Re: [asterisk-users] + dialplan
Hello Adam,
Thank you very much for your info.
Regards,
Jonson.
On Tue, Jun 11, 2013 at 12:34 AM, ad...@3a.hu wrote:
Hi,
On 06/10/2013 22:26, Jonson Player wrote:
Some users of main use + instead of 00 for international dial. Is there
any solution for this problem?
swap the + sign to double zeros if your provider can't handle it
; normal 00 prefix
exten = _00ZZXXX.,1,Macro(**beforealldials)
exten = _00ZZXXX.,n,Dial(SIP/${**EXTEN}@${OUTGOING_LINE})
exten = _00ZZXXX.,n,Hangup()
; swap + prefix to 00
exten = _+ZZXXX.,1,Macro(**beforealldials)
exten = _+ZZXXX.,n,Dial(SIP/00${**EXTEN:1}@${OUTGOING_LINE})
exten = _+ZZXXX.,n,Hangup()
regards
adam
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[asterisk-users] + dialplan
Hello guys, I looking for some dial plan which can mach on +xxx numbers instead of 00xxx numbers. Some users of main use + instead of 00 for international dial. Is there any solution for this problem? As far as i readed in asterisk is some kind of replacement of characters in dial plan command. Could i use that for archiving this option? Thank you for help. Jonson. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + Huawei K3765
Hello, I want to use an Huawei stick model K3765 which support voice with asterisk. I'm begginer with this kind of interaction from asterisk with external devices. Can someone guide me what should i configure to use this device? Thank you for support, Regards, Jonson. --- www.Mobile-Wi.Fi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No more connections allowed.
Hello, I have strange situation with asterisk 1.8.18.0 , randomly i got this message in cli: WARNING[15925] asterisk.c: No more connections allowed All connections freeze and all extensions doesn't work anymore. Is any bug or is any setting that can solve this problem? Thank you. Jonson. http://mobile-sip.tel/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No more connections allowed.
Hello Josua, Thank you for this answers: First of all, yes i run in crontab at 15 min an analyzing script which collect show sip channels with asterisk -rx . This could be my problem... I think that this commands could remain stalled and doesn't logout after execution of command. A friend of main told me that i could put in asterisk init script the following command: ulimit -HSn 135535 You think is useful? Another abortion could be to make something with manager /AMI scripts like that: http://ofps.oreilly.com/titles/9780596517342/asterisk-AMI.html What is your advice? I need this analyzing script to monitor when i have strange rise of use channels to prevent attacks or brute force. On Tue, Nov 20, 2012 at 3:44 PM, Joshua Colp jc...@digium.com wrote: Jonson Player wrote: Hello, Hola, I have strange situation with asterisk 1.8.18.0 , randomly i got this message in cli: WARNING[15925] asterisk.c: No more connections allowed This message is output when the number of Asterisk consoles (asterisk -r instances) has reached the limit. This limit is 128 by default which is quite a lot. Are you doing something that would cause a lot? Could they be hanging around by mistake? All connections freeze and all extensions doesn't work anymore. Is any bug or is any setting that can solve this problem? This definitely shouldn't happen but it would be useful to know exactly what you are doing with the system. Answering my questions above is a good start. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No more connections allowed.
Hello Danny, Could you tell me how can i put time out at execution of remote commands with asterisk -rx show sip channels. I think that is my problem... after i execute asterisk -rx commands something remain stalled and somehow i think that could block my asterisk... I mean all new connections couldn't be made anymore and the old active peers is nonfunctional and logged off, and the strangest thing is that i go on asterisk -rvc and i tap sip show peers i seen old active peers logged in... is clear that asterisk is freezed somehow and i need some workaround at this situation. Thank you. On Tue, Nov 20, 2012 at 4:50 PM, Danny Nicholas da...@debsinc.com wrote: You've exceeded the allowed maximum number of simultaneous remote console connections (128). While it may be a bit aggressive to have it kill all current connected consoles, its also a bit excessive to have 128 connected remote consoles. While the behaviour may not be entirely desirable, this isn't so much a bug as a limitation of the system. -- Why would you need 128 simultaneous remote consoles? IMO the number should be something like 16 and a timeout option needs to be added so you don't have unattended consoles consuming resources and opening holes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No more connections allowed.
