[asterisk-users] Asterisk 1.4 and TE420P
I have a site running 1.4.17 with Zaptel. They want to add a TE420P for additional T1 capacity. I'm 99% sure this will work, anyone aware of a reason it wont? Thanks, James -- _ -- 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] Database postgresql not able to start
I found that on a clean boot, I could not connect to Postgresql either. In my /etc/rc.local, I unload cdr_pgsql.so, sleep 15, then reload the module, and that seems to work. After bootup, cdr_pgsql.so is able to connect immediately. -- James Texter III Sr. Software Engineer NOBLE SYSTEMS 4151 Ashford Dunwoody Road, Suite 600 | Atlanta, GA 30319-1452 (o) 404.851.1331 ext. 357 (f) 404.851.1421 (e) jtext...@noblesys.com (w) www.noblesys.com We succeed when we exceed our customers expectations! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, November 15, 2009 4:09 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Database postgresql not able to start On Sun, Nov 15, 2009 at 06:52:22AM +, aster...@opensourcesolution.in wrote: i have installed database POSTGRESQL for storing call details. when i restart database i get the error. [r...@localhost server]# psql -h 127.0.0.1 -U asterisk Password psql: could not connect to server: Connection refused Is the server running on host 127.0.0.1 and accepting TCP/IP connections on port 5432? I'm not sure about other systems. On my systems working as root is not the simple way to work with database. Try 'su - postgres' and connect as that user. It should work as a PostgreSQL admin user. THIS IS MY /VAR/LIB/PGSQL/DATA/POSTGRESQL.CONF # CONNECTIONS AND AUTHENTICATION Small hint: Text in ALL CAPS is generally considered as shouting. Please try to avoid that if you don't really need it. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CONFIDENTIAL NOTICE : If you have received this email in error, please immediately notify the sender by email at the address shown above. This email may contain confidential or legally privileged information that is intended only for the use of the individual or entity named in this email. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution or reliance upon the contents of this email is strictly prohibited. Please delete from your files if you are not the intended recipient. Thank you for your compliance. ___ -- 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] Database postgresql not able to start
I thought that too, but I already checked. Postgresql is priority 64, Asterisk is priority 90. Watching the boot sequence, I can see that Postgresql is clearly started before Asterisk. It may be that there is something in my config that causes this happen, but it's reproducable on any box I setup. -- James Texter III Sr. Software Engineer NOBLE SYSTEMS 4151 Ashford Dunwoody Road, Suite 600 | Atlanta, GA 30319-1452 (o) 404.851.1331 ext. 357 (f) 404.851.1421 (e) jtext...@noblesys.com (w) www.noblesys.com We succeed when we exceed our customers expectations! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry L. Kline Sent: Monday, November 16, 2009 8:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Database postgresql not able to start James Texter wrote: I found that on a clean boot, I could not connect to Postgresql either. In my /etc/rc.local, I unload cdr_pgsql.so, sleep 15, then reload the module, and that seems to work. After bootup, cdr_pgsql.so is able to connect immediately. This sounds as though you have Asterisk starting before PostgreSQL. If you're using CentOS or RHEL or other RHEL-inspired distro look at /etc/init.d/asterisk and /etc/init.d/postgresql. Compare the line in each that looks like: # chkconfig: 2345 xx yy The 'xx' is the start priority. If the number is lower in the asterisk file than it is in the postgresql file then that's your problem. You need PG to start before Asterisk. man chkconfig for further details on what you can do. Barry ___ -- 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 CONFIDENTIAL NOTICE : If you have received this email in error, please immediately notify the sender by email at the address shown above. This email may contain confidential or legally privileged information that is intended only for the use of the individual or entity named in this email. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution or reliance upon the contents of this email is strictly prohibited. Please delete from your files if you are not the intended recipient. Thank you for your compliance. ___ -- 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] my kernel is dazed and confused
I received this with a Sangoma card and CentOS 5.4. Downgrading to 5.2 resolved the issue. -- James Texter III Sr. Software Engineer NOBLE SYSTEMS 4151 Ashford Dunwoody Road, Suite 600 | Atlanta, GA 30319-1452 (o) 404.851.1331 ext. 357 (f) 404.851.1421 (e) jtext...@noblesys.com (w) www.noblesys.com We succeed when we exceed our customers expectations! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francesco Peeters Sent: Thursday, November 12, 2009 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] my kernel is dazed and confused Dr. Michael J. Chudobiak wrote: Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason a0 on CPU 0. Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely on the PCI bus. Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue Would my Digium TDM410P cause an NMI, or is my computer failing? - Mike Googling for the error seems to indicate a possible kernel bug... Are you using Ubuntu or Debian?... -- Francesco ___ -- 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 CONFIDENTIAL NOTICE : If you have received this email in error, please immediately notify the sender by email at the address shown above. This email may contain confidential or legally privileged information that is intended only for the use of the individual or entity named in this email. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution or reliance upon the contents of this email is strictly prohibited. Please delete from your files if you are not the intended recipient. Thank you for your compliance. ___ -- 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] Desperately need help with Asterisk setup
