Re: [asterisk-users] Polycom IP331 Configuration
Thanks David. I will check it out. -Original message- From: Klaverstyn, David C david.klavers...@intergraph.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Mon, Feb 13, 2012 04:34:30 GMT+00:00 Subject: Re: [asterisk-users] Polycom IP331 Configuration This may help you -- http://www.klaverstyn.com.au/david/wiki/index.php?title=Provision_Polycom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Johnson Sent: Monday, 13 February 2012 5:57 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom IP331 Configuration I hope this doesn't already exist, but I couldn't find anything to help. I am installing a brand new Asterisk server, and want to use the Polycom IP331 phones. Does anyone have any steps on how to configure these? I have softphones working just fine, but for some reason I can't find a clear step by step on provisioning the Polycoms. Any help is greatly appreciated! Mark J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom IP331 Configuration
I hope this doesn't already exist, but I couldn't find anything to help. I am installing a brand new Asterisk server, and want to use the Polycom IP331 phones. Does anyone have any steps on how to configure these? I have softphones working just fine, but for some reason I can't find a clear step by step on provisioning the Polycoms. Any help is greatly appreciated! Mark J. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] display time on Cisco 79xx
Chris Carey wrote: They get the time from their NTP server On Mon, Mar 10, 2008 at 11:59 AM, Don Smith [EMAIL PROTECTED] wrote: I am running Asterisk 1.4.5 on a debian Linux server. Saturday night/Sunday Morning Daylight Savings time occurred. The server shows Mon Mar 10 10:59:42 PDT 2008 when I do a date command, but the Cisco 7940 and 7960 show 09:59 10/03/08. How do I update the time display on the telephones please? Edit your SIPDefault.cnf file on your tftp server and do something like this: time_zone: EST ; Time Zone Phone is in dst_offset: 1 ; Offset from Phone's time when DST is in effect dst_start_month: March ; Month in which DST starts dst_start_day:; Day of month in which DST starts dst_start_day_of_week: Sun ; Day of week in which DST starts dst_start_week_of_month: 2 ; Week of month in which DST starts dst_start_time: 02 ; Time of day in which DST starts dst_stop_month: Nov ; Month in which DST stops dst_stop_day: ; Day of month in which DST stops dst_stop_day_of_week: Sunday; Day of week in which DST stops dst_stop_week_of_month: 1 ; Week of month in which DST stops 8=last week of month dst_stop_time: 2; Time of day in which DST stops dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment -- Mark Johnson http://www.astroshapes.com/information-technology/blog/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rxfax does not work (anymore)
Ronald Wiplinger wrote: [Jan 27 16:03:32] -- Executing RxFAX(SIP/88621001-00728610, /var/spool/asterisk-fax/3000/1201421004.8.tif) in new stack vpbx*CLI Disconnected from Asterisk server I have no idea why it disconnects and hope somebody can help me to get to work. bye Ronald You subject says it doesn't work anymore. Did you change something? Upgrade Asterisk and not SpanDSP? This is the same message you would see if the asterisk service wrecked. Have you tried turning on full debugging? If not, edit the /etc/asterisk/logger.conf and make sure there is a line in there like: full = notice,warning,error,debug,verbose Then, go through your fax process and check the file /var/log/asterisk/full. It will have tons of information about what went on. When you are done with this, be sure to disable the full logging because it will eat a lot of drive space. -- Mark Johnson http://www.astroshapes.com/information-technology/blog/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Your favorite Asterisk application.
Ken D'Ambrosio wrote: Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So I'm going to have to try to sell him. I figured one place to start would be some of the really cool applications that Asterisk has that -- generally, at least -- don't require licensing. Some of my favorites are follow-me, meetme, voicemail-to-e-mail and fax-to-e-mail. What are some of your favorite features/applications, be ith native or third-party? Thanks, -Ken We moved from a Cisco Call Manager about 2.5 years ago to Asterisk. One of the hurdles I had was that the Call Manager had a receptionist panel so they could see who was on the phone, transfer calls, etc... I set up a demo of of the Flash Operator Panel and it alleviated that sticking point. It's a little slower than an executable would be, but it's web based and flash so it's runs on just about every browser and OS. You can even do some slick things like pop up windows in the browser to provide information about who is calling. Works good for a CMS system where a customer service rep can automatically be shown information about the customer who is on the line. http://www.asternic.org/ -- Mark Johnson http://www.astroshapes.com/information-technology/blog/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding difficulty in installing Asterisk
[EMAIL PROTECTED] wrote: Hi all, Please help me in installing Asterisk. I am getting the following error when trying to install Libpri Please help me out. Thanking you, Preeta Pandey You aren't compiling the latest version of 1.4.3. Have you tried that? If that doesn't work, what are the specs of the machine you are on? OS? 32 or 64bit? etc... -- Mark Johnson http://www.astroshapes.com/information-technology/blog ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer Question
I'm having a tough time figuring out how to do something. If I have an operator (which could potentially be in their own context) and an internal-only context, is it possible to make it so the operator can call the internal-only context but *NOT* transfer calls to it? The idea is that the internal-only context should not be allowed to make or receive outside calls. The only concern is that the operator and other office users can transfer outside calls to these internal-only extensions. Also, the operator and office extensions need to be able to call the internal-only extensions directly. Thanks! Mark ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Question
Noah Miller wrote: Sort of. You can create a special extension in the operator's context with a Goto() statement. Something like this: [operator] exten = 100,1,Goto(internal,prompt,1) Then in the internal context: [internal] exten = prompt,1,Background(who-do-you-want-to-call) exten = prompt,2,Waitexten(10) So, when the operator dials 100, he/she can then dial an extension in the internal context. Normal transfer from [operator] to [internal] would not be allowed. - Noah This might work, but I don't want people to have to remember to dial 100 if they need to call a certain set of extensions. I know that the internal numbers all have 3 digits in their caller-id. Maybe have a different action if the caller-id is not exactly 3 digits? Mark ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Phone CID
Rob Schall wrote: This might sound like an odd question but here it is anyways... We currently have Polycom 501 phones. We have Asterisk with Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone dials another, the receiving end does in fact see the callers ID. But... our old phone system set the caller id on the senders phone to show who they called. Example... If Sally calls Jim, then Sally's phone should just say 1001, it should say Jim 1001. Any know if this is possible. Our old PBX did this, and the bosses were curious if this is possible. Thanks, Rob I have tried over and over to figure out how to do this and it doesn't seem possible at the moment. I know this can be done with chan_sccp and maybe even chan_skinny (haven't tried that in a few years), but you'd need Cisco phones to do it. Is this something on anyone's To-Do list? Thanks, Mark ___ --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] Red: Sip Phone CID
Jason Fuermann wrote: I'm not sure about the sippeer stuff, or where they get that variable. We lookup our info in a database to set it. Also to use sipcalledrpid you'll probably need the patch at http://bugs2.digium.com/view.php?id=6643 . I looked at this in the past and never made it work correctly. Does this work in the newest version of 1.2? Mark ___ --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] SpanDSP and Asterisk 1.4
Has anyone made this combination work together? I've tried everything and can't seem to get it work right. It all compiles fine, but when rxfax is called, I get an unknown symbol error. From my reading, everything points to me having multiple copies of spandsp and it's maybe calling the wrong one. I went through the directories and they all look clean when I install. Here's what I'm trying: Asterisk-1.4.0 spandsp-20061217 (from the snapshots) The patchfile from the snapshots works except for one hunk, so I manually apply that one part. Anyone got this working? Any pointers? I had a previous copy of spandsp-0.0.2pre26 prior to this but I really think I got it all removed. Thanks! Mark ___ --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] 1.4.0, IMAP and Dovecot
Dan Austin wrote:
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
The imap server is also the Asterisk server, so connections are
on the localhost.
The error posted to the logs is:
IMAP Error: Can't open mailbox
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX: invalid remote
specification
Digging in the code and the c-client documentation the '//' is where
additional flags would go. I've tried a number of the flags supported
by the c-client library, but the results are the same.
Has anyone managed to get IMAP working in Asterisk with Dovecotas the
backend?
I've been attempting the same with Cyrus and get the same results. The
interesting thing is if I take the same string (like
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX) and plug it
into the 'mtest' command from the c-client package, it works OK. I have
not tried this with the production release with Asterisk. Only beta's 1-4.