Hello Matthew, Could I rise with some option the number of simultaneous console? I don't have simultaneous console but is good to know in case i didn't get any other workaround to fix this problem. Thank you. On Tue, Nov 20, 2012 at 3:56 PM, Matthew Jordan mjor...@digium.com wrote: On 11/20/2012 03:32 AM, Jonson Player wrote: Hello, I have strange situation with asterisk 1.8.18.0 , randomly i got this message in cli: WARNING[15925] asterisk.c: No more connections allowed All connections freeze and all extensions doesn't work anymore. Is any bug or is any setting that can solve this problem? Thank you. You've exceeded the allowed maximum number of simultaneous remote console connections (128). While it may be a bit aggressive to have it kill all current connected consoles, its also a bit excessive to have 128 connected remote consoles. While the behaviour may not be entirely desirable, this isn't so much a bug as a limitation of the system. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Advertising oportunity.
Hello, I don't know if this list is appropriated to this subject but I want to ask you if there's some list where I can make an advertising announce for a new sip web site that was just launched. hank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Questions about SPA3102.
Hello, I got a SPA3102 and everything works fine except calling from voip to phone on fxo port. The phone ring but doesn't get any sound. I connected SPA at my asterisk server and i want to call from asterisk through SPA to fxo port where i have a regular phone. Thank you for support. ___ --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] Strange problem with channel allocation
Hello I just settup a realtime mysql table for sip_peers. All peers
(friends) is autenticateing but when i want to initiate a call between them
i got the following error. Someone have some ideea? Thank you.
---Cut Here---
pbx*CLIconsole dial 1014
== Console is full duplex
-- Executing [EMAIL PROTECTED]:1] Dial(OSS/dsp, SIP/1014|40|t) in new
stack
[2007-06-03 20:16:10] DEBUG[27424]: res_config_mysql.c:650 mysql_reconnect:
MySQL RealTime: Everything is fine.
[2007-06-03 20:16:10] DEBUG[27424]: res_config_mysql.c:138 realtime_mysql:
MySQL RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = '1014'
-- Called 1014
[2007-06-03 20:16:10] WARNING[27424]: channel.c:3222
ast_channel_make_compatible: No path to translate from
SIP/1014-081e93c0(256) to OSS/dsp(64)
[2007-06-03 20:16:10] WARNING[27424]: channel.c:3222
ast_channel_make_compatible: No path to translate from
SIP/1014-081e93c0(256) to OSS/dsp(64)
^ ??
[2007-06-03 20:16:18] NOTICE[27408]: chan_sip.c:2758 auto_congest:
Auto-congesting SIP/1014-081e93c0
[2007-06-03 20:16:18] NOTICE[27408]: chan_sip.c:2758 auto_congest:
Auto-congesting SIP/1014-081e93c0
-- SIP/1014-081e93c0 is circuit-busy
[2007-06-03 20:16:18] DEBUG[27424]: cdr_addon_mysql.c:210 mysql_log:
cdr_mysql: inserting a CDR record.
[2007-06-03 20:16:18] DEBUG[27424]: cdr_addon_mysql.c:226 mysql_log:
cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,userfield)
VALUES ('2007-06-03 20:16:10','','','s','default',
'SIP/1014-081e93c0','','','',8,0,'NO ANSWER',3,'','')
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:2] VoiceMail(OSS/dsp, u1014) in new stack
Console call has been answered
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:6798 vm_exec:
Prefixing the mailbox with an option is deprecated ('u1014').
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:6798 vm_exec:
Prefixing the mailbox with an option is deprecated ('u1014').
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:6799 vm_exec: Please
move all leading options to the second argument.
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:6799 vm_exec: Please
move all leading options to the second argument.
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:2854 leave_voicemail:
No entry in voicemail config file for '1014'
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:2854 leave_voicemail:
No entry in voicemail config file for '1014'
-- Executing [EMAIL PROTECTED]:3] Hangup(OSS/dsp, ) in new stack
== Spawn extension (default, 1014, 3) exited non-zero on 'OSS/dsp'
[2007-06-03 20:16:18] DEBUG[27424]: cdr_addon_mysql.c:210 mysql_log:
cdr_mysql: inserting a CDR record.
