Re: [asterisk-users] Trouble Playing message file via Perl AGI
On Wed, 23 Jul 2008, Mike Diehl wrote: What you sent me is almost exactly what I had, which indicated that that part of my code was correct. So, I moved that block of code to the top of my program and it worked. Eventually, I found a debug print() statement that I had forgotten to take out. Once it was gone, my code worked as expected. Thank you for your time. The agi debug command (1.2) would have shown you where you violated the protocol. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel won´t recognizes sources installed
On Wed, Jul 23, 2008 at 11:37:01PM -0300, Felipe Trevisan wrote: I´m installing zaptel and asterisk on the CEntos 3.9. I´ve installed the kernel-devel which on the kernel 2.4.21 is called kernel-source, but when I run the pre requisites test, the zaptel won´t recognize it. Can I rename the kernel-source to kernel-devel? I'm not sure if kernel-source provides the same as kernel-devel . kernel-devel provides a partial kernel tree that is already configured, and hence external modules could be built with it just as well as with a fully-built kernel source tree. With kernel-source I suspect you'll also have to configure (copy the respective .config file) and run at least a partial build (I don't know if theer's a special target for that in 2.4). I suspect that this is the case because there are several kernel packages with several configrations in 3.9: kernel, kernel-BOOT, kernel-hugemem . But there is just one kernel-source. I don't have a test-system to try it on. Should I try to install the zaptel anyways? Last time I did it, the make comand didn´t work because of the missing devel files. While kernel 2.4 has been removed from DAHDI, it is still supoprted in DAHDI. Some bugs for 2.4 kernel systems were reported and resolved on on older slackwares, Debian Sarge, and, of course, CentOS 3.x . Specifically one of the many RPM packages Axel Thimm maintains is Zaptel, and is also vs. RHEL3: http://atrpms.net/dist/el3/zaptel/ . He reported several breakages in the past (which were fixed). I see that the latest version there is 1.4.11 . -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel won´t recognizes sources installed
On Thu, Jul 24, 2008 at 10:53:59AM +0300, Tzafrir Cohen wrote: On Wed, Jul 23, 2008 at 11:37:01PM -0300, Felipe Trevisan wrote: I´m installing zaptel and asterisk on the CEntos 3.9. I´ve installed the kernel-devel which on the kernel 2.4.21 is called kernel-source, but when I run the pre requisites test, the zaptel won´t recognize it. [snip] Should I try to install the zaptel anyways? Last time I did it, the make comand didn´t work because of the missing devel files. While kernel 2.4 has been removed from DAHDI, it is still supoprted in DAHDI. It is stil supported in Zaptel, that is. -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IP door opening devices
Greetings list, We have a client with an analogue door intercom/opening unit which we're attempting to replace with an IP variant. The existing unit has the following functionality: 1) Intercom - visitor hits call, talks to operator 2) Door opening - operator can open the door by dialling a 4-digit PIN followed by * (the door unit interprets the DTMF tones) 3) Door opening - the door unit has a numeric keypad to enable approved persons to enter by entering the 4-digit PIN on the keypad We've tried getting the existing unit working with an ATA, but it's only about 50% reliable (hangup not always detected, DTMF not always detected, etc.), so it's probably time to look at fully IP alternatives. Any suggestions gratefully appreciated. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime + SIP + MySQL: md5secret BROKEN
I'm quite sure there's a bug somewhere in SIP + realtime + MySQL.
To update, since last post we've integrated with our existing users
database using MySQL views. Our legacy database uses md5
password hashes, and does not store plaintext.
During testing this morning I could swear it was all working, however
for some reason, after going out to lunch today and coming back (no
config changes at all!) authentication would not succeed no matter
what I tried:
- toggling rt* settings in sip.conf
- re-creating MySQL view
- reverting to static table
- sip reload on command line
- recompiling / re-installing asterisk and asterisk-addons
- probably a bunch more
Most of these I tried multiple times in various combinations.
The issue appears in the debug log like this:
[Jul 24 17:16:43] DEBUG[8732] chan_sip.c: = No match Their Call ID:
[EMAIL PROTECTED] Their Tag Our tag: as2f38c31a
[Jul 24 17:16:43] DEBUG[8732] chan_sip.c: Allocating new SIP dialog
for [EMAIL PROTECTED] - REGISTER (No RTP)
[Jul 24 17:16:43] DEBUG[8732] chan_sip.c: Received REGISTER (2) -
Command in SIP REGISTER
[Jul 24 17:16:43] DEBUG[8732] res_config_mysql.c: MySQL RealTime:
Everything is fine.
[Jul 24 17:16:43] DEBUG[8732] res_config_mysql.c: MySQL RealTime:
Retrieve SQL: SELECT * FROM sip_buddies WHERE name = 'walter' AND host
= 'dynamic'
[Jul 24 17:16:43] DEBUG[8732] db.c: Unable to find key 'walter' in
family 'SIP/Registry'
No matter what I tried I could not fix this.
Finally I found out that after dropping md5secret authentication instantly
began to succeed.
mysql select * from sip_buddies;
+++-+-++-+--++---+--++-+-+---+--+--++--+--+-+--+--++--+-+-+-+--+-+++++--+-+-++--+---++--+-+--+
| id | name | host| nat | type | accountcode | amaflags |
call-limit | callgroup | callerid | cancallforward | canreinvite |
context | defaultip | dtmfmode | fromuser | fromdomain | insecure |
language | mailbox | md5secret| deny | permit
| mask | musiconhold | pickupgroup | qualify | regexten | restrictcid
| rtptimeout | rtpholdtimeout | secret | setvar | disallow | allow
| fullcontact | ipaddr | port | regserver | regseconds |
username | defaultuser | subscribecontext |
+++-+-++-+--++---+--++-+-+---+--+--++--+--+-+--+--++--+-+-+-+--+-+++++--+-+-++--+---++--+-+--+
| 1 | walter | dynamic | no | friend | NULL| NULL |
NULL | NULL | NULL | yes| yes | NULL
| NULL | NULL | NULL | NULL | NULL | NULL |
NULL| 4d27b7677bd96f7ba00c4bd0541c9588 | NULL | NULL | NULL |
NULL| NULL| NULL| NULL | NULL| NULL
| NULL | qwedsa | NULL | all |
g729;ilbc;gsm;ulaw;alaw | ||0 | NULL |
0 | walter | | NULL |
+++-+-++-+--++---+--++-+-+---+--+--++--+--+-+--+--++--+-+-+-+--+-+++++--+-+-++--+---++--+-+--+
1 row in set (0.00 sec)
mysql alter table sip_buddies drop regserver;
Query OK, 1 row affected (0.01 sec)
Records: 1 Duplicates: 0 Warnings: 0
(retry auth - no luck yet)
mysql alter table sip_buddies drop regseconds;
Query OK, 1 row affected (0.00 sec)
Records: 1 Duplicates: 0 Warnings: 0
(retry auth - no luck yet)
mysql alter table sip_buddies drop md5secret;
Query OK, 1 row affected (0.00 sec)
Records: 1 Duplicates: 0 Warnings: 0
Suddenly, authentication works!