Mark
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Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
Matt Gibson wrote: Hi Pavel, Thanks for the config! I modified mine so it was more minimal like yours, and it registers just fine now. So much nicer without those big red X's! MG This modified config works sweet!! Any tricks to get the MWI working? Mark ___ --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] OT: MWI on Treo 600/650
Andrew Kohlsmith wrote: On Thursday 13 April 2006 09:02, David Cook wrote: My cell vm goes to asterisk, not the carrier. Apparently MWI is turned on/off with specially formatted SMS messages. Anyone know how to do this on a Treo 600? Having the phone light from Asterisk would be HUGE ... not to mention extremely cool. I've been working on this off and on for AGES. There are some SMS portal sites that claim to be able to do this as well, but I have not managed to find one. -A. I know this thread is probably a little aged, but I'm intrigued... How are you forwarding cell vm to asterisk? When busy or unavailable, do you forward to a DID set up to go directly to your asterisk voicemail? I get so many complaints about how the buttons to navigate Asterisk voicemail are different from the company's cell phones and different again from their personal cell phones. I could combine at least two of them this way! Mark ___ --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] Cisco 79xx and SIP 7.5 Problems
C F wrote: I recently updated my phones Cisco 7960 phones (3 of them) in a high volume call place, where the Secretaries use the 7960 phones to answer inbound calls, as many as 15 simultaneous calls between all three of them. Since then I have had only constant problems, mainly that after 3 calls on a phone, if they try to xfer or do any ohter things (sometimes just answer the 4th call) the phone freezes, they have had this happen to them throughout the week. Until yesterday I decided it must be a frimware problem, so I downgraded them to 7.1. Since then (around 5PM EST yesteday) it didn't happen *yet*. So I'm assuming it has to do with the firmware. So my question is, is anybody else using 7.5 firmware? If yes, do you have all the line buttons configured to the same SIP account? If yes, do you see the same problem? I also noticed that with 7.5 firmware callwaiting has to be enabled for the second call to be able to come in, otherwise the phone returns a Busy here, while with the older versions it could have been disabled and it worked fine, the phone only returned busy here on the 7th call. So I had to enable call waiting, the way I did it was that in the SIPmac.cfg file I added call_waiting: 3 I'm not sure if this is related or not, but that was the only change I had to do to the config files. It's nothing you did... I did the same thing. Went from 7.4 to 7.5 and all sorts of weird things started happening. The biggest of which was lines 2-6 wouldn't register or display the same busy message you got. I also got double ringing which someone told me how to fix. The phones locked up... I rolled back to 7.4 and have had no issues since. Good luck! Mark ___ --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 1.2.4 seg faulting today had been working fine since update
I upgraded to 1.2.4 today and am having issues and can't figure this out. Here's the bottom part of a gdb and a backtrace. Any thoughts? May roll back to 1.2.3? Mark Reading symbols from /usr/lib/asterisk/modules/app_saycountpl.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_saycountpl.so #0 0x080c8cf0 in __ast_device_state_changed_literal (buf=0xbf44d974 SIP/Operator1) at lock.h:611 611 lock.h: No such file or directory. in lock.h (gdb) bt #0 0x080c8cf0 in __ast_device_state_changed_literal (buf=0xbf44d974 SIP/Operator1) at lock.h:611 #1 0x080c8934 in ast_device_state_changed (fmt=0x0) at devicestate.c:243 #2 0x00322313 in register_verify (p=0xbf460538, sin=0x4cbba4, req=0x4cbbb4, uri=0x4cbdd5 sip:asterisk.astroshapes.com, ignore=0) at chan_sip.c:6438 #3 0x0032000e in handle_request (p=0xbf460538, req=0x4cbbb4, sin=0x4cbba4, recount=0x0, nounlock=0x0) at chan_sip.c:10850 #4 0x0031df80 in sipsock_read (id=0x99b41c8, fd=18, events=1, ignore=0x0) at chan_sip.c:11135 #5 0x0805581d in ast_io_wait (ioc=0x99543e8, howlong=0) at io.c:284 #6 0x00313e31 in do_monitor (data=0x0) at chan_sip.c:11284 #7 0x00f3adb2 in pthread_start_thread () from /lib/i686/libpthread.so.0 #8 0x0042f35a in clone () from /lib/i686/libc.so.6 I'm having some trouble here. I really thought chan_sccp was the problem, but now I'm not so sure. Is anyone running 1.2.4 in a production environment without issues? Here's what happened today: (gdb) bt #0 0x0025d8e4 in _int_malloc () from /lib/i686/libc.so.6 #1 0x0025ca23 in malloc () from /lib/i686/libc.so.6 #2 0x0063b269 in sccp_process_data (s=0x325340) at sccp_socket.c:229 #3 0x0063b5a2 in sccp_socket_thread (ignore=0x0) at sccp_socket.c:295 #4 0x00519db2 in pthread_start_thread () from /lib/i686/libpthread.so.0 #5 0x002cb35a in clone () from /lib/i686/libc.so.6 ___ --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 Changes
I just ran into this today, on 1.2.3 with Polycom IP 501 phones. Message was from a potential customer looking for a PBX too... imagine that embarrassment :) Anyone know how to get this resolved? Thanks, Nathan I had this happen today, also. I've seen it happen in the past, but became a problem today. A user missed a call. The caller started to leave a message when his MWI came on. He went to listen to the message and there was nothing to listen to. The voicemail system seems that have attempted to move the files to the ../Old directory but could only deal the the .txt file, leaving a .gsm, .wav and .WAV file in the INBOX. Another voicemail was left and the user could not listen to it. The system kept playing the 0 byte msg.gsm file instead of the latest msg0001.gsm file. I had to remove all of the msg.??? files and then rename all of the msg0001.??? to msg.??? for him to retrieve voicemails again. Mark ___ --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 1.2.4 seg faulting today had been working fine since update
Jerry Geis wrote: All, I had updated to 1.2.4 right when it came out. I had been working just fine. Today I seem to be having recuring seg faults. can explain it. How can I find why? Anyone else experiencing this? I am running (2) TDM04B cards (has been working since 1.0.9) I have a handfull of UIP200 phones and 1 cisco 7960. I have a unused broadvoic connection that I commented out the registration statement but made no difference I have a NuFone account that is rarely used. Jerry I upgraded to 1.2.4 today and am having issues and can't figure this out. Here's the bottom part of a gdb and a backtrace. Any thoughts? May roll back to 1.2.3? Mark Reading symbols from /usr/lib/asterisk/modules/app_saycountpl.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_saycountpl.so #0 0x080c8cf0 in __ast_device_state_changed_literal (buf=0xbf44d974 SIP/Operator1) at lock.h:611 611 lock.h: No such file or directory. in lock.h (gdb) bt #0 0x080c8cf0 in __ast_device_state_changed_literal (buf=0xbf44d974 SIP/Operator1) at lock.h:611 #1 0x080c8934 in ast_device_state_changed (fmt=0x0) at devicestate.c:243 #2 0x00322313 in register_verify (p=0xbf460538, sin=0x4cbba4, req=0x4cbbb4, uri=0x4cbdd5 sip:asterisk.astroshapes.com, ignore=0) at chan_sip.c:6438 #3 0x0032000e in handle_request (p=0xbf460538, req=0x4cbbb4, sin=0x4cbba4, recount=0x0, nounlock=0x0) at chan_sip.c:10850 #4 0x0031df80 in sipsock_read (id=0x99b41c8, fd=18, events=1, ignore=0x0) at chan_sip.c:11135 #5 0x0805581d in ast_io_wait (ioc=0x99543e8, howlong=0) at io.c:284 #6 0x00313e31 in do_monitor (data=0x0) at chan_sip.c:11284 #7 0x00f3adb2 in pthread_start_thread () from /lib/i686/libpthread.so.0 #8 0x0042f35a in clone () from /lib/i686/libc.so.6 ___ --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 hangs on 1.2.1
Mark Johnson wrote: Anyone have any idea what's causing this or how to debug it? I'm pretty sure the root cause is with chan_sccp.so, but not sure how to prove it. I recently upgraded from CVS-head to 1.2.1 and the chan_sccp from 12-17-2005. Now, once or twice a week, I get this on the console: Jan 31 10:39:08 WARNING[10586]: channel.c:784 channel_find_locked: Avoided deadlock for '0xbf1013e0', 10 retries! Once this happens, all of my sccp phones drop offline and attempt to register. I get no sccp messages on the console. There's really nothing on the console to indicate any sort of problem. If I try to do an unload chan_sccp.so and then load it back, all of my SIP phones lose their registrations, none of my Zap channels work and I have to kill Asterisk and restart it. Is this an Asterisk problem or an SCCP problem? Help!! It did it to me again. I enabled full logging and here's what I get. All the 7910's drop off line and try to reregister. All SCCP messages on the CLI stop. Anytime I try a show channels I get the Avoided deadlock message. Here's what the logfile shows. Any ideas? And is there a way to fix the deadlock without restarting Asterisk? Feb 1 09:10:33 WARNING[5327]: channel.c:784 channel_find_locked: Avoided deadlock for '0xbf002d10', 10 retries! Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:09 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:09 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:09 WARNING[6606] channel.c: Avoided deadlock for '0xbf002d10', 10 retries! ___ --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] changing cisco 7940/7960 standard menus ?
Alex Ongena wrote: Hi, We are using Asterisk 1.2.1 with Cisco 7940 and 7960 phones. Most things are running fine ;-) But, when you are calling and you want to Transfer, you need to press first on the 'more' button (4th), then you have the label 'Trnsfr' to Transfer. these are the lables on the softkeys when having a phone call: Holt / EndCall / Confrn / more press more and you get Transfer / / BlndXfr / more We do more 'Transfers' than 'Confrn', so I which to siwtch the 2 softkeys on the phone. Can you do that ? How ? Thanks alex I went though the same thing. I don't think you can change the menus. I simply set up Asterisk to Blind Xfer with the # key. So instead of using the softkeys, you hit # and then the extension and off the call goes. It works out nice because if you go to a different phone, the procedure stays the same. Mark ___ --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] changing cisco 7940/7960 standard menus ?