[2007-06-03 20:16:18] DEBUG[27424]: cdr_addon_mysql.c:226 mysql_log:
cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,userfield)
VALUES ('2007-06-03 20:16:10','','','1014','default',
'OSS/dsp','SIP/1014-081e93c0','Hangup','',8,0,'ANSWERED',3,'','')
Hangup on console
[2007-06-03 20:16:18] DEBUG[27370]: res_config_mysql.c:650 mysql_reconnect:
MySQL RealTime: Everything is fine.
[2007-06-03 20:16:18] DEBUG[27370]: res_config_mysql.c:138 realtime_mysql:
MySQL RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = '1014'
---And Here---
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[asterisk-users] SIP accounts from MYSQL.
Hello, I just want to put all my sip accounts in mysql and asterisk use it from mysql. How can I do that, could you be more specific because I readed alot on wiki and i'm lost... I don't know what to modify in Makefile from channel directory. I use asterisk 1.4.4, that is already compiled and i also have CDR in mysql. I must create manny accounts and I want to realize that from mysql. Thank you for your support guys. ___ --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 accounts from MYSQL.
Than you Joss, the links was very usefull. On 5/27/07, Yossi Ben Hagai [EMAIL PROTECTED] wrote: Asterisk realtime is what you are looking for. the subject is explained very clearly including configuration examples and DB schema on the following links: http://www.voip-info.org/wiki-Asterisk+RealTime http://www.asteriskdocs.org/modules/news/article.php?storyid=28 I won't go over the process as it is detailed in the links above, but basically you should compile the asterisk-addons, configure the res_mysql with the proper DB details, create a table to hold sip.conf and optionally extensions.conf then configure extconfig to map the newly created tables. Joss. On 5/27/07, Jonson Player [EMAIL PROTECTED] wrote: Hello, I just want to put all my sip accounts in mysql and asterisk use it from mysql. How can I do that, could you be more specific because I readed alot on wiki and i'm lost... I don't know what to modify in Makefile from channel directory. I use asterisk 1.4.4, that is already compiled and i also have CDR in mysql. I must create manny accounts and I want to realize that from mysql. Thank you for your support guys. ___ --Bandwidth and Colocation provided by Easynews.comhttp://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 http://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] Local SMS how-to.
Can you tell me how may i do that? On 5/22/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Anselm Martin Hoffmeister [EMAIL PROTECTED] Date: Tue, 22 May 2007 13:41:43 +0200 Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player: Hello, i just want to activate SMS service between my asterisk local sip accounts and between asterisk and local sip accounts. How can i do this thin? Also i tried smsq to an account but all i obtained is a error message: ---Cut Here--- May 22 13:09:37 WARNING[4829] pbx_spool.c: Unable to open /var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1: Permission denied, deleting May 22 13:09:37 WARNING[4829] pbx_spool.c: Failed to scan service '/var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1' ---And Here--- Is necessary supplementary settings in /etc/asterisk/extensions.conf and /etc/asterisk/sip.conf ? Is necessary special module? I checked apps_sms.so is already loaded. Thank you for your support guys. No special change in sip.conf required. I've transmitted SMS over local SIP channel and it's be quire reliable - over LAN. Yuan Liu The SMSq stuff is for landline-type SMS, like those that never became really popular here in Europe ;-) I do not know of any SIP hardphone that supports them, but regular analog and ISDN handsets behind a SIP-to-analog/ISDN gateway work for me. The point of this SMS transfer method is calling the destination handset with a certain callerid set (which differs between countries - whatever number the telco prefers to choose - this can also be configured in the phone). The phone will not ring but instead immediately answer the call and receive the short message at 1200bps whatever modem standard they chose to use. For sending SMS, the handset will call a similarly telco-provided number (premium-rate numbers here in Germany - maybe that is the reason for the lack of popularity of this service) and do that 1200bps talk. If you still think you can make use of it, make sure to call smsq with the user id that asterisk is running as. That _might_ already do the trick. If you do not get it running, ask again - I might have a working setup somewhere around ;-) Nevertheless, for me, landline SMS is a PITA. The only great thing is you can upload Ringtones to Siemens gigaset phones. BR Anselm ___ --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] Local SMS how-to.
