[Asterisk-Users] asterisk locked up
Earlier today, I hung up a call and then tried to call another number with no success, after the 5th try, I tried calling my voicemail. No luck there either. Any number I tried to call resulted in nothing happening. I got into the server and the CLI and it was not logging any events when I tried to make a call. What can cause this? Would a log tell me what happened? Any thoughts would be appreciated. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --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 locked up
Yes, we are using SIP. How can I tell if I am getting sip deadlocks? voiceserver*CLI show version Asterisk CVS-HEAD-04/12/05-21:44:31 built by [EMAIL PROTECTED] on a i686 running Linux Thanks -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Friday, October 21, 2005 4:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] asterisk locked up Are you using SIP? If so, what version of Asterisk are you using? In older CVS-HEAD versions, I _was_ getting sip deadlocks, but I haven't had that problem in over a month... Sherwood --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Andy Goss -Sent: Friday, October 21, 2005 4:01 PM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: [Asterisk-Users] asterisk locked up - -Earlier today, I hung up a call and then tried to call -another number with no success, after the 5th try, I tried -calling my voicemail. No luck there either. Any number I -tried to call resulted in nothing happening. I got into the -server and the CLI and it was not logging any events when I -tried to make a call. - -What can cause this? - -Would a log tell me what happened? - -Any thoughts would be appreciated. - -Thanks, -Andy - --- -H. Andy Goss -Network Engineer -Network Advocates Inc. -Main: 502.412.1050 -DID: 502.992.5933 -Mobile: 502.387.8216 -[EMAIL PROTECTED] - - -___ ---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 ___ --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 locked up
Thanks for your help. any command I issued from the CLI resulted in nothing. Reload, extensions reload, etc. I would update to the latest CVS-HEAD, however I am concerned that this will break something. Any suggestions on this? Thanks, Andy -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Friday, October 21, 2005 4:33 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] asterisk locked up Yeah you're using a REALLY old CVS-HEAD... An easy way to find out if it's a deadlock is to try SIP SHOW PEERS...if you get nuthin'you're deadlocked... There's also a page on Asterisk debugging on voip-info... (http://www.voip-info.org/wiki-Asterisk+debugging) Basically, this is fixed as far as I can tell. Has been for a little while. You may just want to upgrade to the latest CVS-HEAD. Many many bugs were fixed, and your deadlocks won't go away until you either patch the sip channel code, or upgrade. Hope this was informative.. Sherwood --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Andy Goss -Sent: Friday, October 21, 2005 4:17 PM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: RE: [Asterisk-Users] asterisk locked up - -Yes, we are using SIP. How can I tell if I am getting sip deadlocks? - -voiceserver*CLI show version -Asterisk CVS-HEAD-04/12/05-21:44:31 built by -[EMAIL PROTECTED] on a i686 running Linux - -Thanks --- -H. Andy Goss -Network Engineer -Network Advocates Inc. -Main: 502.412.1050 -DID: 502.992.5933 -Mobile: 502.387.8216 -[EMAIL PROTECTED] - - - -Original Message- - From: [EMAIL PROTECTED] -[mailto:asterisk-users- - [EMAIL PROTECTED] On Behalf Of Sherwood McGowan - Sent: Friday, October 21, 2005 4:06 PM - To: 'Asterisk Users Mailing List - Non-Commercial Discussion' - Subject: RE: [Asterisk-Users] asterisk locked up - - Are you using SIP? If so, what version of Asterisk are you using? - - In older CVS-HEAD versions, I _was_ getting sip deadlocks, but I -haven't - had - that problem in over a month... - - Sherwood - - --Original Message- - -From: [EMAIL PROTECTED] - -[mailto:[EMAIL PROTECTED] On -Behalf Of Andy - -Goss - -Sent: Friday, October 21, 2005 4:01 PM - -To: Asterisk Users Mailing List - Non-Commercial Discussion - -Subject: [Asterisk-Users] asterisk locked up - - - -Earlier today, I hung up a call and then tried to call another - -number with no success, after the 5th try, I tried calling my - -voicemail. No luck there either. Any number I tried to call - -resulted in nothing happening. I got into the server and the CLI - -and it was not logging any events when I tried to make a call. - - - -What can cause this? - - - -Would a log tell me what happened? - - - -Any thoughts would be appreciated. - - - -Thanks, - -Andy - - - --- - -H. Andy Goss - -Network Engineer - -Network Advocates Inc. - -Main: 502.412.1050 - -DID: 502.992.5933 - -Mobile: 502.387.8216 - -[EMAIL PROTECTED] - - - - - -___ - ---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 -___ ---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 ___ --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] possible bug, what do you think?