Try putting in a wait after you answer. It's possible the message is playing before the RTP is setup. I would change your dialplan to be exten = 333,1,Answer() exten = 333,n,Wait(1) exten = 333,n,Playback(vm-goodbye) exten = 333,n,Hangup() HTH, James On Mar 17, 2008, at 5:47 AM, Anselm Martin Hoffmeister wrote: Am Montag, den 17.03.2008, 15:08 +0800 schrieb Pete Kay: Hi, I am new to Asterisk and I am having a setup problem that I am trying to resolved for the last couple days without any success. I am pretty much desperated on this issue and I don't know why. Can someone please kindly help me to troubleshoot this? I can't hear any audio from Asterisk when running Playback or VoiceMail tests. Dear Pete, my first idea would be that something with your codecs is borken (TM). I personally use a setup quite similar to yours, with the one visible difference that I also allow the gsm codec, owing to the fact that at least my home-recorded prompts are gsm only. I _guess_ asterisk could or should handle format conversion from audio files automagically, but for making sure, please try adding gsm, at least for now. You might also want to setup the [sipclient] stanza in sip.conf such that nat is set to no, although I do not see why that should break things. Especially as Echo works. The externip is set to your current external IP, right? (Knowing full well that some DSL lines get a new IP as often as 6 times a day, or as a P2P bandwidth countermeasure down to five minute intervals at certain restrictive providers once your fair use volume is used up). Again this should not be the culprit... Poking with a stick in the swamps, but perhaps hitting the bug :-P BR Anselm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unison
No, but I agree with a lot of the comments I saw on Digg. 1.) We're amazed no-one has done this before -- build both a client and a server - Zimbra has had both for quite some time 2.) Nothing of any value on their site 3.) Good luck finding their site. A google on Unison and VoIP only shows press releases. I had to dig through 2 or 3 before I found a link to their actual site. Beyond that, looks like they rolled their own PBX and email. While they mention open source, it looks like its only for libraries, but not the total package (save the email client, as they do mention Thunderbird): http://www.unison.com/opensource/ Thanks, James Texter On Mar 11, 2008, at 8:59 AM, Tzafrir Cohen wrote: On Tue, Mar 11, 2008 at 09:48:04AM -0400, Dean Collins wrote: http://www.pcworld.com/article/id,143198-pg,1/article.html anyone know anything about it? No, but I have heard about http://freshmeat.net/projects/unison http://www.cis.upenn.edu/~bcpierce/unison/ Nothing to do with VoIP. Does relate to synchronization. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simultaneous Inbound and Outbound calls on analog lines...
In the telephony world, this is called glare, it's most prevalent on Analog (though you can have the same thing happen with robbed-bit T1). There really isn't much you can do to prevent it, only minimize it. You need to have your inbound and outbound starting at opposite ends. If your incoming calls are coming top down, then you need to use Gyour group number in your Dial app so that outbound calls go bottom up, or vice versa. HTH, James Texter On Feb 27, 2008, at 2:55 PM, Tim Nelson wrote: Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this installation gets more and more use, we are finding it happens more often. How can this situation be prevented? Shouldn't zaptel see an incoming call and simply choose another trunk? We are running Asterisk 1.2.12.1 and Zaptel 1.2.22.1. Any ideas?!? Tim Nelson Systems/Network Support Rockbochs Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on multi-homed systems
Hi Chris, I have a multi-homed setup, and haven't had any issues, though it's two separate network segments. My Asterisk server has one NIC connected to our voice network (10.0.0.x), and one to our data network (192.168.x.x). Most of my phones are connected to the voice network, but any remote workers connect via VPN, and the phone registers on the data network. Thanks, James Texter On Nov 28, 2007, at 3:58 AM, Chris Bagnall wrote: Greetings list, I remember a discussion many months ago which ISTR concluded that asterisk didn't play nicely at all in multi-homed setups (e.g. SIP packets not being sent out through the same interface they were received on, etc.). Is this still the case, or are there versions which have resolved the issue? Even if it's still the case, is this only a problem for SIP, or does it affect asterisk in general? I have a number of servers with dual NICs, each with an independent net connection. After a few recent failures with one provider, it'd be very useful to be able to use the other connection simultaneously, but only if it's not going to cause problems with the rest of the setup. Any suggestions gratefully appreciated. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mantis 10659 - Make it configurable?
Hello listers, I went to pull some CDR's from my PBX, and noticed they were a bit light. I also noticed output on the console about CDR's not being posted. I am currently running 1.4.13, and in looking at the change log, this was a change in behavior as part of mantis 10659. Personally, I think the old behavior was more correct, but obviously at least one person disagrees. I think it's worth making it configurable so that the old behavior can be restored for those of us who would like to have it. If others agree, I'll work on getting a patch put together to make this happen. Thoughts? Thanks, James ___ --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] ODBC version for cdr?
I believe libtool-ltdl-devel is what you need. On Mon, 2007-10-01 at 13:22 -0500, Chris Stinson wrote: The libtool-ltdl package is installed. On 10/1/07, Jared Smith [EMAIL PROTECTED] wrote: On Mon, 2007-10-01 at 12:52 -0500, Chris Stinson wrote: I'm having an error when I try to ./configure asterisk using --with-odbc=/usr/lib. Below is the version of each application and the ./configure below that. Any help would be appreciated. The autoconf magic in Asterisk looks for a shared library provided by the libtool-ltdl package (at least under Red Hat, CentOS, and Fedora), and won't detect the ODBC libraries without it. (Yes, the build system *should* be a little more informative about this.) -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.24 simultaneous call limits.