Was for [EMAIL PROTECTED] On 5/23/07, Jonson Player [EMAIL PROTECTED] wrote: Can you tell me how may i do that? On 5/22/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Anselm Martin Hoffmeister [EMAIL PROTECTED] Date: Tue, 22 May 2007 13:41:43 +0200 Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player: Hello, i just want to activate SMS service between my asterisk local sip accounts and between asterisk and local sip accounts. How can i do this thin? Also i tried smsq to an account but all i obtained is a error message: ---Cut Here--- May 22 13:09:37 WARNING[4829] pbx_spool.c: Unable to open /var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1: Permission denied, deleting May 22 13:09:37 WARNING[4829] pbx_spool.c: Failed to scan service '/var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1' ---And Here--- Is necessary supplementary settings in /etc/asterisk/extensions.conf and /etc/asterisk/sip.conf ? Is necessary special module? I checked apps_sms.so is already loaded. Thank you for your support guys. No special change in sip.conf required. I've transmitted SMS over local SIP channel and it's be quire reliable - over LAN. Yuan Liu The SMSq stuff is for landline-type SMS, like those that never became really popular here in Europe ;-) I do not know of any SIP hardphone that supports them, but regular analog and ISDN handsets behind a SIP-to-analog/ISDN gateway work for me. The point of this SMS transfer method is calling the destination handset with a certain callerid set (which differs between countries - whatever number the telco prefers to choose - this can also be configured in the phone). The phone will not ring but instead immediately answer the call and receive the short message at 1200bps whatever modem standard they chose to use. For sending SMS, the handset will call a similarly telco-provided number (premium-rate numbers here in Germany - maybe that is the reason for the lack of popularity of this service) and do that 1200bps talk. If you still think you can make use of it, make sure to call smsq with the user id that asterisk is running as. That _might_ already do the trick. If you do not get it running, ask again - I might have a working setup somewhere around ;-) Nevertheless, for me, landline SMS is a PITA. The only great thing is you can upload Ringtones to Siemens gigaset phones. BR Anselm ___ --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] Local SMS how-to.
I tried ... still same errors: ---Cut Here--- May 23 10:56:35 WARNING[31660]: pbx_spool.c:347 scan_service: Unable to open /var/spool/asterisk/outgoing/smsq.mttx.0.1179906994-32569.1: Permission denied, deleting May 23 10:56:35 WARNING[31660]: pbx_spool.c:389 scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/smsq.mttx.0.1179906994-32569.1' May 23 10:57:31 WARNING[31660]: pbx_spool.c:347 scan_service: Unable to open /var/spool/asterisk/outgoing/smsq.mttx.0.1179907051-32570.1: Permission denied, deleting May 23 10:57:31 WARNING[31660]: pbx_spool.c:389 scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/smsq.mttx.0.1179907051-32570.1' ---And Here--- On 5/22/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Dienstag, den 22.05.2007, 17:35 +0300 schrieb Jonson Player: Thank you for reply. Can you send me some working configs? I'm still confusing about this sms option. Just to get you started, try this: Find out which user asterisk runs as. Get a shell for that user. Run (all in one line) smsq --mt --oa=321 --mttx-callerid=01930101 --mttx-channel=SIP/abcde message text goes here where 321 will displayed as sender id on the handset, and 01930101 will have to replaced by the mobile center known to your phone, plus 1 at the end - the German T-Com seems to use 0193010, and this setting works for me. Further, SIP/abcde must be the channel that a SMS-capable handset is available on: If you have some ATA with a DECT handset connected, or similar, use the channel name exactly as you would in the Dial() command. First thing to find out is if this works. Be sure to have asterisk in extra-verbose running a console to see what happens. If the mobile handset rings (instead of getting the SMS) either the 01930101 number has not been set correctly or it probably is not compatible with Asterisk SMS. Once you get this far, you would need the other way round. When your mobile phone tries to _send_ a text message, it will go to 01930100 (sms center number plus 0). You will have to care for that in your extensions.conf, like this exten = 01930100,1,Wait(2) exten = 01930100,2,Answer() exten = 01930100,3,Wait(2) exten = 01930100,4,SMS(01930100,as) exten = 01930100,5,Wait(2) exten = 01930100,6,Hangup() In my experience those Wait(2) improve reliability over internet connections, they probably are superfluous if you have reliable low-latency LAN. For me, they made the difference between 10/100 and 95/100 successfuly sent messages. You will have to write your own scriptwork to play with the files that will be created from those commands. Their structure is simple, you will find out. Sending EMS (for ringtones and bitmaps) is a bit more complex, you will need the UDH flag for that. I think I documented that once on this ML but am not sure. However, it is possible with some Siemens Gigaset devices, and pictures or monophonic ringtones. BR Anselm ___ --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] voicemail notification.