We recently changed file formats on our server to wav49 from gsm. Several users had saved messages in gsm format. When a user attempts to forward an old message to a user and they prepend the message with a recording, the process seems to be flawed. From what I can tell, the prepend message is recorded to a temporary file, in my case msg-prepend.WAV then after the prepend is finished recording, asterisk attempts to merge the two audio files into one. Since it cannot find a msg.WAV file (the file is msg.gsm) it throws an error. The end result is that the user gets a new message envelope in their INBOX (msg.txt) but there is no associated .WAV file to go along with it. The desired behavior here is to a) notify the user who is attempting to forward this message that the process failed so that the asterisk admin (me) can fix the issue or b) convert the file to the proper format and then merge the two together. What do you all think? Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --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] Polycom MWI
Yes, if you look in the cfg files for the phone (either sip.cfg or ipmid.cfg) you will see something similar to this (I use polycom 501s): MESSAGE_WAITING se.pat.misc.1.name=message waiting se.pat.misc.1.inst.1.type=silence se.pat.misc.1.inst.1.value=1 se.pat.misc.1.inst.2.type=silence se.pat.misc.1.inst.2.value=2 se.pat.misc.1.inst.3.type=silence se.pat.misc.1.inst.3.value=1/ I didn't bother taking out the unnecessary stuff, I just changed where it said chord to silence, this way if I needed to bring it back I could just change silence back to chord. Hope this helps. Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chris Coulthurst Sent: Monday, October 17, 2005 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom MWI I think I have an idea of what dto do here. Look in your sip.cfg file for a line starting with MSG_WAITING under the CALLPROGRESS section. It defines the tone chirp you hear for message waiting notification. I'll bet if you zero out the values it would stop alerting you. P.S. It might be in ipmid.cfg if you have that file instead Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Wilson Pickett [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 17, 2005 12:33 AM Subject: [Asterisk-Users] Polycom MWI Hi, I have lookedaround and don't see this anywhere. Is there a way to tell the ip500 to not make the aural MWI blips? ___ --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 ___ --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] initiate call recording from phone.
I am looking for a way to allow a user to record a call simply by pressing a button or some combination of buttons on their phone. Is this possible? I have read the stuff about the monitor command; however, this doesn't seem to be very interactive for the user. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --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] initiate call recording from phone.
Thanks everyone for the help so far. I figured it out using automon, howver my next question is this: Is there a way to make the recording go to the voicemail directory for the user that records it. This way, they can access it from their phone. -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andy Goss Sent: Monday, October 17, 2005 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] initiate call recording from phone. I am looking for a way to allow a user to record a call simply by pressing a button or some combination of buttons on their phone. Is this possible? I have read the stuff about the monitor command; however, this doesn't seem to be very interactive for the user. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --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] warning message when reloading chan_sap.so