Chris Bagnall wrote: Is this specific to the SIP firmware? I'm using chan_sccp with a few 7960s and Transfer is definitely on one of the initial softkeys when on a call. If it's a SIP thing, you might want to consider using SCCP. Regards, Chris Yes, the SIP image did some pretty strange things. The worst change they made was hot dial feature went away. You have to lift the handset or go on speakerphone to start dialing the number. Mark ___ --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] Re: Asterisk hangs on 1.2.1
Peter Fern wrote: I'm pretty sure I've seen some commits dealing with channel locking since 1.2.1 Brent Torrenga wrote: Might it be related to the memory leak bug? Upgrade to 1.2.4? (shot in the dark, a brainstorm on my part is all) Here's what the logfile shows. Any ideas? And is there a way to fix the deadlock without restarting Asterisk? Feb 1 09:10:33 WARNING[5327]: channel.c:784 channel_find_locked: Avoided deadlock for '0xbf002d10', 10 retries! Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Thanks for the suggestions! I'll try the production box this weekend. I just installed the latest in lab and it looks OK. Mark ___ --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] Asterisk hangs on 1.2.1
Anyone have any idea what's causing this or how to debug it? I'm pretty sure the root cause is with chan_sccp.so, but not sure how to prove it. I recently upgraded from CVS-head to 1.2.1 and the chan_sccp from 12-17-2005. Now, once or twice a week, I get this on the console: Jan 31 10:39:08 WARNING[10586]: channel.c:784 channel_find_locked: Avoided deadlock for '0xbf1013e0', 10 retries! Once this happens, all of my sccp phones drop offline and attempt to register. I get no sccp messages on the console. There's really nothing on the console to indicate any sort of problem. If I try to do an unload chan_sccp.so and then load it back, all of my SIP phones lose their registrations, none of my Zap channels work and I have to kill Asterisk and restart it. Is this an Asterisk problem or an SCCP problem? Help!! Mark ___ --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 and VPN
Anyone out there got a SIP phone (mine's a Cisco 7940) to work through a VPN with a Netscreen 5gt? It has always worked for me with any ScreenOS version 4.x. I had the need to upgrade it to ScreenOS 5.x and it breaks the phone. Here's the goofy part, it works enough to still register with the phone system and check if there is voicemail waiting. But I get no audio on outbound calls. Inbound calls seem to work OK. The netscreen is not in NAT mode, but in route mode. When the phone system talks to the phone at home, it uses the home LAN address. In debug mode, the phone system doesn't seem to notice anything is wrong. I don't know if this means anything or not, but... On the phone system, if I do a nmap -sU -p5060 homephoneip it comes back with the port is open. If I do the same thing from my home PC and nmap the SIP port on the phone system, it comes back open|filtered which I think means no UDP packet is returning. SSH to the phone system works fine from home. I also noticed that NTP os broken on the phone, so something is wrong with UDP. I found a really good article from someone having the same issues but it made no difference for me. I have a support contract through Juniper, but they still have not found any resolution. Here's the sip.conf section. I tried some variations with canreinvite and some things, but it didn't help. This has worked for me over a year like this. Anyone got any ideas? Thanks! Mark [1426] type=friend username=123456 secret=123456 host=dynamic ;canreinvite=no ;disallow=all ;allow=ulaw,alaw ;dtmfmode=inband ;nat=never context=office [EMAIL PROTECTED] linelabel=First Last callerid=First Last 1426 line = 1426 ___ --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] SIP and VPN
Lists Pleasants wrote: ScreenOS 5.0x and 5.1x has some issues wit SIP. Try the policies I have listed below. set policcy id 1001 from Trust to Trust Local Remote SIP permit log count set policy id 1001 application IGNORE set policy id 1002 from Trust to Trust Remote Local SIP permit log count set policy id 1002 application IGNORE I am running 5.2r1 without any issues but I have turned off any application deep scanning. unset alg sql unset alg q931 unset alg h245 unset alg ras unset alg sip -Chip I tried adding the above and it made no difference. My unset alg lines look a little different. They end in enable, but that could be the software version. I'm still getting stumped as to how it can register correctly and not have audio on outbound calls. I double checked and if I call from the phone system to the home phone, audio is fine! Mark ___ --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] SIP and VPN
Lists Pleasants wrote: ScreenOS 5.0x and 5.1x has some issues wit SIP. Try the policies I have listed below. set policcy id 1001 from Trust to Trust Local Remote SIP permit log count set policy id 1001 application IGNORE set policy id 1002 from Trust to Trust Remote Local SIP permit log count set policy id 1002 application IGNORE I am running 5.2r1 without any issues but I have turned off any application deep scanning. unset alg sql unset alg q931 unset alg h245 unset alg ras unset alg sip -Chip Why do you go from Trust to Trust in your policies? I tried that and the phone won't work at all. The only way to get it to register is for me to put Remote as an Untrust zone. Thanks! Mark ___ --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] SIP and VPN
cp wrote: The example I gave was going over a VPN with tunnel terminating in the trusted zone. Put the polices how our traffic traverse through the netscreen. I would config a policy for trust to untrust traffic and for untrust to trust or untrust to global if you have MIPing going on. -chip I tried everything and can't figure this out. I can talk all day on the phone if the call originates from somewhere else. I watched the packets with ethereal and the issue seems to be something with the RTP packets. Mark ___ --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] Cisco 7960 Password Recovery
Polycom User wrote: i appear to misplaced my password for my cisco 7960 SIP Phone. Does anyone know the procedure to recover this? I have read in the past that you can use cisco or Cisco but this does not appear to work. Thanks in advance. Is this phone setup using tftp? If so, I would check in the SIPDefault.cnf file or the SIPxxx.cnf file that matches the phone's MAC address on the tftp server. Mark ___ --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] libbluetooth
Victor Alvarez wrote: Hi, I found a problem when trying to install the module chan_bluetooth from 'the crazy greek'. Most of installation seems fine, chan_bluetooth.so is created and located in /usr/src/asterisk/channels/. But when I try to start up asterisk, I get the following message: [chan_bluetooth.so]Jan 8 16:55:07 WARNING[18861]: loader.c:258 ast_load_resource: libbluetooth.so.1: cannot open shared object file: No such file or directory Jan 8 16:55:07 WARNING[18861]: loader.c:440 load_modules: Loading module chan_bluetooth.so failed! That file (libbluetooth.so.1) is in /usr/local/lib/. Should I copy it somewhere else? Where is it trying to find the file? Thanks, Victor. First, check your /etc/ld.so.conf file and make sure /usr/local/lib is in there. When your are certain it's there, run ldconfig -v and all the .so objects should scroll by and you'll see your new bluetooth shared object file! Asterisk should start after that. Mark ___ --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] Moments of silence - take2
Jimmy Smith wrote: seems every 10 sec something is happeneing on your network... make sure your router is rebooted often if you have QOS on it has they tend to get behind on queues.. or UDP crc checksum failing in router.. that happened to me on a linksys your ping is ok 60 is good i would also test my lan quality .. or wan.. some providers cut connections every xx seconds to deter peer sharing THE KEY HERE is you said 2 providers.. meaning i higlhy doubt its them.. 1 ok 2 no way.. its on your side.. solution #1 try another router. #2 try to do a line quality test see if its regular interval something is hapepning.. check your mta also.. I had something happen almost identical to me a few weeks ago. After alot of hair pulling, it turned out it was my own fault. I went to debug a core dump and instead of typing gdb, I typed gdm. It was attempting to do something with X and the Gnome Display Manager and kept failing. It repeatedly tried once every 10 seconds and chewed the processor while it was doing it. Once I killed the process, everything was fine. DOH!!! Mark ___ --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] Slightly OT: Cisco 7960/7940 and AsteriskReg istration Issues ove r a WAN
Geoff Manning wrote: Info relating to the 7.5 firmware version and it failing to register. Thus needing a reboot to fix: I don't have any documentation, but I can tell you that the 7.5 image caused me ALL sorts of headaches. I rolled it out to a few phones to test, one being our receptionist. On a 7960 with 6 lines, I have Asterisk configured to roll new calls to the first available line. The only line that would register correctly was line 1. Lines 5-6 I never did get to register correctly. I rolled all of the phones back to 7.4 and all is well again. Mark ___ --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] Double Ringing for PRI Calls
Matt wrote: Yes, Go into sip.conf and add this line: progressinband=no Thank you!!! My Cisco 7960's started acting weird with SIP version 7.5, so I kept them at 7.4 for this reason. Works great now! Mark ___ --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] ast_fax with sendmail
Technical Support wrote: Has anyone configured ast_fax (sending faxes via asterisk) with sendmail? The creation of rules to trap all numbers [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] seems too complicated. Does anyone have setup details to share? (I don't want to switch MTA's). As a workaround, I could launch the app automatically from sendmail using an alias like: fax:| /ast_fax That way sending to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] is easily handled by the sendmail program (without complex rules for numbers at the TO:) and launches the ast_fax app. The phone numbers to fax to could be entered into to subject line of the email. I looked at eps (address.c) and and ast_fax (ast_fax.c) and it looks like all that is needed is modifying address.c (or copy it so subject.c) to extract from the Subject line instead of the To line. Making this even more useful would be adding a parameter to the .call file which tells ast_fax to extract phone numbers from either the TO: line (default) or the SUBJECT line. I'm wondering if something like this has already been done? (I wouldn't want to reinvent the wheel) Alternatively, does someone have a working sendmail config to share? Thanks MD On my mail server, I added this to the virtusertable [EMAIL PROTECTED][EMAIL PROTECTED] Then on the Asterisk server, I put this in the virtusertable: [EMAIL PROTECTED]fax Then on the Asterisk server, put this in your aliases: fax: |/ast_fax The only drawback to this is the email address is formatted like this: [EMAIL PROTECTED] I could live with this as it actually makes for a more flexible solution. You can use an address like call+8005551234 and it do something different from faxing. Mark ___ --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] Is this normal?
This is off list... I was really concerned about this, too!! It turns out that it is some sort of clean up routine that runs once an hour. If you have calls in progress on channels 3 and 4, those won't show up as restarted!! Good Luck! Mark Matthew T. O'Connor wrote: Hey, I'm up and running fine with 30 Polycom 500s connected to Asterisk 1.2Beta on Cent OS 4.1 with a Digium TE110 connected to a PRI line. Nearly every hour, almost on the hour I get this: Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/1 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/2 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/3 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/4 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/5 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/6 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/7 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/8 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/9 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/10 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/11 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/12 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/13 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/14 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/15 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/16 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/17 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/18 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/19 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/20 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/21 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/22 successfully restarted on span 1 Sep 29 23:01:39 VERBOSE[3567] logger.c: -- B-channel 0/23 successfully restarted on span 1 Is this normal? Thanks, Matthew ___ --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] Cisco 7960 Locking Up
Ok... I asked a question a few months back about a 7960 that a user claims to be shocking her in her ear from time to time. A few others indicated they had similiar issues and alot of them seemed to stem from power over ethernet. Here's what we've done... We replaced the phone, ran two new cat5 cables to a different switch, put in a power brick and disabled power over ethernet. Over the last few months, the number of incidents of her getting shocked have reduced to almost never, but the phone is displaying the same symtoms as when she was getting the shock. The phone seems to lock up. We can not establish any type of pattern as to what causes it, but here's what we do know. She can be on a call and not touch any buttons. The soft keys will blank out and she loses audio as does the person on the other end. This has happened over both Zap and Sip channels. The strange thing is that if she waits about 20 seconds, the LCD panel will sort of flash and she gets the call back!! I never see anything in the CLI that makes me think Asterisk is even aware it is happening. I've done some research and I found some people have had issues with cell phone radiation locking up or rebooting a 7960. Has anyone else experienced this? We tried removing her cell phone from the room and it doesn't seem to make any difference. We do, however, have a cell phone repeater set up, but it's closer to alot of other users than her. Anyone have any suggestions on how to debug this? Is there some type of logging meter we can buy or rent that we could stick over there and monitor the environment for a week or so? As always, thanks for the help!! Mark ___ --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] chan_skinny issue
Jason wrote: Hey all, I have set up my cisco 30vip using chan_skinny because chan_sccp wont register. The problem I am having is, everytime a call is sent to the phone Skinny/[EMAIL PROTECTED] it rings once, then asterisk segfaults. Man... Use chan_sccp from Sergio at: ftp://ftp.berlios.de/pub/chan-sccp/ He is the most helpful person I've ever met. If you find a bug, report it to him, and it's usually fixed by the next day!! I don't have the same phone, but I've used 7910/40/60 with sccp and it works! Mark ___ 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] Cisco ATA and a PayPhone
Andres wrote: Help is on the way:) This is quite simple to achieve on Sipura units. There is a parameter called Dial Tone: [EMAIL PROTECTED],[EMAIL PROTECTED];10(*/0/1+2) It defines the frequencies and duration of the tone. The 10 you see near the end is the duration. Simply change it to 60 like this and you're done: Dial Tone: [EMAIL PROTECTED],[EMAIL PROTECTED];60(*/0/1+2) I just tried it and it works like you want it. I'm not the OP and do plan on deploying several spa3k's, is there somewhere this kind of stuff is documented for the spa's? The Sipura Admin guide covers also the spa3k. The Dial Tone parameter is the same for all SPAs. You can ask your reseller for the Admin guide if you don't have it. Cheers. These are great suggestions!! I will try them on Monday! Mark ___ 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] Cisco ATA and a PayPhone
Hi Mark, I've done this using SPA-2000, SPA-2000 can generate polarity reversal signal, The pay-phone detects call answer and hangup by revesal signal. also the pay-phone must be supported polarity reversal detection. http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800a6210.shtml http://michigantelephone.mi.org/distribute.html Cheers, ~Madhawa I tried the reverse polarity and it didn't seem to make a difference. Let me back up and take the payphone out of the equation. If hook up a house phone to the ATA and take it offhook, it get a busy signal after 10 seconds. I really need this to be more like 1 minute. Any ideas on how to do that? Mark ___ 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] Cisco ATA and a PayPhone
Hi Mark, I've done this using SPA-2000, SPA-2000 can generate polarity reversal signal, The pay-phone detects call answer and hangup by revesal signal. also the pay-phone must be supported polarity reversal detection. Anyone got any suggestions? I need to know what piece of hardware I need (ATA preferably) that allows me to pick up an analog phone, sit idle and not get the reorder tones for at least 1 minute. I am currently using a Cisco ATA-188 and I get them at 10 seconds. I've monkeyed with every single bit of the config file and can't seem to extend or disable it. HELP!! Mark ___ 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] Cisco ATA and a PayPhone
I have an interesting problem. I am attempting to install a payphone utilizing a Cisco ATA-188. The payphone actually works, but there are some timing issues. What happens is you pick up the payphone and the ATA grabs a line and goes offhook. While you monkey with putting money in and dialing the number, you are eating up the time before you get the offhook reorder tones (or howler tones I think). If you can put the money in and dial real fast, it works!! I have been screwing with the ATA configs for days now and can't come up with a way to extend the timeout or to even disable it. Anyone have any suggestions or could recommend another method? FX ports may be an option, but they are pretty far from where these phones are going to go. As always, thanks for any input!! Mark ___ 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] Cisco ATA and a PayPhone
Hi Mark, I think ATA-188 supports polarity reversal. Cheers, ~Madhawa I hope I don't sound stupid, but what does that mean? I can't find a definition for polarity reversal and how it would help me. I do see the 188 supports it, but I'm not sure what to do with it. Thanks!! Mark ___ 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] Setting Caller ID after Dial
Bryce Chidester wrote:
The CallerID that is seen by others on calls originating from your
PRI is set by your PRI provider; you have no control from Asterisk
about this as it gets overridden by the provider. You must contact
your carrier and ask them to set the CallerID for all PRI lines to
the desired name/number.