What do you have ulimit -n and ulimit -x set to? Thanks, James Texter On Fri, 2007-09-21 at 08:51 -0400, Wai Wu wrote: I am not so sure if the interrupts has any thing to do with it. I run some more test just now and I am getting these error on the console of the call receiving machine. All it does is wait for 45 seconds. I think there is more can be done on the Linux configuration, but I just don't know what. Sep 21 08:42:30 WARNING[22820]: channel.c:565 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! Sep 21 08:42:30 WARNING[22820]: chan_sip.c:2797 sip_new: Unable to allocate SIP channel structure Sep 21 08:42:30 NOTICE[22820]: chan_sip.c:10843 handle_request_invite: Unable to create/find channel Sep 21 08:43:03 WARNING[22820]: channel.c:565 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! Sep 21 08:43:03 WARNING[22820]: chan_sip.c:2797 sip_new: Unable to allocate SIP channel structure Sep 21 08:43:03 NOTICE[22820]: chan_sip.c:10843 handle_request_invite: Unable to create/find channel -Original Message- From: [EMAIL PROTECTED] on behalf of Gordon Henderson Sent: Fri 9/21/2007 3:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.2.24 simultaneous call limits. On Thu, 20 Sep 2007, Wai Wu wrote: Hi everyone, I am running into wall today with simultaneous call limits. I have two Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a lot of sip calls from one machine to the other by issuing AMI Originate commands to one machine. The machine that makes calls plays a message (demo-intruct) upon the other machine answer. The machine receives the calls just waits for 40 seconds then hangs up. Throught the manager connection, I was creating 10 calls per-second. I also have sip phone registered with the calling machine. At around 150 to 200 calls. When I call the machine that's making all the calls, most of the calls couldn't go through. For the ones that went through, most of them will drop off within seconds of the call. But here is catch. When I run 'top', the cpu is idling 97%. My question is. Is there a limit on the number of simultaneous calls Asterisk can handle? I know I have very fast systems. Shouldn't they be able to handle that many calls? What is your take? 200 calls using g711 needs 16Mb/sec of network bandwidth - each way. (200 * 80Kbs) This is well within the limits of a 100Mb network interface. However it also needs 50 packets per second of 160 bytes + IP overhead each way, per call, so thats 20,000 packets/second, and that might well be the bottleneck for your system, not just in the hardware issues required to shovel that many packets over the various buses, but the Linux overhead of schedulling each of the 200 threads to take/send that data in real-time. You might want to run iperf on each machine with nothing else going and see just how many UDP packets of 160 bytes you can push between the machines before packet loss starts. Gordon ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Zap channels unavailable?
Have you tried setting resetinterval=never in zapata.conf? On Tue, 2007-07-17 at 15:43 +0200, [EMAIL PROTECTED] wrote: Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the pbx but the machine should take it just fine (2x3,0 ghz xeon with 1 gb ram etc). Restarting asterisk makes the zaps' available again but they get locked later again. It seems it's always the same channels that are unavailable too? This one is unavailable and not being used... It's been in PRI Flags state resetting for hours now. Channel: 1 File Descriptor: 11 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: 702821667 Calling TON: 33 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Resetting PRI Logical Span: Implicit Hookstate (FXS only): Onhook If anyone can help me with this I'd be really glad. Thanks. Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql
Do you have the mysql client and header files installed? On Thu, 2007-06-21 at 04:11 -0700, Khaled Chehab wrote: Yes mysql installed [EMAIL PROTECTED] asterisk-1.4.5]# rpm -q mysql mysql-4.1.20-2.RHEL4.1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leonardo Kamache (Gmail) Sent: Thursday, June 21, 2007 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql Do you have MySQL installed in your machine??? On 6/21/07, Khaled Chehab [EMAIL PROTECTED] wrote: No one faced a problem like this !! From: Khaled Chehab [mailto:[EMAIL PROTECTED] Sent: Thursday, June 21, 2007 12:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: asterisk 1.4.1 app_addon_sql_mysql I am using centos 4.4 updated using yum when I enter asterisk-addons-1.4.1 directory and make menuselect * Asterisk-addons Module Selection * Press 'h' for help. XXX 1. app_addon_sql_mysql [*] 2. app_saycountpl XXX 3. cdr_addon_mysql [ ] 4. chan_ooh323 [*] 5. format_mp3 XXX 6. res_config_mysql Cannot install app_addon_sql_mysql .. Any dependencies required ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel wct2xxp driver causes LEDs to go black and RED alarm
Have you checked to ensure the card in server #2 is jumpered for E1? On Thu, 2007-06-21 at 09:34 -0500, Jason K. Carter wrote: Hi there, I've got two Asterisk hosted PBX servers with Digium TE210P cards attached by a E1 cable to Port 1 on each. On startup, both cards flash red, alternating between ports 1 and 2. When server #1 loads the Zaptel module and drivers, Port 1 status LED goes green. When server #2 loads the same module and drivers, Port 1 status LED goes completely blank. Unloading the wct2xxp module causes the flashing red LEDs to come back. I've tried swapping cable ends and cards between the two machines, but the problem LED always stays with server #2. So, I think there is something misconfigured with server #2, but the configuration file on both servers is identical. zaptel.conf: loadzone=uk defaultzone=uk span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 Any clues what could cause a GREEN alarm on one end with a RED alarm on the other and no LED light as soon as the wct2xxp driver is loaded? Thanks for the help, Jason Carter DLS Internet Services ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [asterisk-users] Starting Asterisk on Ubuntu 7.04