Does anyone know what this error means? WARNING[12632]: chan_zap.c:10084 setup_zap: Ignoring signalling Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --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] help with broken voicemail
drwxr-xr-x6 root root 1024 Oct 12 01:10 . drwxr-xr-x6 root root 1024 Oct 12 01:15 .. -rwxr-xr-x1 root root12801 Oct 11 21:28 busy.wav drwxr-xr-x2 root root 1024 Oct 11 21:28 cust3 -rwxr-xr-x1 root root 3051 Oct 11 21:28 greet.wav drwxr-xr-x2 root root 1024 Oct 11 21:28 inbox drwxr-xr-x2 root root 1024 Oct 12 01:10 old -rwxr-xr-x1 root root29895 Oct 11 21:28 unavail.wav drwxr-xr-x2 root root 1024 Oct 11 21:28 work -bash-2.05b# file unavail.wav unavail.wav: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz From: [EMAIL PROTECTED] on behalf of Tzafrir Cohen Sent: Wed 10/12/2005 2:38 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] help with broken voicemail On Tue, Oct 11, 2005 at 10:09:34PM -0400, Andy Goss wrote: Update - I made a backup of my entire voicemail directory then deleted it. If I then try and record a greeting, it works. Asterisk creates the folder structure and records the greeting. If I try to copy the old file back into the directory, it wont work. It's the same file name and everything. The only thing I can figure might be an issue is that the voicemail drive is mounted as msdos so maybe there is something permissions different about the files that I cant see. Any help would be appreciated. Please post the output of the following two commands: ls -l /path/to/message.wav file /path/to/message.wav Is it indeed a valid wav/RIFF file? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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 winmail.dat___ --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] error message when accessing voicemail
-bash-2.05b# ls -la /var/spool/asterisk/voicemail/default/5933/ total 288 drwxr-xr-x6 root root32768 Oct 12 01:18 . drwxr-xr-x 19 root root32768 Oct 12 01:17 .. -rwxr-xr-x1 root root12936 Oct 12 01:14 busy.gsm drwxr-xr-x2 root root32768 Oct 12 01:14 cust3 -rwxr-xr-x1 root root 3036 Oct 12 01:14 greet.gsm drwxr-xr-x2 root root32768 Oct 12 07:54 inbox drwxr-xr-x2 root root32768 Oct 12 02:01 old -rwxr-xr-x1 root root30294 Oct 12 01:14 unavail.gsm drwxr-xr-x2 root root32768 Oct 12 01:14 work -bash-2.05b# ls -la /var/spool/asterisk/voicemail/default/5933/inbox/ total 128 drwxr-xr-x2 root root32768 Oct 12 07:54 . drwxr-xr-x6 root root32768 Oct 12 01:18 .. -rwxr-xr-x1 root root22110 Oct 12 07:54 msg.gsm -rwxr-xr-x1 root root 264 Oct 12 07:54 msg.txt Asterisk runs under root. I fixed my other errors by converting the wav's to gsm, however they still dont make sense to me. Any thoughts? Thanks, Andy From: [EMAIL PROTECTED] on behalf of Tzafrir Cohen Sent: Wed 10/12/2005 2:55 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] error message when accessing voicemail On Wed, Oct 12, 2005 at 01:44:34AM -0400, Andy Goss wrote: If anyone could tell me what this error is all about, I would be very grateful. Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not permitted Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/Old': Operation not permitted Now, goodnight and thank you in advance Under what user does Asterisk run? ls -la /var/spool/asterisk/voicemail/default/5933 -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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 winmail.dat___ --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] Polycom: Button Remapping, HELP!