Regards,
Bryce Chidester
There must be different types of PRI lines because I was really shocked
when I started testing my Asterisk box on my PRI and the people
receiving the calls were flipping out because their caller id display
was showing my 3 digit SIP extensions. I wanted all outbound calls to
have the same callerid so I did it like this:
extensions.conf
[trunklocal]
exten = _6NX,1,SetCallerID(youroutboundnumber)
exten = _6NX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _6NX,3,Congestion
There was also a callerid option in zapata.conf, but I don't think it
had any affect for me. Good luck!!
Mark
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Re: [Asterisk-Users] Setting Caller ID after Dial
Chee Foong Chiew wrote: Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number (which is one of the number selected from the 200 DID numbers). This I can be achieved in asterisk by calling SetCallerID before Dial command. However in the CDR, the caller id number of the number that i set using SetCallerID is always logged and there is no trace of which sip extension is making the call since the caller is always the same. This has become a serious trouble for billing. I have been searching around and could not seems to get a solution. I have tried DIAL_STATUS variable (only work if call is not answered), using 'g' option in Dial command (does not work if calling party hangup first), etc. Is there a solution or work around for this? Thanks in advance CCF I forgot in my last post to mention that I use Postgres for my CDR, and the SIP extension can be pulled from the channel column. That way, the callerid is still the way it appeared when the calls were placed. I just strip everything from the '-' to the right and it's worked great for me! Mark ___ 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] Setting Caller ID after Dial
Chee Foong Chiew wrote: Hey Mark, Have you tested on doing transfer (blind and attended)? Are the extensions in the CDR still correct? CCF --- Mark Johnson [EMAIL PROTECTED] wrote: Actually, I don't think they are. That was something I wanted to research a little farther. I wish the CDR would show calls how they happened. Outside to the autoattendant. Transfer from attendant to extension. Transfer from extension to voicemail. I'm pretty sure I only see what the final result was. But I still have been able to figure out how the call went by the channel name. Mark ___ 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] MGCP Groups
I am looking into using a Cisco T1 device that uses MGCP. Asterisk is talking to it fine, but I am having a hard time figuring out how to handle channel grouping like Zap does. With Zap, I can take channels 1-23 and make a group g1 out of it and then simply dial Zap/g1. Does MGCP have this type of functionality? Everything I've tried points to no... Thanks! Mark ___ 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] Cisco 7750
I have read of people attempting to do this, and I just wanted everyone to know about what we've discovered about the Cisco 7750. If you don't know what it is, it's basically a blade server. I have 1 power blade, 1 alarm processor, 2 system processing engines and 1 multi-service route processor. We just got asterisk running on this today!!! We haven't tested the T1 with it, yet, but I pretty sure it will work OK. All of the FX ports work beautifully right now. The big deal about this for me is that I have battled over and over again with interrupt issues with Digium hardware. This is sweet because all the T1 processing including echo cancellation should be done on the route processor. Asterisk doesn't have to do much of anything. Thought you guys might want to know. I'll keep you posted as to how it works for us!! Mark ___ 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] Cisco 7750
Trey Scarborough wrote: - Original Message - From: Mark Johnson [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, June 21, 2005 8:56 AM Subject: [Asterisk-Users] Cisco 7750 I have read of people attempting to do this, and I just wanted everyone to know about what we've discovered about the Cisco 7750. If you don't know what it is, it's basically a blade server. I have 1 power blade, 1 alarm processor, 2 system processing engines and 1 multi-service route processor. We just got asterisk running on this today!!! Just dont let cisco know We haven't tested the T1 with it, yet, but I pretty sure it will work OK. All of the FX ports work beautifully right now. The big deal about this for me is that I have battled over and over again with interrupt issues with Digium hardware. This is sweet because all the T1 processing including echo cancellation should be done on the route processor. Asterisk doesn't have to do much of anything. so im guessing that all of the t1/fx ports are configured in the system processor and just talk sip/mgcp to the route proccessor. That sounds like a pretty sweet setup If you could only get cisco to sell you the hardware without having to buy the software. I'm seeing that these things are on E-Bay pretty often. They still want way too much money for what it is. But if you where trying to get away from Call Manger and already owned one... Mark ___ 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] DID Issue
I have a pretty strange problem. I have about 100 DID's that come down a PRI from SBC in the United States. On Friday afternoon, one of my DID's flipped out. When you call it, the SBC operator comes on and says that the line has been disconnected. I contacted them and they ran test and they are telling me the problem has to be on my end. My problem is that the CLI never shows the number as called. It seems to me it would show that ZAP channel ring and then say what it decided to with it. I've got nothing. I even shut the * box down and brought it back up, same problem... In the past, if I shut down a SIP device and you try to call the DID, I'm pretty sure you got a busy signal, not an SBC operator. Anyone have idea how to troubleshoot this one? I pretty sure it's a problem with the phone company, some type of translation issue. Mark ___ 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] DID Issue
Chris Coulthurst wrote: If you have a loopback plug, I would take that PRI down, unplug the NIU from the Asterisk box, and plug that RJ45 loopback plug in to the NIU, and call the telco, have them run a loop test on your circuit. Out here in Qwest-land they can usually get a tester on it and get results to you in less than an hour. Sounds to me like that problem is theirs, this would help prove it. Chris Coulthurst [EMAIL PROTECTED] After arguing with them for the last few days, they finally discovered the problem was on their end. They somehow lost the DID in the translation database. They simply added it back and it works. What upsets me is that they insisted my equipment was telling them it was an unlocated number. It's tough to argue with a large phone company that they are wrong!! Mark ___ 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] CallerID/chan_sccp
Joseph wrote:
On Thu, 2005-06-09 at 02:24 +1000, Julien Goodwin wrote:
On 8/06/2005 11:37 PM, Sergio Chersovani wrote:
Joseph ha scritto:
When sending a call to a line defined on chan_sccp, there is an
error on the console that says:
Jun 7 08:22:29 WARNING[3924]: sccp_channel.c:79
sccp_channel_send_callinfo: Incoming call SCCP/Line1-0008 doesn't
have CallerId name
Fixed, you can find the patch here
http://www.c-net.it/chan_sccp/
And this has been committed, should work through in about 5 hours
(thanks sourceforge)
It works.
Thanks.
I just downloaded the latest chan_sccp and am having problems with
internal to internal calls with callerid. I added a few debug lines to
the code to help sort it out, but here's what happens... Exten 581
calls 580. On the display 581 shows Unknown number to 580. On exten
580, the display shows Test Phone2 to Unknown number. Here are some
of the lines from the CLI including my added debug lines:
-- Set calledParty Name: Test Phone1 Number 580
-- Executing Dial(SCCP/581-0005, SCCP/580|15|Ttr) in new stack
SCCP trying to call SCCP, format 4, data, 580
-- --* 581
-- New channel context: office
-- Asterisk request to call: SCCP/580-0006
-- Set callingParty Name: Test Phone2 Number 581
== Sending Packet Type SetLampMessage (16 bytes)
== Sending Packet Type SetRingerMessage (8 bytes)
== {CallStateMessage} callState=RingIn(4), lineInstance=1, callReference=6
== Sending Packet Type CallStateMessage (28 bytes)
*** Calling Party Name: Test Phone2
*** Calling Party Number: 581
*** Called Party Name:
*** Called Party Number:
== Sending Packet Type CallInfoMessage (208 bytes)
== Sending Packet Type DisplayPromptStatusMessage (48 bytes)
== {SelectSoftKeysMessage} lineInstance=1 callReference=6
softKeySetIndex=3 validKeyMask=65535/65535
== Sending Packet Type SelectSoftKeysMessage (20 bytes)
-- Called 580
-- Asked to indicate '3' (Dialing) condition on channel
SCCP/581-0005
-- Current tone (36) is equiv to wanted tone (36). Ignoring.
== Sending Packet Type DisplayPromptStatusMessage (48 bytes)
== {CallStateMessage} callState=RingOut(3), lineInstance=1,
callReference=5
== Sending Packet Type CallStateMessage (28 bytes)
*** Calling Party Name:
*** Calling Party Number:
*** Called Party Name: Test Phone1
*** Called Party Number: 580
The lines beginning with *** are the debug lines I added inside the
sccp_channel_send_callinfo function. Any ideas?