Hello, I'm wandering how can I make voicemail notification when i got a messages in asterisk mailboxes. For the moment i have e-mail notifications, but I readed that I can do also a sms notification to local sip accounts. Also I'm wandering if i can make something like callback from asterisk to sip account, and play voicemail check, when the user log in. Is there someone that use this feature? Thank you. ___ --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 limit per sip account user.
Hello, I want to limit calls per sip account user. How may I realize this setting? For example I want to limit to 10 min all possible calls from an account or to limit external calls to 10 min and local call remain unlimited. Thank you for support guys. ___ --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] Local SMS how-to.
Hello, i just want to activate SMS service between my asterisk local sip accounts and between asterisk and local sip accounts. How can i do this thin? Also i tried smsq to an account but all i obtained is a error message: ---Cut Here--- May 22 13:09:37 WARNING[4829] pbx_spool.c: Unable to open /var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1: Permission denied, deleting May 22 13:09:37 WARNING[4829] pbx_spool.c: Failed to scan service '/var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1' ---And Here--- Is necessary supplementary settings in /etc/asterisk/extensions.conf and /etc/asterisk/sip.conf ? Is necessary special module? I checked apps_sms.so is already loaded. Thank you for your support guys. ___ --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] Local SMS how-to.
Thank you for reply. Can you send me some working configs? I'm still confusing about this sms option. On 5/22/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player: Hello, i just want to activate SMS service between my asterisk local sip accounts and between asterisk and local sip accounts. How can i do this thin? Also i tried smsq to an account but all i obtained is a error message: ---Cut Here--- May 22 13:09:37 WARNING[4829] pbx_spool.c: Unable to open /var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1: Permission denied, deleting May 22 13:09:37 WARNING[4829] pbx_spool.c: Failed to scan service '/var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1' ---And Here--- Is necessary supplementary settings in /etc/asterisk/extensions.conf and /etc/asterisk/sip.conf ? Is necessary special module? I checked apps_sms.so is already loaded. Thank you for your support guys. The SMSq stuff is for landline-type SMS, like those that never became really popular here in Europe ;-) I do not know of any SIP hardphone that supports them, but regular analog and ISDN handsets behind a SIP-to-analog/ISDN gateway work for me. The point of this SMS transfer method is calling the destination handset with a certain callerid set (which differs between countries - whatever number the telco prefers to choose - this can also be configured in the phone). The phone will not ring but instead immediately answer the call and receive the short message at 1200bps whatever modem standard they chose to use. For sending SMS, the handset will call a similarly telco-provided number (premium-rate numbers here in Germany - maybe that is the reason for the lack of popularity of this service) and do that 1200bps talk. If you still think you can make use of it, make sure to call smsq with the user id that asterisk is running as. That _might_ already do the trick. If you do not get it running, ask again - I might have a working setup somewhere around ;-) Nevertheless, for me, landline SMS is a PITA. The only great thing is you can upload Ringtones to Siemens gigaset phones. BR Anselm ___ --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] Problems with SPA3102
Hello, i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with cdr. Well all I want is to receive incoming calls from pstn on specified sip account (suppose 8000), and to initiate outgoing calls from all my asterisk sip accounts through SPA3102 device. Someone can explain me what may i set on SPA and asterisk to do this thing. Thank you for your support. ___ --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] Problems witch SPA3102.