The md5secret used was the md5 of 'qwedsa', and the value was correct.
mysql select md5('qwedsa');
+--+
| md5('qwedsa')|
+--+
| 4d27b7677bd96f7ba00c4bd0541c9588 |
+--+
1 row in set
Re: [asterisk-users] IP door opening devices
On Thu, 24 Jul 2008, Chris Bagnall wrote: Greetings list, We have a client with an analogue door intercom/opening unit which we're attempting to replace with an IP variant. The existing unit has the following functionality: 1) Intercom - visitor hits call, talks to operator 2) Door opening - operator can open the door by dialling a 4-digit PIN followed by * (the door unit interprets the DTMF tones) 3) Door opening - the door unit has a numeric keypad to enable approved persons to enter by entering the 4-digit PIN on the keypad We've tried getting the existing unit working with an ATA, but it's only about 50% reliable (hangup not always detected, DTMF not always detected, etc.), so it's probably time to look at fully IP alternatives. Any suggestions gratefully appreciated. There was talk of this a week or 2 ago on the list - look into the archives. I don't think there was anything that successfull though... I have to say though - if you have such an integrated unit that needs nothing more than an analogue connection (and power, presumably), I'd love to know the make - for me, (or rather one of my clients) it would be worthwhile trying to find an ATA that would work with it.. Got a name/website for the opener device? Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP door opening devices
On Thu, Jul 24, 2008 at 10:25:34AM +0100, Chris Bagnall wrote: Greetings list, We have a client with an analogue door intercom/opening unit which we're attempting to replace with an IP variant. The existing unit has the following functionality: 1) Intercom - visitor hits call, talks to operator 2) Door opening - operator can open the door by dialling a 4-digit PIN followed by * (the door unit interprets the DTMF tones) 3) Door opening - the door unit has a numeric keypad to enable approved persons to enter by entering the 4-digit PIN on the keypad What would it take to move the logic into Asterisk? (1) is naturally trivial. As for sending the actual signal to open and close the door: that may take a separate out-of-band operation. e.g. System(/usr/sbin/open-sesame) But what about typing the PIN? Hello, you have reached Treasure Cave Inc. If you know the access code, dial it now. If not, press 0 for the operator. The point is to separate communication from door opening. Both are different problems to solve. I'm not sure you will gain much by bundeling them together. -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP door opening devices
Hi all, maybe there is no opener device at all. Anyway take a look here : http://www.barix.com/ On Thu, Jul 24, 2008 at 12:11 PM, Gordon Henderson [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Thu, 24 Jul 2008, Chris Bagnall wrote: Greetings list, We have a client with an analogue door intercom/opening unit which we're attempting to replace with an IP variant. The existing unit has the following functionality: 1) Intercom - visitor hits call, talks to operator 2) Door opening - operator can open the door by dialling a 4-digit PIN followed by * (the door unit interprets the DTMF tones) 3) Door opening - the door unit has a numeric keypad to enable approved persons to enter by entering the 4-digit PIN on the keypad We've tried getting the existing unit working with an ATA, but it's only about 50% reliable (hangup not always detected, DTMF not always detected, etc.), so it's probably time to look at fully IP alternatives. Any suggestions gratefully appreciated. There was talk of this a week or 2 ago on the list - look into the archives. I don't think there was anything that successfull though... I have to say though - if you have such an integrated unit that needs nothing more than an analogue connection (and power, presumably), I'd love to know the make - for me, (or rather one of my clients) it would be worthwhile trying to find an ATA that would work with it.. Got a name/website for the opener device? Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP door opening devices
Siemens HC 450 Dect intercom does exactly what you want it doesn't come cheap, but works like a dream.. Gordon Henderson schreef: On Thu, 24 Jul 2008, Chris Bagnall wrote: Greetings list, We have a client with an analogue door intercom/opening unit which we're attempting to replace with an IP variant. The existing unit has the following functionality: 1) Intercom - visitor hits call, talks to operator 2) Door opening - operator can open the door by dialling a 4-digit PIN followed by * (the door unit interprets the DTMF tones) 3) Door opening - the door unit has a numeric keypad to enable approved persons to enter by entering the 4-digit PIN on the keypad We've tried getting the existing unit working with an ATA, but it's only about 50% reliable (hangup not always detected, DTMF not always detected, etc.), so it's probably time to look at fully IP alternatives. Any suggestions gratefully appreciated. There was talk of this a week or 2 ago on the list - look into the archives. I don't think there was anything that successfull though... I have to say though - if you have such an integrated unit that needs nothing more than an analogue connection (and power, presumably), I'd love to know the make - for me, (or rather one of my clients) it would be worthwhile trying to find an ATA that would work with it.. Got a name/website for the opener device? Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime + SIP + MySQL: md5secret BROKEN
On Thu, Jul 24, 2008 at 11:04 AM, Walter Stanish [EMAIL PROTECTED] wrote: If someone could sort out this bug (or let me know if I'm missing something 'obvious' - a hard call with realtime documentation this sparse...) I'd be most grateful, since we require md5secret support to integrate with our existing users database. Welcome to Asterisk! It's highly unlikely you'll find anyone to find the bug for you unless someone is experiencing the same thing. There's no guarantee the bug is actually with Asterisk it could be with your database or somewhere in between. That's not to say it's not with Asterisk but there are a lot of people using realtime with MySQL so if it was a galring bug it would have been seen and logged already. If you do manage to track down the bug it will generally at lest get a response in a short amount of time once it's on the bug tracker. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP door opening devices
On Thu, 24 Jul 2008, Fons van der Beek wrote: Siemens HC 450 Dect intercom does exactly what you want it doesn't come cheap, but works like a dream.. Not avalable in the UK, and there's an intersting comment about being able to trivially take the unit apart with a screwdriver and connect up a battery to open the door.. Hmm.. Gordon Gordon Henderson schreef: On Thu, 24 Jul 2008, Chris Bagnall wrote: Greetings list, We have a client with an analogue door intercom/opening unit which we're attempting to replace with an IP variant. The existing unit has the following functionality: 1) Intercom - visitor hits call, talks to operator 2) Door opening - operator can open the door by dialling a 4-digit PIN followed by * (the door unit interprets the DTMF tones) 3) Door opening - the door unit has a numeric keypad to enable approved persons to enter by entering the 4-digit PIN on the keypad We've tried getting the existing unit working with an ATA, but it's only about 50% reliable (hangup not always detected, DTMF not always detected, etc.), so it's probably time to look at fully IP alternatives. Any suggestions gratefully appreciated. There was talk of this a week or 2 ago on the list - look into the archives. I don't think there was anything that successfull though... I have to say though - if you have such an integrated unit that needs nothing more than an analogue connection (and power, presumably), I'd love to know the make - for me, (or rather one of my clients) it would be worthwhile trying to find an ATA that would work with it.. Got a name/website for the opener device? Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audiocodes MP-11X configuration to work with Asterisk
I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It registers fine and I can call between the MP-114 and other extensions, but I'm not having much luck with the FXO ports. syslog shows the problem to be in the MP-114 configuration. Can anyone help? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP door opening devices
On Thu, 24 Jul 2008, Tzafrir Cohen wrote: On Thu, Jul 24, 2008 at 10:25:34AM +0100, Chris Bagnall wrote: Greetings list, We have a client with an analogue door intercom/opening unit which we're attempting to replace with an IP variant. The existing unit has the following functionality: 1) Intercom - visitor hits call, talks to operator 2) Door opening - operator can open the door by dialling a 4-digit PIN followed by * (the door unit interprets the DTMF tones) 3) Door opening - the door unit has a numeric keypad to enable approved persons to enter by entering the 4-digit PIN on the keypad What would it take to move the logic into Asterisk? (1) is naturally trivial. As for sending the actual signal to open and close the door: that may take a separate out-of-band operation. e.g. System(/usr/sbin/open-sesame) But what about typing the PIN? Hello, you have reached Treasure Cave Inc. If you know the access code, dial it now. If not, press 0 for the operator. The point is to separate communication from door opening. Both are different problems to solve. I'm not sure you will gain much by bundeling them together. I wish Xorcom would provide a separate relay box, rather than add it onto an channel bank :) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk automatic hold
Hi, I want to make an insertion in a communication; A et B are in communication, an other C wants talk to A, how can i set B on hold state and make a call to A?. Thanks. Rachid ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes MP-11X configuration to work withAsterisk
