RE: [Asterisk-Users] Polycom IP501 Buddy List
Hi Michael -- There is a hardcoded limit of 7 buddies that the Polycom IP phones support with the current firmware. Polycom is rumoured to be increasing this limit to 42 in a new version of the firmware due for release next quarter. Mike. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Welter Sent: Thursday, 23 March 2006 1:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom IP501 Buddy List I have a problem with my Polycom phones. In the buddy list, the phone displays all but three employees. For those three employees, there is no difference in any of the configurations. Is there a secret to getting all employees into the buddy list? Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --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 hints
Hi Garth -- Other users have also reported problems with the status being set by the SwissVoice phones - oh wait a minute... that was you! Have you tried setting call-limit=1 in sip.conf? Also check that you're running the latest firmware on the Swissvoice (I think it's build 18), since I know they've been tinkering with the presence features recently. Cheers, Mike. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth van Sittert Sent: Thursday, 23 February 2006 6:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk hints I am using Swissvoice IP10S phones. Garth Mike Pollitt wrote: Garth -- What kind of phones are you using? Mike. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth van Sittert Sent: Wednesday, 22 February 2006 7:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk hints Hi All Does anyone know how the hints in asterisk works? How does a SIP phone interact with the hints? I am having a problem with certain phone models that do not set the hints correctly when I list the hints with a 'show hints'. Thanks Garth ___ --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 ___ --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] username as extension
You want regexten/regcontext in sip.conf under each peer. http://www.voip-info.org/wiki-Asterisk+sip+regcontext -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan Alberti Sent: Thursday, 23 February 2006 7:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] username as extension Is there a way to have extensions automatically created for registered sip users ? I did some investigation and found some hope in chan_sip with relation to the somewhat undocumented usereqphone option but i may be totally off track. All i want to be able to do is send a call to [EMAIL PROTECTED] where the number is the username configured on the phone that has registered with asterisk on ip_address. From what I understand this should be pretty standard sip functionality no ? Regards, Nathan. ___ --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] auto provision of IP501 polycom
In dhcpd.conf: option tftp-server-name x.x.x.x Yes, I know it says tftp but actually this is the entry used for ftp as well. Also, for reasons known only to your chosen deity, Polycom have chosen to use a mixed-case username for the default ftp user. Not all FTP servers will accommodate this. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Friday, 24 February 2006 12:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] auto provision of IP501 polycom Has anyone been able to get the IP501 to discover the FTP server IP address (via dhcp or dns) and download 100% of the config from a provisioning server? We are still having to touch each unit to enter the ftp server address and password, as well as set many of the options that will not take from the config file. Have a sample config file you are willing to share? What is required in the way of dhcp options or dns entries to get the polycom to discover the ftp boot server? What about changing default passwords via ftp? ___ --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 query a table from the keypad?
Hi Richard What you want is AGI: http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI You could write a perl script to read the PO number from stdin and spit back the balance (or whatever). To read the PO number from the user, use the Read() dialplan application. To play back the balance, you could use SayDigits() (but theres probably a more elegant solution specifically for speaking amounts of money). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Reina Sent: Friday, 24 February 2006 9:34 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to query a table from the keypad? I am trying to give users the option to query our accts. payable database by supplying their PO number. I able to write queries via perl-DBI-mysql but have no idea how to get * to do it from the IVR. Is this possible? Can anyone point me in the right direction for help or examples? Thanks, Richard What are the most popular cars? Find out at Yahoo! Autos ___ --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 hints
Garth -- What kind of phones are you using? Mike. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth van Sittert Sent: Wednesday, 22 February 2006 7:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk hints Hi All Does anyone know how the hints in asterisk works? How does a SIP phone interact with the hints? I am having a problem with certain phone models that do not set the hints correctly when I list the hints with a 'show hints'. Thanks Garth ___ --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] Dial timeouts and SIP 302 redirects
