Re: [asterisk-users] Call Placed through Manager connecting before the call connects.
I am using Asterisk 1.4.19.2 on Fedora Core 8, with a Sangoma A200 card. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Mailing List Asterisk asterisk-users@lists.digium.com Sent: Thursday, May 22, 2008 6:16:17 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: [asterisk-users] Call Placed through Manager connecting before the call connects. Hello, I am trying to place call through the Manager, using the Zap Card the call connect to the designated Extension before the call is actually Answered by someone or the Voicemail. The message that I am sending is Action: Originate Channel: ZAP/G0/1XX MaxRetries: 0 Context: Test Exten: 6563 Priority: 1 CallerID: TEST 1234 The Events that I get from Manger are 1. Newchannel 2. Newcallerid 3. Newcallerid 4. Newstate [Here State is changed to Dialing] 5. Newstate [Here State is changed to Up] 6. Newexten [Here call is bridged to 6563] Once the call is Bridged to 6563, the phone is actually not Answered, you can hear the Ring on the Phone after Bridging. If I try the same for SIP channel I get addition events as Ringing. I want to play a message once the call connects, In this case the message is Played while the phone is Ringing. Please help. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Placed through Manager connecting before the call connects.
I have noticed the same on the CLI while calling out Directly, the CLI does not show Ringing event.. -- Executing [EMAIL PROTECTED]:1] Dial(SIP/sanjay-09a0a970, ZAP/G0/1 XX ) -- Called G0/1 XX -- Zap/4-1 answered SIP/sanjay-09a0a970 -- Hungup 'Zap/4-1' In the above case, when the CLI prints that Zap/4-1 answered SIP/sanjay-09a0a970 actually call has not yet been picked by anyone, it is still ringing. Where as one of our other server where we have T1, the CLI looks like below when calling out -- Executing [ 91XX @internal:1] Dial(SIP/sanjay-09a0a970, ZAP/G2/1 XX ) -- Called G2/ 1XX -- Zap/23-1 is proceeding passing it to SIP/sanjay-08f58048 -- Zap/23-1 is ringing -- Hungup 'Zap/23-1' This one properly works as it should. I am not able to find whether this is Asterisk problem or Zaptel problem. Can someone please suggest what can be wrong? Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Mailing List Asterisk asterisk-users@lists.digium.com Sent: Thursday, May 22, 2008 7:04:34 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Call Placed through Manager connecting before the call connects. I am using Asterisk 1.4.19.2 on Fedora Core 8, with a Sangoma A200 card. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Mailing List Asterisk asterisk-users@lists.digium.com Sent: Thursday, May 22, 2008 6:16:17 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: [asterisk-users] Call Placed through Manager connecting before the call connects. Hello, I am trying to place call through the Manager, using the Zap Card the call connect to the designated Extension before the call is actually Answered by someone or the Voicemail. The message that I am sending is Action: Originate Channel: ZAP/G0/1XX MaxRetries: 0 Context: Test Exten: 6563 Priority: 1 CallerID: TEST 1234 The Events that I get from Manger are 1. Newchannel 2. Newcallerid 3. Newcallerid 4. Newstate [Here State is changed to Dialing] 5. Newstate [Here State is changed to Up] 6. Newexten [Here call is bridged to 6563] Once the call is Bridged to 6563, the phone is actually not Answered, you can hear the Ring on the Phone after Bridging. If I try the same for SIP channel I get addition events as Ringing. I want to play a message once the call connects, In this case the message is Played while the phone is Ringing. Please help. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Placed through Manager connecting before the call connects.
Is there no one who can even comment on below? Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 22, 2008 8:47:13 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Call Placed through Manager connecting before the call connects. I have noticed the same on the CLI while calling out Directly, the CLI does not show Ringing event.. -- Executing [EMAIL PROTECTED]:1] Dial(SIP/sanjay-09a0a970, ZAP/G0/1 XX ) -- Called G0/1 XX -- Zap/4-1 answered SIP/sanjay-09a0a970 -- Hungup 'Zap/4-1' In the above case, when the CLI prints that Zap/4-1 answered SIP/sanjay-09a0a970 actually call has not yet been picked by anyone, it is still ringing. Where as one of our other server where we have T1, the CLI looks like below when calling out -- Executing [EMAIL PROTECTED]:1] Dial(SIP/sanjay- 08f58048 , ZAP/G2/1XX) -- Called G2/ 1XX -- Zap/23-1 is proceeding passing it to SIP/sanjay-08f58048 -- Zap/23-1 is ringing -- Hungup 'Zap/23-1' This one properly works as it should. I am not able to find whether this is Asterisk problem or Zaptel problem. Can someone please suggest what can be wrong? Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Mailing List Asterisk asterisk-users@lists.digium.com Sent: Thursday, May 22, 2008 7:04:34 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Call Placed through Manager connecting before the call connects. I am using Asterisk 1.4.19.2 on Fedora Core 8, with a Sangoma A200 card. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Mailing List Asterisk asterisk-users@lists.digium.com Sent: Thursday, May 22, 2008 6:16:17 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: [asterisk-users] Call Placed through Manager connecting before the call connects. Hello, I am trying to place call through the Manager, using the Zap Card the call connect to the designated Extension before the call is actually Answered by someone or the Voicemail. The message that I am sending is Action: Originate Channel: ZAP/G0/1XX MaxRetries: 0 Context: Test Exten: 6563 Priority: 1 CallerID: TEST 1234 The Events that I get from Manger are 1. Newchannel 2. Newcallerid 3. Newcallerid 4. Newstate [Here State is changed to Dialing] 5. Newstate [Here State is changed to Up] 6. Newexten [Here call is bridged to 6563] Once the call is Bridged to 6563, the phone is actually not Answered, you can hear the Ring on the Phone after Bridging. If I try the same for SIP channel I get addition events as Ringing. I want to play a message once the call connects, In this case the message is Played while the phone is Ringing. Please help. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax solution for Asterisk
We are using Asterisk 1.4.13 on FC6 and have a T1 card with 20 DID's. We are wanting to use one of the DID's for Fax, is this possible or do we have to add some addition Hardware and what is the best way to do this. I know that similar thing would have been asked multiple time already, but I was not able to find anything that could answer my questions. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax solution for Asterisk
How about outbound faxing. Regards, Sanjay Rajdev - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2008 8:04:46 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Fax solution for Asterisk On Wed, May 21, 2008 at 10:26 AM, Lee Howard [EMAIL PROTECTED] wrote: Sanjay Rajdev wrote: We are using Asterisk 1.4.13 on FC6 and have a T1 card with 20 DID's. We are wanting to use one of the DID's for Fax, is this possible or do we have to add some addition Hardware and what is the best way to do this. http://iaxmodem.sourceforge.net Thanks, Lee. It depends on the amount of faxes and desired capabilities. If you just want a standalone fax, you could get an FXS card (in the same box) and bridge the faxes to that. It has worked quite well for me with a bit of tweaking, echocancelwhenbridged=no helps. Another option that works very well for high density, real fax machines is taking another T1 port and attaching it to a channel bank. Third option, hylafax and iaxmodem. This works pretty well since you are using a T for your inbound fax. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax solution for Asterisk
We would like to do something similar to efax, where we can send mail to send fax or something similar. I tried to install Asterisk Fax http://asterfax.sourceforge.net/ but was not able to compile it with Asterisk 1.4.19.2, I have read that they recommend Asterisk 1.2.X and older version of SpanDSP. Regards, Sanjay Rajdev - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2008 8:12:20 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Fax solution for Asterisk On Wed, May 21, 2008 at 10:40 AM, Sanjay Rajdev [EMAIL PROTECTED] wrote: How about outbound faxing. Regards, Sanjay Rajdev How about it? Describe your needs. There are different ways of doing the same thing, it all depends on needs. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax solution for Asterisk