We have a post this morning on VoIPInsider covering Audiocodes gateway configuration with Asterisk and FreeSwitch, you can find it here http://blog.voipsupply.com/technical-advice/setting-up-an-audiocodes-mp- 114118-fxo-with-asterisk-and-freeswitch Cory J Andrews Director, New Market Initiatives VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE [EMAIL PROTECTED] NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Frank Tarczynski Sent: Thursday, July 24, 2008 8:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Audiocodes MP-11X configuration to work withAsterisk I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It registers fine and I can call between the MP-114 and other extensions, but I'm not having much luck with the FXO ports. syslog shows the problem to be in the MP-114 configuration. Can anyone help? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP door opening devices
It's not super cheap, but Cyberdata makes a SIP enabled intercom that is vandal proof and has a dry contact relay built in to actuate a door strike. http://www.cyberdata.net/products/voip/voip-intercom.html Cory J Andrews Director, New Market Initiatives VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE [EMAIL PROTECTED] NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Thursday, July 24, 2008 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IP door opening devices leave the existing keypad there. as for integrating it with asterisk. use an ata with 2 FXS ports. one FXS port connect to a viking door box http://www.vikingelectronics.com/ and set the ATA to do hotline on it. that door box is a regular analog phone in the shape of a door box that when call is pressed it goes offhook hence the requirement of hotline mode. it also has auto answer that when you call the box it goes off hook automaticaly. then use a relay from http://www.mikesandman.com/ that gets activated on ring connect that to the second FXS and that will unlock the door. On 7/24/08, Chris Bagnall [EMAIL PROTECTED] wrote: Greetings list, We have a client with an analogue door intercom/opening unit which we're attempting to replace with an IP variant. The existing unit has the following functionality: 1) Intercom - visitor hits call, talks to operator 2) Door opening - operator can open the door by dialling a 4-digit PIN followed by * (the door unit interprets the DTMF tones) 3) Door opening - the door unit has a numeric keypad to enable approved persons to enter by entering the 4-digit PIN on the keypad We've tried getting the existing unit working with an ATA, but it's only about 50% reliable (hangup not always detected, DTMF not always detected, etc.), so it's probably time to look at fully IP alternatives. Any suggestions gratefully appreciated. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T1/PRI dialing
When dialing using a T1/PRI with a outgoing call files Like Channel: Zap/1/95551212 is there ever a need to delay or pause in there? I have gotten feedback from a customer that instead of dialing the 95551212 it seems to have dialed 55512 which just happened to be an internal extension. So it seems like it missed the 9 so then it only looked at 5 digits internal extension so the last 12 is dropped resulting in 55512 as the number dialed. Does this seem likely? Does changing the outgoing call file and adding w's to loo something like Channel: Zap/1/ww9ww5551212 work on digital lines? I never thought it was needed for digital. I am using zaptel 1.4.11, libpri 1.4.3 and asterisk 1.4.21.1 Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Automatic Redialing feature
Hi, I'm looking to write a dialplan for Automatic Redialing feature,How to ask asterisk to make a automatic re-dial if a channel is busy?? A simple example will be very useful for me. Thanks. Rachid ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Thu, Jul 24, 2008 at 09:23:44AM -0400, Jay R. Ashworth wrote:
So I have these 4 new PRIs turning up tomorrow. Anyone have any
suggestions on some dialplan that I could use to allow me to manually
dial calls out over each channel for testing?
I assume I'd have to make a separate group for each channel in the
/etc/asterisk/zapata.conf? Or could I just specify the channel number
directly in the dialplan and make 24 trunkgroups there with a
dialpattern for each one? (I know enough to be dangerous, but not
quite enough to implement without a little help. :-)
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
--
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Re: [asterisk-users] Acceptance testing of a new PRI
Jay R. Ashworth wrote:
So I have these 4 new PRIs turning up tomorrow. Anyone have any
suggestions on some dialplan that I could use to allow me to manually
dial calls out over each channel for testing?
I use:
exten = _71NXXNXX,1,Read(ZAPLINE|conf-getchannel)
exten = _71NXXNXX,n,Set(CALLERID(number)=734XXX)
exten = _71NXXNXX,n,Dial(ZAP/${ZAPLINE}/${EXTEN:1})
exten = _71NXXNXX,n,NoOP(${DIALSTATUS})
exten = _71NXXNXX,n,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _71NXXNXX,n,Hangup()
Doug
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Re: [asterisk-users] IP door opening devices
leave the existing keypad there. as for integrating it with asterisk. use an ata with 2 FXS ports. one FXS port connect to a viking door box http://www.vikingelectronics.com/ and set the ATA to do hotline on it. that door box is a regular analog phone in the shape of a door box that when call is pressed it goes offhook hence the requirement of hotline mode. it also has auto answer that when you call the box it goes off hook automaticaly. then use a relay from http://www.mikesandman.com/ that gets activated on ring connect that to the second FXS and that will unlock the door. On 7/24/08, Chris Bagnall [EMAIL PROTECTED] wrote: Greetings list, We have a client with an analogue door intercom/opening unit which we're attempting to replace with an IP variant. The existing unit has the following functionality: 1) Intercom - visitor hits call, talks to operator 2) Door opening - operator can open the door by dialling a 4-digit PIN followed by * (the door unit interprets the DTMF tones) 3) Door opening - the door unit has a numeric keypad to enable approved persons to enter by entering the 4-digit PIN on the keypad We've tried getting the existing unit working with an ATA, but it's only about 50% reliable (hangup not always detected, DTMF not always detected, etc.), so it's probably time to look at fully IP alternatives. Any suggestions gratefully appreciated. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP door opening devices
Your using a Linksys right? you can use the fxo port and send DTMF. Chris Bagnall wrote: Greetings list, We have a client with an analogue door intercom/opening unit which we're attempting to replace with an IP variant. The existing unit has the following functionality: 1) Intercom - visitor hits call, talks to operator 2) Door opening - operator can open the door by dialling a 4-digit PIN followed by * (the door unit interprets the DTMF tones) 3) Door opening - the door unit has a numeric keypad to enable approved persons to enter by entering the 4-digit PIN on the keypad We've tried getting the existing unit working with an ATA, but it's only about 50% reliable (hangup not always detected, DTMF not always detected, etc.), so it's probably time to look at fully IP alternatives. Any suggestions gratefully appreciated. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Implementing an Asterisk Server behinda MeridianNorstar
On Thu, Jul 24, 2008 at 12:23 AM, John Faubion [EMAIL PROTECTED] wrote: Well, I am not sure what is needed to interface between the two. I hoped there was something you could use and from the Joseph, Now I'm pretty sure we are not talking about the same things. Let me see if I have the correct picture in my head. I now think you have a Norstar in one office and an asterisk system in another office and want to allow them to send calls between them. Is this correct? Do they make phones with a gig switch in them? I am told there are phones with 100meg switches in them? The new Polycom 670 has a gig interface but at this point I'm not sure why you need that. Are you thinking that if the Norstar phones and lines can't be used, that you would need the phone to have a switch to share the Ethernet connection? Sorry for the confusion but I just want to make sure I know what you need before making a recommendation. John Citel will never get my business again. Their gateways are simply DTAs (digital terminal adapters) and I had nothing but problems with the Definity boxen, maybe they have got better. Heck, you have many options people are passing right over. Asterisk on a modest server with a quad port T1 card and a couple or few Adtran or Adit channel bank populated with the modules you need. Populate the channel bank with however many lines (FXO) coming from the telco and also populate the same channel bank or more with lines (FXS) that connect to your Meridian (I assume you are using POTS lines now), then connect the channel banks to Asterisk via T1. I have done this many many times and it always words great. Some just want additional functionality, others want a slower migration path and add SIP phones with new hires or when old phones break. Some that don't want VoIP on their LAN or only have CAT3 and others that run the FXS ports (single pair POTS) to the work stations so they can use any kind of regular analog phone that they want. There are place where a free after rebate phone is called for such as the lunch room. There are also some very nice analog handsets. The migration does not have to happen all at once, you can take it slow, make it invisible to the end user, start using VoIP trunks and all that Asterisk has to offer, and have a super flexible migration path. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recordings...