Hi List Well, not getting anywhere, I stumped up for Digium support, and the answer is, unfortunately, that there is currently no way of resetting the timer when the Dial application gets a 302 message back from the SIP handset. In other words, the behaviour exhibited below is standard (even though in my opinion it is undesirable). Ive decided to have a crack at a patch for this myself. Will keep you posted, since I know there are at least a couple of other people out there who have been having this problem. Mike. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Pollitt Sent: Tuesday, 21 February 2006 4:03 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Dial timeouts and SIP 302 redirects I have SIP handsets which allow the user to forward a call to another number after a specified interval of ringing time. On the SwissVoice this is refered to as CFNR (Call Forward on No Response). What actually happens is that after a specified period of time (default 15 seconds), the handset sends back a 302 Moved Temporarily response to Asterisk. The problem is that when Asterisk receives the 302 message, it doesnt reset the ringing timer in the Dial command. Lets say Ive issued a Dial command such as: exten = _34XX,1,Dial(SIP/fred|20) exten = _34XX,n,Voicemail(fred) What happens is that the SIP handset rings for the default time of 15 seconds, then sends back the 302 message with the new number to forward to. Asterisk faithfully drops into the Local context with this number, but after a further 5 seconds of ringing the new number, the original Dial command exits and proceeds to the next priority, namely the Voicemail command. The problem with this is that the forwarded number only rings for 5 seconds (or not at all if it takes a few seconds to actually make the new outgoing call, as can happen often with cellphones), which is not enough time for them to answer it. Has anyone else had this problem, and is there a solution? ___ --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] Dial timeouts and SIP 302 redirects
I have SIP handsets which allow the user to forward a call to another number after a specified interval of ringing time. On the SwissVoice this is refered to as CFNR (Call Forward on No Response). What actually happens is that after a specified period of time (default 15 seconds), the handset sends back a 302 Moved Temporarily response to Asterisk. The problem is that when Asterisk receives the 302 message, it doesnt reset the ringing timer in the Dial command. Lets say Ive issued a Dial command such as: exten = _34XX,1,Dial(SIP/fred|20) exten = _34XX,n,Voicemail(fred) What happens is that the SIP handset rings for the default time of 15 seconds, then sends back the 302 message with the new number to forward to. Asterisk faithfully drops into the Local context with this number, but after a further 5 seconds of ringing the new number, the original Dial command exits and proceeds to the next priority, namely the Voicemail command. The problem with this is that the forwarded number only rings for 5 seconds (or not at all if it takes a few seconds to actually make the new outgoing call, as can happen often with cellphones), which is not enough time for them to answer it. Has anyone else had this problem, and is there a solution? ___ --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 do I install speex for asterisk?
I used yum to install speex (although you could quite easily build your own from source). # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start This sequence of commands may require variation depending on your flavour of Linux and how your asterisk is installed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesus E Zepeda Sent: Friday, 17 February 2006 10:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How do I install speex for asterisk? Huuu! I never expected you had to recompile asterisk to add a codec. But if that is what it takes, we'll do it. I noticed that asterisk makes reference to some speex.c in the makefile file. In some of those references I saw the actual speex.c file in the paths specified. A couple of them missing by the way. That could be why speex was never taken by asterisk. Mike, does speex have to be copied to a specific directory, then compiled and installed before re-compiling and re-installing asterisk? I appreciate you took your time to reply. Regards, Jesus -Original Message- From: Mike Pollitt [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 15:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How do I install speex for asterisk? You need to recompile Asterisk itself after installing Speex. Do a make clean, make, make install. I usually stop asterisk before that last step, by the way! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesus E Zepeda Sent: Friday, 17 February 2006 5:58 AM To: Asterisk User List Subject: [Asterisk-Users] How do I install speex for asterisk? Hi, everybody: I enabled speex in my asterisk box (iax.conf), but no phone call went throug. At the asterisk console, I used the show modules command and it did not show the speex codec in the list. So, I downloaded the speex codec from speex.org, v1.0.5, compiled and installed in my asterisk machine. What I still don't know is: what do I need to do from the asterisk side to make it available? I just downloaded it to a directory, compiled and installed thinking that by doing a restart to asterisk it would some how know where to load it from. Any hints are appreciated Regards, Jesus E. Zepeda ___ --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 ___ --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] Call forward on unavailable timer issues
I have a pretty standard
setup with Asterisk acting as a PABX for a bunch of SIP handsets (in this case,
SwissVoice IP10S).
My users are complaining
that when they forward their phones to their cellphones on unavailable (i.e.
forward when no-answer), their cellphone only rings once or twice, and then
Asterisk sends the call through to Voicemail.