It is on page 22 and 23 in my admin guide. Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matthew T. O'Connor Sent: Wednesday, October 12, 2005 2:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom: Button Remapping, HELP! While I don't have it working yet, I think I have it figured out. I have to add keys / entries to my sip.conf Based on your example I was able to find the relevant info in the Polycom SIP 1.5 Admin Guide section 4.6.1.15. My next question, which I haven't found in the admin guide (at least not yet) is where to you get a list of the buttons and their respective numbers? Thanks again, Matthew Mojo with Horan Company, LLC wrote: Do you already have an ipmid/ipmid block in your sip.cfg? add the keys ... / in there: Try putting: ipmid ... ... keys key.IP_500.31.function.prim=SpeedDial key.IP_500.31.subPoint.prim=3/ /ipmid Moj Matthew T. O'Connor wrote: Ok, that would be helpful for me with some other problems, however I don't see keys anywhere in my sip.conf or my phoneXX.conf files. I'm using the 1.5.2 Sip firmware the the conf files that came with that, so I don't have an ipmid.cfg file. Is this something I can just add to my sip.conf? Anyone out there any suggestions on how to do the speed dial in-call? Thanks, Matt Mojo with Horan Company, LLC wrote: Matthew, when I tried this, I couldn't get the soundpoints to dial in-call. They thought there were picking up a new line for a new call. I created a speed-dial entry (in MACADDRESS-directory.xml, itemfnPark/fnct#70#/ctsd3/sd/item) and then in ipmid.cfg: keys key.IP_500.31.function.prim=SpeedDial key.IP_500.31.subPoint.prim=3/ This tells the phone to run Speed Dial 3 whenever the Services button (button #31 on a 500/501) is pressed. I hope someone can help us configure them now to dial these digits in-call... Mojo Matthew T. O'Connor wrote: I need to find a way to have the Polycom phones automatically park calls. Right now my users hit #70# (I know the last # is optional but it speeds it up.) to park a call. Personally I think this is easy, but my users would like one button to do this for them. The Polycom has buttons we don't need (Transfer Services), it would be nice if I could remap one of those buttons to dial #70#. Or if I could add a soft button during a call that would work too. Anyone have any suggestions on how to do this? Thanks much, Matthew O'Connor ___ --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 ___ --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] call to a particular 800 number never showsanswered on Zap channel
First, there is nothing unfair or illegal going on. Large toll-free users have enough clout that they can negotiate contracts, where they are not billed during the service selection phase of a call. For example, when you call American Airlines, billing doesn't start until an agent answers, or the caller selects automated flight information or a similar IVR service. Answer supervision is used to tell the carrier when to start billing. This system is quite common and used by hundreds of companies. This makes good sense, thanks for clearing it up. With Asterisk, three things might go wrong: You may have two-way communication with the IVR, but the call gets disconnected before answer supervision is received. Find out if it's your carrier or Asterisk that is timing out. If the latter, just put a longer timeout in your Dial statement; 180 seconds should be enough. This is the situation I am in. If I am really fast, I can navigate the menu system in enough time to be transferred to a real person, or at least the real-person queue and I get the answer supervision message. Is there a way I can tell if it is asterisk or the carrier that is timing out from the CLI? I thought if the timeout was not specified in the Dial statement it was unlimited, but perhaps I am looking in the wrong place. Also, is there a way to force the phone to start the call counter or force the answer on the asterisk-side. Thanks, Andy ___ --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] call to a particular 800 numbernevershowsanswered on Zap channel
Watch the output of 'pri debug span 1' on the Asterisk server while
placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468)
might be relevant.
Yes, this is exactly what is happening. Thanks a lot. I am thinking about
adding a special case for the IBM 800 number since it is the only one my
company is complaining about. Currently I have this in my dialplan:
; outgoing calls
;
; 7 digit
exten = _NXX,1,Dial(Zap/g1/${EXTEN})
exten = _NXX,2,Congestion
; 10 digit
exten = _NXXNXX,1,Dial(Zap/g1/1${EXTEN})
exten = _NXXNXX,2,Congestion
; 11 digit
exten = _1NXXNXX,1,Dial(Zap/g1/${EXTEN})
exten = _1NXXNXX,2,Congestion
If I wanted to add a special case for IBM, with the Answer before the dial as
suggested for an ugly fix, would I need to add this before the 10 and 11 digit
patterns or does it matter? Would adding this work:
exten = 18004267378,1,Answer()
exten = 18004267378,2,Dial(Zap/g1/${EXTEN})
exten = 18004267378,3,Congestion
Thanks,
Andy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andy Goss
Sent: Monday, October 10, 2005 5:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] call to a particular 800 number
nevershowsanswered on Zap channel
I am still looking to solve this problem, does anyone have any ideas?
Thanks,
Andy
-Original Message-
From: Andy Goss
Sent: Friday, October 07, 2005 5:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] call to a particular 800 number
never showsanswered on Zap channel
Thanks for the reply. Forgive me for being naïve, however
have jumped in to this asterisk project at work due to some
circumstances beyond my control and I don't know a lot about
carriers and how this all works. I am figuring it out, but
it's a lot of trial by fire.
As far as I know, we only use 1 carrier for our system. We
have a PRI from NuVox and we use 7 channels for our asterisk
server. So, I have a few questions:
Is asterisk or the carrier causing the disconnect?