Could you explain further? On Thu, Jul 24, 2008 at 4:13 AM, Gregory Malsack [EMAIL PROTECTED] wrote: I resolved this problem. The key was to get the right combination of self/callee and peer/caller. Read the instructions regarding the application map very closely. My problem was that I was not running the StopMixMonitor command against the proper channel. Even though mixmonitor records both channels simultaneously, the recording is only assigned to 1 channel and you have to run the command against the originating channel of the call. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Malsack Sent: Wednesday, July 23, 2008 2:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Recordings... I'm getting close. The idea is based on the same principal as the link below. Here's what I have done thus far: All calls are recorded via mixmonitor. This is part of the initial dialplan when the call comes in. I then created an application map key sequence that is supposed to run stopmixmonitor. However I am unable to locate examples of syntax on that command. Here is what I have: stoprecording = *8,self/callee,StopMixMonitor, This command syntax does not work and the recording continues on. Can anyone provide direction on this? Thanks, Gregory Malsack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Tuesday, July 22, 2008 4:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Recordings... Am Dienstag, den 22.07.2008, 14:53 -0500 schrieb Gregory Malsack: Hello, My boss is asking me to setup the asterisk server to record all calls. (Simple). However, he wants to be able to enter a key sequence during the call to stop the recording. Any ideas on how I would do that? Hi Gregory, I found something about recording at http://www.voip-info.org/wiki/view/Asterisk+config+features.conf (second example). If you combine that with a default_recording_enabled (Monitor() call before Dial(), I would expect), that could be used to turn _off_ recording by pressing a key. I would not know though how to protect against the external call party pressing the same key. Best regards Anselm No virus found in this incoming message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.5/1568 - Release Date: 7/23/2008 6:55 AM No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.5/1568 - Release Date: 7/23/2008 6:55 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.5/1568 - Release Date: 7/23/2008 6:55 AM No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.5/1568 - Release Date: 7/23/2008 6:55 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Thu, Jul 24, 2008 at 09:39:42AM -0400, Doug Lytle wrote:
Jay R. Ashworth wrote:
So I have these 4 new PRIs turning up tomorrow. Anyone have any
suggestions on some dialplan that I could use to allow me to manually
dial calls out over each channel for testing?
I use:
exten = _71NXXNXX,1,Read(ZAPLINE|conf-getchannel)
exten = _71NXXNXX,n,Set(CALLERID(number)=734XXX)
exten = _71NXXNXX,n,Dial(ZAP/${ZAPLINE}/${EXTEN:1})
exten = _71NXXNXX,n,NoOP(${DIALSTATUS})
exten = _71NXXNXX,n,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _71NXXNXX,n,Hangup()
Nice. I assume the Noop's capture the text in the log, then? (See?
Told you I was fresh caught :-)
Hold it: how do I specify the channel? Ah, no, I see what you're
doing. I wanted to actually dial the channel number.
I came up with this:
; dial a long-distance call; allow the user to select a Zap channel
manually
exten = _88XX1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _88XX1NXXNXX,2,Dial(Zap/${EXTEN:3:2}-1/${EXTEN:4},30,o)
exten = _88XX1NXXNXX,3,Hangup
But I'll add the noops.
Course I have to fix the dialplan in my Poly600, too.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
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[asterisk-users] Acceptance testing of a new PRI
So I have these 4 new PRIs turning up tomorrow. Anyone have any suggestions on some dialplan that I could use to allow me to manually dial calls out over each channel for testing? I assume I'd have to make a separate group for each channel in the /etc/asterisk/zapata.conf? Or could I just specify the channel number directly in the dialplan and make 24 trunkgroups there with a dialpattern for each one? (I know enough to be dangerous, but not quite enough to implement without a little help. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly my Asterisk Box Hanged up all calls
On Wed, Jul 23, 2008 at 06:19:58PM +0200, Philipp Kempgen wrote: While it may sound rude that's absolutely correct. As a software developer in many cases you are more or less sure that an issue has already been solved so you expect the user to upgrade to the latest version or at least to the latest minor version. Having to hunt down problems in old versions is annoying especially for issues that have probably already been addressed. Not arguing. But please note the recently added section at the end of How To Ask Questions The Smart Way that says, in effect, Please Don't Bite The Newbies. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Thursday 24 July 2008 10:30:26 Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 09:39:42AM -0400, Doug Lytle wrote:
Jay R. Ashworth wrote:
So I have these 4 new PRIs turning up tomorrow. Anyone have any
suggestions on some dialplan that I could use to allow me to manually
dial calls out over each channel for testing?
I use:
exten = _71NXXNXX,1,Read(ZAPLINE|conf-getchannel)
exten = _71NXXNXX,n,Set(CALLERID(number)=734XXX)
exten = _71NXXNXX,n,Dial(ZAP/${ZAPLINE}/${EXTEN:1})
exten = _71NXXNXX,n,NoOP(${DIALSTATUS})
exten = _71NXXNXX,n,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _71NXXNXX,n,Hangup()
Nice. I assume the Noop's capture the text in the log, then? (See?
Told you I was fresh caught :-)
NoOp doesn't capture anything, unless you have Verbose logging turned on and
the verbose level is high enough (3 or higher). If you want direct logging,
use the Log() application in 1.6.