Mark
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Re: [Asterisk-Users] CallerID/chan_sccp
Joseph wrote:
On Thu, 2005-06-09 at 11:57 -0400, Mark Johnson wrote:
I just downloaded the latest chan_sccp and am having problems with
internal to internal calls with callerid. I added a few debug lines to
the code to help sort it out, but here's what happens... Exten 581
calls 580. On the display 581 shows Unknown number to 580. On exten
580, the display shows Test Phone2 to Unknown number. Here are some
of the lines from the CLI including my added debug lines:
-- Set calledParty Name: Test Phone1 Number 580
-- Executing Dial(SCCP/581-0005, SCCP/580|15|Ttr) in new stack
SCCP trying to call SCCP, format 4, data, 580
-- --* 581
-- New channel context: office
-- Asterisk request to call: SCCP/580-0006
-- Set callingParty Name: Test Phone2 Number 581
== Sending Packet Type SetLampMessage (16 bytes)
== Sending Packet Type SetRingerMessage (8 bytes)
== {CallStateMessage} callState=RingIn(4), lineInstance=1, callReference=6
== Sending Packet Type CallStateMessage (28 bytes)
*** Calling Party Name: Test Phone2
*** Calling Party Number: 581
*** Called Party Name:
*** Called Party Number:
== Sending Packet Type CallInfoMessage (208 bytes)
== Sending Packet Type DisplayPromptStatusMessage (48 bytes)
== {SelectSoftKeysMessage} lineInstance=1 callReference=6
softKeySetIndex=3 validKeyMask=65535/65535
== Sending Packet Type SelectSoftKeysMessage (20 bytes)
-- Called 580
-- Asked to indicate '3' (Dialing) condition on channel
SCCP/581-0005
-- Current tone (36) is equiv to wanted tone (36). Ignoring.
== Sending Packet Type DisplayPromptStatusMessage (48 bytes)
== {CallStateMessage} callState=RingOut(3), lineInstance=1,
callReference=5
== Sending Packet Type CallStateMessage (28 bytes)
*** Calling Party Name:
*** Calling Party Number:
*** Called Party Name: Test Phone1
*** Called Party Number: 580
The lines beginning with *** are the debug lines I added inside the
sccp_channel_send_callinfo function. Any ideas?
Mark
Is this CVS-HEAD?
It seems to work fine on cvs head.
I think there where some changes in the current cvs head vs the stable
that may make stable caller id not work.
Cisco 7910's and CVS-HEAD from 06/03/05.
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[Asterisk-Users] Features.conf - atxfer
I am trying out the new atxfer feature from CVS-HEAD. I set atxfer equal to *7 and it seems to work OK. I am having a problem getting it to work the way a receptionist would want. If an extension calls me, I hit *7 and I hear the voice say transfer. I dial another extension. If the newly dialed extension goes to voicemail, I can't figure out how to get the original call back to tell them the person they are trying to reach is unavailable. Anything I try bridges the call and the caller go into like the 2nd half of the voicemail greeting. Is there some trick to this? Thanks! Mark ___ 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] Ring but now audio on answer
Garth Brown wrote: I have my Asterisk server all setup. But have an odd problem and hope someone here can help. I have a Polycom IP 300, a Grandstream GXP-2000, and an installation of X-Lite. They can each call each other just fine (extension-to-extension). I can also dial-in from the outside (via Broadvoice) and can leave and retrieve voicemails. When I set ANY of the extensions (clients mentioned above) to the default extension from the SIP provider, the phone rings and shows CID BUT, when I answer the phone, there is no audio either way. I thought this was a firewall issue but the clients ring and I CAN leave and retrieve voicemail. My next assumption is that it is some codec issue. The Polycom defaults to G.711u. Ive tried changing this to G.729AB but there problem persisted. Any ideas? Thanks in advance. I recently got a Polycom IP 300 and am having a similiar problem. I normally use Cisco 7940's and 60's but decided to try a less expensive phone. On the same LAN, the 300 can make calls just fine, and check voicemail. If a Cisco phone calls the Polycom, the phone rings, but when answered there is no audio either direction. The codecs appear to match up between the phones. The * CLI shows the Polycom as registered fine, also. I'll keep you posted if I figure something out. Mark ___ 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: Stange question...
Vikram Rangnekar wrote: static ! Get your carpets washed and use static guard on it. Thank you everyone for the replies. After doing some testing, it has been determined that it was the phone that was the cause of the user being shocked. We could relocate the phone, switch to a power brick and we could still get the cracking noises in the 7960. Thanks everyone!! Mark ___ 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] Stange question...
Ok, guys... Please be gentle with me. I have what is going to be the strangest question you will have ever heard, but I have no idea what to tell this person. I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My receptionist has told me on two different occasions that she tried to transfer a call by pressing #, and she heard a buzz noise in the phone and the phone then SHOCKED her in her ear. She wasn't able to do anything with the phone for a few seconds as the buttons didn't respond, then she could go back to picking up calls and whatnot. This is a Cisco 7960, SIP 7.4 on power over ethernet. I don't see how it would be possible for her to get physically shocked by the phone. Has anyone ever heard of this happening on any type of voip hardware? Mark ___ 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] Stange question...
Dan Austin wrote: Yup. I even suspected it was a 7960 before I got that far in your email. It hasn't happened to any of my users, but I heard about it at a Cisco users group meeting, from a number of people representing a different companies. Cisco was present and stumped, I have heard any more about it though. Dan This is interesting. I thought she had fallen off her rocker because she said the one today actually hurt, where the one before she couldn't tell if she got shocked or not. And to answer the last response, she is being nice about it, but I think I'm going to switch out her phone before it happens again!! Mark ___ 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] Stange question...
Eric Alexander wrote: Are you using POE from a 3550? We have had similar problems, upgrading the firmware on the switch has reduced the occurrences. The Cisco phones are not always nice in an environment with a lot of static electricity. POE is coming from a 3500XL I think. It just weird that this has never happened until I changed from Call Manager to Asterisk. I know this has to be a hardware issue but they are blaiming it on Asterisk... ___ 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] DEBUG output on sip extensions
Marty Mastera wrote: Can anyone help me to understand what the significance of this output is? May 17 10:50:23 DEBUG[2030]: Didn't get a frame from channel: SIP/105-1ae4 May 17 10:50:23 DEBUG[2030]: Bridge stops bridging channels SIP/105-1ae4 and SIP/outbound-7dc3 I searched for these phrases but am coming up short on what they really mean. I'm trying to investigate problems we are having with two separate asterisk installations both using Polycom IP-500 phones. These type of messages appear in the logs of both servers. It almost appears as though these messages are normal following completion of a call (a hangup), but we are troubleshooting bad audio in both locations and the wording of these messages doesn't appear benign. I am noticing these in my logs also. I looks like it is the result of the person hanging up, but I have had a few comlaints of dropped calls the last few days. These messages also appear at the times of the dropped calls. I have been watching CPU usage and it doesn't look like my machine was really loaded or anything. Mark ___ 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] Cisco contract for 7940/7960 firmware access
Mark Brown wrote: Hi Everyone! Is there any hope for us newbie plebs who want to also get hold of the updated Cisco firmware? I need to get a 7910G updated to work on SIP.. Any help on obtaining the updated firmware quickly and painlessly would be great J Cheers M 7910 does not have a SIP image and looks like it never will. I have about 40 of these stupid things that I can't get to work 100% with skinny or sccp. If you ever figure out how, be sure to let me know! Mark ___ 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] Cisco contract for 7940/7960 firmware access
Joseph wrote: On Tue, 2005-05-17 at 14:30 +0100, Mark Brown wrote: Thanks for that Mark... doesn't sound promising then :( 7910 does not have a SIP image and looks like it never will. I have about 40 of these stupid things that I can't get to work 100% with skinny or sccp. If you ever figure out how, be sure to let me know! Mark The sccp looks promising, if some more of us would work on chan_sccp maybe we could get somewhere. It would be good if it could be merged into the main cvs tree and have a good bug tracker on it. Mark, have you tried the latest chan_sccp? Yeah... It has good features but I get deadlocks with the phones and it crashes the whole phone system. I am sending the developer a 7910 but I had an emergency * install and I had to get the 7910's working immediately. Skinny has been completely stable for me, but alot of the features are missing/or broken. I am attempting to build a 2nd * server so I can do more testing and mail results to the right people so maybe we can get sccp working better with the 7910's. ___ 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] VoipSupply.com
Manjit Riat wrote: I am going to buy some IP phones from them but I sent them an email couple of weeks ago and got no reply. Has anyone ordered anything from them? Any other places that I can buy from? Sorry if its a wrong post. I have ordered from them with their web shopping cart and it went very well. Got emails explaining when things were shipped and when to expect them. I would recommend! Mark ___ 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] Zaptel and zttest
I am having trouble with zttest on a Tyan board, dual AMD Opteron's on
FC3. Tried different kernels, no IRQ sharing, everything looks in
order. My zaptel modules load fine, but if I run zttest, it just
hangs. Below is the strace output and you can see where it stops.
Anyone have any ideas?
[EMAIL PROTECTED] zaptel]# strace ./zttest
execve(./zttest, [./zttest], [/* 24 vars */]) = 0
uname({sys=Linux, node=asterisktest.astroshapes.com, ...}) = 0
brk(0) = 0x502000
mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0)
= 0x2aac
access(/etc/ld.so.preload, R_OK) = -1 ENOENT (No such file or
directory)
open(/etc/ld.so.cache, O_RDONLY) = 3
fstat(3, {st_mode=S_IFREG|0644, st_size=150975, ...}) = 0
mmap(NULL, 150975, PROT_READ, MAP_PRIVATE, 3, 0) = 0x2aac1000
close(3)= 0
open(/lib64/tls/libc.so.6, O_RDONLY) = 3# ./zttest
Opened pseudo zap interface, measuring accuracy...
--- Results after 0 passes ---
Best: 0.00 -- Worst: 100.00 -- Average: 100.00
read(3, \177ELF\2\1\1\0\0\0\0\0\0\0\0\0\3\0\0\1\0\0\0p\305\1\0...,
640) = 640
lseek(3, 624, SEEK_SET) = 624
read(3, \4\0\0\0\20\0\0\0\1\0\0\0GNU\0\0\0\0\0\2\0\0\0\4\0\0\0..., 32)
= 32
fstat(3, {st_mode=S_IFREG|0755, st_size=1605832, ...}) = 0
mmap(NULL, 2297832, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 3,
0) = 0x2abc1000
mprotect(0x2ace9000, 1085416, PROT_NONE) = 0
mmap(0x2adc1000, 184320, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x10) = 0x2adc1000
mmap(0x2adee000, 16360, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x2adee000
close(3)= 0
mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0)
= 0x2adf2000
mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0)
= 0x2adf3000
mprotect(0x2ade8000, 16384, PROT_READ) = 0
mprotect(0x2abbf000, 4096, PROT_READ) = 0
arch_prctl(0x1002, 0x2adf2b00) = 0
munmap(0x2aac1000, 150975) = 0
open(/dev/zap/pseudo, O_RDWR) = 3
fstat(1, {st_mode=S_IFCHR|0620, st_rdev=makedev(136, 1), ...}) = 0
mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0)
= 0x2aac1000
write(1, Opened pseudo zap interface, mea..., 51Opened pseudo zap
interface, measuring accuracy...