If you do make config when compiling zaptel and asterisk, it should put the script in /etc/init.d, and add the relevant entries to the various start levels. Thanks, James Texter On Fri, 2007-05-04 at 18:44 +0200, Christian wrote: Hi, I have already done: apt-get build-dep asterisk and then installed libpri, zaptel and asterisk from the latest sources. So what should i do then? New to Ubuntu. many thanks, Christian On 2007-05-04 at 17:00 Tzafrir Cohen wrote: On Fri, May 04, 2007 at 02:04:55PM +0200, Christian wrote: Hi all, Could someone please tell me how to make Asterisk start at boot on Ubuntu Feisty 7.04? Many thanks, Christian apt-get install asterisk Look at the init.d scripts. Note that in Ubuntu, subdirectories under /var/run are deleted at boot, and hence that script generates /var/run/asterisk (with proper ownership) at boot time. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.14 and Audiocodes Mediant 1000
I sent this yesterday, but saw zero traffic, so I think it got lost in the ether, so I'm sending again. I'm having an issue using Asterisk 1.2.14 and an Audiocodes Mediant 1000 ISDN gateway. For the most part, everything is working except for attended transfers. When I do an attended transfer, and complete the transfer before the 3rd party answers, the PSTN side hears dead air until the PSTN party answers or the transfer goes to voicemail. This happens regardless of whether I use the phone to do the transfer, or *2 to have Asterisk do it. Originally, it was actually disconnecting the call, but I fixed that by telling it not to disconnect on a broken connection, however that fact makes me think something is not quite right. Anyone else have experience with the Mediant gateways? Thanks, James ___ --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] Audiocodes Mediant 1000, Polycom, and no ringback on transfer
On Mon, 2007-01-15 at 15:26 -0600, David Gomillion wrote: I don't think you can do that. Here's why: on the Polycom's, the Transfer button doesn't reappear until the transferree picks up the phone. Unless something changed in the firmware recently. But, if you're completing it before the 3rd party answers, it's not an attended transfer. I found it all depends on the dialplan, and the sip.cfg for the phone. If you call Answer() before Dial(), it will allow it. There is also a setting in sip.cfg for the phones, voIpProt.SIP.allowTransferOnProceeding that I think allows that as well. I should have mentioned in my original post, but MOH works just fine. When I complete the transfer, the MOH stops, and that's when the dead air starts. Anyone else have any suggestions? Thanks, James ___ --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] What's up with the Manager Interface?!?!
Doug, Your issue isn't with the manager. It's with the CLI output you are trying to hijack via manager :D If you run sip show peer 2944093 in the CLI, you'll see a blank line, followed by a line that is * Name. It appears what you really want is a manager Action to show a sip peer, in which case I would recommend adding a new manager command that returns a string which is much more machine readable. Remember, CLI output is designed to be human readable. Just my $0.02. On 11/29/06 3:36 PM, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Douglas Garstang Sent: Wednesday, November 29, 2006 12:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! -Original Message- From: Michael Collins [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 29, 2006 11:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug, What language(s) are you using? Just curious. I've been tinkering with Perl, POE, and POE::Component::Client::Asterisk::Manager. These have abstracted away the lowest level of programming. I know you've done Python in the past - I hear that there's a module for AMI called py-Asterisk. Have you seen or tried that? Ditto with Ruby - a module called RAMI. Both are on sourceforge. Also, could you hum a few bars about what you're trying to accomplish with your API? I'm curious about the big picture. Michael, I'm using python. Here's a good example. I'm trying to get SIP blf. I managed to split my result into a list of lines by splitting on ANY of \r\n, \n or \r. I was going use the column headings from the third line as my keys for my dictionary/hash, rather than hard coding them. Notice anything? The 'Call ID' column has a space right in the middle which means I can't simply split this up by white-space. Response: Follows Privilege: Command Peer UserCall ID Extension Last state Type xxx.187.128.105 2944090 f7ee98da-6d 2944006 InUse xpidf+xml xxx.187.128.105 2944090 111e388b-6b 2944077 Idle xpidf+xml I think I looked at the python module and was underwhelmed by it. G. Here's another example... Action: Command Command: sip show peer 2944093 Response: Follows Privilege: Command * Name : 2944093 Secret : Set MD5Secret: Not set Context : 180o_CallStart Subscr.Cont. : 180o_WatchBLF Why the HELL is there an asterisk before 'Name'? Now I have to strip the bloody thing out! And why is there TWO empty lines before it? Good grief! Doug. ___ --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 -- James Texter ___ --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] Astricon - post show Saturday?