--
Tilghman
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[asterisk-users] CallerId show with IP address appended
Hello, Asterisk 1.4.21.1 Well it seems like my month for questions. I have a situation where the CallerID num shows as [EMAIL PROTECTED](the ip of the asterisk box) on calls to any of the internal phones. This prevents the ability to dial out from the missed call list. I have not been able to find out why this is happing. To further confuse the issue when i register and extension to the public IP from outside the firewall I get only 16035551212 as the clid. I have several NoOps in the dial plan and they all show the clid as 16035551212, which is also what is in the cdr, but when it gets to the Polycom it has the IP appended. The phones are all polycoms but have also tested with x-lite and it gets the ip appended also. Any pointers as to where to look? JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect Asterisk PBX to Traditional PBX and retain functionality
If I understand you, then yes you can. I do this now. All our telco lines come through our analog NEC phone switch and then through FXO/ FXS ports to my Asterisk. Asterisk handles voicemail and the menu system so when somebody dials 6 to get my extension the asterisk does the following: Flash(); Wait(0.4); SendDTMF(268); Hangup(); I added the Wait(0.4) as I found that under heavy load the NEC would not catch the first DTMF digit after the Flash. This solution has worked for us for over a year now. Some bonus information that may or may not be relevant to what you are doing: We have a few SIP phones that we needed to be able to do the same kind of thing. We couldn't flash transfer to the Asterisk, but in the NEC I setup a outgoing trunk line (dial 8 to access) that goes to the Asterisk box. Then I setup a forward all calls on extension 268 (when I have my SIP phone active) to dial out to 8268. That way when somebody calls my extension it automatically forwards then to extension 268 on the Asterisk box. Daniel On Jul 23, 2008, at 3:57 PM, Ricardo Melendez wrote: Hi to All, I have a PBX (MAINPBX) from a Telecomm Provider, which have the feature to transfer calls (Incoming call - Answer - FLASH - Dial Number to transfer - Answer - FLASH+4) and the call is transferred, but I have the need to implement an internal ACD using Asterisk as the PBX, the trunks connected to my Asterisk FXO ports are Extensions of my MAINPBX (ex., 5437, 5440 etc), all features work fine, but I have the need to make asterisk act as a normal telephone when transferring calls, I need to release the line (FXO port in my Asterisk) and make the transfer via the MAINPBX feature. Otherwise I will use 2 lines of my Asterisk PBX to make the transfer and it reduce the incoming lines available for my ACD. It’s possible send the commands FLASH, FLASH+4 using the incoming line to my MAINPBX via Asterisk like a normal telephone? Thanks in Advance. Ricardo Melendez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Implementing an Asterisk Server behinda MeridianNorstar
The migration does not have to happen all at once, you can take it slow, make it invisible to the end user, start using VoIP trunks and all that Asterisk has to offer, and have a super flexible migration path. Steve, Lots of good info! So if I put a T1 card in an Asterisk Server, and a T1 card in the Norstar How does a user on the Norstar dial 221 and reach a voip only user connected to asterisk via ip only? That assumes as you mentioned new users are added as voip users in the future? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday at 12 Noon EDT (9 AM Pacific) Asterisk and VoIP User Groups Worldwide
This is a recovery from last week's fiasco. Tech issues prevented the conference from having our full complement of voices. If you are in an Asterisk Users Group, you'll want John Todd to know about it. If nothing else, he may get you a Digium beachball or my personal favorite, the Digium screwdriver that saved my life. Please join this call and weigh in with what a user group means to you, why they're important, how Digium can help you find other interested users in your area. After all, it's in their interest to foster your group's growth, just as they have with our weekly VUC. To listen or participate: http://bit.ly/voip a little before 12 Noon EDT (9AM PDT, 11 Central, 4PM GMT) phone sip:[EMAIL PROTECTED] or PSTN: Call (724) 444-7444 Enter 22622# then 1# or your PIN if you registered at Talkshoe. There are several guest PIN available if you grab my attention on the IRC channel below. The PIN lets me know who's speaking. IRC is on Freenode.net #voip-users-conference Forums, blogs, etc: http://bit.ly/voipusers Recordings: http://bit.ly/archives If http://bit.ly is down, just see http://voipusersconference.org for the info :) no animals were used in creating or testing of this message ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue show name - callerID
There's not any direct way of which I am aware in a single command, but from
the shell you could do the following (and yes, this is a bit of a hack):
for i in `rasterisk -x queue show |grep wait |awk -F '{print $2}'`; do
rasterisk -x core show channel $i | grep Caller ID;done
That will return (in order) the calls in your queue, their caller-ids and
caller id names. If the caller-id and caller-id-name are the same, each
entry will just be 2 repeating lines.
To skip the second entry (caller-id name), you can just add a colon to the
last grep command, like so:
for i in `rasterisk -x queue show |grep wait |awk -F '{print $2}'`; do
rasterisk -x core show channel $i | grep Caller ID:;done
What this does, essentially, is check the callers' channels in the queue,
and then check each individual channel for the caller-id.
Best regards,
Örn
On Tue, Jul 1, 2008 at 7:20 PM, Marcin J. Kowalczyk
[EMAIL PROTECTED] wrote:
Hi,
Is there a way to show callerID of calls waiting in queue?
queue show
shows only channel not callerID
Cheers,
Marcin
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Re: [asterisk-users] Realtime + SIP + MySQL: md5secret BROKEN
If someone could sort out this bug (or let me know if I'm missing something 'obvious' - a hard call with realtime documentation this sparse...) I'd be most grateful, since we require md5secret support to integrate with our existing users database. Welcome to Asterisk! It's highly unlikely you'll find anyone to find the bug for you unless someone is experiencing the same thing. A quick google search for the issue reveals a number people have run in to something similar before, some report 'after upgrade', others seem to be just trying to get realtime working for the first time. http://www.google.com/search?q=db.c+unable+to+find+key+asterisk There's no guarantee the bug is actually with Asterisk it could be with your database or somewhere in between. This seems unlikely for the following reasons. 1. I can trigger the same SQL queries via the console via realtime load sippeers username walter and they return data fine. 2. I have tried upgrading asterisk to root SQL access, with no difference in asterisk's behaviour. 3. The bug is reliably controlled by the presence of the single column 'md5secret' That's not to say it's not with Asterisk but there are a lot of people using realtime with MySQL so if it was a galring bug it would have been seen and logged already. Could I see a show of hands for who's using asterisk realtime with mysel and md5secret? If you could post your asterisk / asterisk-addons / mysql versions too that'd be great. I don't mind changing all three of these if someone's found a stable combination. If you do manage to track down the bug it will generally at lest get a response in a short amount of time once it's on the bug tracker. I'm happy to post the current material to the bug tracker if it's considered enough information for a report. Regards, Walter Stanish Owner / Director Occident Systems (+86 15808 700 801) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco Call Manager to Asterisk conversion
I need to replace a cisco call manager with an asterisk box. Phones are cisco 7940 and 7910. I know the 40's can use SIP but the 7910's have to use the skinny/sccp driver. Its been quite awhile since I did anything with asterisk, so I am looking for some assistance with the configuration and am willing to pay. Its a basic setup, 30+ phones, incoming lines via PRI, 1 dial plan for incoming and outgoing - nothing fancy there, voicemail for each phone and DID number for each phone. -- Chad Whitten [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel 1.4/1.2 on RHEL3 (wa s: Zaptel won´t recognizes sources installed)
Hi, On Thu, Jul 24, 2008 at 10:53:59AM +0300, Tzafrir Cohen wrote: Specifically one of the many RPM packages Axel Thimm maintains is Zaptel, and is also vs. RHEL3: http://atrpms.net/dist/el3/zaptel/ . He reported several breakages in the past (which were fixed). I see that the latest version there is 1.4.11 . Actually it's just 1.4.7.1. :( Newer zaptels break on RHEL3 with (1.4.11) In file included from base.c:46: vpm450m.h:27:28: linux/firmware.h: No such file or directory In file included from base.c:46: vpm450m.h:36: warning: `struct firmware' declared inside parameter list vpm450m.h:36: warning: its scope is only this definition or declaration, which is probably not what you want base.c: In function `t4_shutdown': base.c:1418: warning: implicit declaration of function `msleep' base.c: In function `t4_interrupt_gen2': base.c:2916: warning: implicit declaration of function `IRQ_RETVAL' base.c:2916: warning: `return' with a value, in function returning void base.c: In function `t4_vpm450_init': base.c:3121: storage size of `embedded_firmware' isn't known base.c:3200: warning: passing arg 4 of `init_vpm450m' from incompatible pointer type base.c:3203: warning: implicit declaration of function `release_firmware' base.c:3121: warning: unused variable `embedded_firmware' (1.2.26) zaptel-base.c: In function `calc_fcs': zaptel-base.c:734: `fcstab' undeclared (first use in this function) zaptel-base.c:734: (Each undeclared identifier is reported only once zaptel-base.c:734: for each function it appears in.) zaptel-base.c: In function `__zt_putbuf_chunk': zaptel-base.c:6003: `fcstab' undeclared (first use in this function) -- Axel.Thimm at ATrpms.net pgpYDenkUGEi2.pgp Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Raw asterisk x FreePbx .conf
my best offer to you is to read more about the dial plan to understand what happens.. or try to understand what does freepbx do and what does it write and understand the applications.. Date: Wed, 23 Jul 2008 20:53:45 -0300 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] Raw asterisk x FreePbx .conf All my experience on asterisk was gained through configuring Trixbox, so a ready to go Asterisk system. Now i´m trying to install a server from scratch, so this question arose. When installing freepbx, the .conf files are written, when installing asterisk nothing is written unless I run the Make Samples. Basic features like parking calls, transfering calls, contexts, and so on have to be code written by hand, line by line, if I don´t have the freepbx or the make samples to write it for me?What exactly does the make samples command writes to asterisk? Thanks, Felipe _ Keep your kids safer online with Windows Live Family Safety. http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_family_safety_072008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco vs Asterisk