) = 51
rt_sigaction(SIGHUP, {0x400910, [HUP], SA_RESTART|0x400}, {SIG_DFL},
8) = 0
rt_sigaction(SIGINT, {0x400910, [INT], SA_RESTART|0x400}, {SIG_DFL},
8) = 0
read(3,
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Re: [Asterisk-Users] Something every TDMP user should know
Damian Funnell wrote: 1. Check that the TDMP is on it's own IRQ (much to our embarrassment our card wasn't at the time, so we had to play with it a bit to get it to occupy a unique IRQ). 2. Disable hyper threading on the Xeon CPU. 3. Uninstall our SCSI hardware and replace it with IDE hardware. 4. Upgrade to the latest stable releases of Asterisk, Zaptel and Libpri. We made changes 1 and 2 in the above list and are prepared to make changes 3 and 4 if we find the problem hasn't gone away. It hasn't happened in over two weeks now (after occuring many times per day for a while), so we hopefully won't have to throw out our SCSI hardware. After we made each change (1 and 2 were made about two weeks apart from each other) we found that the quality improved, with the incidence of the issue halving after '1' and disappearing (hopefully for good) after '2'. Incidentally the results of zttest *did not* noticeably improve after making these changes (it is still below 99.98%). This is great info. I am running on an Intel box and attempting to go to a dual AMD Opteron setup on a Tyan board. I am not having luck luck getting my numbers above 99.6%. I've disabled every hardware gadget and service not needed and still haven't had much luck. I'm going to try a custom kernel as opposed to the stock one's I've tried, but that's been about 4 different OS's with the same results. Is there something to disable on Opteron's that would be the equivalent of disabling hyperthreading? Oh, and I even tried setting the pci latencies and it made no noticeable difference. Mark ___ 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 crashes
Can someone please help me. I am currently HEAD as of about 5 days ago (stable was giving me all sort of problems, upgraded per other users suggestions) on an Intel mainboard using a mix of Cisco 7960/40 SIP and 7910 SCCP. Can someone please explain what the following means? When this happens, I am about 1 minute from Asterisk going downhill. All of the SCCP phones quit, while the SIP phones can do calling to some degree. I get kicked out of any consoles and can't reconnect without restarting asterisk. Mark May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 WARNING[28400] channel.c: Avoided deadlock for 'SCCP/118-001a', 10 retries! ___ 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] CVS question
Is there a way to get a download of asterisk from cvs-head as of like 3 weeks ago? Having some weird problems and most people say that alot of these things have been introduced over the last few weeks. Mark ___ 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] HINT
Sean Kennedy wrote: Anton Krall wrote: Guys, what does hint do in a dialplan and how do you use it? I have been trying to figure this out for a while now, even posted a question on the list, to which no one replied. Any details would be apprecaited if you find this one out. I want to use it, but I don't know how. Sean I haven't used it, but I think it's used to notify another phone whether you are busy or not. The Cisco 7914 would be an example. If you set up extensions on it, they would light red when that extension is on the phone. This sound right? Mark ___ 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] PRI timing problems: Fax Voice
Andrew Kohlsmith wrote: On May 4, 2005 12:05 pm, Matthew Boehm wrote: May 4 10:57:04 WARNING[25650]: chan_zap.c:4394 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 2 - audio may have been lost I think that something in asterisk (not zaptel) changed in the last week to create this problem; see my last message to -dev. I also ugraded to head within the last few days and am noticing the same message. I haven't seen or heard of any problems with audio quality. Is this something to be concerned about? Mark ___ 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] Chan_sccp - status
Steve Hanselman wrote: I think it's displaying the name of the line that the call is coming in on, but you're expecting the name of the calling party (as I was!) Steve I looked and there was a change in the sccp.conf file for head with the addition of cid_num and cid_name. I am going to test that tonight. In the newest chan_sccp, is the MWI supposed to work? It's not for me. Here is a sample frmm my sccp.conf: [SEP000XEX9X0X2X] description = First Last type = 7910 context = office autologin = 100 [100] id = 100 pin = label = First Last description = First Last context = office callwaiting = 1 mailbox = 100 cid_num = 100 cid_name= First Last ___ 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] SCCP and channel question
I am running Asterisk HEAD and the latest mayday version of chan_sccp. Everything is going fairly smooth but every once in a while I get a 7910 to lock up. If I do a show channels in the CLI, I get the following and it never goes away. While this is happening, the phone can not be reached. The only way to release it is to stop * and restart it. Is there a way to do a magic kill channel SCCP/227-0007 and fix this? I've poked around and can't seem to find a way to help this. SCCP/227-0007 (floor s1 ) Dialing (None) (None) Mark ___ 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] SIP and CVS Head
I upgraded to CVS Head last night to help fix my SCCP problems and now my SIP installation is having issues. If I restart Asterisk, my SIP phones may take up to an hour to register correctly so I can place calls to them. They immediately go to voicemail as being busy. If I do a sip reload I get: -- Got SIP response 400 Bad Request back from xxx.xxx.xxx.1 -- Got SIP response 400 Bad Request back from xxx.xxx.xxx.2 -- Got SIP response 400 Bad Request back from xxx.xxx.xxx.3 -- Got SIP response 400 Bad Request back from xxx.xxx.xxx.4 -- snip -- Here is some sip debug info: Answering/Requesting with root capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 12 lines Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060: INVITE sip SIP/2.0 Via: SIP/2.0/UDP 10.1.1.2:5060;branch=z9hG4bK3e4409aa From: First Last sip:[EMAIL PROTECTED];tag=as77dd1f77 To: sip Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 03 May 2005 06:42:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 255 v=0 o=root 13863 13863 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 12338 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 122 asterisk*CLI -- SIP read from xxx.xxx.xxx.xxx:50634: SIP/2.0 400 Bad Request Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3e4409aa From: First Last sip:@xxx.xxx.xxx.xxx;tag=as77dd1f77 To: sip Call-ID: [EMAIL PROTECTED] Date: Tue, 03 May 2005 06:42:50 GMT CSeq: 102 INVITE Content-Length: 0 --- (8 headers 0 lines)--- -- Got SIP response 400 Bad Request back from xxx.xxx.xxx.xxx Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060: ACK sip SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3e4409aa From: Craig Deering sip:[EMAIL PROTECTED];tag=as77dd1f77 To: sip Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/123-3428 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) HELP! Mark ___ 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] Chan_sccp - status
Julien Goodwin wrote: Then why haven't you sent a backtrace? If I can see why it's crashing then I can fix it. Thanks, Julien chan_sccp project lead The general consensus was that I needed to be running HEAD to make this work properly. I upraded last night to HEAD and my SCCP stuff seems to working perfect!! Thank you!! Also, I saw you are in need of a 7910 from your announcement. If you email me your shipping info offlist, I will make sure you get a couple of them. Mark ___ 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] Chan_sccp - status
Mark Johnson wrote: Julien Goodwin wrote: Then why haven't you sent a backtrace? If I can see why it's crashing then I can fix it. Thanks, Julien chan_sccp project lead The general consensus was that I needed to be running HEAD to make this work properly. I upraded last night to HEAD and my SCCP stuff seems to working perfect!! Thank you!! Also, I saw you are in need of a 7910 from your announcement. If you email me your shipping info offlist, I will make sure you get a couple of them. Mark While on the topic, I'm having some weird issues with the 7910's and the callerid. I got them to display the outgoing calls correctly, but if I call from an internal SIP phone to an internal SCCP 7910, the display shows that that SCCP phone is calling itself until you answer. After you pick up, it changes to read Unknown Number to sccp ext# Anyone have luck getting this to work? Mark ___ 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] SIP and CVS Head
Mark Johnson wrote: I upgraded to CVS Head last night to help fix my SCCP problems and now my SIP installation is having issues. If I restart Asterisk, my SIP phones may take up to an hour to register correctly so I can place calls to them. They immediately go to voicemail as being busy. If I do a sip reload I get: -- Got SIP response 400 Bad Request back from xxx.xxx.xxx.1 -- Got SIP response 400 Bad Request back from xxx.xxx.xxx.2 -- Got SIP response 400 Bad Request back from xxx.xxx.xxx.3 -- Got SIP response 400 Bad Request back from xxx.xxx.xxx.4 -- snip -- Here is some sip debug info: Answering/Requesting with root capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 12 lines Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060: INVITE sip SIP/2.0 Via: SIP/2.0/UDP 10.1.1.2:5060;branch=z9hG4bK3e4409aa From: First Last sip:[EMAIL PROTECTED];tag=as77dd1f77 To: sip Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 03 May 2005 06:42:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 255 v=0 o=root 13863 13863 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 12338 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 122 asterisk*CLI -- SIP read from xxx.xxx.xxx.xxx:50634: SIP/2.0 400 Bad Request Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3e4409aa From: First Last sip:@xxx.xxx.xxx.xxx;tag=as77dd1f77 To: sip Call-ID: [EMAIL PROTECTED] Date: Tue, 03 May 2005 06:42:50 GMT CSeq: 102 INVITE Content-Length: 0 --- (8 headers 0 lines)--- -- Got SIP response 400 Bad Request back from xxx.xxx.xxx.xxx Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060: ACK sip SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3e4409aa From: Craig Deering sip:[EMAIL PROTECTED];tag=as77dd1f77 To: sip Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/123-3428 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) HELP! Mark Anyone?? This is killing me!!! Mark ___ 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] A good SIP receptionist phone
Adam Goryachev wrote: The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model) phones can all do what he wants. ie, have multiple lines with blinking red lights when a call arrives on that line. The polycom ip600 and cisco 7960 both have 6 lines available. Regards, Adam I am currently having the same problem with our receptionist. We use 7960's, which I really like. The problem with it is that when you are trying to manage 6 lines with it, it has a tendancy to make you mess up. Example, you are talking on line 3 and about to transfer the call or put them hold when line 4 rings. The SIP image will move to line 4 and you inadvertantly answer line 4 instead of transfering line 3. It would be nice if it would stay on the current button and let you select the line you want as opposed to it just jumping around to whatever the newest call happens to be. The Skinny image was a little better in this respect. Mark ___ 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] Chan_sccp - status
Joseph wrote: The cisco 7960 works well with * and SIP. Out of curiosity I loaded the ccm version 7.1 and tested it briefly with CVS HEAD * and latest chan_sccp. The interface when using ccm load on the phone is certainly different. Things I don't see how to fix are: o Setting the date and time on the phone o The vm button makes a msg on * saying VM Button is not yet handled o When on a call there is no transfer button. This must be something the chan hast to tell it display Or not? o It seems like the * console is very busy with messages constantly on it. This likely means more processor power needed for large #s of these phones. Just a thot. Someone may have some real life experience. o I don't see any way of making * read changes to sccp.conf. Tried a * reload. And a module reload. But had to stop * completely to get it to reread the config change. o The phone wants DISTINCTIVERINGLIST.XML. What does that look like? Is anyone using them in real life? The wiki seems to have little information. Like how to setup the ring tone file, the locale etc. Thoughts? I have not had much luck with chan_sccp, yet. I have about 40 7910's and everytime I try them, Asterisk crashes after about 15 minutes. Skinny seems to work a little better, but it's missing some features like WMI and the phone does give a ringing status when calling out on it. I tried the newest version of chan_sccp today and it also bombed on me. It seems like once a get a few of the following in the log file is when it chokes: ERROR[19583]: Erp, tried to hangup when we didn't have an active channel?! Running this on 1.0.7. Any else got chan_sccp working? Mark ___ 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] Chan_sccp - status
Joseph wrote: What if you run it on HEAD? I've been scared to try it. I just went live with this last week. Everything is great except the 7910's. I'm downloading HEAD as we speak. Anything to be aware of or look out for? Mark ___ 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] 7910 and Skinny
I just had a very successful installation of Asterisk and have a
question. On my 7910's using the Skinny protocol, the user does not
hear ringing when they make another call. I found a patch that makes
the ringing work, but something is still wrong with it. If I use the
7910 to make internal Skinny to other internal Skinny or SIP phones, the
ringing works. Once they make an outside call, they can not hear
ringing again until I shutdown Asterisk and start it back up. I'm using
1.0.7. Anyone have any ideas? I also tried chan_sccp and that was a
real disaster. Asterisk kept crashing after a period of about 30
minutes. It was like when the phones reregistered so many times, it
started claiming that some of the phones were dead and that others
couldn't be registered because they already were, then it crashed.