Title: Re: [asterisk-users] Astricon - post show Saturday? I really recommend Pappas Brothers Steakhouse. My wife and I went there for our first anniversary, and it was really nice. Beyond that, check out the west end, its pretty nice place to be for the night scene. On 10/20/06 8:45 AM, Dean Collins [EMAIL PROTECTED] wrote: Is anyone on this list familiar with Dallas? Anyone want to recommend something to do on the Saturday/Sunday? Never been to Dallas so Im hoping for a restaurant recommendation for Saturday night.(somewhere a little more up market would be good) also any sights that have to be visited while in town? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Texter ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom IP 501 phone randomly resets itself (loses Received call log, Missed calls, placed calls)
Mike, Is this using PoE, or the injector? When I've had this happen, I've been using PoE, it's been in my wiring, specifically in the cable between the wall jack and the phone. Replacing that cable has always sorted it out for me. Thanks, James On 10/13/06 9:40 AM, Mike Garey [EMAIL PROTECTED] wrote: I've been noticing that my group of Polycom IP 501 phones seems to randomly reset themselves nearly every night (I guess it usually happens at night, since I've never seen it happen while I've been at work during the day).. When I say reset, I mean, the hands free volume and ring volume are set to the default and the call logs (received calls, missed calls, placed calls) are all reset. It does, however, keep certain settings such as the specific ring tone used for incoming calls.. But most other settings are being reset.. Has anyone else experienced this, or know why it might be happening? Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Texter ___ --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 Directory listing
On the entry to hide, add hidefromdir=yes as an option. On 10/3/06 5:24 PM, asterisk-user [EMAIL PROTECTED] wrote: How do I take out few extensions (vm enabled extensions) from the default company directory listing? thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Texter ___ --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] Running Multiple Instances of Asterisk
Doug, I actually see this as a pretty logical way to solve the problem. Please keep us posted if you have any luck sorting out running multiple instances, or mail me off-list if no one else is interested. Thanks, On 9/25/06 1:52 PM, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Brian Rogan [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Doug, Why do you want to do this to begin with? I think the best solution is Because we are trying to build a hosted IPT solution, not an enterprise solution. to use the realtime stuff, and build your own management tools, which would allow you to do this (you could drastically cut the complexity with the right tools). Even if you could run them together, how would you put everything on the appropriate ports? How would you deal with multiple instances accessing hardware? Realtime is resource intensive, requiring many queries to perform simple lookups. We can easily create multiple virtual IP address, and since each virtual IP address can bind to port 5060, each phone can register with domain.com:5060 without a problem. We don't need multiple instances to access hardware as this is a SIP only solution. Our PSTN access is via external Audiocodes gateways, not via Digium T1 cards. The dial plan was not able to handle the complexity we needed (for example the MySQL() application command could not do nested queries), and so right now, we have a 2000 line python script and several very complex MySQL stored procedures in order to fulfull our requirements. I'm not convinced that maintaining the config files, binaries and other components of multiple asterisk's is easier than just building better tools to configure one. I am. I look at our configuration which is currently for one customer, and there's already several dozen contexts in order to cover a lot of complexity. Multiply that by a couple of hundred, and I won't want to be administering it! You could also try User-Mode-Linux or something like that. I was going to give v-servers a try. There's a guide at: http://www.telephreak.org/papers/vpa/ ___ --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 -- James Texter ___ --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] Running Multiple Instances of Asterisk
But if I segment my zap channels, that shouldn't be an issue, correct? I.e. Instance 1 = Port 1, Instance 2 = Port 2, etc. Of course, you are also assuming there is Zap channels, as I believe he is using a gateway, which takes that out of the equation. On 9/25/06 2:23 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Best of luck getting multiple instances of Asterisk to play nice when accessing Zap channels. James Texter wrote: Doug, I actually see this as a pretty logical way to solve the problem. Please keep us posted if you have any luck sorting out running multiple instances, or mail me off-list if no one else is interested. Thanks, On 9/25/06 1:52 PM, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Brian Rogan [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Doug, Why do you want to do this to begin with? I think the best solution is Because we are trying to build a hosted IPT solution, not an enterprise solution. to use the realtime stuff, and build your own management tools, which would allow you to do this (you could drastically cut the complexity with the right tools). Even if you could run them together, how would you put everything on the appropriate ports? How would you deal with multiple instances accessing hardware? Realtime is resource intensive, requiring many queries to perform simple lookups. We can easily create multiple virtual IP address, and since each virtual IP address can bind to port 5060, each phone can register with domain.com:5060 without a problem. We don't need multiple instances to access hardware as this is a SIP only solution. Our PSTN access is via external Audiocodes gateways, not via Digium T1 cards. The dial plan was not able to handle the complexity we needed (for example the MySQL() application command could not do nested queries), and so right now, we have a 2000 line python script and several very complex MySQL stored procedures in order to fulfull our requirements. I'm not convinced that maintaining the config files, binaries and other components of multiple asterisk's is easier than just building better tools to configure one. I am. I look at our configuration which is currently for one customer, and there's already several dozen contexts in order to cover a lot of complexity. Multiply that by a couple of hundred, and I won't want to be administering it! You could also try User-Mode-Linux or something like that. I was going to give v-servers a try. There's a guide at: http://www.telephreak.org/papers/vpa/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Texter ___ --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] Chanspy crashing the server, again