Im using the
standard extension Macro thus:
[macro-stdexten]
; ${ARG1} - Extension
(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to
ring
; ${ARG3} - Voicemail
context
exten = s,1,Dial(${ARG2},20)
; Ring the interface, 20 seconds maximum
exten =
s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten =
s-NOANSWER,1,Voicemail([EMAIL PROTECTED]) ; If unavailable, send to
voicemail w/ unavail announce
exten =
s-BUSY,1,Voicemail([EMAIL PROTECTED]) ; If busy, send to voicemail w/
busy announce
exten =
_s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten = a,1,VoicemailMain([EMAIL PROTECTED])
; If they press *, send the user into VoicemailMain
Now clearly my problem is
that when the Dial application gets back a Temporarily Moved response from the
SIP phone (after the users preset period to wait before no-answer
forwarding), and drops back into the dialplan as Local/forwarded
number, the 20 second timer on the Dial command is still active.
I think what I need is a
way to reset or cancel this timer when a Temporarily Moved response comes back
in.
Surely this must be a
fairly common problem does anyone have a solution?
Thanks!
Mike.
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Looping through variables, or sort of but not really arrays [was, bizarrely, RE: [Asterisk-Users] is there a web interface to this mailing list?]
Actually, you can do this:
exten = s,1,Set(TRUNK1=foo)
exten = s,n,Set(TRUNK2=bar)
exten = s,n,Set(TRUNK3=gak)
exten = s,n,Set(INDEX=1)
exten = s,n,Set(CURRTRUNK=${TRUNK${INDEX}})
exten = s,n,Dial(${CURRTRUNK}/555|60)
and you could increment INDEX (although these are local, (are you local?) so
you'd have to do it locally).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, 16 February 2006 5:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users
Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] is there a web interface to this mailing list?
Wow. I'm thinking I've got some agreeance with this. The similarity to
assembler actually hit me today like a freight train, after I once again,
'hit the wall' trying to implement something in extensions.conf.
In a shell script, you can do something like this:
$var$num
and if var=NUM and num=1, you'd get NUM1, I was trying to loop through some
variables called NUM1, NUM2, NUM3 etc. G
-Original Message-
From: Johnathan Corgan [mailto:[EMAIL PROTECTED]
Sent: Wed 2/15/2006 10:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] is there a web interface to this
mailing list?
Douglas Garstang wrote:
Yes, programming the dialplan is akin to programming assembler.
Too funny. But true.
The first time I did a 'show dialplan' after trying out AEL, I felt
like
I was seeing an assembler dump of C++ :-)
-Johnathan
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RE: [Asterisk-Users] show calls
CLI show channels -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jonny hashem Sent: Friday, 17 February 2006 12:46 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] show calls HI: what is command on console to know how many calls are sending in the same time? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 do I install speex for asterisk?
You need to recompile Asterisk itself after installing Speex. Do a make clean, make, make install. I usually stop asterisk before that last step, by the way! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesus E Zepeda Sent: Friday, 17 February 2006 5:58 AM To: Asterisk User List Subject: [Asterisk-Users] How do I install speex for asterisk? Hi, everybody: I enabled speex in my asterisk box (iax.conf), but no phone call went throug. At the asterisk console, I used the show modules command and it did not show the speex codec in the list. So, I downloaded the speex codec from speex.org, v1.0.5, compiled and installed in my asterisk machine. What I still don't know is: what do I need to do from the asterisk side to make it available? I just downloaded it to a directory, compiled and installed thinking that by doing a restart to asterisk it would some how know where to load it from. Any hints are appreciated Regards, Jesus E. Zepeda ___ --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] DID's
Wrong list. You want asterisk-biz. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURA Sent: Friday, 17 February 2006 9:06 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DID's I need 10 DID's for it those country's Nicaragua El salvador Costa Rica Panama Honduras Thank's João Carlos Moura NiNeTel Telecommunications 7382 N.W. 35 Terrace Miami, FL 33122 USA ___ --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] Speex echo cancellation
Can the new echo cancellation features in Speex 1.1.9 and higher be activated when using the codec within Asterisk? ___ --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] Polycom buddy watch limit of 7
Hi All -- I've got a Polycom 601 with the sidecar unit all working with extension hints and what Polycom calls the Buddy Watch feature. I can see the state of extensions, but there seems to be a limit of 7 that I can monitor at any one time. I've put in a call to my distributor (this is how Polycom provides support). So far no response. I've seen other people have had this issue (http://voxilla.com/PNphpBB2-viewtopic-t-6350.html) but not whether anyone has successfully resolved it. Cheers, Mike. ___ --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 expire 60