Anyone have any ideas? Below is the patch code I found.
Mark
/@@ -1715,14 +1756,17 @@
}
switch(ind) {
case AST_CONTROL_RINGING:
- if (ast-_state == AST_STATE_RINGING) {
+ ast_verbose(VERBOSE_PREFIX_3 State AST_CONTROL_RINGINGn);
+ // if (ast-_state == AST_STATE_RINGING) {
+ ast_verbose(VERBOSE_PREFIX_3 State AST_STATE_RINGINGn);
if (!sub-progress) {
transmit_tone(s, SKINNY_ALERT);
transmit_callstate(s, l-instance, SKINNY_RINGOU
T, sub-callid);
sub-ringing = 1;
+ ast_verbose(VERBOSE_PREFIX_3 Started Ringingn
);
break;
}
- }
+ // }
/
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Re: [Asterisk-Users] Static and echo on PRI
Michael Welter wrote: Do SIP-SIP calls have static? If you don't have SIP phone then you can use X-lite. Arrange you dial plan so an incoming PSTN call can call an outside number--from outside dial your system and then make an outside call. This call will be bridged on the Digium card. Do you get static? If not then it's not the PRI. 2ยข Well... Per everyone's advice I changed the motherboard (still Intel for now. Ordering an AMD of some sort) and the static WENT AWAY!!! I am still finding it amazing that I could go to a slower, crappier different Intel board and the problems go away. Thank you everyone for all of your help Mark ___ 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] Static and echo on PRI
Jorge Mendoza wrote: Mark, Could you please post the models of your first and second mobo? Thanks The first, that didn't work correctly the the TE400P was an Asus P4 2.4 Ghz. The model that does work correctly is an AOpen P4 2.0 Ghz. ___ 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] Static and echo on PRI
Michael Welter wrote: Mark Johnson wrote: Michael Welter wrote: Try 'vmstat 1'--are you getting 40% system utilization every n seconds? If so, unload the wcfxo and wcfxs modules and test again. Does anyone have some suggestions on how to get rid of this static on my Digium card? I am supposed to go live tomorrow night and will get shot if it's like this!! Mark ___ 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] Static and echo on PRI
Andrew Kohlsmith wrote: On April 26, 2005 06:19 pm, Mark Johnson wrote: Does anyone have some suggestions on how to get rid of this static on my Digium card? I am supposed to go live tomorrow night and will get shot if it's like this!! Lack of planning on your part does not constitute an emergency on our part. There were a number of suggestions given to you over the past week or so and a great number of them (including some given by myself) have gone unanswered. Perhaps you should read over this thread and make sure you haven't missed anything. -A. Um... If you read my orginal post, this was unplanned as I had a Cisco hardware failure. I have been working on building Asterisk for over 6 months and don't have the luxury of forking out over $5,000 for a test T1. I also have noticed that in looking through this particular thread that I have never seen your name in it. Just double checked the archives and, nope, you aren't there... I have tried every suggestion and replied my results. If you don't have any facts to share, please don't bother. I am desperate and don't have alot of time left and am begging for the list's advice. I left probably the largest post this month with EXACTLY what I have tried, the results, debug information, etc... I have removed drivers, swapped cards, changed IRQ's... I am open to any suggestions. If you tell me to go buy a different card, I will do that. You guys know more about than I do. What do you suggest, exactly? Mark ___ 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] Static and echo on PRI
Andrew Kohlsmith wrote: Try these things: Software: - don't play with gains on PRI or T1 unless you have echo or too loud/quiet. Static isn't caused by screwy gains and on digital circuits it technically shouldn't ever need to be adjusted - turn echocancel off for now - I notice you've got span=1,0,0 -- if you're talking to the telco make sure you're synchronizing the clock to them. Use span=1,1,0. - remove all modules except those absolutely necessary - Have you tried span 2, 3 or 4 instead of 1? Also is this a *stock* kernel or some distro-enhanced version? Grab a stock kernel of the same version from ftp.kernel.org. Finally, don't use the agressive canceller unless you REALLY can't get rid of it any other way (I seem to have very good performance with MARK2, using the MMX-friendly implementation (zconfig.h) and making sure my CFLAGS for the zaptel code was optimized for my processor (-march=pentium4). Also see http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html. Hardware: - *remove* the TDM22P from the system. Don't just unload the modules. - pull the TE405P out and put it in another (not same motherboard) system. I've seen this clean things up several times. Wetware: It's getting a little late for this now, but you paid for support from Digium when you bought the card; You might want to give them a call. Unfortunately I don't think this is an issue they will be able to solve over the phone, and their likely recommendation would be to replace the system. I'd love to know what they do find, if you try this route. Again, my apologies, for blasting you; I had you mixed up with someone else. -A. This is perfect stuff!!! Thank you!! I actually pulled the TDM22P today, removed all of those drivers and get the same results. I have built another box and am installing asterisk as we speak. I tried the span=1,1,0 with the same results and have been running that line for a day now. What I find strange is this... If I speak at a normal tone, it sounds OK. I still get static noise when the other person speaks. If I talk louder, I start to get what sounds like a partial echo. If I yell, I get a definite echo. Have not tried a different slot on the quad, will try that tomorrow. When monkeying with the echo cancel, I never really noticed a difference. I would even reboot the machine between changes to see if it made a difference. I am running this on Fedora Core 1. I will try any OS you recommend, but I have always had great luck with RH type distro's. I keep 400 and 500 day uptimes on those machines and they run many, many services. Uptimes would be higher but it seems whenever I find a good place to work, they close up or I move. Admittedly, I don't use RPM's for the core services, I typically compile those myself. I also shut down every module and service I don't need. I did alot of reading and it seemed like Digium cards were the real deal and I also found many users that had luck with the same setup. Should I try a different approach/OS/system? Mark ___ 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] Static and echo on PRI
Matt Klein wrote: ask your upstream. Not sure what you mean. This T1 is in good working order with a different system. Do you mean call the telco or Digium? Mark ___ 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] Static and echo on PRI
Michael Welter wrote: Do SIP-SIP calls have static? If you don't have SIP phone then you can use X-lite. Arrange you dial plan so an incoming PSTN call can call an outside number--from outside dial your system and then make an outside call. This call will be bridged on the Digium card. Do you get static? If not then it's not the PRI. I use Cisco 7940's and 60's. SIP to SIP calls are better than perfect. I also had good luck with my TDM22B, no echo and no static (although it was chewing up the processor, noticed per your advice). Will attempt that and let you know! Mark ___ 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] Static and echo on PRI
Michael Welter wrote: Mark Johnson wrote: I tested and I do in fact get from 40-50% system util every 5 seconds or so. After removing the wctdm module, the system util drops to 0 and stays there. I have not loaded the wcfxs and wcfxo modules because I could never get them to work right. I instead load the wctdm and it has seemed to work fine. I only need to make the fx port to the paging system work and the others can stay idle. What modules and order so you suggest. Here is what I load in this order: wct4xxp wctdm Do you still have the static on the PRI without the TDM modules? I finally got to test... Removing the tdm module makes no difference in the static. I still hear it for any incoming sound. Removing it does, however clean up the CPU usage but quite a bit. One odd thing was with the tdm module removed, it seemed to introduce a little delay in the conversation. I also tried to recompile the zaptel drivers with the aggresive cancellation. This seems to made a HUGE improvement to my echo problem. Any ideas? Mark ___ 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] Static and echo on PRI
I need some serious help!! I have been in the process of building an
Asterisk system to replace a Cisco Call Manager. I have most everything
setup, but only got to test the PRI today. To make a long story short,
my Call Manager is half broken and I need to go live with * a lot sooner
than I expected.
Here's where I am and what I tried. I am using all Cisco phones, mostly
7940's and 60's in a SIP configuration. All internal calls work with no
issues. I have a TE405P for the PRI and a TDM22B for my paging system
and whatnot. I am currently only using one PRI on the quad card. When
calling out on the PRI, I am getting static and some echoing. I have
tried various orders and values for the txand rxgains, echocancellation
and nothing seems to help. I get the staticy noise only when sound is
coming in, like when the other is ringing or when the other person is
talking. Complete silence the rest of the time. I get different
amounts of echo when calling out, the person on the other end says they
hear no echo or static at all, just on the SIP phones. I made sure that
I have no IRQ conflicts (output below) and my CPU usage seems to be
fine, plenty of horsepower remaining.