Title: Re: [asterisk-users] Chanspy crashing the server, again Anybody have a backtrace? On 9/18/06 9:23 AM, Richard [EMAIL PROTECTED] wrote: I am experiencing the same problem. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, September 18, 2006 10:03 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Chanspy crashing the server, again I upgraded to 1.2.12.1 - the problem is much, MUCH worse. Before, it was once or twice per week, now EVERY TIME someone uses chanspy it crashes the machine. Anyone with any ideas? This is killing me. Check Out the new free AIM(R) Mail http://pr.atwola.com/promoclk/100122638x1081283466x1074645346/aol?redir=http%3A%2F%2Fwww%2Eaim%2Ecom%2Ffun%2Fmail%2F -- 2 GB of storage and industry-leading spam and email virus protection. ___ --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 -- James Texter ___ --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: 911 versus 9.911
Title: Re: [asterisk-users] Re: 911 versus 9.911 I ran into the same situation at one of my customers. The recording worked perfectly, and has resolved any accidental 911 calls. Thanks, On 8/30/06 10:05 PM, Steven [EMAIL PROTECTED] wrote: Maybe play a recording before dialing 911. You have dialed 911, if this was not intended hang-up now, otherwise please wait a moment Then connect them to 911. Yes they may not hang up soon enough, but there is not excuse for it. We allow 911 and 9911 (in case they think they need to dial 9). Maybe I will try the recording that I proposed. -- James Texter ___ --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] Playfile waiting for N digits
Title: Playfile waiting for N digits Im not sure if this is the best place to ask, or dev, but here goes. Im writing a custom application module to load into asterisk. Part of my requirement is to play a file until I receive N number of digits. I see ast_waitstream, which looks great for stopping on a specific digit, but is there an existing method I can call that will only stop playing if received a specified number of digits? Thanks, -- James Texter ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom ACD, Asterisk, Kernel 2.6
Title: Re: [asterisk-users] Polycom ACD, Asterisk, Kernel 2.6 What OS are you using? There is a known issue with the kernel sources on CentOS 4.3 and I assume RHEL 4 that will keep Zaptel from compiling? What compilation error are you getting? James On 7/11/06 3:26 PM, Dean @ INKnBITs [EMAIL PROTECTED] wrote: I'm trying to build another asterisk server as I'm having a problem with the current one. Unless anybody can tell me how to compile the meetme app? Everything else works fine, asterisk just will not compile meetme?!? (Under kernel 2.4) I used svn to pull the trunk versions of libpri, zaptel and the polycom_acd_functions (release 30432). I cannot seem to get the zaptel to compile under 2.6, is this correct? Does it only work on 2.4? Is there any release I should be pulling for the zaptel (and libpri)? Or does anybody have it working or know a release version that I could pull? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] when I press transfer - blind - 700 . The user is not able to hear what extension the call was parked on
Title: Re: [Asterisk-Users] when I press transfer - blind - 700 . The user is not able to hear what extension the call was parked on
This is the way blind transfers work. The transferring party doesnt get to hear anything. For call parking, you have no choice but to use supervised transfer if you want the user to hear the parking space. If it worked before, it must have been dumb luck with the timing.
On 6/22/06 10:24 AM, sdgesa gaeharth [EMAIL PROTECTED] wrote:
I am using Polycom 501s with asterisk 1.2.4.
When transfering to call parking wih #1 - 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press transfer - blind - 700 . The user is not able to hear what extension the call was parked on. It seems like the polycom is hanging up before asterisk is able to finish telling the user the extension. I can not tell if this is a problem with the phone, asterisk, transfers or call parking. This problem just started happening a few weeks ago. Before then , blind transfer worked fine. It must be a config issue somewhere
using #1 - 700:
-- Started music on hold, class 'default', on channel 'Zap/1-1'
== Parked Zap/1-1 on 701. Will timeout back to extension [incoming] s, 1 in 45 seconds
-- Added extension '701' priority 1 to parkedcalls
-- Playing 'digits/7' (language 'en')
-- Executing ParkedCall(SIP/1000-300e, 701) in new stack
-- Stopped music on hold on Zap/1-1
-- Channel SIP/1000-300e connected to parked call 701
-- Hungup 'Zap/1-1'
using transfer - blind - 700
-- Started music on hold, class 'default', on channel 'Zap/1-1'
== Parked Zap/1-1 on 701. Will timeout back to extension [incoming] s, 1 in 45 seconds
-- Added extension '701' priority 1 to parkedcalls
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/1' (language 'en')
== Spawn extension (local-access, 97037551131, 1) exited non-zero on 'SIP/1000-d779'
-- Executing ParkedCall(SIP/1000-5f5a, 701) in new stack
-- Stopped music on hold on Zap/1-1
-- Channel SIP/1000-5f5a connected to parked call 701
-- Started music on hold, class 'default', on channel 'Zap/1-1'
-- Stopped music on hold on Zap/1-1
Even though the user can not hear the extension the call was parked on, the call can be retrieved by guessing. Which I am assumming means the call was successfully parked.