Is IBM (the 800 number I am dialing) not passing the answer
supervision or is that a function of the carrier?
Is there a way to make asterisk not drop the call or to force
the answer on this number? Seems like a hard-PBX would have
to be able to handle this type of situation.
Thanks,
Andy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Garth Summey
Sent: Friday, October 07, 2005 5:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call to a particular 800 number
never showsanswered on Zap channel
This one drove me crazy for a while too. I found out that some
companies don't exactly play fair and don't pass answer
supervision on a
call until you are actually speaking with a live person. The
person I
spoke to about this wasn't sure if that was even legal, but
he said it
happens quite a bit. I was lucky in that I use multiple carriers
(voipjet and broadvoice), voipjet disconnected the call after 60
seconds, but broadvoice did not, so when I find one of those
800 numbers
I route it through broadvoice.
Hope that helps,
G
Andy Goss wrote:
Whenever we call IBM, the call counter on the phone never
starts and in
the CLI the zap channel never gets the answered signal from the PRI.
See below.
-- Executing Dial(SIP/5933-645d, Zap/g1/18004267378) in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/18004267378
At this point, I am in IBM's menu system. However the call never
indicates that it is answered either on the phone or in the
CLI. After
60 seconds, the call disconnects.
-- Hungup 'Zap/1-1'
== Spawn extension (main, 18004267378, 1) exited non-zero on
'SIP/5933-7bff'
-- Executing Hangup(SIP/5933-7bff, ) in new stack
== Spawn extension (main, h, 1) exited non-zero on 'SIP/5933-7bff'
{clip}
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[Asterisk-Users] migrated to new ver on voip connection vs1 server voicemail problems
I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on /var/spool/asterisk/voicemail/default/5933/unavail.wav Oct 11 19:57:26 WARNING[6587]: file.c:804 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/5933/unavail (format ulaw): No such file or di -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --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] help with broken voicemail
I can not figure out what the heck is going on. I went back to my old version and I still get errors when the voicemail system tries to load any of the greetings, unavail messages, etc. the normal voicemail prompts work, but any user recording don't work. Leaving a new message appears to work, but the system wont replay them, it throws errors. Here is an example of the errors: Oct 11 20:32:58 WARNING[3048]: format_wav.c:135 check_header: Not a wav file 49 Oct 11 20:32:58 WARNING[3048]: file.c:415 ast_filehelper: Unable to open fd on /var/spool/asterisk/voicemail/default/5926/INBOX/msg.wav Oct 11 20:32:58 WARNING[3048]: file.c:793 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/5926/INBOX/msg (format ulaw): No such file or directory Oct 11 20:33:03 WARNING[3048]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5926/INBOX': File exists Oct 11 20:33:08 WARNING[3048]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5926/Old': File exists Any help would be very much appreciated as I am in a real pickle here, I need to get this up and running quickly. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --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] help with broken voicemail
Yes, the path exists, the files exist, and the permissions all are 755, all owned by root and in group root. I cant figure it out for the life of me. -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of El Flynn Sent: Tuesday, October 11, 2005 8:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] help with broken voicemail Andy Goss wrote: Oct 11 20:32:58 WARNING[3048]: file.c:415 ast_filehelper: Unable to open fd on /var/spool/asterisk/voicemail/default/5926/INBOX/msg.wav Oct 11 20:32:58 WARNING[3048]: file.c:793 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/5926/INBOX/msg (format ulaw): No such file or directory can you check that /var/spool/asterisk exists, and that all its subdirectories are intact? perhaps it got deleted by accident somehow? you might also want to check the file permissions on the directories. flynn ___ --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] help with broken voicemail
Update - I made a backup of my entire voicemail directory then deleted it. If I then try and record a greeting, it works. Asterisk creates the folder structure and records the greeting. If I try to copy the old file back into the directory, it wont work. It's the same file name and everything. The only thing I can figure might be an issue is that the voicemail drive is mounted as msdos so maybe there is something permissions different about the files that I cant see. Any help would be appreciated. -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andy Goss Sent: Tuesday, October 11, 2005 8:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] help with broken voicemail Yes, the path exists, the files exist, and the permissions all are 755, all owned by root and in group root. I cant figure it out for the