Further more I think we will have to license it to for using it on more than one channel. I am looking for something totally open source. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2008 8:21:38 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Fax solution for Asterisk We would like to do something similar to efax, where we can send mail to send fax or something similar. I tried to install Asterisk Fax http://asterfax.sourceforge.net/ but was not able to compile it with Asterisk 1.4.19.2, I have read that they recommend Asterisk 1.2.X and older version of SpanDSP. Regards, Sanjay Rajdev - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2008 8:12:20 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Fax solution for Asterisk On Wed, May 21, 2008 at 10:40 AM, Sanjay Rajdev [EMAIL PROTECTED] wrote: How about outbound faxing. Regards, Sanjay Rajdev How about it? Describe your needs. There are different ways of doing the same thing, it all depends on needs. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Database Handling
I had a similar problem, but in my case we had a custom application that was throwing an segmentation exception which was causing Asterisk to Restart. And in that case It use to miss the log in database. You can determine the same by looking at the UNIQUEID being logged for the call. The UNIQUEID actually comprises of 2 things DateTimeString.CallNumber , everytime Asterisk restarts CallNumber will start from 1. So you can check the 2-3 UNIQUEID before you missed entry in CDR table and 2-3 after the missed entry to determine if Asterisk Restarted. Other way is you can see the kernel logs to see if Asterisk has thrown any exception. they can be found on Linux at /var/log/messages Regards, Sanjay Rajdev - Original Message - From: Alex Balashov [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 22, 2008 3:32:07 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Asterisk Database Handling Douglas Garstang wrote: We are sending CDR's to MySQL via odbc. It seems that Asterisk is sometimes dropping CDR's, and they aren't being sent to the database (they ARE in the Master.csv file though). We suspect that when the MySQL socket is idle, it gets disconnected, either by the MySQL server or by our firewall, and when Asterisk goes to send the next CDR over the socket, does not re-open the database, and drops that CDR. Possibly on the next call, it connects ok and sends the next CDR. Isn't there a keepalive option somewhere for cdr_mysql.conf, or failing that, a keepalive mechanism that can be enabled for TCP connections on the server side? -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Placed through Manager connecting before the call connects.
Hello, I am trying to place call through the Manager, using the Zap Card the call connect to the designated Extension before the call is actually Answered by someone or the Voicemail. The message that I am sending is Action: Originate Channel: ZAP/G0/1XX MaxRetries: 0 Context: Test Exten: 6563 Priority: 1 CallerID: TEST 1234 The Events that I get from Manger are 1. Newchannel 2. Newcallerid 3. Newcallerid 4. Newstate [Here State is changed to Dialing] 5. Newstate [Here State is changed to Up] 6. Newexten [Here call is bridged to 6563] Once the call is Bridged to 6563, the phone is actually not Answered, you can hear the Ring on the Phone after Bridging. If I try the same for SIP channel I get addition events as Ringing. I want to play a message once the call connects, In this case the message is Played while the phone is Ringing. Please help. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fetching Binary data from SQL Server
I am trying to write a customized app using C that would fetch voice file from
SQL Server 2000 using ODBC and FREETDS.
Currently I am only able to fetch first 63 KB chunk from the DB, and not able
to fetch the rest of the file, below is the code that i am using to do so,
fd = open(fullpath, O_RDWR | O_CREAT | O_TRUNC, 0770);
if (fd 0) {
ast_log(LOG_WARNING, Failed to write '%s': %s\n, fullpath, strerror(errno));
res = -1;
goto free_res;
}
res = SQLGetData(stmt, 1, SQL_BINARY, empty, 0, colsize);
fdlen = colsize;
if (option_verbose 2)
ast_verbose(VERBOSE_PREFIX_3 COLSIZE = %d, colsize); //PRINTING COLSIZE ON
CLI
if (fd -1) {
char tmp[1]=;
lseek(fd, fdlen - 1, SEEK_SET);
if (write(fd, tmp, 1) != 1) {
close(fd);
res = -1;
goto free_res;
}
}
if (fd -1){
//Trying to fetch data in chunks
for (offset = 0; offset colsize; offset += CHUNKSIZE) {
if ((fdm = mmap(NULL, CHUNKSIZE, PROT_READ | PROT_WRITE, MAP_SHARED, fd,
offset)) == MAP_FAILED) {
ast_log(LOG_WARNING, Could not mmap the output file: %s (%d)\n,
strerror(errno), errno);
goto free_res;
} else {
res = SQLGetData(stmt, x + 1, SQL_BINARY, fdm, CHUNKSIZE, NULL);
munmap(fdm, CHUNKSIZE);
if ((res != SQL_SUCCESS) (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, SQL Get Data error!\n[%s]\n\n, sql);
unlink(fullpath);
goto free_res;
}
}
}
}
close(fd);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
The value of colsize printed on CLI is 64512, Is there some limitation
somewhere in FREETDS or ODBC.
Can anyone please help me to get this fixed?
Regards,
Sanjay Rajdev
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fetching Binary data from SQL Server
Tilghmanm, Thanks a lot, I have changed the value in FREETDS and it worked. Regards, Sanjay Rajdev - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 17, 2008 4:09:43 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Fetching Binary data from SQL Server On Friday 16 May 2008 16:10:22 Sanjay Rajdev wrote: I am trying to write a customized app using C that would fetch voice file from SQL Server 2000 using ODBC and FREETDS. Currently I am only able to fetch first 63 KB chunk from the DB, and not able to fetch the rest of the file, below is the code that i am using to do so, Actually, if you Google, you'll find that in freetds.conf, the default 'text size' parameter is set to exactly 64512, which is the limit that FreeTDS itself is placing on the data. You might try increasing that (to a maximum of 2GB) and see if that works better for you. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] x100p card or similar in India
We have been using Sangoma A200 for about an year now with BSNL connection. I don't know if you can get it in India directly as in our case it was brought from US directly. Regards, Sanjay Rajdev - Original Message - From: Amit Patel [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, May 12, 2008 8:12:37 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] x100p card or similar in India Hello All, Anyone purchased a asterisk card, x100p or similar in India, if yes from where and what model ? I am interested in setting up a Asterisk Server at home, for single line at the moment and if things work out great, I would like to migrate that to my business and replace the aging pbx solution. Thankx, Amit. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a way to have Manager Bridge Channel without being connected
Hello All, Is there a way to have Manager Bridge Channel to the specified extension without the channel being connected. In the current scenario the channel only bridges once the call get connected, it does not bridge when any service provider (telco) message is played. I want to record all call originated by manager even if a telco message is played. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Linux distribution to use in Asterisk server
I have been using FC6 for the past 1 year without any problem. Regards, Sanjay Rajdev - Original Message - From: equis software [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 9, 2008 8:49:23 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: [asterisk-users] Best Linux distribution to use in Asterisk server Hi, I allways use Gentoo y my Asterisk servers and work well, but what do you think about to use Ubuntu or another distibution?? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Basic modules of Asterisk
Thank Russell, I will try to manage it through the modules.conf file. Regards, Sanjay Rajdev - Original Message - From: Russell Bryant [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 8, 2008 4:11:00 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Basic modules of Asterisk Sanjay Rajdev wrote: I just want to Run Asterisk with the basic required modules, What can I do to achieve so? My only requirement is to run SIP clients and the Dictate Module. 2 options: 1) Before compiling and installing Asterisk, run make menuselect to select only the modules that you want to use. That way, only those modules are compiled and installed. 2) After installing Asterisk, edit /etc/asterisk/modules.conf. By default, Asterisk will load all installed modules. You can turn off the autoload functionality, and explicitly list the modules that you need. You probably want pbx_config, chan_sip, app_dictate, app_dial, probably some others ... -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Restarting due to segfault
I had a problem in the dictate app, which I have fixed. Thanks for the help.
