap channels
Already have that.. On 8/27/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote: > Eric Bishop wrote: > > Hi all, > > > > Our Asterisk box sends calls outbound via either SIP (through our VoIP > > provider) or an E1 PRI (directly connected via a TE410P). When we dial > > a number that is engaged via our VoIP provider we get the following on > > the Asterisk console (numbers and IP addresses changed to protect the > > innocent): > > > > -- Called [EMAIL PROTECTED] > > -- Got SIP response 486 "Busy here" back from 123.123.123.123 > > -- SIP/sip-outbound-af71 is busy > > == Everyone is busy/congested at this time > > > > This is what we want as it then send the call to priority n+101 and we > > can handle it any way we want from there. However if the outbound call > > is made via the PRI (Zap channel) to an enaged number it simply plays an > > enaged > > signal to the caller and never progresses past the Dial priority. I > > know for a fact the call is not actually being answered, because I get > > the following onthe console. > > ; PRI Out of band indications. > ; Enable this to report Busy and Congestion on a PRI using out-of-band > ; notification. Inband indication, as used by Asterisk doesn't seem to work > ; with all telcos. > ; > ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT > ; inband: Signal Busy/Congestion using in-band tones > > priindication = outofband > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial command nor progressing on Zap channels
Hi all,
Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via our VoIP provider we get the following on
the Asterisk console (numbers and IP addresses changed to protect the
innocent):
-- Called [EMAIL PROTECTED]
-- Got SIP response 486 "Busy here" back from 123.123.123.123
-- SIP/sip-outbound-af71 is busy
== Everyone is busy/congested at this time
This is what we want as it then send the call to priority n+101 and we
can handle it any way we want from there. However if the outbound call
is made via the PRI (Zap channel) to an enaged number it simply plays an enaged
signal to the caller and never progresses past the Dial priority. I
know for a fact the call is not actually being answered, because I get
the following onthe console.
Executing Dial("SIP/1001-270b", "Zap/g1/123456789") in new stack
-- Called g1/123456789
pri debug span 1 gives me:
< Protocol Discriminator: Q.931 (8) len=5
< Call Ref: len= 2 (reference 32809/0x8029) (Terminator)
< Message type: RELEASE (77)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request
> Protocol Discriminator: Q.931 (8) len=9
> Call Ref: len= 2 (reference 41/0x29) (Originator)
> Message type: RELEASE COMPLETE (90)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
> Private network serving the local user (1)
> Ext: 1 Cause: Normal Clearing (16), class = Normal Event
> (1) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Executing Macro("SIP/1001-8cdc", "Dial_Telco_ISDN|123456789")
in new stack
-- Executing SetAccount("SIP/1001-8cdc", "TELCO-ISDN") in new stack
-- Executing NoOp("SIP/1001-8cdc", "") in new stack
-- Executing Dial("SIP/1001-8cdc", "Zap/g1/123456789") in new stack
-- Making new call for cr 32810
> Protocol Discriminator: Q.931 (8) len=51
> Call Ref: len= 2 (reference 42/0x2A) (Originator)
> Message type: SETUP (5)
> [04 03 80 90 a3]
> Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability:
> Speech (0)
> Ext: 1 Trans mode/rate: 64kbps, circuit-mode
> (16)
> Ext: 1 User information layer 1: A-Law (35)
> [18 03 a9 83 81]
> Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: > 0
>ChanSel: Reserved
> Ext: 1 Coding: 0 Number Specified Channel Type: 3
> Ext: 1 Channel: 1 ]
> [28 09 41 6c 6c 61 6e 20 44 69 62]
> Display (len= 9) [ Eric Bishop ]
> [6c 09 21 81 33 30 30 31 30 30 31]
> Calling Number (len=11) [ Ext: 0 TON: National Number (2) NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> Presentation: Presentation permitted, user number
> passed network screening (1) '3001001' ]
> [70 0b 80 30 33 39 35 37 30 32 37 30 38]
> Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown
> Number Plan (0) '123456789' ]
> [a1]
> Sending Complete (len= 1)
-- Called g1/123456789
Why is the Dial command not progeessing?