life of me. -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of El Flynn Sent: Tuesday, October 11, 2005 8:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] help with broken voicemail Andy Goss wrote: Oct 11 20:32:58 WARNING[3048]: file.c:415 ast_filehelper: Unable to open fd on /var/spool/asterisk/voicemail/default/5926/INBOX/msg.wav Oct 11 20:32:58 WARNING[3048]: file.c:793 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/5926/INBOX/msg (format ulaw): No such file or directory can you check that /var/spool/asterisk exists, and that all its subdirectories are intact? perhaps it got deleted by accident somehow? you might also want to check the file permissions on the directories. flynn ___ --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 ___ --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] help with broken voicemail
Update - After we converted all the audio to gsm, it miraculously started working. I still don't know why. If anyone knows how the wav49 codec or whatever can get screwed up, your input is still welcome. Thanks and goodnight, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andy Goss Sent: Tuesday, October 11, 2005 10:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] help with broken voicemail Update - I made a backup of my entire voicemail directory then deleted it. If I then try and record a greeting, it works. Asterisk creates the folder structure and records the greeting. If I try to copy the old file back into the directory, it wont work. It's the same file name and everything. The only thing I can figure might be an issue is that the voicemail drive is mounted as msdos so maybe there is something permissions different about the files that I cant see. Any help would be appreciated. -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andy Goss Sent: Tuesday, October 11, 2005 8:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] help with broken voicemail Yes, the path exists, the files exist, and the permissions all are 755, all owned by root and in group root. I cant figure it out for the life of me. -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of El Flynn Sent: Tuesday, October 11, 2005 8:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] help with broken voicemail Andy Goss wrote: Oct 11 20:32:58 WARNING[3048]: file.c:415 ast_filehelper: Unable to open fd on /var/spool/asterisk/voicemail/default/5926/INBOX/msg.wav Oct 11 20:32:58 WARNING[3048]: file.c:793 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/5926/INBOX/msg (format ulaw): No such file or directory can you check that /var/spool/asterisk exists, and that all its subdirectories are intact? perhaps it got deleted by accident somehow? you might also want to check the file permissions on the directories. flynn ___ --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 ___ --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] error message when accessing voicemail
If anyone could tell me what this error is all about, I would be very grateful. Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not permitted Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/Old': Operation not permitted Now, goodnight and thank you in advance Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --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] customize the pager email
I am running CVS-HEAD-04/12/05-21:44:31 and I am curious if it is possible to customize the email message sent to the pager email address. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --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] call to a particular 800 number never showsanswered on Zap channel
I am still looking to solve this problem, does anyone have any ideas? Thanks, Andy -Original Message- From: Andy Goss Sent: Friday, October 07, 2005 5:37 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] call to a particular 800 number never showsanswered on Zap channel Thanks for the reply. Forgive me for being naïve, however have jumped in to this asterisk project at work due to some circumstances beyond my control and I don't know a lot about carriers and how this all works. I am figuring it out, but it's a lot of trial by fire. As far as I know, we only use 1 carrier for our system. We have a PRI from NuVox and we use 7 channels for our asterisk server. So, I have a few questions: Is asterisk or the carrier causing the disconnect? Is IBM (the 800 number I am dialing) not passing the answer supervision or is that a function of the carrier? Is there a way to make asterisk not drop the call or to force the answer on this number? Seems like a hard-PBX would have to be able to handle this type of situation. Thanks, Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth Summey Sent: Friday, October 07, 2005 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call to a particular 800 number never showsanswered on Zap channel This one drove me crazy for a while too. I found out that some companies don't exactly play fair and don't pass answer supervision on a call until you are actually speaking with a live person. The person I spoke to about this wasn't sure if that was even legal, but he said it happens quite a bit. I