By the way here is a description of what was happening.
app_dictate does not close the file descriptor after the call hangs or a new
dictation starts, as and when the dictation increased the count of open file
descriptor increased and forced the asterisk process to reach the limit of
allowed maximum number of open file descriptor.
S o I added ast_closestream(fs), where ever I thought it was necessary and at
the end I checked for
if(fs){
ast_closestream(fs) ;
}
this line was causing the problem, in case the file descriptor was already
closed it was still going into the if and trying to close a closed descriptor.
I have made change to set fs = NULL everywhere after ast_closestream(fs)
I am not a developer for Asterisk and even cannot make changes in the SVN as I
do not know lot about the branches in it, but if someone from your side can
take the effort to change this It would be great help for others.
Regards,
Sanjay Rajdev
- Original Message -
From: Russell Bryant [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May 8, 2008 8:36:14 AM GMT +05:30 Chennai, Kolkata, Mumbai, New
Delhi
Subject: Re: [asterisk-users] Asterisk Restarting due to segfault
Sanjay Rajdev wrote:
I have Asterisk 1.4.15 installed on a Fedora Core 8 machine. Asterisk is
snip
In the dialplan we have used MixMonitor() to record the calls.
Can anyone help me on getting to the root of the problem or fixing it?
We have fixed a _lot_ of issues in that area of the code since 1.4.15. I would
suggest trying the latest version. If it still gives you trouble, please let us
know on http://bugs.digium.com so that we can fix it up for you.
Thanks,
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which Cepstral Voice to license
Which Cepstral voice is best for Asterisk? We need to license one. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which Cepstral Voice to license
We are looking for a female voice. Regards, Sanjay Rajdev - Original Message - From: Matthew Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 9, 2008 2:35:24 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Which Cepstral Voice to license david-8khz and the regular david aren't bad in my experience. On Thu, May 8, 2008 at 4:54 PM, Sanjay Rajdev [EMAIL PROTECTED] wrote: Which Cepstral voice is best for Asterisk? We need to license one. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Basic modules of Asterisk
I just want to Run Asterisk with the basic required modules, What can I do to achieve so? My only requirement is to run SIP clients and the Dictate Module. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
We are using Asterisk 1.4.13 in production, were we have almost 30 SIP users on a Asterisk box, we are also using IAX to communicate between main Asterisk server and the other. we use Queues, Conference too. Regards, Sanjay Rajdev - Original Message - From: Benoit Plessis [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, May 6, 2008 5:08:37 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: [asterisk-users] Asterisk in Production ? Hi, I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some Avoiding IAX destroy deadlock and now that we have added a Queue, it's worse than ever. The queue goes stuck quite often (agent are stuck in 'In use' state and if they logoff they can't log-in till an asterisk restart). regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mixmonitor recording issue
I had a similar problem. In my case Asterisk was crashing due to MixMonitor() and then automatically restarting. I have never found a alternative solution to record the calls. Regards, Sanjay Rajdev - Original Message - From: Rahul Yadav [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, May 6, 2008 10:54:32 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: [asterisk-users] Mixmonitor recording issue Hi All I am using asterisk 1.4.18.1.My problem is that Mixmonitor is not recording complete recording. Suppose i got a call connected and talking for 3 minutes then mixmonitor records only 2 minutes of call. This problem happens randomly.Please help as i am suffering very much due to this problem. Rahul Yadav ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Restarting due to segfault
I have Asterisk 1.4.15 installed on a Fedora Core 8 machine. Asterisk is re-starting intermediately i.e sometime it does not re-start the whole day, and sometime it just re-start after few calls. There is no log of error in /var/log/asterisk/messages but if I see the /var/log/messages I can see lines similar to below for Asterisk Process almost the same time asterisk re-start segfault at fff10098 eip 0809b0b5 esp b793e370 error 4 segfault at 2dfc eip 0809b0c5 esp b781f370 error 4 In the dialplan we have used MixMonitor() to record the calls. Can anyone help me on getting to the root of the problem or fixing it? Thanks in advance. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitor not merging calls
I have setup Asterisk on 2 Fedora Core 8 machines, and have made it to record
all incoming calls. One of the box that have Asterisk 1.4.18 is properly
merging calls and the other box that has Asterisk 1.4.15 is recording the calls
but not merging them, I have made sure that SOX is installed on the box.
Here is the Dialplan of both the machines :
exten = 1234,1,Answer()
exten = 1234,2,Monitor(gsm,/recordings)/${UNIQUEID},m)
Do I have to upgrade and check or is their some other thing I can check?
Regards,
Sanjay Rajdev
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor not merging calls
John, Is their something that I can change on my side to get this working ? Jared, I thought MixMonitor() was for Queue, Can you let me know how to use it? Thanking you for replying. Regards, Sanjay Rajdev - Original Message - From: John covici [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 21, 2008 9:39:39 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Monitor not merging calls Newer version of sox don't seem to have soxmix anymore, but you can use sox -m and I think asterisk should be changed to use that instead. on Monday 04/21/2008 Jared Smith([EMAIL PROTECTED]) wrote On Mon, 2008-04-21 at 21:11 +0530, Sanjay Rajdev wrote: One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. It might be worth giving the MixMonitor() application a try instead. :-) -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitor v/s MixMonitor
What is good for recording all the incoming and outgoing calls, Monitor() or MixMonitor(). Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] C# SIP API to Comiunicate with Asterisk
Do anyone has an idea about an open source SIP API written in C# that can communicate with Asterisk, to call out? Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk
This work with Asterisk Manager Interface. I want to implement basic phone functionality in C#. Regards, Sanjay. - Original Message - From: Matt Watson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 4, 2008 3:48:52 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk There is a .NET 1.1 library out there... I've played with it a little bit, but not enough that I could comment on how feature rich or stable it is... http://www.voip-info.org/wiki/view/Asterisk+.NET It'll more than likely not be compatible with AMI 1.1 however, which I believe is included in ast 1.6 -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, April 03, 2008 5:28 PM To: asterisk-users Subject: [asterisk-users] C# SIP API to Comiunicate with Asterisk Do anyone has an idea about an open source SIP API written in C# that can communicate with Asterisk, to call out? Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk
Can you Please refer me to any, the one that I found are all either in Java/C. Or if they are in C# they are not opensource. Regards, Sanjay. - Original Message - From: Grey Man [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 4, 2008 4:32:44 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk On Thu, Apr 3, 2008 at 11:35 PM, [EMAIL PROTECTED] wrote: Do anyone has an idea about an open source SIP API written in C# that can communicate with Asterisk, to call out? There are a few C# SIP stacks around that will let you do that. Creating a call from such a stack to Asterisk will be the same as to any other SIP server. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk
Thanks a lot, will try this out. Regards, Sanjay. - Original Message - From: Grey Man [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 4, 2008 4:58:56 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk On Fri, Apr 4, 2008 at 12:11 AM, [EMAIL PROTECTED] wrote: Can you Please refer me to any, the one that I found are all either in Java/C. Or if they are in C# they are not opensource. I know www.mysipswitch is written in C# and can place SIP calls to Asterisk servers. The code is open sourced at http://www.codeplex.com/mysipswitch. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Question
No It does not require. Regards, Sanjay. - Original Message - From: Drew Miller [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, March 31, 2008 9:17:19 PM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Simple Question Does AMD (answering machine detect) need ztdummy or some other timer to function properly? -- Drew Miller Iowa Democratic Party Information Technology Director Office: (515) 974-1682 Cell: (515) 451-4509 AIM: ItsDrewMiller MSN: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to register IAX user without password
Create a User and a Peer on both the machines for each other.