Hello, i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with cdr. Well all I want is to receive incoming calls from pstn on specified sip account (suppose 8000), and to initiate outgoing calls from all my asterisk sip accounts through SPA3102 device. Someone can explain me what may i set on SPA and asterisk to do this thing. Thank you for your support. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Could two Asterisk servers connect through VPN
How about required MTU and jitter? I think openvpn will add some latency and frames will be charged with supplementary encapsulation bits. On 08 May 2007 19:03:09 +0200, Benny Amorsen [EMAIL PROTECTED] wrote: NM == Noah Miller [EMAIL PROTECTED] writes: NM If it helps at all, I read a study that said that SSL VPN's can NM actually help with jitter problems. So it might be preferable to NM implement something with OpenVPN (uses SSL) rather than an NM IPSec-based VPN. I found the link: Only if you use gold-plated connectors and oxygen-free copper. /Benny ___ --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] help with Sipura SPA 3000
Hello Francis,
I also hev asterisk and sipura. Can we chat online on gmail/yahoo. Let's
make some experiments... I hev the same problem like you.
On 4/12/07, Francis Augusto Medeiros [EMAIL PROTECTED] wrote:
On 10 de abr de 2007, at 23:05, James Harper wrote:
2 - How can I gain full control to the FXS? I mean, a simple * dialed
is
not sent for asterisk (the server) interpretation, probably because
it's
used by Sipura's suplementary services, I don't know. Also, is it
possible
to get a dial tone from ASterisk, instead of Sipura's? My goal with
this
is to provide users with direct access to the PSTN line pressing 0,
instead of collecting calls and making the call themselves, or at
least
making ignorepat to work!
A dialplan of '(S0:s)' will get your phone to jump straight into the
's' extension in asterisk as soon as someone picks it up. From
there you
can do something like:
It worked perfectly! Thanks!
[sip_ata_incoming]
exten = s,1,Answer
exten = s,n,DISA(no-password|sip_extension_in)
so Asterisk will give you dialtone and do the dialplan stuff for you.
From the 'sip_extension_in' context you can make a single '0' or '*'
call the PSTN line.
On the sip_extension_in, I entered the following
exten = 0,1,NoOp(outgoing call via POTS line to No. ${EXTEN:1})
exten = 0,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],120)
exten = 0,3,Congestion()
exten = 0,4,Hangup
However, when I press the 0, it does gives me a dialtone, but it
doesn't seem to be delivering the tones imediately. I even suspect it
isn't my PSTN tone after the 0. Is there something else?
Cheers,
Francis
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[asterisk-users] Error Message.
Hello, i just installed asterisk 1.2.15. I got this error message. Somebody can help me? Thank You. Feb 27 11:47:43 NOTICE[17086] cdr.c: CDR simple logging enabled. Feb 27 11:47:44 WARNING[17086] pbx.c: Already have an application 'Pickup' Feb 27 11:47:44 WARNING[17086] loader.c: app_directed_pickup.so: load_module failed, returning -1 Feb 27 11:47:44 WARNING[17086] loader.c: Loading module app_directed_pickup.so failed! ___ --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] queue information into db
Hello, I'm interested too in analyzer/statistics/billing system. Can we develop together something simple? What scripts do you recomand me? Thank you, Jonson. On 2/22/07, nik600 [EMAIL PROTECTED] wrote: I am planning to develop an open source (GPL) queue statistic/analyzer. Can i use that to store data into the db? Or shall i wrote some php code to do that? On 2/22/07, lenz [EMAIL PROTECTED] wrote: Not sure about * 1.4, but you can definitely use our Qloaderd script to do that - see http://queuemetrics.com/download.jsp . That script is pretty smart (to be a loader script...) and is able to handle restarts and database disconnections. l. In data Thu, 22 Feb 2007 09:20:59 +0100, nik600 [EMAIL PROTECTED] ha scritto: Hi the new asterisk 1.4 supports to store queue log information directly into a database? (like CDR) ? thanks -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sample Config.