I'm a CCIE and CCVP. I have worked in the Cisco TSBU on both CCM and Telepresence systems I have two IP patents for the VoiP Lite protocols and have been designing and building OSS IPBXs for companies including Google going back to 2001. I'm not mentioning any of that to be jerk I mentioned it to say I'm as qualified as anyone to to compare the CCM and OSS servers. The only fair way to compare the two is a list of weights features, for example if cost is your biggest feature then OSS is better, if support is your biggest feature than Cisco wins. When a customer is comparing the costly (TCO) and best supported systems in the world with hundreds of thousands installed systems for the large global companies on the planted backed by 54,000 employees and over $25b in the bank vs, a FREE system with one layer of support maybe two layers of support, the features don't even come in the evaluation in my opinion. I once asked a manager why did you buy the CCM and he said no one ever got fired for buying Cisco if anything wrong, If push the OSS and it goes I could loose my job. I would get a list of the important features, because there is no answer to your question of which is better. - Original Message - From: Benoit Plessis To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco vs Asterisk Date: Tue, 22 Jul 2008 15:10:50 +0200 voip crazy a écrit : Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? To answer your questions, one would need to know what exactly are all the functionalities of a Cisco Unity server, and more specificaly, what are the needs of your client. But i'm pretty sure the voip-info wiki can answer the asterisk part... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco vs Asterisk
My son owns compoanyn here in San Jose and when a customers says they want Cisco be provides Cisco phones with OSS PBX, it seems to work the lower cost and Cisco phone on the desktop. - Original Message - From: Steve Totaro To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco vs Asterisk Date: Tue, 22 Jul 2008 08:59:24 -0400 On Tue, Jul 22, 2008 at 8:52 AM, voip crazy wrote: Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? Thanks. VoipCrazy. You said migrate to a Cisco, what do they have now? Sell them all Cisco. You will make more money and great residual income for MACs ;-) Anyways, you could ditch the Cisco entirely and use Asterisk. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco vs Asterisk
T G wrote: I'm a CCIE and CCVP. I have worked in the Cisco TSBU on both CCM and Telepresence systems I have two IP patents for the VoiP Lite protocols and have been designing and building OSS IPBXs for companies including Google going back to 2001. I'm not mentioning any of that to be jerk I mentioned it to say I'm as qualified as anyone to to compare the CCM and OSS servers. The only fair way to compare the two is a list of weights features, for example if cost is your biggest feature then OSS is better, if support is your biggest feature than Cisco wins. When a customer is comparing the costly (TCO) and best supported systems in the world with hundreds of thousands installed systems for the large global companies on the planted backed by 54,000 employees and over $25b in the bank vs, a FREE system with one layer of support maybe two layers of support, the features don't even come in the evaluation in my opinion. I once asked a manager why did you buy the CCM and he said no one ever got fired for buying Cisco if anything wrong, If push the OSS and it goes I could loose my job. I would get a list of the important features, because there is no answer to your question of which is better. Yet amazingly (if this is, indeed, a source of amazement for you), CCM and other Cisco software can be just as buggy as anything OSS, if not worse. Depending on how critical the bugs or other support exigencies, the TCO can be driven way up. Except with the OSS community, you report the bug, and usually get a quick fix - even if it's a significant issue for you, not necessarily most of the installed base. If by chance that proves not to be the case, the source code is available, and you can fix it yourself. With Cisco, you pay for expensive support and get to file some complaint with the TAC. Yay. There are many, many angles from which onec an look at this in one's TCO / OPEX formula. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Implementing an Asterisk Server behinda MeridianNorstar
Steve, Lots of good info! So if I put a T1 card in an Asterisk Server, and a T1 card in the Norstar How does a user on the Norstar dial 221 and reach a voip only user connected to asterisk via ip only? That assumes as you mentioned new users are added as voip users in the future? Have the Norstar programmer send all 3 digit, unused extensions to the PRI. Then Asterisk will see 221, etc. and can handle at your dialplan sees fit. Retaining all NXX, NXXNXX, 1NXXNXX etc to the standard treatment they receive now. --- dbc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP door opening devices
See ITS at www.its-tel.com The Pantel and Pancode IP are what you are looking for. Rupert Utteridge Director - Sales Marketing Digital Techniques (Australia) Pty Ltd 4 The Lee Middle Cove, NSW, 2068 Australia Tel: +61 2 9037 4191 Mobile: +61 424 373 516 Web: www.dtasia.com.au ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco vs Asterisk
T G wrote: I'm a CCIE and CCVP. I have worked in the Cisco TSBU on both CCM and Telepresence systems I have two IP patents for the VoiP Lite protocols and have been designing and building OSS IPBXs for companies including Google going back to 2001. I'm not mentioning any of that to be jerk I mentioned it to say I'm as qualified as anyone to to compare the CCM and OSS servers. The only fair way to compare the two is a list of weights features, for example if cost is your biggest feature then OSS is better, if support is your biggest feature than Cisco wins. When a customer is comparing the costly (TCO) and best supported systems in the world with hundreds of thousands installed systems for the large global companies on the planted backed by 54,000 employees and over $25b in the bank vs, a FREE system with one layer of support maybe two layers of support, the features don't even come in the evaluation in my opinion. I once asked a manager why did you buy the CCM and he said no one ever got fired for buying Cisco if anything wrong, If push the OSS and it goes I could loose my job. I would get a list of the important features, because there is no answer to your question of which is better. What you mentioned above is mostly correct presuming you are referencing OSS being provided by an organisation with limited resources and perhaps limited experience in OS. Spin that into a perspective of a well organised company harvesting full potential of OS, adding its own proprietary software level allowing it to offer value products and EXCELLENT support, then I will strongly disagree with you. In particular where customer solution isn't just a solution, but rather its products and people becomes your business's communications partner. Senad www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Implementing an Asterisk Server behi nda MeridianNorstar
On July 24, 2008 04:42:42 pm David Cook wrote: Have the Norstar programmer send all 3 digit, unused extensions to the PRI. Then Asterisk will see 221, etc. and can handle at your dialplan sees fit. Yes, this works, but you won't be able to treat those as regular extensions; the Nortel will treat them as external numbers. -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble Playing message file via Perl AGI
On Thursday 24 July 2008 12:58:29 am Steve Edwards wrote: The agi debug command (1.2) would have shown you where you violated the protocol. Nice to know... -- Mike Diehl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Click to Dial
I have a question about click to dial. Each of my users is going to have a VOIP phone with an assigned extension. Is there a simple way to build a web-based speed-dial list that will allow them to put in their extension, click on the number they want to dial, and have asterisk ring their phone, then as soon as they pick up, start dialing the number from the speed-dial? Thanks, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk automatic hold
So you basically want a call-interrupt feature that puts the interrupted party on hold? rachid wrote: Hi, I want to make an insertion in a communication; A et B are in communication, an other C wants talk to A, how can i set B on hold state and make a call to A?. Thanks. Rachid ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click to Dial
Hello, On Jul/24/2008, Brent Davidson wrote: I have a question about click to dial. Each of my users is going to have a VOIP phone with an assigned extension. Is there a simple way to build a web-based speed-dial list that will allow them to put in their extension, click on the number they want to dial, and have asterisk ring their phone, then as soon as they pick up, start dialing the number from the speed-dial? I think that a good start could be: http://lexatel.com/en/22/Whitepapers (currently, there is only one Whitepaper about click to dial, but of course you would need to do some frontend... easy to do, if you need any help feel free to ask). We are preparing some other Whitepaper too... Sorry for the semi-Spam :-) -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click to Dial
Yep you just build a 'log in' query (eg to identify which extension to send Leg A to or you can just build it into the url with a unique extension id) and then list all the extensions you want (obviously if it's company wide then it will be the same for all - only the Leg A will be different). Then build the url's to generate call files Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: Thursday, 24 July 2008 5:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Click to Dial I have a question about click to dial. Each of my users is going to have a VOIP phone with an assigned extension. Is there a simple way to build a web-based speed-dial list that will allow them to put in their extension, click on the number they want to dial, and have asterisk ring their phone, then as soon as they pick up, start dialing the number from the speed-dial? Thanks, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] increase ring time out
- Fidel Garcia [EMAIL PROTECTED] wrote:
Where exactly do I have to change it?