e.g IAX.conf on PCa
[pca2pcb]
type=peer
host=[IP OF pcb]
username=pca2pcb
serect=pca2pcb12345
qualify=yes
[pcb2pca]
type=user
context=default
auth=md5
secret=pcb2pca12345
deny=0.0.0.0/0.0.0.0
permit=[IP of pcb]
qualify=yes
ON PCb do the reverse in iax.conf
[pcb2pca]
type=peer
host=[IP OF pca]
username=pcb2pca
serect=pcb2pca12345
qualify=yes
[pca2pcb]
type=user
context=default
auth=md5
secret=pca2pcb12345
deny=0.0.0.0/0.0.0.0
permit=[IP of pca]
qualify=yes
NOW in Your extensions.conf you can use as
On PCa
exten=_.,1,Dial(IAX2/pca2pcb/${EXTEN})
exten=_y.,1,Dial(IAX2/pca2pcb/${EXTEN})
exten=_a.,1,Dial(IAX2/pca2pcb/${EXTEN})
and on PCb
exten=_.,1,Dial(IAX2/pcb2pca/${EXTEN})
exten=_y.,1,Dial(IAX2/pcb2pca/${EXTEN})
exten=_a.,1,Dial(IAX2/pcb2pca/${EXTEN})
Let me know if this works.
Regards,
Sanjay.
- Original Message -
From: Mian M Asif [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, March 28, 2008 8:04:08 PM (GMT+0530) Asia/Calcutta
Subject: [asterisk-users] how to register IAX user without password
hi,
i want to call PC2PC between to IAX client without authentication i
want to allow every user to use PC2PC no any password required. Please
let me know what i have need to do in IAX.conf or any other file to
allow any user to call Pc2Pc.
My IAX.conf
[guest]
type=user
context=default
callerid=Guest IAX User
My extensions.conf
[default]
exten=_.,1,Dial(IAX2/${EXTEN})
exten=_y.,1,Dial(IAX2/${EXTEN})
exten=_a.,1,Dial(IAX2/${EXTEN})
below is my Asterisk console logs which i see after making call.
Mar 28 03:25:43 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:6910 socket_read: Rejected
connect attempt from 203.99.57.80, who was trying to reach
'jaffaradvcommnet@'
Mar 28 03:26:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'j' (from 203.99.57.80)
advcomm6*CLI iax2 show channels
Channel Peer UsernameID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None)203.99.57.80 (None) 4/15232
1/1 0ms -0001ms ms unknow
(None)203.99.57.80 jaffaradvc 5/15233
4/4 0ms -0001ms ms unknow
(None)203.99.57.80 (None) 6/18423
1/1 0ms -0001ms ms unknow
3 active IAX channels
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:27:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'jaffaradvcommnet' (from 203.99.57.80)
i am very thankful if some one help me in this regards,
i am getting Registration Refused error when i debug on console.
please tell me how can i registration every user without any username
and password and these user can make calls between each other.
i am very thankful if any body help me in this regards,
advcomm6*CLIiax2 debug
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 3ms SCall: 09398 DCall: 0 [203.99.57.80:47641]
USERNAME: aliadvcommnet
REFRESH : 60
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ
Timestamp: 2ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
CAUSE : Registration Refused
CAUSE CODE : 29
regards,
Asif
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How is uniqueid computed
Can anyone let me know how the uniqueid for a call is computed in asterisk? Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How is uniqueid computed
Thanks Mindaugas. Regards, Sanjay. - Original Message - From: Mindaugas Kezys [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 18, 2008 10:26:37 PM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] How is uniqueid computed Hello, Uniqueid = (call initiation time in unix time format) . (call count since asterisk restart / 2 ) If call is transfered or it is leg2 then: Uniqueid = (call initiation time in unix time format) . (call count since asterisk restart / 2 + 1) This is from observations, i can be mistaken. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing Solution for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, March 18, 2008 6:12 PM To: asterisk-users Subject: [asterisk-users] How is uniqueid computed Can anyone let me know how the uniqueid for a call is computed in asterisk? Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Logs for Call generated by Manager API
I am generating an outbound call through the Manager API and bridging it to an internal Extension, my problem is I am not able to find the logs for the call generated by the Manger API, Since on the same Asterisk server there are many users connected and I am receiving lot of Events back, not able to recognize which was the call generated by me as same time multiple users are dialing out. Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logs for Call generated by Manager API
Is there way to get the logs of the call generated by Manager API, or is there some other way to achieve same scenario so that I can get the status of the call generated by me. Actually I have a scenario where I have to call customers and play a message, I do not want to send messages to Manager to generate all the calls at once, I just want to monitor the status of the call placed by me, so that I do not place more than 2 or 3 calls at the same time, hence not consuming all the available line on the Asterisk Server, leaving some lines for other people too. Please help. Regards, Sanjay. - Original Message - From: sanjay rajdev [EMAIL PROTECTED] To: asterisk-users asterisk-users@lists.digium.com Sent: Friday, March 14, 2008 7:26:00 PM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Logs for Call generated by Manager API I am generating an outbound call through the Manager API and bridging it to an internal Extension, my problem is I am not able to find the logs for the call generated by the Manger API, Since on the same Asterisk server there are many users connected and I am receiving lot of Events back, not able to recognize which was the call generated by me as same time multiple users are dialing out. Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logs for Call generated by Manager API
Thanks Lee, Will try to match on Parameter received in message. Regards, Sanjay. - Original Message - From: Lee Jenkins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 14, 2008 11:27:51 PM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Logs for Call generated by Manager API [EMAIL PROTECTED] wrote: I am generating an outbound call through the Manager API and bridging it to an internal Extension, my problem is I am not able to find the logs for the call generated by the Manger API, Since on the same Asterisk server there are many users connected and I am receiving lot of Events back, not able to recognize which was the call generated by me as same time multiple users are dialing out. Also, I wrote a Windows based utility for viewing AMI packets and testing AMI commands. It's Freeware: http://www.voip-info.org/wiki-Asterisk+GUI#OperatorManagerInterfaces Look for Manager API Test Utility or download it directly from our site: http://www.datatrakpos.com/pos/datatalk/dpdownload.aspx I use it all the time when write apps for AMI. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logs for Call generated by Manager API
I would just try parsing the message in my code to make it work. I know this is not a full proof solution but is ok for now, till I get something better. Regards, Sanjay. - Original Message - From: Mark Hamilton [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 15, 2008 3:11:25 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Logs for Call generated by Manager API I don't think the link that Lee gave works. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: March 14, 2008 5:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Logs for Call generated by Manager API Thanks Lee, Will try to match on Parameter received in message. Regards, Sanjay. - Original Message - From: Lee Jenkins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 14, 2008 11:27:51 PM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Logs for Call generated by Manager API [EMAIL PROTECTED] wrote: I am generating an outbound call through the Manager API and bridging it to an internal Extension, my problem is I am not able to find the logs for the call generated by the Manger API, Since on the same Asterisk server there are many users connected and I am receiving lot of Events back, not able to recognize which was the call generated by me as same time multiple users are dialing out. Also, I wrote a Windows based utility for viewing AMI packets and testing AMI commands. It's Freeware: http://www.voip-info.org/wiki-Asterisk+GUI#OperatorManagerInterfaces Look for Manager API Test Utility or download it directly from our site: http://www.datatrakpos.com/pos/datatalk/dpdownload.aspx I use it all the time when write apps for AMI. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best alternative for getting prompts recorded.