Hi Jerry -- Have you tried adjusting the settings in the SIP device itself? That's where you can adjust how frequently the device will try to register. Regards, Mike. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Tuesday, 14 February 2006 9:50 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] sip expire 60 I am getting messages on the console about Registered SIP ... expires 60 How do I increase that 60 to 3 minutes??? I have tried in [general] of sip.conf to set expirey=300 defaultexpirey=300 nothing seems to affect it. Thanks, Jerry ___ --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] Voicemail Problem
Case sensitivity? The CLI references Goodbye but your filename is goodbye.gsm. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sam Lee Sent: Friday, 10 February 2006 1:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Voicemail Problem Strange thing that , its there ! [EMAIL PROTECTED]:/home/sam# ls /var/lib/asterisk/sounds/goodbye.gsm /var/lib/asterisk/sounds/goodbye.gsm [EMAIL PROTECTED]:/home/sam# That's why i found it very strange. Thanks for replying. Are there any other ideas ? Regards, Sam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wojciech Tryc Sent: Friday, February 10, 2006 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail Problem You don't have 'vm-goodbye' voice file. Check under /var/lib/asterisk/sounds Wojtek - Original Message - From: Sam Lee To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, February 09, 2006 8:38 PM Subject: RE: [Asterisk-Users] Voicemail Problem Hey guys, Any hint at all ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sam Lee Sent: Thursday, February 09, 2006 3:30 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail Problem I have just setup my OPENSER to work with the asterisk 1.2.2. I've set extension 400 in extension.conf to point to the VoicemailMain() application The entire program works fine, but there seems to be some problem whenever the call is hangup, either by pushing # to exit the VoicemailMain() apps or by hanging the phone. If the # button is push, should Asterisk send something back to tell OPENSER to hang up the party ? Here's the log of verbose level 3 Asterisk*CLI -- Playing 'vm-youhave' (language 'en') -- Playing 'vm-no' (language 'en') -- Playing 'vm-messages' (language 'en') -- Playing 'vm-opts' (language 'en') -- Playing 'vm-goodbye' (language 'en') -- Executing Playback(SIP/210.23.1.139-081ee3d8, Goodbye) in new stack Feb 9 15:05:06 WARNING[23242]: file.c:509 ast_openstream_full: File Goodbye does not exist in any format Feb 9 15:05:06 WARNING[23242]: file.c:821 ast_streamfile: Unable to open Goodbye (format alaw): No such file or dire ctory Feb 9 15:05:06 WARNING[23242]: app_playback.c:132 playback_exec: ast_streamfile failed on SIP/203.125.68.66-081ee3d8 for Goodbye -- Executing Hangup(SIP/203.125.68.66-081ee3d8, ) in new stack == Spawn extension (default, 400, 3) exited non-zero on 'SIP/203.125.68.66-081ee3d8' Asterisk*CLI Any idea what is this all about ? Regards, Sam ___ --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] odd 'digital' sound artifacts
Hi Gerard -- I found that I get the really loud buzzing sound in the handset earpiece when I set echocancel=256 instead of echocancel=yes (the default = 128 taps). It seemed to occur irrespective of the actual echo canceller chosen. Mike. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gerard Saraber Sent: Saturday, 11 February 2006 2:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] odd 'digital' sound artifacts So nobody heard these before? or did I do something stupid that anyone should know and nobody wanted to yell at me for it ;) On Wed, 2006-02-08 at 12:54 -0600, Gerard Saraber wrote: Hi, I've got some weird sound artifacts happening during calls, they're very hard to describe, so I have a 122kb recording: http://openprojects.rarcoa.com/~miztic/artifact.wav normally the artifacts are just short blips, not quite as long as the one above, but they sound the same. When using the aggressive echo suppressor, it seems like those artifacts cause a really loud buzzing sound to come out of the cisco phone, pretty much made using the aggressive canceler impossible to use, it's too bad because it worked the best out of all of them, mark3 works ok but still gives echos on at least 20% of the calls. I thought they might be caused by IRQ sharing, so I pulled one of the TDM400P cards out and made sure the remaining two were on their own IRQ, the artifacts were still there. I've also tried running a kernel with all the low-latency stuff turned on, and the same kernel with it all turned off (2.6.16-rc2) doesn't appear to make any difference either. I'm not sure what else to try, any input would be appreciated. Thanks, Gerard Saraber [EMAIL PROTECTED] hardware: AMD64 1.8Ghz 512M ram MSI nforce3 socket 754 mainboard 3 Digium TDM400P cards, 10 FXO + 2 FXS modules /proc/interrupts CPU0 0:2784232IO-APIC-edge timer 1: 8IO-APIC-edge i8042 8: 0IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 177: 71552 IO-APIC-level eth0 185: 9412 IO-APIC-level libata, NVidia CK8S 193: 0 IO-APIC-level ehci_hcd:usb1 201: 0 IO-APIC-level ohci_hcd:usb2 209: 0 IO-APIC-level ohci_hcd:usb3 217:5577811 IO-APIC-level wctdm, wctdm 225:2769262 IO-APIC-level wctdm lspci (for completeness): 02:07.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 217 I/O ports at ac00 [size=256] Memory at fdeff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 02:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 225 I/O ports at a800 [size=256] Memory at fdefe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 02:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 217 I/O ports at a400 [size=256] Memory at fdefd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [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