The GUI on the AA50 generates users via users.conf. These users are added into
the dialplan automatically and are placed into the default context. Calls to
the users are made via the stdexten macro. In that macro is a Dial statement
with a timeout of 20. You would have to adjust that timeout manually and save
it off (Run the save_config script)
One caveat is that the AA50 is not supported when you manually modify the dial
plan. The changes you make are at your own risk.
- Doug Bailey
This is the extensions.conf file:
;! Automatically generated configuration file
;! Filename: extensions.conf (/etc/asterisk/extensions.conf)
;! Generator: Manager
;! Creation Date: Tue Jul 22 15:14:28 2008
;!
[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
priorityjumping = no
[globals]
trunk_1 = Zap/g1
trunk_1_cid = asreceived
[dundi-e164-canonical]
[dundi-e164-customers]
[dundi-e164-via-pstn]
[dundi-e164-local]
include = dundi-e164-canonical
include = dundi-e164-customers
include = dundi-e164-via-pstn
[dundi-e164-switch]
switch = DUNDi/e164
[dundi-e164-lookup]
include = dundi-e164-local
include = dundi-e164-switch
[macro-dundi-e164]
exten = s,1,Goto(${ARG1},1)
include = dundi-e164-lookup
[macro-trunkdial]
exten = s,1,set(CALLERID(all)=${IF(${LEN(${CALLERID(num)})} 6 ?
${CALLERID(al
l)} : ${ARG2})})
exten = s,n,Dial(${ARG1})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Hangup
exten = _s-.,1,NoOp
[iaxtel700]
exten =
_91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[iaxprovider]
[trunkint]
exten = _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten = _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
exten = _91NXXNXX,1,Macro(dundi-e164,${EXTEN:1})
exten = _91NXXNXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
exten = _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[international]
ignorepat = 9
include = longdistance
include = trunkint
[longdistance]
ignorepat = 9
include = local
include = trunkld
[local]
ignorepat = 9
include = default
include = parkedcalls
include = trunklocal
include = iaxtel700
include = trunktollfree
include = iaxprovider
[macro-stdexten]
exten = s,1,Dial(${ARG2},20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[macro-stdPrivacyexten]
exten = s,1,Dial(${ARG2},20|p)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = s-DONTCALL,1,Goto(${ARG3},s,1)
exten = s-TORTURE,1,Goto(${ARG4},s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[macro-page]
exten = s,1,ChanIsAvail(${ARG1}|js)
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
exten = s,n(autoanswer),Set(_ALERT_INFO=RA)
exten = s,n,SIPAddHeader(Call-Info: Answer-After=0)
exten = s,n,NoOp()
exten = s,n,Dial(${ARG1}||)
exten = s,n(fail),Hangup
[demo]
exten = s,1,Wait(1)
exten = s,n,Answer
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n(restart),BackGround(demo-congrats)
exten = s,n(instruct),BackGround(demo-instruct)
exten = s,n,WaitExten
exten = 2,1,BackGround(demo-moreinfo)
exten = 2,n,Goto(s,instruct)
exten = 3,1,Set(LANGUAGE()=fr)
exten = 3,n,Goto(s,restart)
exten = 1000,1,Goto(default,s,1)
exten = 1234,1,Playback(transfer,skip)
exten = 1234,n,Macro(stdexten,1234,${CONSOLE})
exten = 1235,1,Voicemail(u1234)
exten = 1236,1,Dial(Console/dsp)
exten = 1236,n,Voicemail(u1234)
exten = #,1,Playback(demo-thanks)
exten = #,n,Hangup
exten = t,1,Goto(#,1)
exten = i,1,Playback(invalid)
exten = 500,1,Playback(demo-abouttotry)
exten = 500,n,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = 500,n,Playback(demo-nogo)
exten = 500,n,Goto(s,6)
exten = 600,1,Playback(demo-echotest)
exten = 600,n,Echo
exten = 600,n,Playback(demo-echodone)
exten = 600,n,Goto(s,6)
exten = 76245,1,Macro(page,SIP/Grandstream1)
exten = _7XXX,1,Macro(page,SIP/${EXTEN})
exten = 7999,1,Set(TIMEOUT(absolute)=60)
exten =
7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED]/n
|d)
exten = 8500,1,VoicemailMain
exten = 8500,n,Goto(s,6)
[page]
exten = _X.,1,Macro(page,SIP/${EXTEN})
[default]
exten =
Re: [asterisk-users] Click to Dial
this simple php script do whatr you need
should be called by your user pc with url ike
http://ip_your_Asterisk_host/chiama_ora.php?INT=xxNOME=yyNUMERO=xxxCONTESTO=xxx
if you are interested I have developped a complete phonebook
integrated wwith asterisk
the main functions are
- multiple listings user/company
incoming calls lookup in listings and pop up to user plsu
tracking,recording, reporting
outgoing call reservation time date with tracking
etc...