What is the best alternative for getting the IVR and other prompts recorded for Asterisk. Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best alternative for getting prompts recorded.
Thanks everyone for the reply. Till now we had simple IVR so we recorded it ourself. Now I have a requirement where customer needs a customized message to be played to customer. I am basically looking for some Text to Speech software that would be cost effective (most probably a open source) and would convert Text to Speech. I tried Fetival, but the quality of the sound is not good. Can we improve the sound quality of Festival somehow. Regards, Sanjay. - Original Message - From: Ron Joffe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 12, 2008 2:30:44 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Best alternative for getting prompts recorded. On Tuesday 11 March 2008 16:21, [EMAIL PROTECTED] wrote: What is the best alternative for getting the IVR and other prompts recorded for Asterisk. We decided to record our own. We set up a recording studio, and that has worked out very well for us. Let me know if we can help. Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best alternative for getting prompts recorded.
Thanks a lot everyone, I will go ahead and try Cepstral. Regards, Sanjay. - Original Message - From: Steve Edwards [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 12, 2008 2:54:50 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Best alternative for getting prompts recorded. On Wed, 12 Mar 2008, [EMAIL PROTECTED] wrote: Thanks everyone for the reply. Till now we had simple IVR so we recorded it ourself. Now I have a requirement where customer needs a customized message to be played to customer. I am basically looking for some Text to Speech software that would be cost effective (most probably a open source) and would convert Text to Speech. I tried Fetival, but the quality of the sound is not good. Can we improve the sound quality of Festival somehow. Cepstral with Allison is only $30. I did a demo IVR for a potential client and it was hard to tell the TTS bits from the human bits. If I took the time to learn Cepstral's markup language I probably could have fooled myself :) 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Want to know Frequency and lenght of Frame
I am planning to write a module to find if a Special Information was detected or not. Can anyone please help me to figure out the below fields? 1. The Frequency of a frame 2. Length of frame in milliseconds Thanks in advance. Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting SIT (Special Information Tone) on outbound calls
For reference of SIT please check http://en.wikipedia.org/wiki/Special_information_tone Regards, Sanjay. - Original Message - From: sanjay rajdev [EMAIL PROTECTED] To: asterisk-users asterisk-users@lists.digium.com Sent: Friday, February 29, 2008 8:35:08 PM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Detecting SIT (Special Information Tone) on outbound calls Is there a way to detect SIT (Special Information Tone) when making an outbound call. Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detecting SIT (Special Information Tone) on outbound calls
Is there a way to detect SIT (Special Information Tone) when making an outbound call. Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can AMD detect Service Provider Message.
Is there a way to detect Service Provider message such as invalid number, using AMD or some other application. Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question regarding AGI
I have questions about AGI. 1. When Using CONTROL STREAM FILE command with all the parameter, I could not find any way to * or # in the DTMF, it only returns if any digit is pressed, even if I set forward and rewind digits to BLANK () 2. When I call out using ZAP, is their a way to find if the call went to the Person Called, or if their was a message played by his service provider, e.g. Number does not exists, Out of range, Switched off, etc. Can we even achieve this using some Dialplan applications Can someone please help. Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip Version
What version of SIP do Asterisk 1.4.x uses. Regards, Sanjay. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfering IAX context
Hello Everyone, I have a 2 Asterisk Servers, one in US and another in India. Once someone from US calls, call hit US server and then is forwarded to India which then is answered by someone. i.e. Caller -- US Asterisk Server -- India Asterisk Server -- Employee(India) The Employee in India decides that the call was for Employee in US, so he transfer the call to the employee in US. i.e. Caller -- US Asterisk Server -- India Asterisk Server -- Employee(India) -- India Asterisk Server -- US Asterisk Server -- Employee (US) OR Caller -- US Asterisk Server -- India Asterisk Server -- US Asterisk Server -- Employee (US) (Not sure which explanation is correct as per asterisk working, but hopefully should be the second.) The way this type of communication traverse is that the call has to come to India and the reverted back to US. Is their a way that when a Employee in India transfers back the call to US Asterisk Server, the Indian server should completely removed from the picture. This would save our Bandwidth utilization. i.e. flow becomes :: Caller -- US Asterisk Server -- Employee (US) Regards, Sanjay. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detecting card on the PCI Slot
I have some Analog card on a PCI slot of a remote computer, Is their a way I can figure out remotely the name of the card. I have FC6 installed on the machine. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting card on the PCI Slot
I have installed FC6 on it, want to configure it with Asterisk. It had some driver earlier but the machine has been formatted yesterday, so no idea. Also I am new to Linux. Regards, Sanjay Rajdev Phone : +1 (877) 342 2329 x 1702 Fax : +1 (815) 261 5907 http://www.featherstoneinformatics.com Communications from Featherstone Informatics Group (FIG) may transmit information that is confidential and privileged information of Featherstone Informatics Group (FIG). Unless you are the intended addressee, you may not use, copy or disclose to anyone this communication or any information transmitted by this communication. If you have received such communication in error, please advise the sender by e-mail and/or telephone and destroy this communication immediately. This communication and any information transmitted by this communication may also be considered protected health information as defined under the Health Insurance Portability and Accountability Act and its related regulations (a.k.a., HIPAA) or any other similar state law. Please exercise due care and ensure that you comply with its contractual and legal obligations. - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, June 5, 2007 2:19:22 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Detecting card on the PCI Slot On Tue, Jun 05, 2007 at 12:59:37AM +0530, Sanjay Rajdev wrote: I have some Analog card on a PCI slot of a remote computer, Is their a way I can figure out remotely the name of the card. I have FC6 installed on the machine. lspci What driver handles it? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting card on the PCI Slot
Thanks for the suggestion, I figured out the cards. I have 2 Digium TDM400P card and a Sangoma A101 single port card on the machine. Any suggestion on installing them. Regards, Sanjay Rajdev - Original Message - From: Gordon Henderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 5, 2007 2:11:47 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Detecting card on the PCI Slot On Tue, 5 Jun 2007, Sanjay Rajdev wrote: I have some Analog card on a PCI slot of a remote computer, Is their a way I can figure out remotely the name of the card. I have FC6 installed on the machine. Try the 'lspci' command. Eg: :00:14.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface is what a Digium TDM400P card looks like. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can two card be configured on same machine.