RE: [Asterisk-Users] No Voice when canreinvite=no
Hi That's a known problem with 1.2.2. Upgrade to 1.2.4. Mike. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kamran Ahmad Sent: Saturday, 11 February 2006 9:09 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] No Voice when canreinvite=no Hi all I am using Asterisk 1.2.2 on frdora core 4. i have two sip UA. if i put canreinvite=yes voice Ok on both sides. and if i change canreinvite=no there is no voice (media through asterisk) one thing more if i try to use playback application for playing some sound file it is also working (like exten = 500,1,Playback(demo-abouttotry) this is working). here is sip.conf //sip.conf// [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes allow=all nat=no [6000] type=peer host=dynamic context=default canreinvite=yes allow=all [1000] type=peer host=dynamic secret=1000 canreinvite=yes allow=all __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] TE411P Really Bad Echo
Hi Rob Is it possible to disable the onboard echo canceller so that one may try the software cancellers instead? I have the TE110P and am experiencing the same bad echo problems that I cant seem to effect by fiddling with the echo canceller settings in zconfig.h Cheers, Mike. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith Sent: Sunday, 12 February 2006 7:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TE411P Really Bad Echo Sorry, that's correct - so when experimenting with s/w echo can try the different options. Rob On 2/11/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I thought that if the VPM was detected then you didn't have any control as to which algorithm was used. I was under the impression that the algorithms were only used for the software echo cancellation. At this point I'll give anything a try. Stagg Shelton www.oneringnetworks.com -Original Message- From:Rob Lith [EMAIL PROTECTED] Subj:Re: [Asterisk-Users] TE411P Really Bad Echo Date:Sat Feb 11, 2006 10:17 am Size:4K To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Stagg I don't think it's a matter of trying echocancel on and off, it is a matter of tuning your system to your local PSTN - this is a combination of trying the different echocan alogrithims ( i.e. MG2), the echotrainign etc and setting your txgain - too loud outgoing audio will result in echo. http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718 Rob On 2/11/06, Stagg Shelton [EMAIL PROTECTED] wrote: Yes, right now we are only using span 1 on the quad span card with plans to pull in another T1 PRI when we get this echo problem solved.The echo is only experienced when the call terminates to traditional analog circuits both local and long distance.Calls to cell phones, and other known digital circuits do not exhibit the symptoms.I've been sent a few articles off list which discussed the reasons why this may occur, imperfect impedance where the digital circuit is switched to the analog circuit. In testing I've set echocancel=no made calls, and echocancel=yes and made calls with no real audible difference.I haven't done a zap show channel while on a call with echo, but I plan to this weekend.I was informed that the number of taps would be shown.I used a stop watch last night to try to get the delay, and it was about 1/2 to 1 second delay and was continual so long as I was talking.It wasn't affected by differing acoustical variations.I tested this using handset, headset, and speakerphone. Disabling VPM and recompiling zaptel or removing the VPM off the board completely is the only thing that has any effect on the echo.Hopefully the zap show channel will provide to me another data point to help me determine if the HW module is active. More to come... Stagg Shelton www.oneringnetworks.com Cory Andrews wrote: Stagg - I know you get a full 128ms tail of echo can on the Sangoma.I believe that on the TE411P, the 128ms tail is shared by all (4) spans, and as you add additional spans up to maximum of 4, the echo can tail amount decreases accordingly.If you are running 4 spans, you have 32ms of echo can tail on each span, not the full 128ms. Now that I'm reading back through your post thread, it looks like you were only running 1 span on the TE411P, so you should have been getting the full 128ms of echo can tail.You may not find an improvement with the Sangoma. I have never experience, nor heard of a 1-2 full second delay.Trial and error is likely your course of action. Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - *From:* Stagg Shelton [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Sent:* Friday, February 10, 2006 11:34 PM *Subject:* Re: [Asterisk-Users] TE411P Really Bad Echo 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