Hello Mihaela, my name is Catalin. Thank you for you advices, I still hev problems with my spa3102, I just wanna use a xlite to log in to sipura... is that possible? I try to make small steps to understand all options. Thank you for your support. Catalin S. On 1/27/07, Token PBX [EMAIL PROTECTED] wrote: Hi! I don't understand what you mean by : „configure voice part on it, but I can give general guidelines: First you setup SPA3000 web UI: 1) Line1 Tab: Sip settings: SIP port : 5060 Proxy and Registration: Proxy: Asterisk IP Subscriber Information: Display Name: FXS_username Password: FXS password User ID: FXS_username 2) PSTN Line Tab: SIP Settings: SIP port: 5061 Proxy and Registration: Proxy: Asterisk IP Subscriber Information: Display Name: FXO_username Password: FXO_password User ID: FXO_username Dial Plans: Dial Plan 1: (S0:[EMAIL PROTECTED] IP:5060)(may be any other dial plan) VoIP-To-PSTN Gateway Setup: VoIP-To-PSTN Gateway Enable: Yes Line 1 VoIP Caller DP: 1 (or any other setup like Dial Plan 1) VoIP Users and Passwords (HTTP Authentication) VoIP User 1 Auth ID: asterisk VoIP User 1 DP: 1(same as above) PSTN-To-VoIP Gateway Setup: PSTN-To-VoIP Gateway Enable: Yes Then Asterisk sip.conf: [ FXO_username] disallow=all allow=alaw type=friend fromuser= FXO_username username= FXO_username secret= FXO_password host=dynamic dtmfmode=rfc2833 canreinvite=no qualify=1000 context=incoming port=5061 [FXS_username ] disallow=all allow=alaw type=friend username= FXS_username secret= FXS_password host=dynamic dtmfmode=rfc2833 canreinvite=no qualify=1000 context=outgoing Best regards Mihaela MJ On 1/26/07, Jonson Player [EMAIL PROTECTED] wrote: Hello, I just buyed a SPA3102 from Linksys. Can anyone help me or guide me to configure voice part on it. I cannot get it if I can use like peer for my asterisk. Please help me with some tips. Thank you guys. Regards, Jonson. ___ --Bandwidth and Colocation provided by Easynews.comhttp://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 http://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] Sample Config.
Hello, I just buyed a SPA3102 from Linksys. Can anyone help me or guide me to configure voice part on it. I cannot get it if I can use like peer for my asterisk. Please help me with some tips. Thank you guys. Regards, Jonson. ___ --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 about FXO/FXS device.
Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking about SPA3102. What you guys think about it. Is ok, is working with asterisk, can i use it like voip peer. Thank you for your advice. Jonson. ___ --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] Re: [asterisk-dev] Question about FXO/FXS device.
Okay, i'll move my discuss to asterisk-users. Thank you. On 1/17/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jan 17, 2007 at 04:39:03PM +0800, 黄宗宁 wrote: Jonson Player wrote: Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking about SPA3102. What you guys thik about it. Is ok, is working with asterisk, can i use it like voip peer. Thank you for your advice. Generally yes. This type of question should be asked on asterisk-users . This is a list for the development of Asterisk (ugly code stuff). Please follow-up there. Why done you buy FXO/FXS device from China.It is cheaper than other ones and more compact with asterisk. This belongs on either asterisk-users or asterisk-biz (if you happen to promote your own product). BTW: sorry for the messed encoding. I guess that sending a message in UTF-8 would have been safer. -- 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-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --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] Push to Talk settings.
Hello if someone found some method to authentificate to asterisk with nokia push to talk clients please send me all your documentations and the tests results, I really need this for a project of main and i wanna dig deeper to solve this mister. Thank you guys for your cooperation. Alex i put you at cc because i know you find this interesting too and maybe meanwhile you already know more about this... Let's begin a thread. Thank you again. ___ --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] Simple example for call transfer.
Hello, i hev a subscription to a international voip provider and I want all calls for numbers _001xx to go through my voip provider. I tried many settings in sip.conf, extensions.conf and iax.conf. Please give me some simple example for how can i transfer the specified calls to my external voip provider. What may I put and where in witch file. Thank you for your support. ___ --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