I have a 2 sangoma cards that need to be configured on a same server, one is a T1 and another is a for PSTN line. Is this possible, if so please help. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IAX
Can you send IAX.conf of both the systems Regards, Sanjay Rajdev - Original Message - From: Malcom Kemp [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, 30 May, 2007 9:05:34 PM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Help with IAX ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] socket_process: Received mini frame before first full voice frame
Never received a response for this from anyone. This is being seen more frequently now. Please Suggest. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: asterisk-users asterisk-users@lists.digium.com Cc: asterisk-dev [EMAIL PROTECTED] Sent: Friday, May 11, 2007 2:26:30 AM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] socket_process: Received mini frame before first full voice frame Anyone any idea why do we keep on getting chan_iax2.c:7535 socket_process: Received mini frame before first full voice frame Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AgentCallbackLogin not working with 1.4.4
I installed Asterisk 1.4.4. yesterday and was not able to make the queue work, It does not allow me the AgentCallbackLogin. It just does not confirm that the agent has logged in, it ask for the password once you provide the password, it directly hangs up. I i check on the CLI it shows that no agent is online. Is it so that we require to install a T1 Card (or some other) to make this work, just wanted to ask this because I was not able to make 1.4.1 Meetme work until we install a T1 card. Regards, Sanjay Rajdev Phone : +1 (877) 342 2329 x 1702 Fax : +1 (815) 261 5907 http://www.featherstoneinformatics.com Communications from Featherstone Informatics Group (FIG) may transmit information that is confidential and privileged information of Featherstone Informatics Group (FIG). Unless you are the intended addressee, you may not use, copy or disclose to anyone this communication or any information transmitted by this communication. If you have received such communication in error, please advise the sender by e-mail and/or telephone and destroy this communication immediately. This communication and any information transmitted by this communication may also be considered protected health information as defined under the Health Insurance Portability and Accountability Act and its related regulations (a.k.a., HIPAA) or any other similar state law. Please exercise due care and ensure that you comply with its contractual and legal obligations. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] socket_process: Received mini frame before first full voice frame
Anyone any idea why do we keep on getting chan_iax2.c:7535 socket_process: Received mini frame before first full voice frame Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iaxy clicking
What OS (operating system are you using) Regards, Sanjay Rajdev - Original Message - From: Matthew Yingling [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, May 11, 2007 2:59:01 AM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Iaxy clicking Hi, I have three Iaxy devices (s101i) parts. Two of them seem to work fine. The third plays a loud repeating click sound when an analog phone is plugged in. I can provision all of them, and make calls to all of them. The clicking one will blink when a call is incoming, but no audio from the call can be heard on the handset, and the caller only hears silence. The same handset works on the other Iaxys, and other handsets have the same clicking issue. Resetting the Iaxy doesn't seem to fix the problem. Does anyone have any ideas on how to fix this problem, or whether the Iaxy is broken and unfixable (for me as an end-user). Thanks, Matthew Yingling ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call In queue stucks
Hello All, I have a queue with only one agent logged in al the time, but if for some reason the agent cannot pick up the call for 2 full ring, the phone does not ring the 3rd time and all the call in the queue get stuck. Below is my agents.conf [general] persistentagents=yes multiplelogin=no [agents] wrapuptime=5000 musiconhold = default updatecdr=yes agent = 1001,4321,Agent01 While the queue is stuck if I type the agents show online i get the following asterisk*CLI agent show online 1001 (Agent01) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 1 agents online The above states that the agent has not logged off, I dont know then why the call do not come to him. I have also tried adding the autologoff=1000 autologoffunavail=no but it does not work It only ring twice in this case also and the caller keeps on hearing, you are first in line. If the agent re login the queue starts again. I have Asterisk 1.4.2 with zaptel 1.4.1 Can anyone Please help. Thanks in advance. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BSNL caller ID (India)
Yes, As I have mentioned below I tried the link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to make it work. Regards, Sanjay Rajdev - Original Message - From: Steve Murphy [EMAIL PROTECTED] To: sanjay rajdev [EMAIL PROTECTED] Sent: Thursday, April 19, 2007 4:37:39 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) Sanjay-- Did you look at bug 6683? Sorry, I haven't reviewed all the messages on this thread. murf On Wed, 2007-04-18 at 01:01 +0530, Sanjay Rajdev wrote: Tzafrir, Can you Please let me know if the zapata.conf below is correct, or do I have to change something. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: tzafrir cohen [EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 3:38:16 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) Yes below is the zapata.conf [trunkgroups] [channels] context=incoming usecallerid=yes cidsignalling=dtmf cidstart=ring hidecallerid=no callerid=asreceived callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A200 [slot:2 bus:4 span:1] group=0 signalling = fxs_ks channel = 1 group=0 signalling = fxs_ks channel = 2 group=0 signalling = fxs_ks channel = 3 group=0 signalling = fxs_ks channel = 4 Regards, Sanjay Rajdev - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 2:40:32 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) On Mon, Apr 16, 2007 at 11:18:46PM +0530, Sanjay Rajdev wrote: Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Hmmm... are you sure you have configured your system to get callerid from the PSTN? callerid=asrecieved in zapata.conf. -- Steve Murphy [EMAIL PROTECTED] Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BSNL caller ID (India)
Tzafrir, Can you Please let me know if the zapata.conf below is correct, or do I have to change something. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: tzafrir cohen [EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 3:38:16 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) Yes below is the zapata.conf [trunkgroups] [channels] context=incoming usecallerid=yes cidsignalling=dtmf cidstart=ring hidecallerid=no callerid=asreceived callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A200 [slot:2 bus:4 span:1] group=0 signalling = fxs_ks channel = 1 group=0 signalling = fxs_ks channel = 2 group=0 signalling = fxs_ks channel = 3 group=0 signalling = fxs_ks channel = 4 Regards, Sanjay Rajdev - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 2:40:32 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) On Mon, Apr 16, 2007 at 11:18:46PM +0530, Sanjay Rajdev wrote: Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Hmmm... are you sure you have configured your system to get callerid from the PSTN? callerid=asrecieved in zapata.conf. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with queue
Thanks Philipp, I tried making it 5000, and it worked. Once again thank for your help. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: philipp kempgen [EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 5:58:22 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Problem with queue I waited for almost 5 minutes but still did not receive the call. Regards, Sanjay Rajdev - Original Message - From: Philipp Kempgen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 5:36:45 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Problem with queue Sanjay Rajdev wrote: Regards, Sanjay Rajdev Tha i did because i dont want any call to get disconnected. Can you let me know what can be the problem doing so. - Original Message - From: Philipp Kempgen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 4:35:42 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Problem with queue Sanjay Rajdev wrote: I have queue set up in realtime on Asterisk 1.4.2. Below is the senario that is happenening :: I have created a test queue with only one agent. Once I call the test queue the agents phone rings if the aagent is logged on. everything till here is fine. Now if the agent does not pick up the call, the call automaticaly disconnects after 15 secs as set for the queue, till here also it is fine. But the agents phone never rings again for that Call and therefore the caller goes on an infinite wait and listen the wonderfull on hold music. :) Here are few more observations. If I reload the asterisk it ring again for one time. OR If the agent relogin then also it rings for one more time. OR If the caller disconnecs and callback again, it will ring one more time. Here is the agent.conf [general] persistentagents=yes multiplelogin=no [agents] autologoff=150 wrapuptime=6 6/60/60 = 16,67 *hours*! Use something like 5. ---cut--- ; Define wrapuptime. This is the minimum amount of time when ; after disconnecting before the caller can receive a new call ; note this is in milliseconds. ---cut--- Sorry, it's milliseconds. But even 60 seconds is probably quite long. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BSNL caller ID (India)