was lucky in that I use multiple carriers (voipjet and broadvoice), voipjet disconnected the call after 60 seconds, but broadvoice did not, so when I find one of those 800 numbers I route it through broadvoice. Hope that helps, G Andy Goss wrote: Whenever we call IBM, the call counter on the phone never starts and in the CLI the zap channel never gets the answered signal from the PRI. See below. -- Executing Dial(SIP/5933-645d, Zap/g1/18004267378) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/18004267378 At this point, I am in IBM's menu system. However the call never indicates that it is answered either on the phone or in the CLI. After 60 seconds, the call disconnects. -- Hungup 'Zap/1-1' == Spawn extension (main, 18004267378, 1) exited non-zero on 'SIP/5933-7bff' -- Executing Hangup(SIP/5933-7bff, ) in new stack == Spawn extension (main, h, 1) exited non-zero on 'SIP/5933-7bff' Does anyone have any ideas? Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --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 ___ --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] call to a particular 800 number never shows answered on Zap channel
Whenever we call IBM, the call counter on the phone never starts and in the CLI the zap channel never gets the answered signal from the PRI. See below. -- Executing Dial(SIP/5933-645d, Zap/g1/18004267378) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/18004267378 At this point, I am in IBM's menu system. However the call never indicates that it is answered either on the phone or in the CLI. After 60 seconds, the call disconnects. -- Hungup 'Zap/1-1' == Spawn extension (main, 18004267378, 1) exited non-zero on 'SIP/5933-7bff' -- Executing Hangup(SIP/5933-7bff, ) in new stack == Spawn extension (main, h, 1) exited non-zero on 'SIP/5933-7bff' Does anyone have any ideas? Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --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] call to a particular 800 number never showsanswered on Zap channel
Thanks for the reply. Forgive me for being naïve, however have jumped in to this asterisk project at work due to some circumstances beyond my control and I don't know a lot about carriers and how this all works. I am figuring it out, but it's a lot of trial by fire. As far as I know, we only use 1 carrier for our system. We have a PRI from NuVox and we use 7 channels for our asterisk server. So, I have a few questions: Is asterisk or the carrier causing the disconnect? Is IBM (the 800 number I am dialing) not passing the answer supervision or is that a function of the carrier? Is there a way to make asterisk not drop the call or to force the answer on this number? Seems like a hard-PBX would have to be able to handle this type of situation. Thanks, Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth Summey Sent: Friday, October 07, 2005 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call to a particular 800 number never showsanswered on Zap channel This one drove me crazy for a while too. I found out that some companies don't exactly play fair and don't pass answer supervision on a call until you are actually speaking with a live person. The person I spoke to about this wasn't sure if that was even legal, but he said it happens quite a bit. I was lucky in that I use multiple carriers (voipjet and broadvoice), voipjet disconnected the call after 60 seconds, but broadvoice did not, so when I find one of those 800 numbers I route it through broadvoice. Hope that helps, G Andy Goss wrote: Whenever we call IBM, the call counter on the phone never starts and in the CLI the zap channel never gets the answered signal from the PRI. See below. -- Executing Dial(SIP/5933-645d, Zap/g1/18004267378) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/18004267378 At this point, I am in IBM's menu system. However the call never indicates that it is answered either on the phone or in the CLI. After 60 seconds, the call disconnects. -- Hungup 'Zap/1-1' == Spawn extension (main, 18004267378, 1) exited non-zero on 'SIP/5933-7bff' -- Executing Hangup(SIP/5933-7bff, ) in new stack == Spawn extension (main, h, 1) exited non-zero on 'SIP/5933-7bff' Does anyone have any ideas? Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --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 ___ --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] hardware echo cancellation. sangoma?
I have been reassigned from my normal duties to figure out the asterisk echo problems we have been experiencing. We currently use a TE110P card (I think.) I know that the problem is the worst when calling from our office to a residential analog line or a analog PBX. Occasionally the problem will appear when calling anther digital PBX. The Asterisk Guru who just left my company mentioned that there is a Sangoma card that can help with the echo and he gave me the name of David Mandelstam. So David, if you are listening, I would appreciate your thoughts or advice. If anyone else has some great ideas, they would be appreciated too. Thanks, Andy Goss -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 [EMAIL PROTECTED] ___ --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