Here are the parts of my configs that I feel are relavent:
/etc/zaptel.conf
---
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone = us
defaultzone=us
fxoks=97
fxoks=98
fxsks=99
fxsks=100
/etc/asterisk/zapata.conf
--
[trunkgroups]
[channels]
context=incoming
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
callgroup=1
pickupgroup=1
immediate=no
callerid=xx
rxgain=0.0
txgain=0.0
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
;echotraining=800
switchtype = national
signalling = pri_cpe
group = 1
channel = 1-23
signalling=fxo_ks
group = 2
channel = 97
signalling=fxo_ks
group = 3
channel = 98
signalling=fxs_ks
group = 4
channel = 99
signalling=fxs_ks
group = 5
channel = 100
/etc/asterisk/extensions.conf
---
TRUNK=Zap/g1
TRUNKMSD=1
[trunklocal]
exten = _6NX,1,SetCallerID(xx)
exten = _6NX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _6NX,3,Congestion
Here are a few lines from the logs that might mean something to someone:
Apr 24 18:22:40 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:22:40 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:22:41 NOTICE[196620]: PRI got event: 8 on Primary D-channel of
span 1
Apr 24 18:22:41 NOTICE[196620]: PRI got event: 8 on Primary D-channel of
span 1
Apr 24 18:24:31 NOTICE[196620]: PRI got event: 8 on Primary D-channel of
span 1
Apr 24 18:25:53 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:25:54 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:26:09 NOTICE[196620]: PRI got event: 8 on Primary D-channel of
span 1
Apr 24 18:26:09 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:26:09 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:26:10 WARNING[196620]: PRI: !! Got reject for frame 51, but we
only have others!
Apr 24 18:26:10 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:26:11 NOTICE[196620]: PRI got event: 8 on Primary D-channel of
span 1
Apr 24 18:27:01 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:27:41 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
** I tried the line span=1,0,0,esf,b8zs in my zaptel.conf and made no
difference.
Here is a debug section for my PRI when I was getting static and echo:
Enabled debugging on span 1
-- Executing SetCallerID(SIP/226-9fca, 3307551414) in new stack
-- Executing Dial(SIP/226-9fca, Zap/g1/3305596313) in new stack
-- Making new call for cr 32771
Protocol Discriminator: Q.931 (8) len=46
Call Ref: len= 2 (reference 3/0x3) (Originator)
Message type: SETUP (5)
[04 03 80 90 a2]
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: u-Law (34)
[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 ]
[1e 02 80 83]
Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0 Location: User (0)
Ext: 1 Progress Description: Calling
equipment is non-ISDN. (3) ]
[6c 0c 21 80 33 33 30 37 35 35 31 34 31 34]
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Re: [Asterisk-Users] Static and echo on PRI
Michael Welter wrote: Try 'vmstat 1'--are you getting 40% system utilization every n seconds? If so, unload the wcfxo and wcfxs modules and test again. I tested and I do in fact get from 40-50% system util every 5 seconds or so. After removing the wctdm module, the system util drops to 0 and stays there. I have not loaded the wcfxs and wcfxo modules because I could never get them to work right. I instead load the wctdm and it has seemed to work fine. I only need to make the fx port to the paging system work and the others can stay idle. What modules and order so you suggest. Here is what I load in this order: wct4xxp wctdm Thanks! Mark ___ 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] Cisco 7940, Voicemail DTMF
Derek Conniffe wrote: Would anyone know why Voicemail in * doesn't get the DTML keypresses from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to do with dtmf_avt_payload: 101 setting in SIPDefault.cnf in the tftp server? Thanks for any help! Derek I have the same line in my SIPDefault.cnf and my 7940's and 60's work OK using the same POS version as you. I don't have any suggestions. Sorry. ___ 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] How to grab CallerId information
I am building a click to dial and CRM type web page and I'm having
trouble with something. I can make everything in the manager api work
as documented, but I can't seem to get a grip on how to tell what the
callerid is of an active call. Example: I know that on phone SIP/101
that there is an active call that originated from the outside. What's
the best way to get the callerid of that call?
I have attempted to put the callerid into the database with DBPut during
the initial call setup, but I don't really know that the call is
active. I can get the last busy and last unanswered callerid using
${DIALSTATUS}, but not the last or current answered. Anyone have any ideas?
Here's what I want to do (not using the Flash Operator Panel). If a
salesrep is on the phone, I want them to click a link on a webpage that
will open up a window with all of the customer information they would
need, based on the callerid of the active call. I already have a really
nice click to dial application and don't want a separate app. I also
don't want to monitor all of the time like the Flash Operator Panel
does. Anyone?
Thanks!
Mark
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[Asterisk-Users] Cisco 7960 Message Light on multiple phones
Here is what I am attempting to do (which works well on Cisco Call Manger). I have some 7960's that have multiple lines on them. The second line specifically is a helpdesk line that is shared among multiple phones. Here is how I am making that line ring on multiple phones, maybe you have other suggestions: exten = 135,1,Dial(SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED],20,rt) So this rings the second line on the phones that have the first line as 100 and 101. This works great. When someone leaves a voicemail, the messagelight will only light on the phone that was booted up last. Is there a way to make the light come on all of the helpdesk phones, with the second line icon displaying the correct mail icon? Here is the sip.conf section for those particular extensions: [100] type=friend username=100 secret=100 host=dynamic mailbox=100 linelabel=First Last line = 102 [135] type=friend username=135 secret=135 host=dynamic mailbox=135 linelabel=HelpDesk line = 135 [101] type=friend username=101 secret=101 host=dynamic mailbox=101 linelabel=First1 Last1 callerid=First1 Last1 101 line = 101 ___ 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] Cisco 7960 Message Light on multiple phones
Rich Adamson wrote: Here is what I am attempting to do (which works well on Cisco Call Manger). I have some 7960's that have multiple lines on them. The second line specifically is a helpdesk line that is shared among multiple phones. Here is how I am making that line ring on multiple phones, maybe you have other suggestions: exten = 135,1,Dial(SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED],20,rt) So this rings the second line on the phones that have the first line as 100 and 101. This works great. When someone leaves a voicemail, the messagelight will only light on the phone that was booted up last. Is there a way to make the light come on all of the helpdesk phones, with the second line icon displaying the correct mail icon? I believe you'll find the phone that registered 'last' will be the one that gets the vm lite (not the last reboot). If your phones re-register ever 3600 seconds, the last one gets the mwi indicator and that will cause the mwi to move between phones over time. (Snom phones had a similar problem some time ago.) I believe the current implementation for vm notification is to use a sip 'notify' message to turn on the mwi, and the sip protocol implementation within * does not support sending 'notify' messages to multiple phones. (E.g., how would * even know how many phones you are trying to ring via the above dialplan entry?) I was hoping that asterisk would be able to sort that out. The neatest part about this setup is that this shared extension can have multiple calls going on. Example: on Cisco Call Manger if you have a shared extension between three phones and someone picks up the line, none of the other phones can use that extension. With SIP, If the same person picks up the line, so can the other two people. The message light working on all of the phones would be great! ___ 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] Cisco 7960 Message Light on multiple phones
Adi Linden wrote: I believe the current implementation for vm notification is to use a sip 'notify' message to turn on the mwi, and the sip protocol implementation within * does not support sending 'notify' messages to multiple phones. (E.g., how would * even know how many phones you are trying to ring via the above dialplan entry?) This is interesting because I am doing a very similar thing. I have four Cisco phone, two 7940 and two 7905 and a couple of ata186. An incoming call rings all six phones. There is a single voicemail box that is assicate with every phone. The MWI indicator lights up on all phones when a message is received. It also extiguishes from all phones if the voicemail is deleted from any phone. Adi ___ Could you describe how you do that! That's exactly what I am trying to do! ___ 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] Cisco 7960 Message Light on multiple phones
Brian M. Arlinghaus wrote: I've got 25 7960s with different mailboxes set for different lines. The MWI indicator (red light) comes on if there are messages in either of the mailboxes. However, on the display, an envelope shows up next to the line that has the voicemail waiting. Therefore I can tell which line has the voicemail. In my extensions.conf, I have a dial command such as exten = 8900,1,Dial(SIP/89XX-3SIP/89XX-3SIP/89XX-3,,). All of the phones ring at the same time. All of the message waiting indicators will light if someone leaves a message and all of them will go out when the the messages are removed. What does the @ do? Original Example: exten = 135,1,Dial(SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED],20,rt) Brian [89XX-1] type=friend host=dynamic secret=89XX-1 context=local callerid=NAMEXX 859-392-89XX mailbox=89XX;= Mailbox Designation disallow=all allow=ulaw qualify=yes [89XX-2] type=friend host=dynamic secret=89XX-2 context=local callerid=NAMEXX 859-392-89XX disallow=all allow=ulaw [89XX-3] type=friend host=dynamic secret=89XX-3 context=local callerid=NAMEXX 859-392-8900 mailbox=8900; Mailbox Designation disallow=all allow=ulaw I got it working following the above layout. I was attempting to use [135] for all of the phones and that won't work. As in the example above, I used [135-1] and [135-2], setup the tftp config files in the same manner, and modified the exensions.conf file to look like: exten = 135,1,Dial(SIP/135-1SIP/135-2,20,rt) I works great!! Thanks! ___ 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] Cisco 7940/7960
Doug Lytle wrote: Mark Johnson wrote: This may be OT, but I can't seem to find how to do this. I have 7940/7960's with Skinny on them. When you start pressing numbers on the dialpad, you start building a number to dial. When I install SIP, that functionality goes away. You have to hit the speaker button, or lift the handset before you can start dialing. Is there a setting I am missing, or is this just a product of SIP and I have to live with? Mark, I just got a 7940(eBay) and put the 7.3 SIP image on it. To dial, I can either start dialing to build the number and press either the # key to initiate the dial or presss the dial option on the lcd panel. Doug I also have loaded POS3-07-3-00 and hitting any numbers does nothing. I am using the default dialplan.xml file and a really basic SIPxxx.cnf file. This is the same on a couple of phones I am trying. Any ideas? ___ 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] Configuring VLAN takes ages
Asterisk wrote: when booting the cisco 7960 with SIP image 7.3, the Configuring VLAN takes in order of minutes before it issues a DHCP request . Does anyone else have this problem - is there any way of disabling the VLAN configuration at all ? We are not using Cisco switches. Julian I upgraded to 7.3 yesterday and am having the same problem using Cisco switches. ___ 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