Tzafrir, I am sure about both of them in my zapata.conf. I am on Asterisk 1.4.2 and the zapata.conf is in /etc/asterisk directory with all other asterisk configuration files Do you have any other idea which can help me finding out what is wrong. Regards, Sanjay Rajdev - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: Sanjay Rajdev [EMAIL PROTECTED] Sent: Wednesday, April 18, 2007 1:15:31 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) On Tue, Apr 17, 2007 at 03:38:16AM +0530, Sanjay Rajdev wrote: Yes below is the zapata.conf [trunkgroups] [channels] context=incoming usecallerid=yes cidsignalling=dtmf cidstart=ring Are you sure about those two? hidecallerid=no callerid=asreceived This is correct, of course. My typo. callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A200 [slot:2 bus:4 span:1] group=0 signalling = fxs_ks channel = 1 group=0 signalling = fxs_ks channel = 2 group=0 signalling = fxs_ks channel = 3 group=0 signalling = fxs_ks channel = 4 Regards, Sanjay Rajdev - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 2:40:32 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) On Mon, Apr 16, 2007 at 11:18:46PM +0530, Sanjay Rajdev wrote: Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Hmmm... are you sure you have configured your system to get callerid from the PSTN? callerid=asrecieved in zapata.conf. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question regrading IAX
I have a server that only handles the inbound and outbound call and passes everything to the second server using IAX. Sometimes it so happen that a call comes in on the First machine, this machine forward to the second machine as an inbound call using IAX, now the second machine decides that this is an outbound call request so it forward it back to the first machine to make the outbound call. Is it possible once the second machine has decided that this is a outbound call, to intimate the first machine to directly make the outbond call without traversing to the second machine and coming back. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues
You can have the agent login once and newer log out. You can certainly set up your asterisk box to persit the login over the reload and the restart. persistentagents=yes Regards, Sanjay Rajdev Phone : +1 (877) 342 2329 x 1702 Fax : +1 (815) 261 5907 http://www.featherstoneinformatics.com Communications from Featherstone Informatics Group (FIG) may transmit information that is confidential and privileged information of Featherstone Informatics Group (FIG). Unless you are the intended addressee, you may not use, copy or disclose to anyone this communication or any information transmitted by this communication. If you have received such communication in error, please advise the sender by e-mail and/or telephone and destroy this communication immediately. This communication and any information transmitted by this communication may also be considered protected health information as defined under the Health Insurance Portability and Accountability Act and its related regulations (a.k.a., HIPAA) or any other similar state law. Please exercise due care and ensure that you comply with its contractual and legal obligations. - Original Message - From: Voip Asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 18, 2007 5:23:18 AM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] queues ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BSNL caller ID (India)
Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Can someone please help. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ${CALLERIDNUM} DEPRECATED in 1.4.2
${CALLERIDNUM} is DEPRECATED in 1.4.2 what is the alternative to find the same
in extensions.conf for setting a proper dialplan.
Please Suggest
Regards,
Sanjay Rajdev
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BSNL caller ID (India)
Yes below is the zapata.conf [trunkgroups] [channels] context=incoming usecallerid=yes cidsignalling=dtmf cidstart=ring hidecallerid=no callerid=asreceived callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A200 [slot:2 bus:4 span:1] group=0 signalling = fxs_ks channel = 1 group=0 signalling = fxs_ks channel = 2 group=0 signalling = fxs_ks channel = 3 group=0 signalling = fxs_ks channel = 4 Regards, Sanjay Rajdev - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 2:40:32 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) On Mon, Apr 16, 2007 at 11:18:46PM +0530, Sanjay Rajdev wrote: Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Hmmm... are you sure you have configured your system to get callerid from the PSTN? callerid=asrecieved in zapata.conf. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with queue
I have queue set up in realtime on Asterisk 1.4.2. Below is the senario that is happenening :: I have created a test queue with only one agent. Once I call the test queue the agents phone rings if the aagent is logged on. everything till here is fine. Now if the agent does not pick up the call, the call automaticaly disconnects after 15 secs as set for the queue, till here also it is fine. But the agents phone never rings again for that Call and therefore the caller goes on an infinite wait and listen the wonderfull on hold music. :) Here are few more observations. If I reload the asterisk it ring again for one time. OR If the agent relogin then also it rings for one more time. OR If the caller disconnecs and callback again, it will ring one more time. Here is the agent.conf [general] persistentagents=yes multiplelogin=no [agents] autologoff=150 wrapuptime=6 musiconhold = default updatecdr=yes recordagentcalls=yes recordformat=wav agent = 1001,4321,Agent1 Here are the enties in queue table name=test timeout=15 monitor_join=t (yes) monitor_format=wav announce_frequency=60 retry=5 wrapuptime=20 maxlen=0 servicelevel=120 strategy=rrmemory eventwhencalled=t (yes) reportholdtime=t (yes) memberdelay=0 weight=0 Does anyone have idea what is wrong. Please suggest. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with queue
Regards, Sanjay Rajdev Tha i did because i dont want any call to get disconnected. Can you let me know what can be the problem doing so. - Original Message - From: Philipp Kempgen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 4:35:42 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Problem with queue Sanjay Rajdev wrote: I have queue set up in realtime on Asterisk 1.4.2. Below is the senario that is happenening :: I have created a test queue with only one agent. Once I call the test queue the agents phone rings if the aagent is logged on. everything till here is fine. Now if the agent does not pick up the call, the call automaticaly disconnects after 15 secs as set for the queue, till here also it is fine. But the agents phone never rings again for that Call and therefore the caller goes on an infinite wait and listen the wonderfull on hold music. :) Here are few more observations. If I reload the asterisk it ring again for one time. OR If the agent relogin then also it rings for one more time. OR If the caller disconnecs and callback again, it will ring one more time. Here is the agent.conf [general] persistentagents=yes multiplelogin=no [agents] autologoff=150 wrapuptime=6 6/60/60 = 16,67 *hours*! Use something like 5. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with queue
I waited for almost 5 minutes but still did not receive the call. Regards, Sanjay Rajdev - Original Message - From: Philipp Kempgen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 5:36:45 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Problem with queue Sanjay Rajdev wrote: Regards, Sanjay Rajdev Tha i did because i dont want any call to get disconnected. Can you let me know what can be the problem doing so. - Original Message - From: Philipp Kempgen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 4:35:42 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Problem with queue Sanjay Rajdev wrote: I have queue set up in realtime on Asterisk 1.4.2. Below is the senario that is happenening :: I have created a test queue with only one agent. Once I call the test queue the agents phone rings if the aagent is logged on. everything till here is fine. Now if the agent does not pick up the call, the call automaticaly disconnects after 15 secs as set for the queue, till here also it is fine. But the agents phone never rings again for that Call and therefore the caller goes on an infinite wait and listen the wonderfull on hold music. :) Here are few more observations. If I reload the asterisk it ring again for one time. OR If the agent relogin then also it rings for one more time. OR If the caller disconnecs and callback again, it will ring one more time. Here is the agent.conf [general] persistentagents=yes multiplelogin=no [agents] autologoff=150 wrapuptime=6 6/60/60 = 16,67 *hours*! Use something like 5. ---cut--- ; Define wrapuptime. This is the minimum amount of time when ; after disconnecting before the caller can receive a new call ; note this is in milliseconds. ---cut--- Sorry, it's milliseconds. But even 60 seconds is probably quite long. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap failure: cause 66 - Channel not implemented