Digit map:
[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT|1xxxT
extensions.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
ATTENDANT=SIP/1006SIP/1002SIP/1011SIP/1009
OUTBOUNDTRUNK=ZAP/g1
[meetme-ext]
exten = 600,1,MeetMe(1234|Mp|98765)
[extentions]
include = parkedcalls
include = meetme-ext
include = direct-to-voicemail
exten = _10XX,1,Dial(SIP/${EXTEN},20,t)
exten = _10XX,n,Answer
exten = _10XX,n,VoiceMail([EMAIL PROTECTED])
exten = _10XX,n,Hangup()
[voicemail]
exten = _910XX,1,Wait(1)
exten = _910XX,n,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])
[direct-to-voicemail]
exten = _810XX,1,VoiceMail(u${EXTEN:[EMAIL PROTECTED])
exten = _810XX,n,Hangup()
[local]
include = extentions
include = voicemail
[incoming]
exten = s,1,Answer
exten = s,n,Wait(2)
exten = s,n,Set(TIMEOUT(response)=15)
exten = s,n,Background(intro)
exten = s,n,WaitExten()
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup()
exten = 0,1,Dial(${ATTENDANT},20,t)
exten = 0,n,Playback(vm-nobodyavail)
exten = 0,n,Hangup()
exten = 1,1,Directory(voicemail,extentions,f)
exten = 2,1,Directory(voicemail,extentions)
include = meetme-ext
include = extentions
exten = i,1,Playback(pbx-invalid)
exten = i,2,Goto(incoming,s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup()
[outbound]
ignorepat = 9
include = parkedcalls
exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,T)
exten = _9XX,2,Congestion()
exten = _9XX,102,Congestion()
exten = _91900NXX,1,Congestion()
exten = _91976NXX,1,Congestion()
exten = _91[123456789]XXNXX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,T)
exten = _91[123456789]XXNXX,2,Congestion()
exten = _91[123456789]XXNXX,102,Congestion()
exten = 9911,1,Dial(${OUTBOUNDTRUNK}/ww911)
exten = 9411,1,Dial(${OUTBOUNDTRUNK}/ww411)
exten = 0,1,Dial(${OUTBOUNDTRUNK}/ww0)
[local-access]
include = local
include = outbound
features.conf:
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
parkingtime = 45 ; Number of seconds a call can be parked for
[featuremap]
blindxfer = #1
Thanks
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Re: [Asterisk-Users] Echo Problem with T411P
Title: Re: [Asterisk-Users] Echo Problem with T411P Try setting echocancelwhenbridged=yes. Also, in your zaptel, you only need to define one span as the clock master, so should be like zaptel.conf-- span = 1,1,0,ccs,hdb3,crc4 bchan = 1-15,17-31 dchan = 16 span = 2,0,0,ccs,hdb3 bchan = 32-46,48-62 dchan = 47 span = 3,0,0,ccs,hdb3,crc4 bchan = 63-77,79-93 dchan = 78 span = 4,0,0,ccs,hdb3 bchan = 94-108,110-124 dchan = 109 loadzone = nl defaultzone = nl Thanks, James On 6/15/06 6:29 AM, Idris AVCI [EMAIL PROTECTED] wrote: Hello, There are 3 PRIs connected to the card each from different operators. Especially echo occured on span 3 is really annoying. Configuration files are as follows. Is there something wrong in conf ? Zapata.conf -- [channels] context=default switchtype=euroisdn usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes callerid=asreceived echocancel=yes echotraining=yes rxgain=0.0 txgain=0.0 group = 1 signalling=pri_cpe context=default channel = 1-15 channel = 17-31 group = 2 signalling=pri_cpe context=default channel = 32-46 channel = 48-62 group = 3 signalling=pri_cpe context=Satelco channel = 63-77 channel = 79-93 group = 4 signalling=pri_cpe context=default channel = 94-108 channel = 110-124 zaptel.conf-- span = 1,1,0,ccs,hdb3,crc4 bchan = 1-15,17-31 dchan = 16 span = 2,1,0,ccs,hdb3 bchan = 32-46,48-62 dchan = 47 span = 3,1,0,ccs,hdb3,crc4 bchan = 63-77,79-93 dchan = 78 span = 4,1,0,ccs,hdb3 bchan = 94-108,110-124 dchan = 109 loadzone = nl defaultzone = nl ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with zaptel and TE210P
Title: Re: [Asterisk-Users] Problems with zaptel and TE210P Shouldnt your zapata.conf be span=1,1,0,esf,b8zs As it stands, you are not taking timing from the PRI. Changing the second digit of the span entry to 1 will tell Asterisk to use that line as the clock master. HTH, James On 5/1/06 1:58 PM, Dan Brummer [EMAIL PROTECTED] wrote: Thank you for the reply Alexandar. After I restarted my dev machine I recieved these messages from asterisk: chan_zap.c:2290 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! chan_zap.c:8202 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 May 1 10:23:27 WARNING[3882]: chan_zap.c:2290 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! So you are right on the D-Chan. If my telco says everything is ok, what should I look at next? AFAIK this PRI was in working condition before I moved it to the asterisk test machine. Thanks! -Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Monday, May 01, 2006 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Problems with zaptel and TE210P Looks like your D-channel is down. Ztcfg reports all is ok, b/c as far as iut is concerned, it is talking to your card just fine. LibPri handles the PRI implemetaton. Since you are able to see the pri commands from the CLI, Isdn supprt is linked into your asterisk core. Call your telco and ask if they have your D-channel in a loop. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Brummer Sent: Monday, May 01, 2006 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Problems with zaptel and TE210P Some more info to my problem: ipt-dev01*CLI zap show status Description Alarms IRQ bpviol CRC4 T2XXP (PCI) Card 0 Span 1 OK 0 0 0 ipt-dev01*CLI pri show span 1 Primary D-channel: 24 Status: Provisioned, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 Any ideas? ZTCFG looks good. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Brummer Sent: Monday, May 01, 2006 10:41 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problems with zaptel and TE210P Hello, I'm just starting out with asterisk and I'm playing around with the system. Currently I have a Digium TE210P connected to a PRI on the Asterisk server. I have a SIP soft phone on my laptop for testing that is working fine. When I try to place a call from my soft phone I get this from Asterisk: May 1 09:11:41 NOTICE[20098]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/test-3a26' status is 'CONGESTION' #/etc/zaptel.conf: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 #/etc/asterisk/zapata.conf: [channels] switchtype=national context=default signalling=pri_cpe group=1 channel = 1-23 #/etc/asterisk/extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes [default] exten = 123,1,Answer() exten = 123,2,Playback(hello-world) exten = 123,3,Hangup() exten = _9NXX,1,Dial(Zap/g1) Any ideas? Thank you in advance, your help is greatly appreciated. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hinting