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Re: [Asterisk-Users] PRI signaling experts please help
Already am using this option. On 8/25/05, Jens von Bülow <[EMAIL PROTECTED]> wrote: > Hi Eric, > > Don't you need to use out-of-band PRI signaling... > > From /etc/asterisk/zapata.conf > > ; PRI Out of band indications. > ; Enable this to report Busy and Congestion on a PRI using out-of-band > ; notification. Inband indication, as used by Asterisk doesn't seem to work > ; with all telcos. > ; > ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT > ; inband: Signal Busy/Congestion using in-band tones > priindication = outofband > > > Hope that Helps > Jens > > > -Original Message----- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop > Sent: 25 August 2005 09:32 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] PRI signaling experts please help > > Hi all, > > Our Asterisk box sends calls outbound via either SIP (through our VoIP > provider) or an E1 PRI (directly connected via a TE410P). When we dial > a number that is engaged via our VoIP provider we get the following on > the Asterisk console (numbers and IP addresses changed to protect the > innocent): > > -- Called [EMAIL PROTECTED] >-- Got SIP response 486 "Busy here" back from 123.123.123.123 >-- SIP/sip-outbound-af71 is busy > == Everyone is busy/congested at this time > > This is what we want as it then send the call to priority n+101 and we > can handle it any way we want from there. However if the outbound call > is made via the PRI (Zap channel) to an enaged number it simply plays an > enaged > signal to the caller and never progresses to priority n+101. > > Anyone have any suggestions? > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy number signalling
Andres,
Thanks for the suggestion. I did try it but it is not moving to the
next priority after the Dial command. I also do know for a fact that
it is not actually being answered. On the console I just get:
-- Called g1/123456789
On 8/26/05, Andres <[EMAIL PROTECTED]> wrote:
> Eric Bishop wrote:
>
> >Hi all,
> >
> >Our Asterisk box sends calls outbound via either SIP (through our VoIP
> >provider) or an E1 PRI (directly connected via a TE410P). When we dial
> >a number that is engaged via our VoIP provider we get the following on
> >the Asterisk console (numbers and IP addresses changed to protect the
> >innocent):
> >
> > -- Called [EMAIL PROTECTED]
> >-- Got SIP response 486 "Busy here" back from 123.123.123.123
> >-- SIP/sip-outbound-af71 is busy
> > == Everyone is busy/congested at this time
> >
> >This is what we want as it then send the call to priority n+101 and we
> >can handle it any way we want from there. However if the outbound call
> >is made via the PRI to an enaged number it simply plays an enaged
> >signal to the caller and never progresses to priority n+101.
> >
> >Anyone have any suggestions?
> >
> >
> You can try checking for the DIALSTATUS variable. ON our PRIs we do
> something like:
> exten => _1XX,1,Dial(Zap/r1/${EXTEN:1})
> exten => _1XX,2,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?8)
> exten => _1XX,3,GotoIf($["${DIALSTATUS}" = "BUSY"]?8:4)
> exten => _1XX,4,Congestion
> exten => _1XX,8,Busy(1)
>
> But if your call is really getting answered then this won't work.
>
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> >
> >
> >
> >
>
>
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[Asterisk-Users] PRI signaling experts please help
Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called [EMAIL PROTECTED] -- Got SIP response 486 "Busy here" back from 123.123.123.123 -- SIP/sip-outbound-af71 is busy == Everyone is busy/congested at this time This is what we want as it then send the call to priority n+101 and we can handle it any way we want from there. However if the outbound call is made via the PRI (Zap channel) to an enaged number it simply plays an enaged signal to the caller and never progresses to priority n+101. Anyone have any suggestions? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy number signalling
Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called [EMAIL PROTECTED] -- Got SIP response 486 "Busy here" back from 123.123.123.123 -- SIP/sip-outbound-af71 is busy == Everyone is busy/congested at this time This is what we want as it then send the call to priority n+101 and we can handle it any way we want from there. However if the outbound call is made via the PRI to an enaged number it simply plays an enaged signal to the caller and never progresses to priority n+101. Anyone have any suggestions? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