Which version of Zaptel and Asterisk are you using. If you have compiled Asterisk 1.4.2 with Zaptel 1.4.0 or a lesser version of Zaptel, you may face this problem. Regards, Sanjay Rajdev - Original Message - From: Greg Woods [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, April 13, 2007 9:29:43 PM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Zap failure: cause 66 - Channel not implemented On Fri, 2007-04-13 at 07:46 +0200, Jose Limeres wrote: when I try to make a call through the ZAP channel get an error message about NO ZAP CHANNEL AVAILABLE. Ztcfg and zttool show the card correctly installed. When I tried to use the debug command ZAP SHOW, it was not present in the CLI. This almost surely means that when you compiled asterisk, it did not detect that you had the Zaptel drivers. I had this happen to me and I beat my head against the wall for quite a while before I figured it out. You must compile and install the zaptel driver first, then compile asterisk. --Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] LED does not glow on new Voicemail
I have a CISCO 7912 phone, the LED on the phone does not glow when there is new voicemail, can we configure Asterisk to have the LED glow on new Voicemail. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LED does not glow on new Voicemail
I am using Asterisk in realtime with ODBC drivers. I tried setting the [EMAIL PROTECTED] in the sip_users table, but it did not seemed to work. I also do not see any message being sent or reject for the MWI notification on the Asterisk realtime. Furthermore I have Asterisks 1.4.2 installed on my asterisk box with SIP Firmware version 8.0.1 Any ideas Also it would be great if someone could tell me how to configure MWI from step 1, so that I can check if I am missing something. Regards, Sanjay Rajdev - Original Message - From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, April 14, 2007 3:11:47 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] LED does not glow on new Voicemail ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] missing chan_zap.so
Few days back I installed Asterisk 1.4.2 with Zaptel 1.4.0. All SIP accounts were working fine, today I tried to install a fxs Sangoma A200 card and got the following error. [Apr 12 01:15:17] WARNING[31018]: channel.c:3024 ast_request: No channel type registered for 'Zap' [Apr 12 01:15:17] WARNING[31018]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) Searched google and came to conclusion that I was missing chan_zap.so on my machine. Followed the instruction of the bug at http://bugzilla.atrpms.net/show_bug.cgi?id=1165 and downloaded zaptel 1.4.1, after that executed the following commands ./configure make clean make make install Went to asterisk folder ./configure make clean make make upgrade But could not get chan_zap.so then did the make install of asterisk. still missing the chan_zap.so Can someone please help. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ZAP does not disconnect
I have a ZAPTEL interface card with 4 channel. If I call out through the zap channel to my mobile, the mobile starts ringing, but If I disconnect the internal phone that is my SIP client the mobile does not stop ringing. Anyone any suggestion of what am I doing wrong. Regards, Sanjay Rajdev Phone : +1 (877) 342 2329 x 1702 Fax : +1 (815) 261 5907 http://www.featherstoneinformatics.com Communications from Featherstone Informatics Group (FIG) may transmit information that is confidential and privileged information of Featherstone Informatics Group (FIG). Unless you are the intended addressee, you may not use, copy or disclose to anyone this communication or any information transmitted by this communication. If you have received such communication in error, please advise the sender by e-mail and/or telephone and destroy this communication immediately. This communication and any information transmitted by this communication may also be considered protected health information as defined under the Health Insurance Portability and Accountability Act and its related regulations (a.k.a., HIPAA) or any other similar state law. Please exercise due care and ensure that you comply with its contractual and legal obligations. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Require only GSM Codec
Hello All, I would like to only use the gsm codec for all the calls, is it possible I want to use minimum possible bandwidth as we have most of calls over Internet. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem while using asterisk Realtime
Another thing I found is if i do the following # make menuselect go in option 2. Call Detail Recording here the option 4. cdr_odbc and option 5. cdr_pgsql both have XXX marked infront of them. And at the bottom of screen it says ODBC CDR Backend Depends on: unixodbc(E) I donot know why it says so as I have already mentioned below that the odbc connectivity is working fine. Also I have checked other option in Menuselect everywhere it says same for odbc. Can someone please let me know what I is wrong here. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: asterisk-dev@lists.digium.com Sent: Friday, March 30, 2007 4:09:16 AM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Problem while using asterisk Realtime I am having problem while having asterisk work with ODBC (Postgres) The error that I am getting is config.c: Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available I really donot know what has went wrong. I have set the ODBC connection properly I have verified it using :: [EMAIL PROTECTED] ~]# echo select 1 | isql asterisk +---+ | Connected!| | | | sql-statement | | help [tablename] | | quit | | | +---+ SQL ++ | ?column? | ++ | 1 | ++ SQLRowCount returns 1 1 rows fetched Here are the details of the stuff I am using OS :- fedora core 6 kernel 2798 (Was able to build asterisk on it) asterisk-1.4.1 libpri-1.4.0 zaptel-1.4.0 asterisk-addons-1.4.0 (Also tried with or without) Can someone please help, I am very new to asterisk. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem while using asterisk Realtime
I am having problem while having asterisk work with ODBC (Postgres) The error that I am getting is config.c: Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available I really donot know what has went wrong. I have set the ODBC connection properly I have verified it using :: [EMAIL PROTECTED] ~]# echo select 1 | isql asterisk +---+ | Connected!| | | | sql-statement | | help [tablename] | | quit | | | +---+ SQL ++ | ?column? | ++ | 1 | ++ SQLRowCount returns 1 1 rows fetched Here are the details of the stuff I am using OS :- fedora core 6 kernel 2798 (Was able to build asterisk on it) asterisk-1.4.1 libpri-1.4.0 zaptel-1.4.0 asterisk-addons-1.4.0 (Also tried with or without) Can someone please help, I am very new to asterisk. Regards, Sanjay Rajdev Phone : +1 (877) 342 2329 x 1702 Fax : +1 (815) 261 5907 http://www.featherstoneinformatics.com Communications from Featherstone Informatics Group (FIG) may transmit information that is confidential and privileged information of Featherstone Informatics Group (FIG). Unless you are the intended addressee, you may not use, copy or disclose to anyone this communication or any information transmitted by this communication. If you have received such communication in error, please advise the sender by e-mail and/or telephone and destroy this communication immediately. This communication and any information transmitted by this communication may also be considered protected health information as defined under the Health Insurance Portability and Accountability Act and its related regulations (a.k.a., HIPAA) or any other similar state law. Please exercise due care and ensure that you comply with its contractual and legal obligations. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP RTP Tunnel
Try setting canreinvite = no in sip.conf or the database (where you have sipuser setting). Regards, Sanjay Rajdev - Original Message - From: kalle odenthal [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, March 30, 2007 5:52:47 AM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] SIP RTP Tunnel Hello, is it possible to rout ALL RTP Data over Asterisk, like SIP1 ---RTP--- Asterisk ---RTP--- SIP2 I know it seems quite useless. But I want to simulate a IAX - SIP connection and have no Phonecard installed on my computer ;) Thanx, Kalle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