Has anyone been able to overcome the limit of being able to watch 7 people? If so, how? On 4/3/06 5:52 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Mon, 3 Apr 2006, Darrick Hartman wrote: Very good explanation. Additionally, (at least on the Polycom 600's) you need to reboot your phone for this to take effect. just about anything you do to a polycom will make it reboot. sometimes i feel like if I sneeze near the phone it will need to reboot for changes to take effect. shades of microsoft windows. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with T1 card on laptop
I have a 4 PCI slot version. It worked well under Windows, but I could never get my laptop to see any devices I stuck in it when running under Linux. I have tried RHEL3 and RHEL4. They list their last supported Linux version has Redhat 9, so doesn't appear they keep the Linux version very up to date. The driver would compile, but I couldn't get any programs to recognize cards I put in it. HTH, James Rusty Dekema wrote: On 2/28/06, Arnaud [EMAIL PROTECTED] wrote: What are the options to hook a T1 card up to a laptop running * ? Are there USB or PCMCIA T1 cards ? Has anyone tried a USB to PCI adapter as such : http://www.mobl.com/expansion/products/cardbus_expansion/1slot/ It looks nice but cost 1k Those aren't USB to PCI adapters, they are PCMCIA to PCI adapters. They look cool and expensive, and I have never used one. -Rusty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SOLVED - Channel bank woes - no outbound calls
Thanks to the great support at Rhino Equipment and Digium, this has finally been solved. I wanted to post the solution back to the list in case anyone else is having a similiar issue. I started by calling Rhino support so I could eliminate channel bank configuration as the issue. We were able to determine the channel bank and signalling were all working as expected. I then began to monitor the analog POTS line while attempting to place a call. When I did this, I noticed that the DTMF tones were coming through very short and choppy. So, I tried setting toneduration=250 in zapata.conf. Now, instead of getting the Your call did not go through. message I was getting Your call could not be completed as dialed operator intercept message. I monitored the line again, and this time, I found I could hear the first digit quite well, but the other 6 digits were coming through stuttered. At this point, I contacted Digium support. Digium connected to the box, and placed a few test calls to get some debug information. Then, they ended up changing a line in wct4xxp.c of the zaptel branch, recompiling, and trying again. Lo and behold, calls have now begun to complete without issue. To resolve this issue, you need to find the line in wct4xxp.c that looks like the following: static int vpmdtmfsupport = 1; Change this to static int vpmdtmfsupport = 0; According to Digium support, this moves the playing of the DTMF's from hardware to software. Once that change was made, I have not had any other issues. Thanks, James James Texter wrote: So, I'm still having this problem with outbound calls not working when using a channel bank. I've purchased a Rhino FXO channel bank from VoIPSupply.com to make sure it wasn't an equipment problem. I am using a Digium TE411P card, and have simplified it down to just 1 port plugged into the channel bank, with just 1 analog line plugged in. If I place an inbound call on the line, it goes through just fine. However, if I attempt an outbound call, I get Your call did not go through. Please try your call again. After much experimenting, I found out this happens if you dial some digits, but not enough for a full phone number. My zaptel.conf looks like: span=1,0,0,esf,b8zs fxsks=1 loadzone=us defaultzone=us And my zapata.conf looks like: [channels] usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=no transfer=no echocancel=no ; define channels group=1 context=from-pstn signalling=fxs_ks channel = 1 And finally, extensions.conf looks like: [from-pstn] exten = 6080,1,Answer() exten = 6080,2,Playback(hello-world) exten = 6080,3.Hangup() [internal] exten = 5148346,1,Dial(Zap/g1/514836) Anybody out there have any ideas on why all of the digits aren't being sent out? Thanks, James ___ --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] Channel bank woes - no outbound calls
I'm hooked up to a regular analog POTS line. I've tried both loop start and ground start, but no luck either way. Any other thoughts? Thanks, James Doug Lytle wrote: [EMAIL PROTECTED] wrote: So, I'm still having this problem with outbound calls not working when using a channel bank. I've purchased a Rhino FXO channel bank from VoIPSupply.com to make sure it wasn't an equipment problem. I am using a Digium TE411P card, and have simplified it down to just 1 port plugged into the channel bank, with just 1 analog line plugged in. If I place an inbound call on the line, it goes through just fine. However, if I attempt an outbound call, I get Your call did not go through. Please try your call again. After much experimenting, I found out this happens if you James, When I have problems with outbound on my Adit 600, it's usually because I have signaling screwed up on that channel. (i.e. trying to grab a groundstart line with loopstart signaling). Doug ___ --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 overlap dialing (PRI)
In my telecom experience, overlap on a PRI isn't sending digits as INFORMATION messages, but instead means to send the digits as DTMF tones over the B channel. This is pretty common when connecting to the PSTN where the carrier requires an authorization code, either for billing or as an access code. If it's a robbed-bit T1, you just send all digits, including authorization code as DTMF. However, on PRI, you send the phone number in the SETUP message, and once you get a PROCEEDING event back, you send the authorization code as DTMF on the B channel. I would imagine that is what the overlapdial setting does, but can't confirm since I don't have the source code on this machine. HTH, James On Mon, 2005-11-14 at 02:26 +0100, Piotr Dydycz wrote: On 11/13/05, Miloš Kocbek [EMAIL PROTECTED] wrote: overlapdial setting should work but you also have to set immediate=no if you want overlapdial to work greetings mk unfortunatelly it didn't help :( -- dydyczp ___ --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