hope this help
?php
---
parameters to send via request
INT = extension
NOME = called name
NUMERO = called number
CONTESTO = sterisk contexts to be used
**/
//MUST BE CUSTOMIZED
define(CALL_PREFIX,0);
//controllo parametri
if ( !isset($_REQUEST['INT']) )
{
die(chiama errore : interno non prevenuto);
}
else
$interno=$_REQUEST['INT'];
if ( isset($_REQUEST['NOME']) )
$nome=$_REQUEST['NOME'];
else
$nome=;
if ( !isset($_REQUEST['NUMERO']) )
{
die(chiama errore : NUMERO non prevenuto);
}
else
$numero=$_REQUEST['NUMERO'];
if ( !isset($_REQUEST['CONTESTO']) )
{
die(chiama errore : contesto non prevenuto);
}
else
$contesto=$_REQUEST['CONTESTO'];
//impostazione valori
$strChannel = SIP/$interno;
$strExten = CALL_PREFIX .$numero;
$delay=0;
$strContext = $contesto;
$strHost = localhost;
$strUser = admin;
$strSecret = amp111;
$strWaitTime = 05;
$strPriority = 1;
$strMaxRetry = 3;
$strCallerId = ATTIVA:$nome;
$oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or
die(Connection to host failed);
fputs($oSocket, Action: login\r\n);
fputs($oSocket, Events: off\r\n);
fputs($oSocket, Username: $strUser\r\n);
fputs($oSocket, Secret: $strSecret\r\n\r\n);
fputs($oSocket, Action: originate\r\n);
fputs($oSocket, Channel: $strChannel\r\n);
fputs($oSocket, WaitTime: $strWaitTime\r\n);
fputs($oSocket, CallerId: $strCallerId\r\n);
fputs($oSocket, Exten: $strExten\r\n);
fputs($oSocket, Context: $strContext\r\n);
fputs($oSocket, Priority: $strPriority\r\n\r\n);
fputs($oSocket, Action: Logoff\r\n\r\n);
stream_set_blocking($oSocket,0);
$out='';
while (!feof($oSocket)) {
$out .= fgets($oSocket, 128);
}
fclose($oSocket);
print $out;
?
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Re: [asterisk-users] Implementing an Asterisk Server behind a Meridian Norstar
for a similar project I used astandard FXS interface connected to one extension connector (RJ) of the legacy pbx and I could end some commands to he legacy pbx by flashing the line and then send appropriate dtmf bye 2008/7/23 Joseph L. Casale [EMAIL PROTECTED]: We have an older Meridian Norstar system and are thinking of using Asterisk behind it to use a SIP Voip Provider instead of our local telco. Does anyone make an interface card that can integrate with the digital input of the Meridian. Not the optimal solution, but it allows for the current infrastructure to be retained. Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] different gains per channel?
Hi all, How do I set different rx and tx gains for each channel? in my zapata.conf file I have a heading [trunkgroups] and then [channels] under this I have information such as language context signalling etc and also rxgain and txgain. My assumption is that these settings are used for all channels (I have three zap channels). I need to have different gain settings on each channel. Is this easy to achieve? Many thanks Kate ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] finding out on hold channels
I noticed that i' m not getting any manager event for hold and unhold of a channel. is this normal? Also is there any easy way through either CLI or manager to find out which one of the channels are on hold? I checked show channels that did not show a channel being on hold or not, also sip show channels does show that but it has call id instead of channel id. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco vs Asterisk
You are mentionning very particular case here, a company with a very strict hierarchy, where a new ideas and solutions are not advised, i think that in the past they used cisco who has some issues from time to time, and they are prepared for that, but new name scares them, and sometimes people use OSS and forget about support, and then when issue arrives, they claim 'OSS' is bad. That experience acumulates, and we are getting scared managers ;) Dont forget to sign a support contract to avoid crying after. 2008/7/24 T G [EMAIL PROTECTED]: I'm a CCIE and CCVP. I have worked in the Cisco TSBU on both CCM and Telepresence systems I have two IP patents for the VoiP Lite protocols and have been designing and building OSS IPBXs for companies including Google going back to 2001. I'm not mentioning any of that to be jerk I mentioned it to say I'm as qualified as anyone to to compare the CCM and OSS servers. The only fair way to compare the two is a list of weights features, for example if cost is your biggest feature then OSS is better, if support is your biggest feature than Cisco wins. When a customer is comparing the costly (TCO) and best supported systems in the world with hundreds of thousands installed systems for the large global companies on the planted backed by 54,000 employees and over $25b in the bank vs, a FREE system with one layer of support maybe two layers of support, the features don't even come in the evaluation in my opinion. I once asked a manager why did you buy the CCM and he said no one ever got fired for buying Cisco if anything wrong, If push the OSS and it goes I could loose my job. I would get a list of the important features, because there is no answer to your question of which is better. - Original Message - From: Benoit Plessis To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco vs Asterisk Date: Tue, 22 Jul 2008 15:10:50 +0200 voip crazy a écrit : Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? To answer your questions, one would need to know what exactly are all the functionalities of a Cisco Unity server, and more specificaly, what are the needs of your client. But i'm pretty sure the voip-info wiki can answer the asterisk part... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a *Free* Account at www.mail.com http://www.mail.com/Product.aspx! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Call Manager to Asterisk conversion
Search someone in local area, remote configuration of server is possible but configuring the phones is more difficult, you need someone to load firmwares, ect 2008/7/24 Chad Whitten [EMAIL PROTECTED]: I need to replace a cisco call manager with an asterisk box. Phones are cisco 7940 and 7910. I know the 40's can use SIP but the 7910's have to use the skinny/sccp driver. Its been quite awhile since I did anything with asterisk, so I am looking for some assistance with the configuration and am willing to pay. Its a basic setup, 30+ phones, incoming lines via PRI, 1 dial plan for incoming and outgoing - nothing fancy there, voicemail for each phone and DID number for each phone. -- Chad Whitten [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco vs Asterisk
I agree, No manager gets fired even if a Cisco Call Manager goes south. that's not the case with Asterisk. With limited experience that i have with both, i hit more bugs using Asterisk than a CCM, but this is not relevant to your final answer. If you can afford CCM, and you can live with less flexibility and features, i would choose Cisco. If you prefer to have cheaper solution and more features and flexibility, Asterisk is good. With Cisco, everything is cisco, handsets are designed for Cisco, it connects to Exchange much more in depth than even microsoft response point. unlike Asterisk, unfortunately exchange integration is not something you may get in close future and that can be a deal breaker for some companies, but you dont pay per seat license. and so on. On Thu, Jul 24, 2008 at 2:56 PM, Senad Jordanovic [EMAIL PROTECTED] wrote: T G wrote: I'm a CCIE and CCVP. I have worked in the Cisco TSBU on both CCM and Telepresence systems I have two IP patents for the VoiP Lite protocols and have been designing and building OSS IPBXs for companies including Google going back to 2001. I'm not mentioning any of that to be jerk I mentioned it to say I'm as qualified as anyone to to compare the CCM and OSS servers. The only fair way to compare the two is a list of weights features, for example if cost is your biggest feature then OSS is better, if support is your biggest feature than Cisco wins. When a customer is comparing the costly (TCO) and best supported systems in the world with hundreds of thousands installed systems for the large global companies on the planted backed by 54,000 employees and over $25b in the bank vs, a FREE system with one layer of support maybe two layers of support, the features don't even come in the evaluation in my opinion. I once asked a manager why did you buy the CCM and he said no one ever got fired for buying Cisco if anything wrong, If push the OSS and it goes I could loose my job. I would get a list of the important features, because there is no answer to your question of which is better. What you mentioned above is mostly correct presuming you are referencing OSS being provided by an organisation with limited resources and perhaps limited experience in OS. Spin that into a perspective of a well organised company harvesting full potential of OS, adding its own proprietary software level allowing it to offer value products and EXCELLENT support, then I will strongly disagree with you. In particular where customer solution isn't just a solution, but rather its products and people becomes your business's communications partner. Senad www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Arabic IVR
Hello, I want to use an arabic TTS on asterisk. Do you know any arabic TTS Open Source supported by an amd 64? Because I found Mbrolla a free TTS that include arabic and that you can combine it with Festival but it doesn't support and amd64 Thanks' _ Découvrez Windows Live Spaces et créez votre site Web perso en quelques clics ! http://spaces.live.com/signup.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
